From pinghan022 at gmail.com Wed Jun 1 02:48:28 2016 From: pinghan022 at gmail.com (Ping Han) Date: Wed, 1 Jun 2016 10:48:28 +1000 Subject: [OpenSIPS-Users] Question regarding b2b_bridge function In-Reply-To: <574D44F7.9090204@opensips.org> References: <574D44F7.9090204@opensips.org> Message-ID: Hi Bogdan, Thank you very much for your reply. I have tried to use the module parameter "b2bl_key_avp" as described in the document as below. ------------------------------------ modparam("b2b_logic", "b2bl_key_avp", "$avp(99)") ------------------------------------ However, I got the following errors when the Opensips is restarted. ------------------------------------ ERROR:core:set_mod_param_regex: parameter not found in module ------------------------------------ I am using the Opensips version opensips-2.1.2-1.el6.x86_64 (rpm). Thanks, Chris On Tue, May 31, 2016 at 6:01 PM, Bogdan-Andrei Iancu wrote: > Hi Chris, > > The "dialog_id" is actually the b2b key, that is expose by the b2b_logic > via the module parameter b2bl_key_avp. See: > > http://www.opensips.org/html/docs/modules/2.1.x/b2b_logic.html#id294094 > > That key can also be found in the b2b_logic table in DB. > > At signaling level, the key is the Call-ID of the outbound calls from b2b. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 23.05.2016 07:32, Ping Han wrote: > > Hi Bogdan, > > I asked the question a few days ago but have not got a response. > > I am just wondering if I could get some advice from you. > > Any advice will be appreciated. > > Thanks, > Chris > > On Wed, May 18, 2016 at 4:39 PM, Ping Han wrote: > >> Hi, >> >> I would like to use the b2b_bridge fifo function as specified at >> http://www.opensips.org/html/docs/modules/2.1.x/b2b_logic.html#id248916. >> >> The function will be triggered by a third party. I will need to pass the >> parameters to the third party for it to trigger the function. One of the >> parameters is the "dialog-id". >> >> The problem is that I am not sure how the value of the dialog-id can be >> available in the Opensips config. Is there any Opensips modules/function >> that can retrieve the value of the dialog-id? >> >> I tried to get the value from the "b2b_entities" and "b2b_logic" table. >> However, it seems that it does not work this way because the two tables do >> not pop the data in real time. Sometimes I can see the data but sometimes I >> am not able to see it. >> >> It is appreciated that you can give me some idea. >> >> Thanks, >> >> Ping >> > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Wed Jun 1 08:46:18 2016 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Wed, 1 Jun 2016 09:46:18 +0300 Subject: [OpenSIPS-Users] Segfault using Loadbalancer Module. In-Reply-To: References: Message-ID: <61bee8db-047b-93b9-de19-76cfb72b4a91@opensips.org> Hi, Qasim! Yes, please post a backtrace of the core dump on pastebin for further investigation. Best regards, R?zvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 05/31/2016 10:12 PM, qasimakhan at gmail.com wrote: > Hi, > > I was using default script generated by opensips menuconfig and it > gives the following segfault > > http://pastebin.com/6zuimn5N > > I was evaluating opensips 2.2 latest release. Please let me know if > core dump is required > > Regards, > Qasim Ayyaz Khan > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Jun 1 09:10:13 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 1 Jun 2016 10:10:13 +0300 Subject: [OpenSIPS-Users] Out of memory problem In-Reply-To: <1038866634.20160531191636@ptl.ru> References: <899020901.20160229093310@ptl.ru> <20160229063616.5435471.28050.211241@evaristesys.com> <1215773319.20160229094623@ptl.ru> <20160229065024.5435471.72953.211245@evaristesys.com> <1916375360.20160229112009@ptl.ru> <56D422A1.3060605@opensips.org> <126920209.20160229135826@ptl.ru> <56D425F5.9080604@opensips.org> <1038866634.20160531191636@ptl.ru> Message-ID: <574E8A55.7050000@opensips.org> Hi Denis, If you upgrade to 2.2, use the DBG_MALLOC (together with your memory manager like F_MALLOC, HP_MALLOC or QM_MALLOC) to enable memory leak troubleshooting - at shutdown you will get a memory dump, so you will clearly see if there is a leak of not. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 31.05.2016 19:16, Denis wrote: > Re: [OpenSIPS-Users] Out of memory problem Hello! > > I had increased PKG memory twice (to 1000 M) but the problem is still > exists. > > May 31 19:09:29 opensips-main > /usr/local/opensips2.1/sbin/opensips[20681]: > ERROR:core:db_allocate_rows: no memory left > May 31 19:09:29 opensips-main > /usr/local/opensips2.1/sbin/opensips[20681]: > ERROR:db_mysql:db_mysql_fetch_result: no memory left > May 31 19:09:29 opensips-main > /usr/local/opensips2.1/sbin/opensips[20681]: > ERROR:dialplan:dp_load_db: failed to fetch > May 31 19:09:29 opensips-main > /usr/local/opensips2.1/sbin/opensips[20681]: > ERROR:dialplan:dp_load_all_db: unable to load ast_dialplan table > May 31 19:09:29 opensips-main > /usr/local/opensips2.1/sbin/opensips[20681]: > ERROR:dialplan:mi_reload_rules: failed to reload database > May 31 19:09:29 opensips-main > /usr/local/opensips2.1/sbin/opensips[20681]: > ERROR:mi_fifo:mi_fifo_server: command (dp_reload) processing failed > > Haw else can i make memory debug without Opensips stopped? > > Thank you. > > mailto:denis7979 at mail.ru > > > No, more like the 2.2+ LTS, due to be released towards the end of > March, a fork of the current "master" branch on git. > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > On 29.02.2016 12:58, ????? ?????? wrote: > Re: [OpenSIPS-Users] Out of memory problem Newer version = 2.1? > > mailto:denis7979 at mail.ru > > > This is PKG memory (i.e. packaged / private / per-process), so you > should actually increase "-M" CLI switch! > > Note: Newer versions of OpenSIPS will have improved error reporting > for easier troubleshooting of oom (out-of-memory) errors. > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > On 29.02.2016 10:20, ????? ?????? wrote: > Re: [OpenSIPS-Users] Out of memory problem > ERROR:core:db_allocate_rows: no memory left > ERROR:db_mysql:db_mysql_fetch_result: no memory left > ERROR:dialplan:dp_load_db: failed to fetch > ERROR:dialplan:dp_load_all_db: unable to load ast_dialplan table > ERROR:dialplan:mi_reload_rules: failed to reload database > ERROR:mi_fifo:mi_fifo_server: command () processing failed > > > mailto:denis7979 at mail.ru > > /> What is the exact text of the error? > > ? > > -- > > Alex Balashov | Principal | Evariste Systems LLC > > 303 Perimeter Center North, Suite 300 > > Atlanta, GA 30346 > > United States > > > Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) > > Web: /http://www.evaristesys.com//, /http://www.csrpswitch.com/ > > /> Sent from my BlackBerry. > > Original Message > > From: ????? ?????? > > Sent: Monday, February 29, 2016 00:58 > > To: Alex Balashov; OpenSIPS users mailling list > > Subject: Re: [OpenSIPS-Users] Out of memory problem > > > > _______________________________________________ > > Users mailing list > > /Users at lists.opensips.org > /> /http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > /_______________________________________________ > Users mailing list > /Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Jun 1 09:13:30 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 1 Jun 2016 10:13:30 +0300 Subject: [OpenSIPS-Users] Question regarding b2b_bridge function In-Reply-To: References: <574D44F7.9090204@opensips.org> Message-ID: <574E8B1A.7070508@opensips.org> Hi Ping, Indeed, my bad - the docs are not updated, as that param was disabled long time ago (4 years ago): https://sourceforge.net/p/opensips/bugs/502/ Still, there are available option. But the question is : do you need that value in OpenSIPS cfg or outside OpenSIPS ? as there are different way to get the ID. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 01.06.2016 03:48, Ping Han wrote: > Hi Bogdan, > > Thank you very much for your reply. > > I have tried to use the module parameter "b2bl_key_avp" as described > in the document as below. > > ------------------------------------ > modparam("b2b_logic", "b2bl_key_avp", "$avp(99)") > ------------------------------------ > > However, I got the following errors when the Opensips is restarted. > ------------------------------------ > ERROR:core:set_mod_param_regex: parameter not found in > module > ------------------------------------ > > I am using the Opensips version opensips-2.1.2-1.el6.x86_64 (rpm). > > Thanks, > Chris > > On Tue, May 31, 2016 at 6:01 PM, Bogdan-Andrei Iancu > > wrote: > > Hi Chris, > > The "dialog_id" is actually the b2b key, that is expose by the > b2b_logic via the module parameter b2bl_key_avp. See: > http://www.opensips.org/html/docs/modules/2.1.x/b2b_logic.html#id294094 > > That key can also be found in the b2b_logic table in DB. > > At signaling level, the key is the Call-ID of the outbound calls > from b2b. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 23.05.2016 07:32, Ping Han wrote: >> Hi Bogdan, >> >> I asked the question a few days ago but have not got a response. >> >> I am just wondering if I could get some advice from you. >> >> Any advice will be appreciated. >> >> Thanks, >> Chris >> >> On Wed, May 18, 2016 at 4:39 PM, Ping Han > > wrote: >> >> Hi, >> >> I would like to use the b2b_bridge fifo function as specified >> at >> http://www.opensips.org/html/docs/modules/2.1.x/b2b_logic.html#id248916. >> >> The function will be triggered by a third party. I will need >> to pass the parameters to the third party for it to trigger >> the function. One of the parameters is the "dialog-id". >> >> The problem is that I am not sure how the value of the >> dialog-id can be available in the Opensips config. Is there >> any Opensips modules/function that can retrieve the value of >> the dialog-id? >> >> I tried to get the value from the "b2b_entities" and >> "b2b_logic" table. However, it seems that it does not work >> this way because the two tables do not pop the data in real >> time. Sometimes I can see the data but sometimes I am not >> able to see it. >> >> It is appreciated that you can give me some idea. >> >> Thanks, >> >> Ping >> >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From miha at softnet.si Wed Jun 1 10:04:49 2016 From: miha at softnet.si (Miha) Date: Wed, 1 Jun 2016 10:04:49 +0200 Subject: [OpenSIPS-Users] engage_rtp_proxy() In-Reply-To: References: <11f00e05-ef31-214c-115d-da9fedde6451@softnet.si> <6753bb93-1801-d3b7-15a9-c53ec7567cb0@softnet.si> <574D8B6D.4010706@42com.com> <59dbfe5f-20f9-bba9-209a-bc9f5e685179@softnet.si> Message-ID: HI engage_rtp_proxy() work ok. I was having some other issue with dialog. Tnx to @Bogdan I figure it out. br miha On 31/05/2016 15:54, Sasmita Panda wrote: > In my case its working great . So I haven't done such experiments to > know what is happening with dialog module . We are using this form > years . > > If you got to know then let me know also . That may help me in > future . > > */Thanks & Regards/* > /Sasmita Panda/ > /Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > On Tue, May 31, 2016 at 6:56 PM, Miha > wrote: > > @Sasmita I had writen cfg script like this and it works. I tried > than with engage_rtp_proxy() but did not work automatically, that > is why i asked :) > > So, can be some issue with dialog module? Not configured properly? > > > tnx > > miha > > > On 31/05/2016 15:21, Sasmita Panda wrote: >> Yes . This should happen . But I don't know the exact problem . >> What I explain is the way we are using rtpproxy . >> This is clearly mention in the document also .. You can go >> through opensips.org >> >> This is what we are doing . Rest I am not an expertise in >> opensips . >> route { >> ... >> if (is_method("INVITE")) { >> if (has_body("application/sdp")) { >> if (rtpproxy_offer()) >> t_on_reply("1"); >> } else { >> t_on_reply("2"); >> } >> } >> if (is_method("ACK") && has_body("application/sdp")) >> rtpproxy_answer(); >> ... >> } >> >> onreply_route[1] >> { >> ... >> if (has_body("application/sdp")) >> rtpproxy_answer(); >> ... >> } >> >> onreply_route[2] >> { >> ... >> if (has_body("application/sdp")) >> rtpproxy_offer(); >> ... >> } >> >> */Thanks & Regards/* >> /Sasmita Panda/ >> /Network Testing and Software Engineer/ >> /3CLogic , ph:07827611765/ >> >> On Tue, May 31, 2016 at 6:32 PM, Max M?hlbronner > > wrote: >> >> Hi, >> >> >> @Miha: Are you sure that it does not automatically set the >> rtpproxies for 200OK & ACK? >> >> @Sasmita: According to the documentation it is not necessary >> to invoke engage_rtp_proxy() for replies as this is handled >> by the dialog module. >> >> >> "Function must only be called for the initial INVITE and >> internally takes care of rewriting the body of 200 OKs and >> ACKs. " >> >> >> >> Best Regards >> >> Max M. >> >> >> On 31.05.2016 14:42, Miha wrote: >>> @Sasmita, totally clear :) >>> >>> I asked wrong question :) >>> >>> >>> What is the difference between using engage_rtp_proxy() or >>> using rtpproxy_offer(), rtpproxy_answer()? >>> >>> >>> tnx >>> >>> miha >>> >>> >>> On 31/05/2016 14:39, Sasmita Panda wrote: >>>> If you are using in INVITE , then it should be >>>> offer . Because firstly we are offering media to someone . >>>> If its 200 Ok then it will be answer because the 2nd party >>>> is answering the call . >>>> >>>> If there is no sdp in INVITE but in ACK , then it >>>> will get reversed . In 200 OK you should offer and in ACK >>>> you have to answer . >>>> This can be done in loop . >>>> >>>> I hope I make you understand . >>>> >>>> */Thanks & Regards/* >>>> /Sasmita Panda/ >>>> /Network Testing and Software Engineer/ >>>> /3CLogic , ph:07827611765/ >>>> >>>> On Tue, May 31, 2016 at 6:02 PM, Miha >>> >>>> > wrote: >>>> >>>> ok tnx. I understand documentation on wrong way. >>>> >>>> But then, what is the difference with using rtpproxy >>>> offer, answer ? >>>> >>>> >>>> br >>>> >>>> mia >>>> >>>> >>>> On 31/05/2016 14:17, Sasmita Panda wrote: >>>>> If there is sdp in ACK and u wanted to engage rtp >>>>> proxy , the >>>>> you have to write it inside ACK also ... By writing >>>>> for INVITE >>>>> cant help you to update ACK also . For 200 OK , you >>>>> must write it >>>>> in reply route . >>>>> >>>>> */Thanks & Regards/* >>>>> /Sasmita Panda/ >>>>> /Network Testing and Software Engineer/ >>>>> /3CLogic , ph:07827611765/ >>>>> >>>>> On Tue, May 31, 2016 at 5:35 PM, Johan De Clercq >>>>> >>>>> > wrote: >>>>> >>>>> put it also in reply route. >>>>> >>>>> 2016-05-31 13:42 GMT+02:00 Miha >>>> >>>>> >: >>>>> >>>>> HI >>>>> >>>>> if I use engage_rtp_proxy(), I can use it only >>>>> on initial >>>>> INVITE and opensips should automatically >>>>> rewritten also >>>>> 200 OK and ACK with SDP, right? >>>>> But when I am using this function, I can see >>>>> from trace >>>>> that only SDP for initial invite is rewritten, >>>>> 200 ok >>>>> with sdp is not changed. Must I do something >>>>> else? >>>>> >>>>> Rtpproxy is not running in bridge mode. >>>>> >>>>> >>>>> tnx >>>>> miha >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> >>>>> >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> >>>>> >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> >>>>> >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> >>>> >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From qasimakhan at gmail.com Wed Jun 1 12:17:21 2016 From: qasimakhan at gmail.com (qasimakhan at gmail.com) Date: Wed, 1 Jun 2016 15:17:21 +0500 Subject: [OpenSIPS-Users] Segfault using Loadbalancer Module. In-Reply-To: <61bee8db-047b-93b9-de19-76cfb72b4a91@opensips.org> References: <61bee8db-047b-93b9-de19-76cfb72b4a91@opensips.org> Message-ID: Dear Razvan, Please find below backtrace of the core file: http://pastebin.com/WpLtezAS Regards, Qasim Ayyaz Khan On Wed, Jun 1, 2016 at 11:46 AM, R?zvan Crainea wrote: > Hi, Qasim! > > > Yes, please post a backtrace of the core dump on pastebin for further > investigation. > > Best regards, > > R?zvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 05/31/2016 10:12 PM, qasimakhan at gmail.com wrote: > > Hi, > > I was using default script generated by opensips menuconfig and it gives > the following segfault > > http://pastebin.com/6zuimn5N > > I was evaluating opensips 2.2 latest release. Please let me know if core > dump is required > > Regards, > Qasim Ayyaz Khan > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Wed Jun 1 12:17:33 2016 From: spanda at 3clogic.com (Sasmita Panda) Date: Wed, 1 Jun 2016 15:47:33 +0530 Subject: [OpenSIPS-Users] Connecting to mongodb Replica set from opensips . Message-ID: HI Guys , i am trying to use mongodb in replica set . I have tested my opensips-1.11 with a stand alone mongodb instance . Its working fine . Now I am trying to connect to a replica set . But somehow I amnot able to do this . Opensips is getting connected to the replica set but its not able to run raw query . My opensips configuration is like bellow : loadmodule "cachedb_mongodb.so" loadmodule "db_cachedb.so" modparam("cachedb_mongodb", "cachedb_url","mongodb:replicaset1:// root:password at 1.2.3.4:27017,2.3.4.5:27017/jack.db.CallCenter_Info") modparam("db_cachedb","cachedb_url","mongodb:replicaset1:// root:password at 1.2.3.4:27017,2.3.4.5:27017/jack.db.test") modparam("acc", "db_url", "cachedb://mongodb:replicaset1") modparam("acc", "db_flag", 2) modparam("acc", "log_flag", 2) modparam("acc", "cdr_flag", 1) modparam("acc", "log_facility", "LOG_LOCAL7") Opensips is not able to execute version check query for acc module . Bellow is the logs of opensips . INFO:cachedb_mongodb:mongo_new_connection: Connected at server 52.71.216.67:27017,23.21.65.168:27017 with version 3.0.12 , to db db.test DBG:db_cachedb:db_cachedb_init: Succesfully initiated connection to [mongodb:replicaset1] DBG:cachedb_mongodb:mongo_db_query_trans: Running raw mongo query on table db.my_version_table ERROR:cachedb_mongodb:mongo_db_query_trans: Failed to run query. Err = 0, 0 , 0 DBG:cachedb_mongodb:mongo_db_query_trans: Fetched key lastOp DBG:cachedb_mongodb:mongo_db_query_trans: (unknown type 17) DBG:cachedb_mongodb:mongo_db_query_trans: Fetched key connectionId DBG:cachedb_mongodb:mongo_db_query_trans: (int) 516 DBG:cachedb_mongodb:mongo_db_query_trans: Fetched key n DBG:cachedb_mongodb:mongo_db_query_trans: (int) 0 DBG:cachedb_mongodb:mongo_db_query_trans: Fetched key syncMillis DBG:cachedb_mongodb:mongo_db_query_trans: (int) 0 DBG:core:wait_status_code: read code 0 ? rc = 0, errno=Success INFO:core:daemonize: pre-daemon process exiting with -1 In my mongodb instance there is some error printing also . [initandlisten] connection accepted from 104.131.6.7:35857 #565 (7 connections now open) 2016-06-01T10:09:14.802+0000 I QUERY [conn565] assertion 13 not authorized for query on db.my_version_table ns:db.my_version_table query:{ $query: { table_name: "acc" } } By login through a remote machine with same username and password . I am able to execute all query manually . But why my opensips cant do this . If anybody have tried this can you suggest me any solution . What else I should do in my replica set or in opensips to make it happen ? Please help me . As I am quite new to mongodb replica set so not able to understand the actual problem also . *Thanks & Regards* *Sasmita Panda* *Network Testing and Software Engineer* *3CLogic , ph:07827611765* -------------- next part -------------- An HTML attachment was scrubbed... URL: From saioa.perurena at enigmedia.es Wed Jun 1 12:22:30 2016 From: saioa.perurena at enigmedia.es (Saioa Perurena) Date: Wed, 1 Jun 2016 12:22:30 +0200 Subject: [OpenSIPS-Users] Valid tls connection closed on a "dos" simulation Message-ID: <574EB766.9070704@enigmedia.es> Hi, We have an opensips 1.11.6 server with tls and we simulate a dos attack sending an invalid request to the tls port every 10 seconds (echo "giberish" | nc sip-service.example.com 5061 ). We have only two UAC connected correctly through tls, when one of this clients sends an INVITE request when the dos attack is working, then servers close the tls connection of that client with error: ERROR:proto_tls:tls_print_errstack: TLS errstack: error:1408F10B:SSL routines:SSL3_GET_RECORD:wrong version number When client sends MESSAGE or OPTIONS request it does not happen. If we stop the dos attack all works correctly. We can reproduce it so easily, also with Opensips 2.1 version. Any idea of what is happening?? Maybe it is a bug on tls? Any suggestion or idea is welcome. Thanks in advance. Saioa. From denis7979 at mail.ru Wed Jun 1 12:26:15 2016 From: denis7979 at mail.ru (Denis) Date: Wed, 1 Jun 2016 13:26:15 +0300 Subject: [OpenSIPS-Users] Fraud module In-Reply-To: <574DA47E.6070005@opensips.org> References: <1472412649.20160531155045@ptl.ru> <574DA47E.6070005@opensips.org> Message-ID: <469549155.20160601132615@ptl.ru> Hello Liviu! Is this action only for "sequential calls"? And what do mean about "different number"? I have prefix "810" in profile table. And client dials different numbers but all of them begin from 810. And i see that count of "sequential calls" increases for each call. mailto:denis7979 at mail.ru Hi Denis, Currently this statistic is not reset between intervals, but only when a user dials a different number than the one(s) before. This makes it so that the module never loses track of the sequence of calls a particular user has made. In your particular case, does it make more sense for the module to do a reset? Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 31.05.2016 15:50, ????? ?????? wrote: Fraud module Hello! I am using Opensips 2.1 with fraud module. Should the module resets current statistics when new time stamp begin? I use 00:00 - 23:59 time interval for fraud detection and see that, for example, "sequential calls" increases for every call every day. Thank you mailto:denis7979 at mail.ru _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Wed Jun 1 12:56:14 2016 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 1 Jun 2016 13:56:14 +0300 Subject: [OpenSIPS-Users] Fraud module In-Reply-To: <469549155.20160601132615@ptl.ru> References: <1472412649.20160531155045@ptl.ru> <574DA47E.6070005@opensips.org> <469549155.20160601132615@ptl.ru> Message-ID: <574EBF4E.6070903@opensips.org> That is the normal behavior, since by "different number" I actually meant "different matching rule". In your case, that short prefix rule will match a lot more often for a given user, since it's likely that all his contacts are within the same region/country, so you should first assess whether a given statistic is relevant or not for a given rule, before setting its alerting thresholds! Maybe we should add the possibility to disable certain statistics by entering "-1" as value for either of their thresholds. Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 01.06.2016 13:26, Denis wrote: > Re: [OpenSIPS-Users] Fraud module Hello Liviu! > > Is this action only for "sequential calls"? > And what do mean about "different number"? > I have prefix "810" in profile table. And client dials different > numbers but all of them begin from 810. And i see that count of > "sequential calls" increases for each call. > > mailto:denis7979 at mail.ru > > > Hi Denis, > > Currently this statistic is not reset between intervals, but only when > a user dials a different number than the one(s) before. > This makes it so that the module never loses track of the sequence of > calls a particular user has made. > > In your particular case, does it make more sense for the module to do > a reset? > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > On 31.05.2016 15:50, ????? ?????? wrote: > Fraud module Hello! > > I am using Opensips 2.1 with fraud module. > Should the module resets current statistics when new time stamp begin? > I use 00:00 - 23:59 time interval for fraud detection and see that, > for example, "sequential calls" increases for every call every day. > > Thank you > > mailto:denis7979 at mail.ru > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Wed Jun 1 13:18:54 2016 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 1 Jun 2016 14:18:54 +0300 Subject: [OpenSIPS-Users] Valid tls connection closed on a "dos" simulation In-Reply-To: <574EB766.9070704@enigmedia.es> References: <574EB766.9070704@enigmedia.es> Message-ID: <574EC49E.4050800@opensips.org> Hi Saioa, We have addressed this issue somewhere between OpenSIPS 1.11.6 - 1.11.7, and 2.1.2 - 2.1.3. Please update to the latest version (possibly even from GitHub [1], [2]), and let us know if it solved your problem! [1]: https://github.com/OpenSIPS/opensips/tree/1.11 [2]: https://github.com/OpenSIPS/opensips/tree/2.1 Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 01.06.2016 13:22, Saioa Perurena wrote: > Hi, > > We have an opensips 1.11.6 server with tls and we simulate a dos attack > sending an invalid request to the tls port every 10 seconds (echo > "giberish" | nc sip-service.example.com 5061 ). > > We have only two UAC connected correctly through tls, when one of this > clients sends an INVITE request when the dos attack is working, then > servers close the tls connection of that client with error: > ERROR:proto_tls:tls_print_errstack: TLS errstack: error:1408F10B:SSL > routines:SSL3_GET_RECORD:wrong version number > > When client sends MESSAGE or OPTIONS request it does not happen. > > If we stop the dos attack all works correctly. We can reproduce it so > easily, also with Opensips 2.1 version. > > Any idea of what is happening?? Maybe it is a bug on tls? Any suggestion > or idea is welcome. > > Thanks in advance. > > Saioa. > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From denis7979 at mail.ru Wed Jun 1 13:40:43 2016 From: denis7979 at mail.ru (Denis) Date: Wed, 1 Jun 2016 14:40:43 +0300 Subject: [OpenSIPS-Users] Fraud module In-Reply-To: <574EBF4E.6070903@opensips.org> References: <1472412649.20160531155045@ptl.ru> <574DA47E.6070005@opensips.org> <469549155.20160601132615@ptl.ru> <574EBF4E.6070903@opensips.org> Message-ID: <128426099.20160601144043@ptl.ru> I understand. Disabling certain statistics is a quite good idea. An additional offer. To have a way for resetting certain statistic of certain user using, for example, mi interface. It will vary help to administrate fraud detection service. Thank you. mailto:denis7979 at mail.ru That is the normal behavior, since by "different number" I actually meant "different matching rule". In your case, that short prefix rule will match a lot more often for a given user, since it's likely that all his contacts are within the same region/country, so you should first assess whether a given statistic is relevant or not for a given rule, before setting its alerting thresholds! Maybe we should add the possibility to disable certain statistics by entering "-1" as value for either of their thresholds. Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 01.06.2016 13:26, Denis wrote: Re: [OpenSIPS-Users] Fraud module Hello Liviu! Is this action only for "sequential calls"? And what do mean about "different number"? I have prefix "810" in profile table. And client dials different numbers but all of them begin from 810. And i see that count of "sequential calls" increases for each call. mailto:denis7979 at mail.ru Hi Denis, Currently this statistic is not reset between intervals, but only when a user dials a different number than the one(s) before. This makes it so that the module never loses track of the sequence of calls a particular user has made. In your particular case, does it make more sense for the module to do a reset? Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 31.05.2016 15:50, ????? ?????? wrote: Fraud module Hello! I am using Opensips 2.1 with fraud module. Should the module resets current statistics when new time stamp begin? I use 00:00 - 23:59 time interval for fraud detection and see that, for example, "sequential calls" increases for every call every day. Thank you mailto:denis7979 at mail.ru _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Wed Jun 1 13:48:12 2016 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 1 Jun 2016 14:48:12 +0300 Subject: [OpenSIPS-Users] Segfault using Loadbalancer Module. In-Reply-To: References: <61bee8db-047b-93b9-de19-76cfb72b4a91@opensips.org> Message-ID: <574ECB7C.1070408@opensips.org> Thank you for the backtrace, Qasim! It was very helpful. I managed to fix the issue on the "master" and "2.2" git branches [1]. Please pull the latest sources and redo your tests! [1]: https://github.com/OpenSIPS/opensips.git Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 01.06.2016 13:17, qasimakhan at gmail.com wrote: > Dear Razvan, > > Please find below backtrace of the core file: > > http://pastebin.com/WpLtezAS > > Regards, > Qasim Ayyaz Khan > > On Wed, Jun 1, 2016 at 11:46 AM, R?zvan Crainea > wrote: > > Hi, Qasim! > > > Yes, please post a backtrace of the core dump on pastebin for > further investigation. > > Best regards, > > R?zvan Crainea > OpenSIPS Solutions > www.opensips-solutions.com > > On 05/31/2016 10:12 PM, qasimakhan at gmail.com > wrote: >> Hi, >> >> I was using default script generated by opensips menuconfig and >> it gives the following segfault >> >> http://pastebin.com/6zuimn5N >> >> I was evaluating opensips 2.2 latest release. Please let me know >> if core dump is required >> >> Regards, >> Qasim Ayyaz Khan >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From qasimakhan at gmail.com Wed Jun 1 14:35:23 2016 From: qasimakhan at gmail.com (qasimakhan at gmail.com) Date: Wed, 1 Jun 2016 17:35:23 +0500 Subject: [OpenSIPS-Users] Segfault using Loadbalancer Module. In-Reply-To: <574ECB7C.1070408@opensips.org> References: <61bee8db-047b-93b9-de19-76cfb72b4a91@opensips.org> <574ECB7C.1070408@opensips.org> Message-ID: Thank Liviu... Its fixed now. Regards, Qasim On Wed, Jun 1, 2016 at 4:48 PM, Liviu Chircu wrote: > Thank you for the backtrace, Qasim! It was very helpful. > > I managed to fix the issue on the "master" and "2.2" git branches [1]. > Please pull the latest sources and redo your tests! > > [1]: https://github.com/OpenSIPS/opensips.git > > Liviu Chircu > OpenSIPS Developerhttp://www.opensips-solutions.com > > On 01.06.2016 13:17, qasimakhan at gmail.com wrote: > > Dear Razvan, > > Please find below backtrace of the core file: > > http://pastebin.com/WpLtezAS > > Regards, > Qasim Ayyaz Khan > > On Wed, Jun 1, 2016 at 11:46 AM, R?zvan Crainea < > razvan at opensips.org> wrote: > >> Hi, Qasim! >> >> >> Yes, please post a backtrace of the core dump on pastebin for further >> investigation. >> >> Best regards, >> >> R?zvan Crainea >> OpenSIPS Solutionswww.opensips-solutions.com >> >> On 05/31/2016 10:12 PM, qasimakhan at gmail.com wrote: >> >> Hi, >> >> I was using default script generated by opensips menuconfig and it gives >> the following segfault >> >> http://pastebin.com/6zuimn5N >> >> I was evaluating opensips 2.2 latest release. Please let me know if core >> dump is required >> >> Regards, >> Qasim Ayyaz Khan >> >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From saioa.perurena at enigmedia.es Wed Jun 1 15:06:24 2016 From: saioa.perurena at enigmedia.es (Saioa Perurena) Date: Wed, 1 Jun 2016 15:06:24 +0200 Subject: [OpenSIPS-Users] Valid tls connection closed on a "dos" simulation In-Reply-To: <574EC49E.4050800@opensips.org> References: <574EB766.9070704@enigmedia.es> <574EC49E.4050800@opensips.org> Message-ID: <574EDDD0.807@enigmedia.es> Hi, I've tried Opensips 2.1.3 and it seems that the problem is solved. Thank you very much. On 01/06/16 13:18, Liviu Chircu wrote: > Hi Saioa, > > We have addressed this issue somewhere between OpenSIPS 1.11.6 - 1.11.7, > and 2.1.2 - 2.1.3. Please update to the latest version (possibly even > from GitHub [1], [2]), and let us know if it solved your problem! > > [1]: https://github.com/OpenSIPS/opensips/tree/1.11 > [2]: https://github.com/OpenSIPS/opensips/tree/2.1 > > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > > On 01.06.2016 13:22, Saioa Perurena wrote: >> Hi, >> >> We have an opensips 1.11.6 server with tls and we simulate a dos attack >> sending an invalid request to the tls port every 10 seconds (echo >> "giberish" | nc sip-service.example.com 5061 ). >> >> We have only two UAC connected correctly through tls, when one of this >> clients sends an INVITE request when the dos attack is working, then >> servers close the tls connection of that client with error: >> ERROR:proto_tls:tls_print_errstack: TLS errstack: error:1408F10B:SSL >> routines:SSL3_GET_RECORD:wrong version number >> >> When client sends MESSAGE or OPTIONS request it does not happen. >> >> If we stop the dos attack all works correctly. We can reproduce it so >> easily, also with Opensips 2.1 version. >> >> Any idea of what is happening?? Maybe it is a bug on tls? Any suggestion >> or idea is welcome. >> >> Thanks in advance. >> >> Saioa. >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From pimenta at inatel.br Wed Jun 1 20:19:19 2016 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Wed, 1 Jun 2016 18:19:19 +0000 Subject: [OpenSIPS-Users] Does attr_avp (string) decrease the available RAM? Message-ID: Hi. I'm using OpenSIPS 2.1 and the module REGISTRAR. So, in my script I have: modparam("registrar", "attr_avp", "$avp(attr)") ... if (is_method("REGISTER")) { $avp(attr) = "contact_info"; save("location"); exit; } ... lookup("location"); I would like to know whether every time such code is executed the available memory decreases. What happens? Does the avp demand more and more memory to keep its information about lots of "contact_info"? Any hint will be very helpful! Thanks alot. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 -------------- next part -------------- An HTML attachment was scrubbed... URL: From qasimakhan at gmail.com Wed Jun 1 21:21:09 2016 From: qasimakhan at gmail.com (qasimakhan at gmail.com) Date: Thu, 2 Jun 2016 00:21:09 +0500 Subject: [OpenSIPS-Users] Segfault in opensips 2.2 Message-ID: Dear Team, There is another segfault when i try to run with my old configuration from opensips 1.11. as far as i can understand it dosent go beyond loading the modules. Please find below logs and backtrace. syslog: http://pastebin.com/EAqTKu1n backtrace: http://pastebin.com/rP9JDeDW Regards, Qasim -------------- next part -------------- An HTML attachment was scrubbed... URL: From pinghan022 at gmail.com Thu Jun 2 04:13:49 2016 From: pinghan022 at gmail.com (Ping Han) Date: Thu, 2 Jun 2016 12:13:49 +1000 Subject: [OpenSIPS-Users] Question regarding b2b_bridge function In-Reply-To: <574E8B1A.7070508@opensips.org> References: <574D44F7.9090204@opensips.org> <574E8B1A.7070508@opensips.org> Message-ID: Hi Bogdan, Thanks for the information. I need the value in the Opensips cfg. I understand that I can query the b2b_logic or b2b_entities tables to get the value in Opensips config. Apart from that could you tell me other way to easily access the value in Opensips config? Thanks, Ping On Wed, Jun 1, 2016 at 5:13 PM, Bogdan-Andrei Iancu wrote: > Hi Ping, > > Indeed, my bad - the docs are not updated, as that param was disabled long > time ago (4 years ago): > https://sourceforge.net/p/opensips/bugs/502/ > > Still, there are available option. But the question is : do you need that > value in OpenSIPS cfg or outside OpenSIPS ? as there are different way to > get the ID. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 01.06.2016 03:48, Ping Han wrote: > > Hi Bogdan, > > Thank you very much for your reply. > > I have tried to use the module parameter "b2bl_key_avp" as described in > the document as below. > > ------------------------------------ > modparam("b2b_logic", "b2bl_key_avp", "$avp(99)") > ------------------------------------ > > However, I got the following errors when the Opensips is restarted. > ------------------------------------ > ERROR:core:set_mod_param_regex: parameter not found in > module > ------------------------------------ > > I am using the Opensips version opensips-2.1.2-1.el6.x86_64 (rpm). > > Thanks, > Chris > > On Tue, May 31, 2016 at 6:01 PM, Bogdan-Andrei Iancu < > bogdan at opensips.org> wrote: > >> Hi Chris, >> >> The "dialog_id" is actually the b2b key, that is expose by the b2b_logic >> via the module parameter b2bl_key_avp. See: >> >> http://www.opensips.org/html/docs/modules/2.1.x/b2b_logic.html#id294094 >> >> That key can also be found in the b2b_logic table in DB. >> >> At signaling level, the key is the Call-ID of the outbound calls from b2b. >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com >> >> On 23.05.2016 07:32, Ping Han wrote: >> >> Hi Bogdan, >> >> I asked the question a few days ago but have not got a response. >> >> I am just wondering if I could get some advice from you. >> >> Any advice will be appreciated. >> >> Thanks, >> Chris >> >> On Wed, May 18, 2016 at 4:39 PM, Ping Han < >> pinghan022 at gmail.com> wrote: >> >>> Hi, >>> >>> I would like to use the b2b_bridge fifo function as specified at >>> >>> http://www.opensips.org/html/docs/modules/2.1.x/b2b_logic.html#id248916. >>> >>> The function will be triggered by a third party. I will need to pass the >>> parameters to the third party for it to trigger the function. One of the >>> parameters is the "dialog-id". >>> >>> The problem is that I am not sure how the value of the dialog-id can be >>> available in the Opensips config. Is there any Opensips modules/function >>> that can retrieve the value of the dialog-id? >>> >>> I tried to get the value from the "b2b_entities" and "b2b_logic" table. >>> However, it seems that it does not work this way because the two tables do >>> not pop the data in real time. Sometimes I can see the data but sometimes I >>> am not able to see it. >>> >>> It is appreciated that you can give me some idea. >>> >>> Thanks, >>> >>> Ping >>> >> >> >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From sevpal at aol.com Thu Jun 2 05:01:38 2016 From: sevpal at aol.com (sevpal) Date: Wed, 1 Jun 2016 23:01:38 -0400 Subject: [OpenSIPS-Users] Connecting to mongodb Replica set from opensips . In-Reply-To: References: Message-ID: <13D867875F7A463C96C9794315E67762@LenovoPC> Hi, some modules are not safe to use with mongodb, namely; acc (Cannot write to missed_call), msilo (Does not delete stored messages after sent), avpops(do not recall the issue with this one) and there may be others. To connect to the replica set (Minimum 3 replicas for proper master election): All the modules (cachedb_mongodb.so, db_cachedb.so etc.) should point to the same db name in the xdb_url module parameter. loadmodule "cachedb_mongodb.so" loadmodule "db_cachedb.so" modparam("cachedb_mongodb", "cachedb_url","mongodb:replicaset1://root:password at 1.2.3.4:27017,2.3.4.5:27017,5.5.5.5:27017,/jack.db.CallCenter_Info") modparam("db_cachedb","cachedb_url","mongodb:replicaset1://root:password at 1.2.3.4:27017,2.3.4.5:27017,5.5.5.5:27017,/jack.db.CallCenter_Info") loadmodule "usrloc.so" modparam("usrloc", "db_url","mongodb:replicaset1://root:password at 1.2.3.4:27017,2.3.4.5:27017,5.5.5.5:27017,/jack.db.CallCenter_Info") Use the mongo slave_ok,1 parameter to read write via slaves. For the version table, you will need to create a copy in mongodb with the same name as the version table in mysql. Use the db_version_table= global parameter if you want to change the default name of the table. From john.nash778 at gmail.com Thu Jun 2 06:07:12 2016 From: john.nash778 at gmail.com (John Nash) Date: Thu, 2 Jun 2016 09:37:12 +0530 Subject: [OpenSIPS-Users] Rate limit question Message-ID: I am using opensips(2,1) + freeswitch. At opensips doing auth and drouting. Now i plan to test rate limit but should I be checking CPS at opensips or at freeswitch?...as Rate limit uses timers would it be more appropriate to check at freeswitch? -------------- next part -------------- An HTML attachment was scrubbed... URL: From john.nash778 at gmail.com Thu Jun 2 07:49:05 2016 From: john.nash778 at gmail.com (John Nash) Date: Thu, 2 Jun 2016 11:19:05 +0530 Subject: [OpenSIPS-Users] Rate limit question In-Reply-To: References: Message-ID: Also how can I decide which Rate limit algorithm should I choose ? Like RED or TAILDROP or NETWORK On Thu, Jun 2, 2016 at 9:37 AM, John Nash wrote: > I am using opensips(2,1) + freeswitch. At opensips doing auth and > drouting. Now i plan to test rate limit but should I be checking CPS at > opensips or at freeswitch?...as Rate limit uses timers would it be more > appropriate to check at freeswitch? > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Jun 2 09:08:59 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 2 Jun 2016 10:08:59 +0300 Subject: [OpenSIPS-Users] Does attr_avp (string) decrease the available RAM? In-Reply-To: References: Message-ID: <574FDB8B.50708@opensips.org> Hi Rodrigo, The content of the AVP will be stored in memory, along with the registration info. Of course, additional memory will be consumed for that. If the info you attach is 10 bytes long and you have 1000 registrations -> 10KB extra data (not so much after all). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 01.06.2016 21:19, Rodrigo Pimenta Carvalho wrote: > > > Hi. > || > > > I'm using OpenSIPS 2.1 and the module REGISTRAR. > > > So, in my script I have: > > modparam("registrar", "attr_avp", "$avp(attr)") ... if > (is_method("REGISTER")) { $avp(attr) = "contact_info"; > save("location"); exit; } ... lookup("location"); > > > I would like to know whether every time such code is executed > the available memory decreases. What happens? Does the avp > demand more and more memory to keep its information about lots > of "contact_info"? > > > |Any hint will be very helpful!| > > > |Thanks alot.| > > > RODRIGO PIMENTA CARVALHO > Inatel Competence Center > Software > Ph: +55 35 3471 9200 RAMAL 979 > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Jun 2 09:25:17 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 2 Jun 2016 10:25:17 +0300 Subject: [OpenSIPS-Users] Segfault in opensips 2.2 In-Reply-To: References: Message-ID: <574FDF5D.3040000@opensips.org> Hi Qasim, Thank you for your report. In your opensips cfg, do you have any save() or lookup() ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 01.06.2016 22:21, qasimakhan at gmail.com wrote: > Dear Team, > > There is another segfault when i try to run with my old configuration > from opensips 1.11. as far as i can understand it dosent go beyond > loading the modules. Please find below logs and backtrace. > > syslog: http://pastebin.com/EAqTKu1n > backtrace: http://pastebin.com/rP9JDeDW > > Regards, > Qasim > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Jun 2 09:35:28 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 2 Jun 2016 10:35:28 +0300 Subject: [OpenSIPS-Users] Question regarding b2b_bridge function In-Reply-To: References: <574D44F7.9090204@opensips.org> <574E8B1A.7070508@opensips.org> Message-ID: <574FE1C0.9000904@opensips.org> Hi Ping, In script, in a b2b route, you can look at the callid or TO tag (depending on the direction) to get the key : http://www.opensips.org/html/docs/modules/2.2.x/b2b_logic.html#id294094 The B2B uses that key as Call-ID when acting as UAC and as To tag when acting as UAS. You can run a SIP capture to see the traffic. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 02.06.2016 05:13, Ping Han wrote: > Hi Bogdan, > > Thanks for the information. > > I need the value in the Opensips cfg. > > I understand that I can query the b2b_logic or b2b_entities tables to > get the value in Opensips config. Apart from that could you tell me > other way to easily access the value in Opensips config? > > Thanks, > Ping > > > > On Wed, Jun 1, 2016 at 5:13 PM, Bogdan-Andrei Iancu > > wrote: > > Hi Ping, > > Indeed, my bad - the docs are not updated, as that param was > disabled long time ago (4 years ago): > https://sourceforge.net/p/opensips/bugs/502/ > > Still, there are available option. But the question is : do you > need that value in OpenSIPS cfg or outside OpenSIPS ? as there are > different way to get the ID. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 01.06.2016 03:48, Ping Han wrote: >> Hi Bogdan, >> >> Thank you very much for your reply. >> >> I have tried to use the module parameter "b2bl_key_avp" as >> described in the document as below. >> >> ------------------------------------ >> modparam("b2b_logic", "b2bl_key_avp", "$avp(99)") >> ------------------------------------ >> >> However, I got the following errors when the Opensips is restarted. >> ------------------------------------ >> ERROR:core:set_mod_param_regex: parameter not >> found in module >> ------------------------------------ >> >> I am using the Opensips version opensips-2.1.2-1.el6.x86_64 (rpm). >> >> Thanks, >> Chris >> >> On Tue, May 31, 2016 at 6:01 PM, Bogdan-Andrei Iancu >> > wrote: >> >> Hi Chris, >> >> The "dialog_id" is actually the b2b key, that is expose by >> the b2b_logic via the module parameter b2bl_key_avp. See: >> http://www.opensips.org/html/docs/modules/2.1.x/b2b_logic.html#id294094 >> >> That key can also be found in the b2b_logic table in DB. >> >> At signaling level, the key is the Call-ID of the outbound >> calls from b2b. >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> On 23.05.2016 07:32, Ping Han wrote: >>> Hi Bogdan, >>> >>> I asked the question a few days ago but have not got a >>> response. >>> >>> I am just wondering if I could get some advice from you. >>> >>> Any advice will be appreciated. >>> >>> Thanks, >>> Chris >>> >>> On Wed, May 18, 2016 at 4:39 PM, Ping Han >>> > wrote: >>> >>> Hi, >>> >>> I would like to use the b2b_bridge fifo function as >>> specified at >>> http://www.opensips.org/html/docs/modules/2.1.x/b2b_logic.html#id248916. >>> >>> The function will be triggered by a third party. I will >>> need to pass the parameters to the third party for it to >>> trigger the function. One of the parameters is the >>> "dialog-id". >>> >>> The problem is that I am not sure how the value of the >>> dialog-id can be available in the Opensips config. Is >>> there any Opensips modules/function that can retrieve >>> the value of the dialog-id? >>> >>> I tried to get the value from the "b2b_entities" and >>> "b2b_logic" table. However, it seems that it does not >>> work this way because the two tables do not pop the data >>> in real time. Sometimes I can see the data but sometimes >>> I am not able to see it. >>> >>> It is appreciated that you can give me some idea. >>> >>> Thanks, >>> >>> Ping >>> >>> >> >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Thu Jun 2 10:18:21 2016 From: liviu at opensips.org (Liviu Chircu) Date: Thu, 2 Jun 2016 11:18:21 +0300 Subject: [OpenSIPS-Users] Does attr_avp (string) decrease the available RAM? In-Reply-To: References: Message-ID: <574FEBCD.7000003@opensips.org> AVPs eat up shared memory. For each "lookup()" you perform on a different INVITE request, a different $avp(attr) will be populated with the attributes of each contact behind the AoR you looked up. All AVPs of a transaction are then freed along with the transaction itself. Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 01.06.2016 21:19, Rodrigo Pimenta Carvalho wrote: > > > Hi. > || > > > I'm using OpenSIPS 2.1 and the module REGISTRAR. > > > So, in my script I have: > > modparam("registrar", "attr_avp", "$avp(attr)") ... if > (is_method("REGISTER")) { $avp(attr) = "contact_info"; > save("location"); exit; } ... lookup("location"); > > > I would like to know whether every time such code is executed > the available memory decreases. What happens? Does the avp > demand more and more memory to keep its information about lots > of "contact_info"? > > > |Any hint will be very helpful!| > > > |Thanks alot.| > > > RODRIGO PIMENTA CARVALHO > Inatel Competence Center > Software > Ph: +55 35 3471 9200 RAMAL 979 > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Thu Jun 2 14:23:19 2016 From: spanda at 3clogic.com (Sasmita Panda) Date: Thu, 2 Jun 2016 17:53:19 +0530 Subject: [OpenSIPS-Users] mongodb replica set connection . Message-ID: Hi All , I am using opensips-1.11 and I am using cachedb_mongodb module . I am able to connect to mongodb replica set without authentication . modparam("cachedb_mongodb", "cachedb_url","mongodb:replicaset1:// 1.2.3.4:27017 , 2.3.4.5:27017/jack.db.CallCenter_Info") modparam("db_cachedb","cachedb_url","mongodb:replicaset1://1.2.3.4:27017 ,2.3.4.5:27017/jack.db. db") modparam("acc", "db_url", "cachedb://mongodb:replicaset1") I am able to connect to mongodb for account module . By this line . But I am not able to store any data through raw query like bellow . cache_store("mongodb:group1","$ci","$ci,$var(c)"); Its giving bellow error . DBG:core:cachedb_store: from script [mongodb] - with grp [group1] ERROR:core:cachedb_store: failed to get connection for grp name [group1] What does it mean ? If I am changing the group1 to replicaset1 then also its giving error . What should I do ? Please help me . Am I doing something wrong in the opensips config file or I need to do something in the mongodb replica set ? Thanks in advance . any kind of suggestion is welcome . *Thanks & Regards* *Sasmita Panda* *Network Testing and Software Engineer* *3CLogic , ph:07827611765* -------------- next part -------------- An HTML attachment was scrubbed... URL: From qasimakhan at gmail.com Thu Jun 2 18:23:11 2016 From: qasimakhan at gmail.com (qasimakhan at gmail.com) Date: Thu, 2 Jun 2016 21:23:11 +0500 Subject: [OpenSIPS-Users] Segfault in opensips 2.2 In-Reply-To: <574FDF5D.3040000@opensips.org> References: <574FDF5D.3040000@opensips.org> Message-ID: Hi, No we are not using any save() or lookup() in our code. Regards, Qasim On Thu, Jun 2, 2016 at 12:25 PM, Bogdan-Andrei Iancu wrote: > Hi Qasim, > > Thank you for your report. > > In your opensips cfg, do you have any save() or lookup() ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 01.06.2016 22:21, qasimakhan at gmail.com wrote: > > Dear Team, > > There is another segfault when i try to run with my old configuration from > opensips 1.11. as far as i can understand it dosent go beyond loading the > modules. Please find below logs and backtrace. > > syslog: http://pastebin.com/EAqTKu1n > backtrace: http://pastebin.com/rP9JDeDW > > Regards, > Qasim > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From pinghan022 at gmail.com Fri Jun 3 05:27:27 2016 From: pinghan022 at gmail.com (Ping Han) Date: Fri, 3 Jun 2016 13:27:27 +1000 Subject: [OpenSIPS-Users] Question regarding b2b_bridge function In-Reply-To: <574FE1C0.9000904@opensips.org> References: <574D44F7.9090204@opensips.org> <574E8B1A.7070508@opensips.org> <574FE1C0.9000904@opensips.org> Message-ID: Thanks, Bogdan, It seems the function you mentioned is the internal function "1.4.2 b2b_bridge_request(b2bl_key,entity_no)". Actually the function that I am trying to use is the "b2b_bridge" (Exported MI Functions). It is defined as below ---------------------------------- http://www.opensips.org/html/docs/modules/2.1.x/b2b_logic.html#id294210 1.5.2. b2b_bridge Example: opensipsctl b2b_bridge 1020.30 sip:alice at opensips.org ---------------------------------- In the example "1020.30" is the "dialog-id". This is the parameter that I am not sure how to easily access in the Opensips config. What I am trying to do is to get the value and deliver to the next hop via a custom SIP header. When the next hop tries to transfer the call to a new destination. It can run the b2b_bridge command straight away with the "dialog-id" without rechieving the value from the Opensips database (from b2b_logic or b2b_entities tables). Any advice will be appreciated. Thanks, Ping On Thu, Jun 2, 2016 at 5:35 PM, Bogdan-Andrei Iancu wrote: > Hi Ping, > > In script, in a b2b route, you can look at the callid or TO tag (depending > on the direction) to get the key : > > http://www.opensips.org/html/docs/modules/2.2.x/b2b_logic.html#id294094 > > The B2B uses that key as Call-ID when acting as UAC and as To tag when > acting as UAS. You can run a SIP capture to see the traffic. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 02.06.2016 05:13, Ping Han wrote: > > Hi Bogdan, > > Thanks for the information. > > I need the value in the Opensips cfg. > > I understand that I can query the b2b_logic or b2b_entities tables to get > the value in Opensips config. Apart from that could you tell me other way > to easily access the value in Opensips config? > > Thanks, > Ping > > > > On Wed, Jun 1, 2016 at 5:13 PM, Bogdan-Andrei Iancu < > bogdan at opensips.org> wrote: > >> Hi Ping, >> >> Indeed, my bad - the docs are not updated, as that param was disabled >> long time ago (4 years ago): >> https://sourceforge.net/p/opensips/bugs/502/ >> >> Still, there are available option. But the question is : do you need that >> value in OpenSIPS cfg or outside OpenSIPS ? as there are different way to >> get the ID. >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com >> >> On 01.06.2016 03:48, Ping Han wrote: >> >> Hi Bogdan, >> >> Thank you very much for your reply. >> >> I have tried to use the module parameter "b2bl_key_avp" as described in >> the document as below. >> >> ------------------------------------ >> modparam("b2b_logic", "b2bl_key_avp", "$avp(99)") >> ------------------------------------ >> >> However, I got the following errors when the Opensips is restarted. >> ------------------------------------ >> ERROR:core:set_mod_param_regex: parameter not found in >> module >> ------------------------------------ >> >> I am using the Opensips version opensips-2.1.2-1.el6.x86_64 (rpm). >> >> Thanks, >> Chris >> >> On Tue, May 31, 2016 at 6:01 PM, Bogdan-Andrei Iancu < >> bogdan at opensips.org> wrote: >> >>> Hi Chris, >>> >>> The "dialog_id" is actually the b2b key, that is expose by the b2b_logic >>> via the module parameter b2bl_key_avp. See: >>> >>> http://www.opensips.org/html/docs/modules/2.1.x/b2b_logic.html#id294094 >>> >>> That key can also be found in the b2b_logic table in DB. >>> >>> At signaling level, the key is the Call-ID of the outbound calls from >>> b2b. >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com >>> >>> On 23.05.2016 07:32, Ping Han wrote: >>> >>> Hi Bogdan, >>> >>> I asked the question a few days ago but have not got a response. >>> >>> I am just wondering if I could get some advice from you. >>> >>> Any advice will be appreciated. >>> >>> Thanks, >>> Chris >>> >>> On Wed, May 18, 2016 at 4:39 PM, Ping Han < >>> pinghan022 at gmail.com> wrote: >>> >>>> Hi, >>>> >>>> I would like to use the b2b_bridge fifo function as specified at >>>> >>>> http://www.opensips.org/html/docs/modules/2.1.x/b2b_logic.html#id248916 >>>> . >>>> >>>> The function will be triggered by a third party. I will need to pass >>>> the parameters to the third party for it to trigger the function. One of >>>> the parameters is the "dialog-id". >>>> >>>> The problem is that I am not sure how the value of the dialog-id can be >>>> available in the Opensips config. Is there any Opensips modules/function >>>> that can retrieve the value of the dialog-id? >>>> >>>> I tried to get the value from the "b2b_entities" and "b2b_logic" table. >>>> However, it seems that it does not work this way because the two tables do >>>> not pop the data in real time. Sometimes I can see the data but sometimes I >>>> am not able to see it. >>>> >>>> It is appreciated that you can give me some idea. >>>> >>>> Thanks, >>>> >>>> Ping >>>> >>> >>> >>> >> >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Fri Jun 3 09:17:33 2016 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Fri, 3 Jun 2016 10:17:33 +0300 Subject: [OpenSIPS-Users] Rate limit question In-Reply-To: References: Message-ID: Hi, John! It depends on the logic of your platform. If you have multiple FreeSWITCH servers behind OpenSIPS, and you want to have a "global" limit, then you must do it on OpenSIPS side. It is also recommended to do it there because for example if the threashold is reached, you don't have to load FreeSWITCH with traffic that's about to be denied. However, if you have local (per FreeSWITCH) limits, then you can do it on the FreeSWITCH side (with the earlier observations). Regarding the algorithm, it depends on what you want to limit. If you want to do CPS, then TAILDROP or RED is more suitable. If you want to do a limitation based on the network load, then use NETWORK. Also take into account that the way RED and TAILDROP algorithm work, they are not so accurate. That's why in OpenSIPS 2.2 we added a new algorithm, SBT[1], which is very accurate and custamizable. [1] http://www.opensips.org/html/docs/modules/2.2.x/ratelimit.html#id293435 Best regards, R?zvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 06/02/2016 08:49 AM, John Nash wrote: > Also how can I decide which Rate limit algorithm should I choose ? > Like RED or TAILDROP or NETWORK > > On Thu, Jun 2, 2016 at 9:37 AM, John Nash > wrote: > > I am using opensips(2,1) + freeswitch. At opensips doing auth and > drouting. Now i plan to test rate limit but should I be checking > CPS at opensips or at freeswitch?...as Rate limit uses timers > would it be more appropriate to check at freeswitch? > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Fri Jun 3 09:26:35 2016 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Fri, 3 Jun 2016 10:26:35 +0300 Subject: [OpenSIPS-Users] mongodb replica set connection . In-Reply-To: References: Message-ID: <90b0c601-fd77-f4b5-a498-977f69959513@opensips.org> Hi, Sasmita! Are these two the only errors you see in the logs? Can you confirm that replicaset "jack" exists and that the "CallCenter_Info" table exists? What about the "db" collection? Have you tried to set the cachedb_url like this: modparam("db_cachedb","cachedb_url","mongodb:replicaset1://1.2.3.4:27017 /jack.db ") Best regards, R?zvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 06/02/2016 03:23 PM, Sasmita Panda wrote: > Hi All , > > I am using opensips-1.11 and I am using cachedb_mongodb module > . I am able to connect to mongodb replica set without authentication . > > modparam("cachedb_mongodb", > "cachedb_url","mongodb:replicaset1://1.2.3.4:27017 > ,2.3.4.5:27017/jack.db.CallCenter_Info > ") > > modparam("db_cachedb","cachedb_url","mongodb:replicaset1://1.2.3.4:27017 > ,2.3.4.5:27017/jack.db. > db") > > modparam("acc", "db_url", "cachedb://mongodb:replicaset1") > I am able to connect to mongodb for account module . By this line . > > But I am not able to store any data through raw query like bellow . > cache_store("mongodb:group1","$ci","$ci,$var(c)"); > > Its giving bellow error . > > DBG:core:cachedb_store: from script [mongodb] - with grp [group1] > ERROR:core:cachedb_store: failed to get connection for grp name [group1] > > > What does it mean ? If I am changing the group1 to replicaset1 then > also its giving error . What should I do ? Please help me . Am I doing > something wrong in the opensips config file or I need to do something > in the mongodb replica set ? > > Thanks in advance . any kind of suggestion is welcome . > > */Thanks & Regards/* > /Sasmita Panda/ > /Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Jun 3 10:54:33 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 3 Jun 2016 11:54:33 +0300 Subject: [OpenSIPS-Users] Question regarding b2b_bridge function In-Reply-To: References: <574D44F7.9090204@opensips.org> <574E8B1A.7070508@opensips.org> <574FE1C0.9000904@opensips.org> Message-ID: <575145C9.9080406@opensips.org> Hi Ping, b2b_bridge_request() is a script function: http://www.opensips.org/html/docs/modules/2.1.x/b2b_logic.html#id294094 I pointed to this function as you mentioned (on my question) that you want to do the bridging from script level. Indeed, the equivalent MI function is b2b_bridge: http://www.opensips.org/html/docs/modules/2.1.x/b2b_logic.html#id294210 If you want to get that "1020.30", you can get it from Call-ID or To tag, where you have B2B.1020.30 (so you have to strip that B2B prefix). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 03.06.2016 06:27, Ping Han wrote: > Thanks, Bogdan, > > It seems the function you mentioned is the internal function "1.4.2 > b2b_bridge_request(b2bl_key,entity_no)". > > Actually the function that I am trying to use is the "b2b_bridge" > (Exported MI Functions). It is defined as below > ---------------------------------- > http://www.opensips.org/html/docs/modules/2.1.x/b2b_logic.html#id294210 > > > 1.5.2. b2b_bridge > > Example: opensipsctl b2b_bridge 1020.30 sip:alice at opensips.org > > ---------------------------------- > > In the example "1020.30" is the "dialog-id". This is the parameter > that I am not sure how to easily access in the Opensips config. > > What I am trying to do is to get the value and deliver to the next hop > via a custom SIP header. When the next hop tries to transfer the call > to a new destination. It can run the b2b_bridge command straight away > with the "dialog-id" without rechieving the value from the Opensips > database (from b2b_logic or b2b_entities tables). > > Any advice will be appreciated. > > Thanks, > Ping > > > On Thu, Jun 2, 2016 at 5:35 PM, Bogdan-Andrei Iancu > > wrote: > > Hi Ping, > > In script, in a b2b route, you can look at the callid or TO tag > (depending on the direction) to get the key : > http://www.opensips.org/html/docs/modules/2.2.x/b2b_logic.html#id294094 > > The B2B uses that key as Call-ID when acting as UAC and as To tag > when acting as UAS. You can run a SIP capture to see the traffic. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 02.06.2016 05:13, Ping Han wrote: >> Hi Bogdan, >> >> Thanks for the information. >> >> I need the value in the Opensips cfg. >> >> I understand that I can query the b2b_logic or b2b_entities >> tables to get the value in Opensips config. Apart from that could >> you tell me other way to easily access the value in Opensips config? >> >> Thanks, >> Ping >> >> >> >> On Wed, Jun 1, 2016 at 5:13 PM, Bogdan-Andrei Iancu >> > wrote: >> >> Hi Ping, >> >> Indeed, my bad - the docs are not updated, as that param was >> disabled long time ago (4 years ago): >> https://sourceforge.net/p/opensips/bugs/502/ >> >> Still, there are available option. But the question is : do >> you need that value in OpenSIPS cfg or outside OpenSIPS ? as >> there are different way to get the ID. >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> On 01.06.2016 03:48, Ping Han wrote: >>> Hi Bogdan, >>> >>> Thank you very much for your reply. >>> >>> I have tried to use the module parameter "b2bl_key_avp" as >>> described in the document as below. >>> >>> ------------------------------------ >>> modparam("b2b_logic", "b2bl_key_avp", "$avp(99)") >>> ------------------------------------ >>> >>> However, I got the following errors when the Opensips is >>> restarted. >>> ------------------------------------ >>> ERROR:core:set_mod_param_regex: parameter not >>> found in module >>> ------------------------------------ >>> >>> I am using the Opensips version opensips-2.1.2-1.el6.x86_64 >>> (rpm). >>> >>> Thanks, >>> Chris >>> >>> On Tue, May 31, 2016 at 6:01 PM, Bogdan-Andrei Iancu >>> > wrote: >>> >>> Hi Chris, >>> >>> The "dialog_id" is actually the b2b key, that is expose >>> by the b2b_logic via the module parameter b2bl_key_avp. See: >>> http://www.opensips.org/html/docs/modules/2.1.x/b2b_logic.html#id294094 >>> >>> That key can also be found in the b2b_logic table in DB. >>> >>> At signaling level, the key is the Call-ID of the >>> outbound calls from b2b. >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> OpenSIPS Founder and Developer >>> http://www.opensips-solutions.com >>> >>> On 23.05.2016 07:32, Ping Han wrote: >>>> Hi Bogdan, >>>> >>>> I asked the question a few days ago but have not got a >>>> response. >>>> >>>> I am just wondering if I could get some advice from you. >>>> >>>> Any advice will be appreciated. >>>> >>>> Thanks, >>>> Chris >>>> >>>> On Wed, May 18, 2016 at 4:39 PM, Ping Han >>>> > wrote: >>>> >>>> Hi, >>>> >>>> I would like to use the b2b_bridge fifo function as >>>> specified at >>>> http://www.opensips.org/html/docs/modules/2.1.x/b2b_logic.html#id248916. >>>> >>>> The function will be triggered by a third party. I >>>> will need to pass the parameters to the third party >>>> for it to trigger the function. One of the >>>> parameters is the "dialog-id". >>>> >>>> The problem is that I am not sure how the value of >>>> the dialog-id can be available in the Opensips >>>> config. Is there any Opensips modules/function that >>>> can retrieve the value of the dialog-id? >>>> >>>> I tried to get the value from the "b2b_entities" >>>> and "b2b_logic" table. However, it seems that it >>>> does not work this way because the two tables do >>>> not pop the data in real time. Sometimes I can see >>>> the data but sometimes I am not able to see it. >>>> >>>> It is appreciated that you can give me some idea. >>>> >>>> Thanks, >>>> >>>> Ping >>>> >>>> >>> >>> >> >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Fri Jun 3 11:01:52 2016 From: spanda at 3clogic.com (Sasmita Panda) Date: Fri, 3 Jun 2016 14:31:52 +0530 Subject: [OpenSIPS-Users] mongodb replica set connection . In-Reply-To: <90b0c601-fd77-f4b5-a498-977f69959513@opensips.org> References: <90b0c601-fd77-f4b5-a498-977f69959513@opensips.org> Message-ID: I have tried this . But its not working . Now I got something interesting . I changed the cache_store command and now its working . This is " replicasetname.db.collection " this is the format what I get from opensips documentation . modparam("cachedb_mongodb", "cachedb_url","mongodb:replicaset1:// 1.2.3.4:27017 , 2.3.4.5:27017/jack.db.CallCenter_Info") cache_store("mongodb:replicaset1","$ci","$ci,$var(c)"); What opensips is doing is . Its creating a database jack and inside that a collection db.CallCenter_Info and storing data on that collections . But I have created a database db and i have also created a collection in that CallCenter_Info and I was expecting the data will goes to this collection . I am doing anything wrong ? I am getting the data so I think my problem is almost solved . But my acc and sip_trace collection and CallCenter_Info collection are in different data base . I wanted them to present inside a single data base so that it will be easier for me which searching something . Let me know if I am doing anything wrong . One problem i am facing is , I am not able to connect to the replica set with authentication . Without authentication its working like above . *Thanks & Regards* *Sasmita Panda* *Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Fri, Jun 3, 2016 at 12:56 PM, R?zvan Crainea wrote: > Hi, Sasmita! > > Are these two the only errors you see in the logs? > > Can you confirm that replicaset "jack" exists and that the > "CallCenter_Info" table exists? What about the "db" collection? Have you > tried to set the cachedb_url like this: > modparam("db_cachedb","cachedb_url","mongodb:replicaset1://1.2.3.4:27017 > /jack.db > ") > > Best regards, > > R?zvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 06/02/2016 03:23 PM, Sasmita Panda wrote: > > Hi All , > > I am using opensips-1.11 and I am using cachedb_mongodb module . I > am able to connect to mongodb replica set without authentication . > > modparam("cachedb_mongodb", "cachedb_url","mongodb:replicaset1:// > 1.2.3.4:27017 , > 2.3.4.5:27017/jack.db.CallCenter_Info") > > modparam("db_cachedb","cachedb_url","mongodb:replicaset1://1.2.3.4:27017 > ,2.3.4.5:27017/jack.db. > db") > > modparam("acc", "db_url", "cachedb://mongodb:replicaset1") > I am able to connect to mongodb for account module . By this line . > > But I am not able to store any data through raw query like bellow . > cache_store("mongodb:group1","$ci","$ci,$var(c)"); > > Its giving bellow error . > > DBG:core:cachedb_store: from script [mongodb] - with grp [group1] > ERROR:core:cachedb_store: failed to get connection for grp name [group1] > > > What does it mean ? If I am changing the group1 to replicaset1 then also > its giving error . What should I do ? Please help me . Am I doing something > wrong in the opensips config file or I need to do something in the mongodb > replica set ? > > Thanks in advance . any kind of suggestion is welcome . > > *Thanks & Regards* > *Sasmita Panda* > *Network Testing and Software Engineer* > *3CLogic , ph:07827611765* > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From igor.pavlov1987 at gmail.com Fri Jun 3 11:02:28 2016 From: igor.pavlov1987 at gmail.com (Igor Pavlov) Date: Fri, 3 Jun 2016 13:02:28 +0400 Subject: [OpenSIPS-Users] opensips 1.11.7 tcp_send failed Message-ID: <38329c63-8d88-0927-5717-366f5630a3fd@gmail.com> Hi, all. I have a problem with TCP in opensips 1.11.7. All users are registered with tcp proto. Call flow: userA -> (tcp) opensips -> (udp) freeswitch -> (udp) opensips -> (tcp) userB. When call is coming from FS I get error: > Jun 3 11:53:57 ERROR:tm:msg_send: tcp_send failed > Jun 3 11:53:57 ERROR:tm:t_forward_nonack: sending request failed opensips.cfg contain tcp_no_new_conn_bflag = 6 and I'm setting this flag when INVITE is coming from FS. ____________ Best regards, Igor Pavlov From john.nash778 at gmail.com Fri Jun 3 11:15:02 2016 From: john.nash778 at gmail.com (John Nash) Date: Fri, 3 Jun 2016 14:45:02 +0530 Subject: [OpenSIPS-Users] Rate limit question In-Reply-To: References: Message-ID: Thank you for detailed reply. I will choose to implement on Opensips. I would also prefer performance over accuracy as my objective is to stop customers from flooding with too many calls so I can select RED? On Fri, Jun 3, 2016 at 12:47 PM, R?zvan Crainea wrote: > Hi, John! > > It depends on the logic of your platform. If you have multiple FreeSWITCH > servers behind OpenSIPS, and you want to have a "global" limit, then you > must do it on OpenSIPS side. It is also recommended to do it there because > for example if the threashold is reached, you don't have to load FreeSWITCH > with traffic that's about to be denied. However, if you have local (per > FreeSWITCH) limits, then you can do it on the FreeSWITCH side (with the > earlier observations). > > Regarding the algorithm, it depends on what you want to limit. If you want > to do CPS, then TAILDROP or RED is more suitable. If you want to do a > limitation based on the network load, then use NETWORK. > > Also take into account that the way RED and TAILDROP algorithm work, they > are not so accurate. That's why in OpenSIPS 2.2 we added a new algorithm, > SBT[1], which is very accurate and custamizable. > > [1] > http://www.opensips.org/html/docs/modules/2.2.x/ratelimit.html#id293435 > > Best regards, > > R?zvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 06/02/2016 08:49 AM, John Nash wrote: > > Also how can I decide which Rate limit algorithm should I choose ? Like > RED or TAILDROP or NETWORK > > On Thu, Jun 2, 2016 at 9:37 AM, John Nash wrote: > >> I am using opensips(2,1) + freeswitch. At opensips doing auth and >> drouting. Now i plan to test rate limit but should I be checking CPS at >> opensips or at freeswitch?...as Rate limit uses timers would it be more >> appropriate to check at freeswitch? >> >> >> > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Fri Jun 3 11:22:21 2016 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Fri, 3 Jun 2016 12:22:21 +0300 Subject: [OpenSIPS-Users] Rate limit question In-Reply-To: References: Message-ID: Yes, RED seems to be what you need. Best regards, R?zvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 06/03/2016 12:15 PM, John Nash wrote: > Thank you for detailed reply. I will choose to implement on Opensips. > I would also prefer performance over accuracy as my objective is to > stop customers from flooding with too many calls so I can select RED? > > > > > > On Fri, Jun 3, 2016 at 12:47 PM, R?zvan Crainea > wrote: > > Hi, John! > > It depends on the logic of your platform. If you have multiple > FreeSWITCH servers behind OpenSIPS, and you want to have a > "global" limit, then you must do it on OpenSIPS side. It is also > recommended to do it there because for example if the threashold > is reached, you don't have to load FreeSWITCH with traffic that's > about to be denied. However, if you have local (per FreeSWITCH) > limits, then you can do it on the FreeSWITCH side (with the > earlier observations). > > Regarding the algorithm, it depends on what you want to limit. If > you want to do CPS, then TAILDROP or RED is more suitable. If you > want to do a limitation based on the network load, then use NETWORK. > > Also take into account that the way RED and TAILDROP algorithm > work, they are not so accurate. That's why in OpenSIPS 2.2 we > added a new algorithm, SBT[1], which is very accurate and > custamizable. > > [1] > http://www.opensips.org/html/docs/modules/2.2.x/ratelimit.html#id293435 > > Best regards, > > R?zvan Crainea > OpenSIPS Solutions > www.opensips-solutions.com > > On 06/02/2016 08:49 AM, John Nash wrote: >> Also how can I decide which Rate limit algorithm should I choose >> ? Like RED or TAILDROP or NETWORK >> >> On Thu, Jun 2, 2016 at 9:37 AM, John Nash > > wrote: >> >> I am using opensips(2,1) + freeswitch. At opensips doing auth >> and drouting. Now i plan to test rate limit but should I be >> checking CPS at opensips or at freeswitch?...as Rate limit >> uses timers would it be more appropriate to check at freeswitch? >> >> >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Jun 3 11:42:29 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 3 Jun 2016 12:42:29 +0300 Subject: [OpenSIPS-Users] Segfault in opensips 2.2 In-Reply-To: References: <574FDF5D.3040000@opensips.org> Message-ID: <57515105.8050207@opensips.org> Hi Qasim, Yes, you came across an unforeseen case, where you enable pinging via nathelper, but you do not use any usrloc domains. This is not specific to 2.2, it is in all versions. I pushed a fix (trunk, 2.2, 2.1, 1.11) on GIT - please update and try again. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 02.06.2016 19:23, qasimakhan at gmail.com wrote: > Hi, > > No we are not using any save() or lookup() in our code. > > Regards, > Qasim > > On Thu, Jun 2, 2016 at 12:25 PM, Bogdan-Andrei Iancu > > wrote: > > Hi Qasim, > > Thank you for your report. > > In your opensips cfg, do you have any save() or lookup() ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 01.06.2016 22:21, qasimakhan at gmail.com > wrote: >> Dear Team, >> >> There is another segfault when i try to run with my old >> configuration from opensips 1.11. as far as i can understand it >> dosent go beyond loading the modules. Please find below logs and >> backtrace. >> >> syslog: http://pastebin.com/EAqTKu1n >> backtrace: http://pastebin.com/rP9JDeDW >> >> Regards, >> Qasim >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Fri Jun 3 12:03:21 2016 From: liviu at opensips.org (Liviu Chircu) Date: Fri, 3 Jun 2016 13:03:21 +0300 Subject: [OpenSIPS-Users] opensips 1.11.7 tcp_send failed In-Reply-To: <38329c63-8d88-0927-5717-366f5630a3fd@gmail.com> References: <38329c63-8d88-0927-5717-366f5630a3fd@gmail.com> Message-ID: <575155E9.4060507@opensips.org> Hi Igor, Are you sure you are using "setbflag(6)" and not "setflag(6)"? It's a very important detail, since doing a "setflag(6)" would be completely irrelevant, and useless. Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 03.06.2016 12:02, Igor Pavlov wrote: > Hi, all. I have a problem with TCP in opensips 1.11.7. All users are > registered with tcp proto. Call flow: userA -> (tcp) opensips -> (udp) > freeswitch -> (udp) opensips -> (tcp) userB. > > When call is coming from FS I get error: > >> Jun 3 11:53:57 ERROR:tm:msg_send: tcp_send failed >> Jun 3 11:53:57 ERROR:tm:t_forward_nonack: sending request failed > opensips.cfg contain tcp_no_new_conn_bflag = 6 and I'm setting this > flag when INVITE is coming from FS. > > > ____________ > Best regards, > Igor Pavlov > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From razvan at opensips.org Fri Jun 3 12:20:14 2016 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Fri, 3 Jun 2016 13:20:14 +0300 Subject: [OpenSIPS-Users] mongodb replica set connection . In-Reply-To: References: <90b0c601-fd77-f4b5-a498-977f69959513@opensips.org> Message-ID: <91eaaddd-2d8d-9801-b7b0-acfb08971ff4@opensips.org> Hi, Sasmita! Are you using multiple IPs in the replica set, or only one? Because if you are using only one, the replica set feature is not enabled, and the "jack.db.CallCenter_Info" is treated as database "jack" and collection "db.CallCenter_Info", as in your case. Is this your problem? Are you setting multiple IPs in the cachedb_url? Best regards, R?zvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 06/03/2016 12:01 PM, Sasmita Panda wrote: > I have tried this . But its not working . > > Now I got something interesting . I changed the cache_store command > and now its working . > This is " replicasetname.db.collection " this is the format what I get > from opensips documentation . > > modparam("cachedb_mongodb", > "cachedb_url","mongodb:replicaset1://1.2.3.4:27017 > ,2.3.4.5:27017/jack.db.CallCenter_Info > ") > cache_store("mongodb:replicaset1","$ci","$ci,$var(c)"); > What opensips is doing is . Its creating a database jack and > inside that a collection db.CallCenter_Info and storing data on that > collections . > > But I have created a database db and i have also created a > collection in that CallCenter_Info and I was expecting the data will > goes to this collection . > > I am doing anything wrong ? I am getting the data so I think my > problem is almost solved . But my acc and sip_trace collection and > CallCenter_Info collection are in different data base . I wanted them > to present inside a single data base so that it will be easier for me > which searching something . > > Let me know if I am doing anything wrong . > > One problem i am facing is , I am not able to connect to the > replica set with authentication . Without authentication its working > like above . > > > */Thanks & Regards/* > /Sasmita Panda/ > /Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > On Fri, Jun 3, 2016 at 12:56 PM, R?zvan Crainea > wrote: > > Hi, Sasmita! > > Are these two the only errors you see in the logs? > > Can you confirm that replicaset "jack" exists and that the > "CallCenter_Info" table exists? What about the "db" collection? > Have you tried to set the cachedb_url like this: > > modparam("db_cachedb","cachedb_url","mongodb:replicaset1://1.2.3.4:27017 > /jack.db > ") > > Best regards, > > R?zvan Crainea > OpenSIPS Solutions > www.opensips-solutions.com > > On 06/02/2016 03:23 PM, Sasmita Panda wrote: >> Hi All , >> >> I am using opensips-1.11 and I am using cachedb_mongodb >> module . I am able to connect to mongodb replica set without >> authentication . >> >> modparam("cachedb_mongodb", >> "cachedb_url","mongodb:replicaset1://1.2.3.4:27017 >> ,2.3.4.5:27017/jack.db.CallCenter_Info >> ") >> >> modparam("db_cachedb","cachedb_url","mongodb:replicaset1://1.2.3.4:27017 >> ,2.3.4.5:27017/jack.db. >> db") >> >> modparam("acc", "db_url", "cachedb://mongodb:replicaset1") >> I am able to connect to mongodb for account module . By this line . >> >> But I am not able to store any data through raw query like bellow . >> cache_store("mongodb:group1","$ci","$ci,$var(c)"); >> >> Its giving bellow error . >> >> DBG:core:cachedb_store: from script [mongodb] - with grp [group1] >> ERROR:core:cachedb_store: failed to get connection for grp name >> [group1] >> >> >> What does it mean ? If I am changing the group1 to replicaset1 >> then also its giving error . What should I do ? Please help me . >> Am I doing something wrong in the opensips config file or I need >> to do something in the mongodb replica set ? >> >> Thanks in advance . any kind of suggestion is welcome . >> >> */Thanks & Regards/* >> /Sasmita Panda/ >> /Network Testing and Software Engineer/ >> /3CLogic , ph:07827611765/ >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From johan at democon.be Fri Jun 3 12:25:04 2016 From: johan at democon.be (johan de clercq) Date: Fri, 3 Jun 2016 12:25:04 +0200 Subject: [OpenSIPS-Users] opensips.org Message-ID: <003601d1bd82$338b4b60$9aa1e220$@democon.be> Hi everybody, Can it be that opensips.org is down ? BR, -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Fri Jun 3 12:26:47 2016 From: spanda at 3clogic.com (Sasmita Panda) Date: Fri, 3 Jun 2016 15:56:47 +0530 Subject: [OpenSIPS-Users] mongodb replica set connection . In-Reply-To: <91eaaddd-2d8d-9801-b7b0-acfb08971ff4@opensips.org> References: <90b0c601-fd77-f4b5-a498-977f69959513@opensips.org> <91eaaddd-2d8d-9801-b7b0-acfb08971ff4@opensips.org> Message-ID: modparam("cachedb_mongodb", "cachedb_url","mongodb:replicaset1:// 1.2.3.4:27017 , 2.3.4.5:27017/jack.db.CallCenter_Info") This says , I am using two IPs for the replica set . If its like jack is the data base . In this case If I am setting " jack.callCenter_Info " , Its not working , and if I am setting "db.CallCenter_Info" its also not working . Opensips is not even get started in these two cases . This is the only case what I have written is working . *Thanks & Regards* *Sasmita Panda* *Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Fri, Jun 3, 2016 at 3:50 PM, R?zvan Crainea wrote: > Hi, Sasmita! > > Are you using multiple IPs in the replica set, or only one? Because if you > are using only one, the replica set feature is not enabled, and the > "jack.db.CallCenter_Info" is treated as database "jack" and collection > "db.CallCenter_Info", as in your case. Is this your problem? Are you > setting multiple IPs in the cachedb_url? > > Best regards, > > R?zvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 06/03/2016 12:01 PM, Sasmita Panda wrote: > > I have tried this . But its not working . > > Now I got something interesting . I changed the cache_store command and > now its working . > This is " replicasetname.db.collection " this is the format what I get > from opensips documentation . > > modparam("cachedb_mongodb", "cachedb_url","mongodb:replicaset1:// > 1.2.3.4:27017 , > 2.3.4.5:27017/jack.db.CallCenter_Info") > cache_store("mongodb:replicaset1","$ci","$ci,$var(c)"); > > What opensips is doing is . Its creating a database jack and inside > that a collection db.CallCenter_Info and storing data on that collections . > > But I have created a database db and i have also created a collection > in that CallCenter_Info and I was expecting the data will goes to this > collection . > > I am doing anything wrong ? I am getting the data so I think my > problem is almost solved . But my acc and sip_trace collection and > CallCenter_Info collection are in different data base . I wanted them to > present inside a single data base so that it will be easier for me which > searching something . > > Let me know if I am doing anything wrong . > > One problem i am facing is , I am not able to connect to the replica > set with authentication . Without authentication its working like above . > > > > > *Thanks & Regards* > *Sasmita Panda* > *Network Testing and Software Engineer* > *3CLogic , ph:07827611765* > > On Fri, Jun 3, 2016 at 12:56 PM, R?zvan Crainea > wrote: > >> Hi, Sasmita! >> >> Are these two the only errors you see in the logs? >> >> Can you confirm that replicaset "jack" exists and that the >> "CallCenter_Info" table exists? What about the "db" collection? Have you >> tried to set the cachedb_url like this: >> modparam("db_cachedb","cachedb_url","mongodb:replicaset1://1.2.3.4:27017 >> /jack.db >> ") >> >> Best regards, >> >> R?zvan Crainea >> OpenSIPS Solutionswww.opensips-solutions.com >> >> On 06/02/2016 03:23 PM, Sasmita Panda wrote: >> >> Hi All , >> >> I am using opensips-1.11 and I am using cachedb_mongodb module . I >> am able to connect to mongodb replica set without authentication . >> >> modparam("cachedb_mongodb", "cachedb_url","mongodb:replicaset1:// >> 1.2.3.4:27017 , >> 2.3.4.5:27017/jack.db.CallCenter_Info") >> >> modparam("db_cachedb","cachedb_url","mongodb:replicaset1://1.2.3.4:27017 >> ,2.3.4.5:27017/jack.db. >> db") >> >> modparam("acc", "db_url", "cachedb://mongodb:replicaset1") >> I am able to connect to mongodb for account module . By this line . >> >> But I am not able to store any data through raw query like bellow . >> cache_store("mongodb:group1","$ci","$ci,$var(c)"); >> >> Its giving bellow error . >> >> DBG:core:cachedb_store: from script [mongodb] - with grp [group1] >> ERROR:core:cachedb_store: failed to get connection for grp name [group1] >> >> >> What does it mean ? If I am changing the group1 to replicaset1 then also >> its giving error . What should I do ? Please help me . Am I doing something >> wrong in the opensips config file or I need to do something in the mongodb >> replica set ? >> >> Thanks in advance . any kind of suggestion is welcome . >> >> *Thanks & Regards* >> *Sasmita Panda* >> *Network Testing and Software Engineer* >> *3CLogic , ph:07827611765* >> >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Fri Jun 3 12:28:14 2016 From: spanda at 3clogic.com (Sasmita Panda) Date: Fri, 3 Jun 2016 15:58:14 +0530 Subject: [OpenSIPS-Users] opensips.org In-Reply-To: <003601d1bd82$338b4b60$9aa1e220$@democon.be> References: <003601d1bd82$338b4b60$9aa1e220$@democon.be> Message-ID: Its working for me .. *Thanks & Regards* *Sasmita Panda* *Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Fri, Jun 3, 2016 at 3:55 PM, johan de clercq wrote: > Hi everybody, > > > > Can it be that opensips.org is down ? > > > > BR, > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From rrobson at greenlightcrm.com Fri Jun 3 12:28:52 2016 From: rrobson at greenlightcrm.com (Richard Robson) Date: Fri, 3 Jun 2016 11:28:52 +0100 Subject: [OpenSIPS-Users] opensips.org In-Reply-To: <003601d1bd82$338b4b60$9aa1e220$@democon.be> References: <003601d1bd82$338b4b60$9aa1e220$@democon.be> Message-ID: <5072c65b-e0a0-fd5c-6b50-eab992292e89@greenlightcrm.com> An HTML attachment was scrubbed... URL: From razvan at opensips.org Fri Jun 3 13:14:58 2016 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Fri, 3 Jun 2016 14:14:58 +0300 Subject: [OpenSIPS-Users] mongodb replica set connection . In-Reply-To: References: <90b0c601-fd77-f4b5-a498-977f69959513@opensips.org> <91eaaddd-2d8d-9801-b7b0-acfb08971ff4@opensips.org> Message-ID: When starting OpenSIPS, are you seeing any INFO logs that contain "Connecting to"? Can you turn on the logging and put it on pastebin or something? Regards, R?zvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 06/03/2016 01:26 PM, Sasmita Panda wrote: > modparam("cachedb_mongodb", > "cachedb_url","mongodb:replicaset1://1.2.3.4:27017 > ,2.3.4.5:27017/jack.db.CallCenter_Info > ") > > This says , I am using two IPs for the replica set . If its like jack > is the data base . In this case If I am setting " > jack.callCenter_Info " , Its not working , and if I am setting > "db.CallCenter_Info" its also not working . Opensips is not even get > started in these two cases . > > This is the only case what I have written is working . > > */Thanks & Regards/* > /Sasmita Panda/ > /Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > On Fri, Jun 3, 2016 at 3:50 PM, R?zvan Crainea > wrote: > > Hi, Sasmita! > > Are you using multiple IPs in the replica set, or only one? > Because if you are using only one, the replica set feature is not > enabled, and the "jack.db.CallCenter_Info" is treated as database > "jack" and collection "db.CallCenter_Info", as in your case. Is > this your problem? Are you setting multiple IPs in the cachedb_url? > > Best regards, > > R?zvan Crainea > OpenSIPS Solutions > www.opensips-solutions.com > > On 06/03/2016 12:01 PM, Sasmita Panda wrote: >> I have tried this . But its not working . >> >> Now I got something interesting . I changed the cache_store >> command and now its working . >> This is " replicasetname.db.collection " this is the format what >> I get from opensips documentation . >> >> modparam("cachedb_mongodb", >> "cachedb_url","mongodb:replicaset1://1.2.3.4:27017 >> ,2.3.4.5:27017/jack.db.CallCenter_Info >> ") >> cache_store("mongodb:replicaset1","$ci","$ci,$var(c)"); >> What opensips is doing is . Its creating a database jack and >> inside that a collection db.CallCenter_Info and storing data on >> that collections . >> >> But I have created a database db and i have also created a >> collection in that CallCenter_Info and I was expecting the data >> will goes to this collection . >> >> I am doing anything wrong ? I am getting the data so I think >> my problem is almost solved . But my acc and sip_trace collection >> and CallCenter_Info collection are in different data base . I >> wanted them to present inside a single data base so that it will >> be easier for me which searching something . >> >> Let me know if I am doing anything wrong . >> >> One problem i am facing is , I am not able to connect to >> the replica set with authentication . Without authentication its >> working like above . >> >> >> */Thanks & Regards/* >> /Sasmita Panda/ >> /Network Testing and Software Engineer/ >> /3CLogic , ph:07827611765/ >> >> On Fri, Jun 3, 2016 at 12:56 PM, R?zvan Crainea >> > wrote: >> >> Hi, Sasmita! >> >> Are these two the only errors you see in the logs? >> >> Can you confirm that replicaset "jack" exists and that the >> "CallCenter_Info" table exists? What about the "db" >> collection? Have you tried to set the cachedb_url like this: >> >> modparam("db_cachedb","cachedb_url","mongodb:replicaset1://1.2.3.4:27017 >> /jack.db >> ") >> >> Best regards, >> >> R?zvan Crainea >> OpenSIPS Solutions >> www.opensips-solutions.com >> >> On 06/02/2016 03:23 PM, Sasmita Panda wrote: >>> Hi All , >>> >>> I am using opensips-1.11 and I am using >>> cachedb_mongodb module . I am able to connect to mongodb >>> replica set without authentication . >>> >>> modparam("cachedb_mongodb", >>> "cachedb_url","mongodb:replicaset1://1.2.3.4:27017 >>> ,2.3.4.5:27017/jack.db.CallCenter_Info >>> ") >>> >>> modparam("db_cachedb","cachedb_url","mongodb:replicaset1://1.2.3.4:27017 >>> ,2.3.4.5:27017/jack.db. >>> db") >>> >>> modparam("acc", "db_url", "cachedb://mongodb:replicaset1") >>> I am able to connect to mongodb for account module . By this >>> line . >>> >>> But I am not able to store any data through raw query like >>> bellow . >>> cache_store("mongodb:group1","$ci","$ci,$var(c)"); >>> >>> Its giving bellow error . >>> >>> DBG:core:cachedb_store: from script [mongodb] - with grp >>> [group1] >>> ERROR:core:cachedb_store: failed to get connection for grp >>> name [group1] >>> >>> >>> What does it mean ? If I am changing the group1 to >>> replicaset1 then also its giving error . What should I do ? >>> Please help me . Am I doing something wrong in the opensips >>> config file or I need to do something in the mongodb replica >>> set ? >>> >>> Thanks in advance . any kind of suggestion is welcome . >>> >>> */Thanks & Regards/* >>> /Sasmita Panda/ >>> /Network Testing and Software Engineer/ >>> /3CLogic , ph:07827611765/ >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From igor.pavlov1987 at gmail.com Fri Jun 3 13:21:27 2016 From: igor.pavlov1987 at gmail.com (Igor Pavlov) Date: Fri, 3 Jun 2016 15:21:27 +0400 Subject: [OpenSIPS-Users] opensips 1.11.7 tcp_send failed In-Reply-To: <575155E9.4060507@opensips.org> References: <38329c63-8d88-0927-5717-366f5630a3fd@gmail.com> <575155E9.4060507@opensips.org> Message-ID: <3eed2e47-fa2a-23e7-4821-f94c4e46f515@gmail.com> Yes, it's setbflag(6) used. if ($si == "192.168.101.5") { setbflag(6); ... } if (isbflagset(6)) xlog("L_INFO","BFLAG SET"); > Jun 3 14:11:52 vs01 /usr/sbin/opensips[27233]: BFLAG SET 03.06.2016 14:03, Liviu Chircu ?????: > Hi Igor, > > Are you sure you are using "setbflag(6)" and not "setflag(6)"? It's a > very important detail, since doing a "setflag(6)" would be completely > irrelevant, and useless. > > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > > On 03.06.2016 12:02, Igor Pavlov wrote: >> Hi, all. I have a problem with TCP in opensips 1.11.7. All users are >> registered with tcp proto. Call flow: userA -> (tcp) opensips -> >> (udp) freeswitch -> (udp) opensips -> (tcp) userB. >> >> When call is coming from FS I get error: >> >>> Jun 3 11:53:57 ERROR:tm:msg_send: tcp_send failed >>> Jun 3 11:53:57 ERROR:tm:t_forward_nonack: sending request failed >> opensips.cfg contain tcp_no_new_conn_bflag = 6 and I'm setting this >> flag when INVITE is coming from FS. >> >> >> ____________ >> Best regards, >> Igor Pavlov >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- ____________ Best regards, Igor Pavlov From spanda at 3clogic.com Fri Jun 3 13:37:06 2016 From: spanda at 3clogic.com (Sasmita Panda) Date: Fri, 3 Jun 2016 17:07:06 +0530 Subject: [OpenSIPS-Users] mongodb replica set connection . In-Reply-To: References: <90b0c601-fd77-f4b5-a498-977f69959513@opensips.org> <91eaaddd-2d8d-9801-b7b0-acfb08971ff4@opensips.org> Message-ID: bellow is the like for pastebin to see the logs . This is the log in loglevel=6 . http://pastebin.com/jCnRBMcG *Thanks & Regards* *Sasmita Panda* *Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Fri, Jun 3, 2016 at 4:44 PM, R?zvan Crainea wrote: > When starting OpenSIPS, are you seeing any INFO logs that contain > "Connecting to"? Can you turn on the logging and put it on pastebin or > something? > Regards, > > R?zvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 06/03/2016 01:26 PM, Sasmita Panda wrote: > > modparam("cachedb_mongodb", "cachedb_url","mongodb:replicaset1:// > 1.2.3.4:27017 , > 2.3.4.5:27017/jack.db.CallCenter_Info") > > This says , I am using two IPs for the replica set . If its like jack is > the data base . In this case If I am setting " jack.callCenter_Info " , > Its not working , and if I am setting "db.CallCenter_Info" its also not > working . Opensips is not even get started in these two cases . > > This is the only case what I have written is working . > > *Thanks & Regards* > *Sasmita Panda* > *Network Testing and Software Engineer* > *3CLogic , ph:07827611765* > > On Fri, Jun 3, 2016 at 3:50 PM, R?zvan Crainea > wrote: > >> Hi, Sasmita! >> >> Are you using multiple IPs in the replica set, or only one? Because if >> you are using only one, the replica set feature is not enabled, and the >> "jack.db.CallCenter_Info" is treated as database "jack" and collection >> "db.CallCenter_Info", as in your case. Is this your problem? Are you >> setting multiple IPs in the cachedb_url? >> >> Best regards, >> >> R?zvan Crainea >> OpenSIPS Solutionswww.opensips-solutions.com >> >> On 06/03/2016 12:01 PM, Sasmita Panda wrote: >> >> I have tried this . But its not working . >> >> Now I got something interesting . I changed the cache_store command and >> now its working . >> This is " replicasetname.db.collection " this is the format what I get >> from opensips documentation . >> >> modparam("cachedb_mongodb", "cachedb_url","mongodb:replicaset1:// >> 1.2.3.4:27017 , >> 2.3.4.5:27017/jack.db.CallCenter_Info") >> cache_store("mongodb:replicaset1","$ci","$ci,$var(c)"); >> >> What opensips is doing is . Its creating a database jack and inside >> that a collection db.CallCenter_Info and storing data on that collections . >> >> But I have created a database db and i have also created a >> collection in that CallCenter_Info and I was expecting the data will goes >> to this collection . >> >> I am doing anything wrong ? I am getting the data so I think my >> problem is almost solved . But my acc and sip_trace collection and >> CallCenter_Info collection are in different data base . I wanted them to >> present inside a single data base so that it will be easier for me which >> searching something . >> >> Let me know if I am doing anything wrong . >> >> One problem i am facing is , I am not able to connect to the >> replica set with authentication . Without authentication its working like >> above . >> >> >> >> >> *Thanks & Regards* >> *Sasmita Panda* >> *Network Testing and Software Engineer* >> *3CLogic , ph:07827611765* >> >> On Fri, Jun 3, 2016 at 12:56 PM, R?zvan Crainea < >> razvan at opensips.org> wrote: >> >>> Hi, Sasmita! >>> >>> Are these two the only errors you see in the logs? >>> >>> Can you confirm that replicaset "jack" exists and that the >>> "CallCenter_Info" table exists? What about the "db" collection? Have you >>> tried to set the cachedb_url like this: >>> modparam("db_cachedb","cachedb_url","mongodb:replicaset1://1.2.3.4:27017 >>> /jack.db >>> ") >>> >>> Best regards, >>> >>> R?zvan Crainea >>> OpenSIPS Solutionswww.opensips-solutions.com >>> >>> On 06/02/2016 03:23 PM, Sasmita Panda wrote: >>> >>> Hi All , >>> >>> I am using opensips-1.11 and I am using cachedb_mongodb module . >>> I am able to connect to mongodb replica set without authentication . >>> >>> modparam("cachedb_mongodb", "cachedb_url","mongodb:replicaset1:// >>> 1.2.3.4:27017 , >>> 2.3.4.5:27017/jack.db.CallCenter_Info") >>> >>> modparam("db_cachedb","cachedb_url","mongodb:replicaset1://1.2.3.4:27017 >>> ,2.3.4.5:27017/jack.db. >>> db") >>> >>> modparam("acc", "db_url", "cachedb://mongodb:replicaset1") >>> I am able to connect to mongodb for account module . By this line . >>> >>> But I am not able to store any data through raw query like bellow . >>> cache_store("mongodb:group1","$ci","$ci,$var(c)"); >>> >>> Its giving bellow error . >>> >>> DBG:core:cachedb_store: from script [mongodb] - with grp [group1] >>> ERROR:core:cachedb_store: failed to get connection for grp name >>> [group1] >>> >>> >>> What does it mean ? If I am changing the group1 to replicaset1 then >>> also its giving error . What should I do ? Please help me . Am I doing >>> something wrong in the opensips config file or I need to do something in >>> the mongodb replica set ? >>> >>> Thanks in advance . any kind of suggestion is welcome . >>> >>> *Thanks & Regards* >>> *Sasmita Panda* >>> *Network Testing and Software Engineer* >>> *3CLogic , ph:07827611765* >>> >>> >>> _______________________________________________ >>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Fri Jun 3 13:45:04 2016 From: spanda at 3clogic.com (Sasmita Panda) Date: Fri, 3 Jun 2016 17:15:04 +0530 Subject: [OpenSIPS-Users] mongodb replica set connection . In-Reply-To: References: <90b0c601-fd77-f4b5-a498-977f69959513@opensips.org> <91eaaddd-2d8d-9801-b7b0-acfb08971ff4@opensips.org> Message-ID: Hi , whether its not getting store in right database but its working somehow . But when I am doing authentication in my mongodb , Opensips is not able to connect to mongodb . In mongodb I am getting bellow error . assertion 13 not authorized for query on db.my_version_table ns:db.my_version_table query:{ $query: { table_name: "acc" } } I am trying to login through user having Role : root , still I am facing this issue . Can you tell me what is the right way to connect with authentication . In single standalone mongodb instance I am able to do authentication . Facing problem in replica set . Please help me in this . *Thanks & Regards* *Sasmita Panda* *Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Fri, Jun 3, 2016 at 5:07 PM, Sasmita Panda wrote: > bellow is the like for pastebin to see the logs . This is the log in > loglevel=6 . > > http://pastebin.com/jCnRBMcG > > *Thanks & Regards* > *Sasmita Panda* > *Network Testing and Software Engineer* > *3CLogic , ph:07827611765* > > On Fri, Jun 3, 2016 at 4:44 PM, R?zvan Crainea > wrote: > >> When starting OpenSIPS, are you seeing any INFO logs that contain >> "Connecting to"? Can you turn on the logging and put it on pastebin or >> something? >> Regards, >> >> R?zvan Crainea >> OpenSIPS Solutionswww.opensips-solutions.com >> >> On 06/03/2016 01:26 PM, Sasmita Panda wrote: >> >> modparam("cachedb_mongodb", "cachedb_url","mongodb:replicaset1:// >> 1.2.3.4:27017 , >> 2.3.4.5:27017/jack.db.CallCenter_Info") >> >> This says , I am using two IPs for the replica set . If its like jack is >> the data base . In this case If I am setting " jack.callCenter_Info " , >> Its not working , and if I am setting "db.CallCenter_Info" its also not >> working . Opensips is not even get started in these two cases . >> >> This is the only case what I have written is working . >> >> *Thanks & Regards* >> *Sasmita Panda* >> *Network Testing and Software Engineer* >> *3CLogic , ph:07827611765* >> >> On Fri, Jun 3, 2016 at 3:50 PM, R?zvan Crainea >> wrote: >> >>> Hi, Sasmita! >>> >>> Are you using multiple IPs in the replica set, or only one? Because if >>> you are using only one, the replica set feature is not enabled, and the >>> "jack.db.CallCenter_Info" is treated as database "jack" and collection >>> "db.CallCenter_Info", as in your case. Is this your problem? Are you >>> setting multiple IPs in the cachedb_url? >>> >>> Best regards, >>> >>> R?zvan Crainea >>> OpenSIPS Solutionswww.opensips-solutions.com >>> >>> On 06/03/2016 12:01 PM, Sasmita Panda wrote: >>> >>> I have tried this . But its not working . >>> >>> Now I got something interesting . I changed the cache_store command and >>> now its working . >>> This is " replicasetname.db.collection " this is the format what I get >>> from opensips documentation . >>> >>> modparam("cachedb_mongodb", "cachedb_url","mongodb:replicaset1:// >>> 1.2.3.4:27017 , >>> 2.3.4.5:27017/jack.db.CallCenter_Info") >>> cache_store("mongodb:replicaset1","$ci","$ci,$var(c)"); >>> >>> What opensips is doing is . Its creating a database jack and inside >>> that a collection db.CallCenter_Info and storing data on that collections . >>> >>> But I have created a database db and i have also created a >>> collection in that CallCenter_Info and I was expecting the data will goes >>> to this collection . >>> >>> I am doing anything wrong ? I am getting the data so I think my >>> problem is almost solved . But my acc and sip_trace collection and >>> CallCenter_Info collection are in different data base . I wanted them to >>> present inside a single data base so that it will be easier for me which >>> searching something . >>> >>> Let me know if I am doing anything wrong . >>> >>> One problem i am facing is , I am not able to connect to the >>> replica set with authentication . Without authentication its working like >>> above . >>> >>> >>> >>> >>> *Thanks & Regards* >>> *Sasmita Panda* >>> *Network Testing and Software Engineer* >>> *3CLogic , ph:07827611765* >>> >>> On Fri, Jun 3, 2016 at 12:56 PM, R?zvan Crainea < >>> razvan at opensips.org> wrote: >>> >>>> Hi, Sasmita! >>>> >>>> Are these two the only errors you see in the logs? >>>> >>>> Can you confirm that replicaset "jack" exists and that the >>>> "CallCenter_Info" table exists? What about the "db" collection? Have you >>>> tried to set the cachedb_url like this: >>>> modparam("db_cachedb","cachedb_url","mongodb:replicaset1:// >>>> 1.2.3.4:27017 /jack.db >>>> ") >>>> >>>> Best regards, >>>> >>>> R?zvan Crainea >>>> OpenSIPS Solutionswww.opensips-solutions.com >>>> >>>> On 06/02/2016 03:23 PM, Sasmita Panda wrote: >>>> >>>> Hi All , >>>> >>>> I am using opensips-1.11 and I am using cachedb_mongodb module . >>>> I am able to connect to mongodb replica set without authentication . >>>> >>>> modparam("cachedb_mongodb", "cachedb_url","mongodb:replicaset1:// >>>> 1.2.3.4:27017 , >>>> 2.3.4.5:27017/jack.db.CallCenter_Info") >>>> >>>> modparam("db_cachedb","cachedb_url","mongodb:replicaset1:// >>>> 1.2.3.4:27017 , >>>> 2.3.4.5:27017/jack.db. db") >>>> >>>> modparam("acc", "db_url", "cachedb://mongodb:replicaset1") >>>> I am able to connect to mongodb for account module . By this line . >>>> >>>> But I am not able to store any data through raw query like bellow . >>>> cache_store("mongodb:group1","$ci","$ci,$var(c)"); >>>> >>>> Its giving bellow error . >>>> >>>> DBG:core:cachedb_store: from script [mongodb] - with grp [group1] >>>> ERROR:core:cachedb_store: failed to get connection for grp name >>>> [group1] >>>> >>>> >>>> What does it mean ? If I am changing the group1 to replicaset1 then >>>> also its giving error . What should I do ? Please help me . Am I doing >>>> something wrong in the opensips config file or I need to do something in >>>> the mongodb replica set ? >>>> >>>> Thanks in advance . any kind of suggestion is welcome . >>>> >>>> *Thanks & Regards* >>>> *Sasmita Panda* >>>> *Network Testing and Software Engineer* >>>> *3CLogic , ph:07827611765* >>>> >>>> >>>> _______________________________________________ >>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>> >>> >>> _______________________________________________ >>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From 35633 at heb.be Fri Jun 3 13:50:48 2016 From: 35633 at heb.be (Francjos) Date: Fri, 3 Jun 2016 04:50:48 -0700 (MST) Subject: [OpenSIPS-Users] Too many Hops error In-Reply-To: References: <1464325697044-7603151.post@n2.nabble.com> <5AB35FAD44B8F4E0.F6654531-775D-455F-9487-647AE0B43214@mail.outlook.com> <1464430509795-7603171.post@n2.nabble.com> Message-ID: <1464954648662-7603300.post@n2.nabble.com> Hello, I really failed to use lb_is_destination (.....) function. In fact, when i do this: if(!load_balance("1", "calls", "1")){ sl_send_reply("500", "Service full"); exit; } without any other condiiton before load balancing, it success on the first server, but it failed on the second server saying "Service Full". I think this error is caused by the sipsak i'm using to send OPTIONS. I dont understand why it is ok on one server and fails on the other? -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Too-many-Hops-error-tp7603151p7603300.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. From johan at democon.be Fri Jun 3 13:52:15 2016 From: johan at democon.be (johan de clercq) Date: Fri, 3 Jun 2016 13:52:15 +0200 Subject: [OpenSIPS-Users] Too many Hops error In-Reply-To: <1464954648662-7603300.post@n2.nabble.com> References: <1464325697044-7603151.post@n2.nabble.com> <5AB35FAD44B8F4E0.F6654531-775D-455F-9487-647AE0B43214@mail.outlook.com> <1464430509795-7603171.post@n2.nabble.com> <1464954648662-7603300.post@n2.nabble.com> Message-ID: <02c801d1bd8e$615dadb0$24190910$@democon.be> You need to configure the number of sessions per server. -----Original Message----- From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Francjos Sent: Friday, June 3, 2016 1:51 PM To: users at lists.opensips.org Subject: Re: [OpenSIPS-Users] Too many Hops error Hello, I really failed to use lb_is_destination (.....) function. In fact, when i do this: if(!load_balance("1", "calls", "1")){ sl_send_reply("500", "Service full"); exit; } without any other condiiton before load balancing, it success on the first server, but it failed on the second server saying "Service Full". I think this error is caused by the sipsak i'm using to send OPTIONS. I dont understand why it is ok on one server and fails on the other? -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Too-many-Hops-error-tp 7603151p7603300.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users From pimenta at inatel.br Fri Jun 3 14:54:41 2016 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Fri, 3 Jun 2016 12:54:41 +0000 Subject: [OpenSIPS-Users] opensips.org In-Reply-To: <5072c65b-e0a0-fd5c-6b50-eab992292e89@greenlightcrm.com> References: <003601d1bd82$338b4b60$9aa1e220$@democon.be>, <5072c65b-e0a0-fd5c-6b50-eab992292e89@greenlightcrm.com> Message-ID: Hi. It is ok for me in Brazil too. Regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________ De: users-bounces at lists.opensips.org em nome de Richard Robson Enviado: sexta-feira, 3 de junho de 2016 07:28 Para: users at lists.opensips.org Assunto: Re: [OpenSIPS-Users] opensips.org Site is working for me in the UK Regards, Richard On 03/06/2016 11:25, johan de clercq wrote: Hi everybody, Can it be that opensips.org is down ? BR, _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Richard Robson Greenlight Support 01382 843843 support at greenlightcrm.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From 35633 at heb.be Sat Jun 4 11:21:36 2016 From: 35633 at heb.be (Francjos) Date: Sat, 4 Jun 2016 02:21:36 -0700 (MST) Subject: [OpenSIPS-Users] Too many Hops error In-Reply-To: <02c801d1bd8e$615dadb0$24190910$@democon.be> References: <1464325697044-7603151.post@n2.nabble.com> <5AB35FAD44B8F4E0.F6654531-775D-455F-9487-647AE0B43214@mail.outlook.com> <1464430509795-7603171.post@n2.nabble.com> <1464954648662-7603300.post@n2.nabble.com> <02c801d1bd8e$615dadb0$24190910$@democon.be> Message-ID: <1465032096588-7603314.post@n2.nabble.com> Hello, Yes I configure up to 100 sessions per server, but i still have the same error on the second server. I'm gonna explain the scenario once again, what i'm trying to do is to put in place redundancy with Opensips servers and do load balancing. I'm using sipsak to send OPTIONS to Opensips servers and load_balancer module to do load balancing. And i think that the error "service fuul" occurs when OPTIONS are sent, because Opensips is supposed to sent OPTIONS to FeeSwitch servers only. So how can i tell Opensips server to sent OPTIONS to the other Opensips server without generating any error? Thanks. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Too-many-Hops-error-tp7603151p7603314.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. From spanda at 3clogic.com Mon Jun 6 10:23:43 2016 From: spanda at 3clogic.com (Sasmita Panda) Date: Mon, 6 Jun 2016 13:53:43 +0530 Subject: [OpenSIPS-Users] Opensips -1.11 crash with mongodb replica set . Message-ID: Hi All , I am using opesips-1.11 with mongodb replica set . I have 3 members in the replica set , promary , secondary and arbitrary . Problem 1 : Sometime , If primay is not rechable , the secondary becomes primary , But opensips loss connection from mongodb . Its wont put any data in the data base . My call goes on but their is not data in the mongodb database . Problme 2: If the primay machine is down then secondary becomes primary within some millisecond time , but opensips crashes giving bellow error ERROR:cachedb_mongodb:mongo_con_get: Failed to run query. Err = 6, 0 , 0 CRITICAL:core:receive_fd: EOF on 10 INFO:core:handle_sigs: child process 5278 exited by a signal 11 INFO:core:handle_sigs: core was generated INFO:core:handle_sigs: terminating due to SIGCHLD INFO:core:sig_usr: signal 15 received INFO:core:sig_usr: signal 15 received INFO:core:sig_usr: signal 15 received INFO:core:sig_usr: signal 15 received INFO:core:sig_usr: signal 15 received INFO:core:sig_usr: signal 15 received INFO:core:sig_usr: signal 15 received INFO:core:sig_usr: signal 15 received INFO:core:sig_usr: signal 15 received INFO:core:sig_usr: signal 15 received INFO:core:sig_usr: signal 15 received INFO:core:sig_usr: signal 15 received INFO:core:sig_usr: signal 15 received INFO:core:cleanup: cleanup NOTICE:db_cachedb:destroy: destroy module db_cachedb ... NOTICE:cachedb_mongodb:destroy: destroy module cachedb_mongodb ... If my secondary mongodb machine is not reachable then rather connection with db breaks for sometime but opensips wont crashes . But in case of Primary opensips crashes with above error . Is this an expected behavior or I am doing anything wrong . Data loss can be bearable but application cant be . So please let me know whats the problem . *Thanks & Regards* *Sasmita Panda* *Network Testing and Software Engineer* *3CLogic , ph:07827611765* -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Mon Jun 6 10:44:10 2016 From: liviu at opensips.org (Liviu Chircu) Date: Mon, 6 Jun 2016 11:44:10 +0300 Subject: [OpenSIPS-Users] Opensips -1.11 crash with mongodb replica set . In-Reply-To: References: Message-ID: <575537DA.60002@opensips.org> Hi Sasmita! We have some plans to do some more testing on the module, including your "connect directly to replica set" usage case, and also bringing it up to date with the latest mongo driver. However, it is hard to give you an estimation for when this work will start. Currently, the fastest way to fix your problems is for you to set up a "mongos" instance, along with 3 config server instances (these are needed by mongos, and ensure proper write consistency), and configure "cachedb_mongodb" to use this new "mongos" node. A mongos instance is aware of all the replica sets (aka "shards"). In your case, it will initially shard the keys to your single replica set - thus, it will hold 100% of the data. As your data set grows, you may partition it over to additional replica sets, which you can configure into mongos without modifying anything on the OpenSIPS side. This way, failover within a replica set will also happen transparently, again, without impacting OpenSIPS at all. PS: if you have a way to 100% reproduce a crash, please open a GH ticket describing the steps, and we'll go from there! [1] [1]: https://github.com/OpenSIPS/opensips/issues Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 06.06.2016 11:23, Sasmita Panda wrote: > Hi All , > > I am using opesips-1.11 with mongodb replica set . I have 3 > members in the replica set , promary , secondary and arbitrary . > > Problem 1 : Sometime , If primay is not rechable , the > secondary becomes primary , But opensips loss connection from mongodb > . Its wont put any data in the data base . My call goes on but their > is not data in the mongodb database . > > Problme 2: If the primay machine is down then secondary > becomes primary within some millisecond time , but opensips crashes > giving bellow error > > ERROR:cachedb_mongodb:mongo_con_get: Failed to run query. Err = > 6, 0 , 0 > CRITICAL:core:receive_fd: EOF on 10 > INFO:core:handle_sigs: child process 5278 exited by a signal 11 > INFO:core:handle_sigs: core was generated > INFO:core:handle_sigs: terminating due to SIGCHLD > INFO:core:sig_usr: signal 15 received > INFO:core:sig_usr: signal 15 received > INFO:core:sig_usr: signal 15 received > INFO:core:sig_usr: signal 15 received > INFO:core:sig_usr: signal 15 received > INFO:core:sig_usr: signal 15 received > INFO:core:sig_usr: signal 15 received > INFO:core:sig_usr: signal 15 received > INFO:core:sig_usr: signal 15 received > INFO:core:sig_usr: signal 15 received > INFO:core:sig_usr: signal 15 received > INFO:core:sig_usr: signal 15 received > INFO:core:sig_usr: signal 15 received > INFO:core:cleanup: cleanup > NOTICE:db_cachedb:destroy: destroy module db_cachedb ... > NOTICE:cachedb_mongodb:destroy: destroy module cachedb_mongodb ... > > If my secondary mongodb machine is not reachable then rather > connection with db breaks for sometime but opensips wont crashes . But > in case of Primary opensips crashes with above error . Is this an > expected behavior or I am doing anything wrong . > Data loss can be bearable but application cant be . So please > let me know whats the problem . > > */Thanks & Regards/* > /Sasmita Panda/ > /Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From pinghan022 at gmail.com Mon Jun 6 10:46:40 2016 From: pinghan022 at gmail.com (Ping Han) Date: Mon, 6 Jun 2016 18:46:40 +1000 Subject: [OpenSIPS-Users] Question regarding b2b_bridge function In-Reply-To: <575145C9.9080406@opensips.org> References: <574D44F7.9090204@opensips.org> <574E8B1A.7070508@opensips.org> <574FE1C0.9000904@opensips.org> <575145C9.9080406@opensips.org> Message-ID: Thanks, Bogdan, I have done a few testing. It seems only the value of the param works for this function. However, it does not look like the callid and the to-tag. Is it possible to get the param value in the Opensips config? Please see the test results below. --------------------------------- [root at opensips-02 ~]# opensipsctl fifo b2be_list dlg:: 37 param=237.0 state=5 last_invite_cseq=1 last_method=0 last_reply_code=200 db_flag=2 ruri:: sip:2401012350 at 10.216.235.38:5060 callid:: NjEyZWEyOTEwZmVlZDIzOTViMTc3YjJiMmJmY2RjODI. from:: "Tropo test" uri=sip:2401012350 at 10.216.235.115:5060 tag=1e53cf61 to:: "2401012350" uri=sip:2401012350 at 10.216.235.115:5060 tag= B2B.297.37 cseq:: caller=1 callee=1 route_set:: caller=, contact:: caller=sip:2401012350 at 10.203.1.196:22238 callee=sip: 10.216.235.72:5060 send_sock:: 10.216.235.72 dlg:: 6715348 param=237.0 state=5 last_invite_cseq=2 last_method=4 db_flag=2 callid:: B2B.237.6715348 from:: "Tropo test" uri=sip:2401012350 at 10.216.235.115:5060 tag=44623c403b25fd7905bfa7a7325b2b8f to:: uri=sip:dialog at 10.216.235.38:5060 tag=40711690 cseq:: caller=2 callee=1 contact:: caller=sip:10.216.235.72:5060 callee=sip: 10.216.235.38:5060 send_sock:: 10.216.235.72 LEGS:: leg:: 0 tag=40711690 cseq=2 contact=sip:10.216.235.38:5060 [root at opensips-02 ~]# opensipsctl fifo b2b_bridge 237.6715348 sip:123 at 10.203.1.196 (this is the callid with out the "B2B" prefix) 500 command 'b2b_bridge' failed [root at opensips-02 ~]# opensipsctl fifo b2b_bridge 297.37 sip:123 at 10.203.1.196 (this is the To-tag) 500 command 'b2b_bridge' failed [root at opensips-02 ~]# opensipsctl fifo b2b_bridge 237.0 sip:123 at 10.203.1.196 (this is the value of param, only this works) --------------------------------- Regards, Ping On Fri, Jun 3, 2016 at 6:54 PM, Bogdan-Andrei Iancu wrote: > Hi Ping, > > b2b_bridge_request() is a script function: > > http://www.opensips.org/html/docs/modules/2.1.x/b2b_logic.html#id294094 > > I pointed to this function as you mentioned (on my question) that you want > to do the bridging from script level. > > Indeed, the equivalent MI function is b2b_bridge: > > http://www.opensips.org/html/docs/modules/2.1.x/b2b_logic.html#id294210 > > If you want to get that "1020.30", you can get it from Call-ID or To tag, > where you have B2B.1020.30 (so you have to strip that B2B prefix). > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 03.06.2016 06:27, Ping Han wrote: > > Thanks, Bogdan, > > It seems the function you mentioned is the internal function "1.4.2 > b2b_bridge_request(b2bl_key,entity_no)". > > Actually the function that I am trying to use is the "b2b_bridge" > (Exported MI Functions). It is defined as below > ---------------------------------- > http://www.opensips.org/html/docs/modules/2.1.x/b2b_logic.html#id294210 > > > 1.5.2. b2b_bridge > > Example: opensipsctl b2b_bridge 1020.30 > sip:alice at opensips.org > ---------------------------------- > > In the example "1020.30" is the "dialog-id". This is the parameter that I > am not sure how to easily access in the Opensips config. > > What I am trying to do is to get the value and deliver to the next hop via > a custom SIP header. When the next hop tries to transfer the call to a new > destination. It can run the b2b_bridge command straight away with the > "dialog-id" without rechieving the value from the Opensips database (from > b2b_logic or b2b_entities tables). > > Any advice will be appreciated. > > Thanks, > Ping > > > On Thu, Jun 2, 2016 at 5:35 PM, Bogdan-Andrei Iancu < > bogdan at opensips.org> wrote: > >> Hi Ping, >> >> In script, in a b2b route, you can look at the callid or TO tag >> (depending on the direction) to get the key : >> >> http://www.opensips.org/html/docs/modules/2.2.x/b2b_logic.html#id294094 >> >> The B2B uses that key as Call-ID when acting as UAC and as To tag when >> acting as UAS. You can run a SIP capture to see the traffic. >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com >> >> On 02.06.2016 05:13, Ping Han wrote: >> >> Hi Bogdan, >> >> Thanks for the information. >> >> I need the value in the Opensips cfg. >> >> I understand that I can query the b2b_logic or b2b_entities tables to get >> the value in Opensips config. Apart from that could you tell me other way >> to easily access the value in Opensips config? >> >> Thanks, >> Ping >> >> >> >> On Wed, Jun 1, 2016 at 5:13 PM, Bogdan-Andrei Iancu < >> bogdan at opensips.org> wrote: >> >>> Hi Ping, >>> >>> Indeed, my bad - the docs are not updated, as that param was disabled >>> long time ago (4 years ago): >>> https://sourceforge.net/p/opensips/bugs/502/ >>> >>> Still, there are available option. But the question is : do you need >>> that value in OpenSIPS cfg or outside OpenSIPS ? as there are different way >>> to get the ID. >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com >>> >>> On 01.06.2016 03:48, Ping Han wrote: >>> >>> Hi Bogdan, >>> >>> Thank you very much for your reply. >>> >>> I have tried to use the module parameter "b2bl_key_avp" as described in >>> the document as below. >>> >>> ------------------------------------ >>> modparam("b2b_logic", "b2bl_key_avp", "$avp(99)") >>> ------------------------------------ >>> >>> However, I got the following errors when the Opensips is restarted. >>> ------------------------------------ >>> ERROR:core:set_mod_param_regex: parameter not found in >>> module >>> ------------------------------------ >>> >>> I am using the Opensips version opensips-2.1.2-1.el6.x86_64 (rpm). >>> >>> Thanks, >>> Chris >>> >>> On Tue, May 31, 2016 at 6:01 PM, Bogdan-Andrei Iancu < >>> bogdan at opensips.org> wrote: >>> >>>> Hi Chris, >>>> >>>> The "dialog_id" is actually the b2b key, that is expose by the >>>> b2b_logic via the module parameter b2bl_key_avp. See: >>>> >>>> >>>> http://www.opensips.org/html/docs/modules/2.1.x/b2b_logic.html#id294094 >>>> >>>> That key can also be found in the b2b_logic table in DB. >>>> >>>> At signaling level, the key is the Call-ID of the outbound calls from >>>> b2b. >>>> >>>> Regards, >>>> >>>> Bogdan-Andrei Iancu >>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com >>>> >>>> On 23.05.2016 07:32, Ping Han wrote: >>>> >>>> Hi Bogdan, >>>> >>>> I asked the question a few days ago but have not got a response. >>>> >>>> I am just wondering if I could get some advice from you. >>>> >>>> Any advice will be appreciated. >>>> >>>> Thanks, >>>> Chris >>>> >>>> On Wed, May 18, 2016 at 4:39 PM, Ping Han < >>>> pinghan022 at gmail.com> wrote: >>>> >>>>> Hi, >>>>> >>>>> I would like to use the b2b_bridge fifo function as specified at >>>>> >>>>> http://www.opensips.org/html/docs/modules/2.1.x/b2b_logic.html#id248916 >>>>> . >>>>> >>>>> The function will be triggered by a third party. I will need to pass >>>>> the parameters to the third party for it to trigger the function. One of >>>>> the parameters is the "dialog-id". >>>>> >>>>> The problem is that I am not sure how the value of the dialog-id can >>>>> be available in the Opensips config. Is there any Opensips modules/function >>>>> that can retrieve the value of the dialog-id? >>>>> >>>>> I tried to get the value from the "b2b_entities" and "b2b_logic" >>>>> table. However, it seems that it does not work this way because the two >>>>> tables do not pop the data in real time. Sometimes I can see the data but >>>>> sometimes I am not able to see it. >>>>> >>>>> It is appreciated that you can give me some idea. >>>>> >>>>> Thanks, >>>>> >>>>> Ping >>>>> >>>> >>>> >>>> >>> >>> >> >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Mon Jun 6 12:02:31 2016 From: spanda at 3clogic.com (Sasmita Panda) Date: Mon, 6 Jun 2016 15:32:31 +0530 Subject: [OpenSIPS-Users] Opensips -1.11 crash with mongodb replica set . In-Reply-To: <575537DA.60002@opensips.org> References: <575537DA.60002@opensips.org> Message-ID: Thank you so much for the information . Let me try to deploy with mongos server . I will post the detailed steps for opensips crash as soon as I will be free . *Thanks & Regards* *Sasmita Panda* *Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Mon, Jun 6, 2016 at 2:14 PM, Liviu Chircu wrote: > Hi Sasmita! > > We have some plans to do some more testing on the module, including your > "connect directly to replica set" usage case, and also bringing it up to > date with the latest mongo driver. However, it is hard to give you an > estimation for when this work will start. > > Currently, the fastest way to fix your problems is for you to set up a > "mongos" instance, along with 3 config server instances (these are needed > by mongos, and ensure proper write consistency), and configure > "cachedb_mongodb" to use this new "mongos" node. > > A mongos instance is aware of all the replica sets (aka "shards"). In your > case, it will initially shard the keys to your single replica set - thus, > it will hold 100% of the data. As your data set grows, you may partition it > over to additional replica sets, which you can configure into mongos > without modifying anything on the OpenSIPS side. This way, failover within > a replica set will also happen transparently, again, without impacting > OpenSIPS at all. > > PS: if you have a way to 100% reproduce a crash, please open a GH ticket > describing the steps, and we'll go from there! [1] > > [1]: https://github.com/OpenSIPS/opensips/issues > > Liviu Chircu > OpenSIPS Developerhttp://www.opensips-solutions.com > > On 06.06.2016 11:23, Sasmita Panda wrote: > > Hi All , > > I am using opesips-1.11 with mongodb replica set . I have 3 members > in the replica set , promary , secondary and arbitrary . > > Problem 1 : Sometime , If primay is not rechable , the secondary > becomes primary , But opensips loss connection from mongodb . Its wont put > any data in the data base . My call goes on but their is not data in the > mongodb database . > > Problme 2: If the primay machine is down then secondary becomes > primary within some millisecond time , but opensips crashes giving bellow > error > > ERROR:cachedb_mongodb:mongo_con_get: Failed to run query. Err = 6, 0 > , 0 > CRITICAL:core:receive_fd: EOF on 10 > INFO:core:handle_sigs: child process 5278 exited by a signal 11 > INFO:core:handle_sigs: core was generated > INFO:core:handle_sigs: terminating due to SIGCHLD > INFO:core:sig_usr: signal 15 received > INFO:core:sig_usr: signal 15 received > INFO:core:sig_usr: signal 15 received > INFO:core:sig_usr: signal 15 received > INFO:core:sig_usr: signal 15 received > INFO:core:sig_usr: signal 15 received > INFO:core:sig_usr: signal 15 received > INFO:core:sig_usr: signal 15 received > INFO:core:sig_usr: signal 15 received > INFO:core:sig_usr: signal 15 received > INFO:core:sig_usr: signal 15 received > INFO:core:sig_usr: signal 15 received > INFO:core:sig_usr: signal 15 received > INFO:core:cleanup: cleanup > NOTICE:db_cachedb:destroy: destroy module db_cachedb ... > NOTICE:cachedb_mongodb:destroy: destroy module cachedb_mongodb ... > > If my secondary mongodb machine is not reachable then rather > connection with db breaks for sometime but opensips wont crashes . But in > case of Primary opensips crashes with above error . Is this an expected > behavior or I am doing anything wrong . > > Data loss can be bearable but application cant be . So please let me > know whats the problem . > > *Thanks & Regards* > *Sasmita Panda* > *Network Testing and Software Engineer* > *3CLogic , ph:07827611765* > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Mon Jun 6 12:55:55 2016 From: spanda at 3clogic.com (Sasmita Panda) Date: Mon, 6 Jun 2016 16:25:55 +0530 Subject: [OpenSIPS-Users] Opensips -1.11 crash with mongodb replica set . In-Reply-To: References: <575537DA.60002@opensips.org> Message-ID: Hi , I have created an issue with detailed steps . Bellow is the issue ID . Let me know if the issue will get fixed . https://github.com/OpenSIPS/opensips/issues/895 Thanks *Thanks & Regards* *Sasmita Panda* *Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Mon, Jun 6, 2016 at 3:32 PM, Sasmita Panda wrote: > Thank you so much for the information . Let me try to deploy with mongos > server . > > I will post the detailed steps for opensips crash as soon as I will be > free . > > *Thanks & Regards* > *Sasmita Panda* > *Network Testing and Software Engineer* > *3CLogic , ph:07827611765* > > On Mon, Jun 6, 2016 at 2:14 PM, Liviu Chircu wrote: > >> Hi Sasmita! >> >> We have some plans to do some more testing on the module, including your >> "connect directly to replica set" usage case, and also bringing it up to >> date with the latest mongo driver. However, it is hard to give you an >> estimation for when this work will start. >> >> Currently, the fastest way to fix your problems is for you to set up a >> "mongos" instance, along with 3 config server instances (these are needed >> by mongos, and ensure proper write consistency), and configure >> "cachedb_mongodb" to use this new "mongos" node. >> >> A mongos instance is aware of all the replica sets (aka "shards"). In >> your case, it will initially shard the keys to your single replica set - >> thus, it will hold 100% of the data. As your data set grows, you may >> partition it over to additional replica sets, which you can configure into >> mongos without modifying anything on the OpenSIPS side. This way, failover >> within a replica set will also happen transparently, again, without >> impacting OpenSIPS at all. >> >> PS: if you have a way to 100% reproduce a crash, please open a GH ticket >> describing the steps, and we'll go from there! [1] >> >> [1]: https://github.com/OpenSIPS/opensips/issues >> >> Liviu Chircu >> OpenSIPS Developerhttp://www.opensips-solutions.com >> >> On 06.06.2016 11:23, Sasmita Panda wrote: >> >> Hi All , >> >> I am using opesips-1.11 with mongodb replica set . I have 3 >> members in the replica set , promary , secondary and arbitrary . >> >> Problem 1 : Sometime , If primay is not rechable , the secondary >> becomes primary , But opensips loss connection from mongodb . Its wont put >> any data in the data base . My call goes on but their is not data in the >> mongodb database . >> >> Problme 2: If the primay machine is down then secondary becomes >> primary within some millisecond time , but opensips crashes giving bellow >> error >> >> ERROR:cachedb_mongodb:mongo_con_get: Failed to run query. Err = 6, >> 0 , 0 >> CRITICAL:core:receive_fd: EOF on 10 >> INFO:core:handle_sigs: child process 5278 exited by a signal 11 >> INFO:core:handle_sigs: core was generated >> INFO:core:handle_sigs: terminating due to SIGCHLD >> INFO:core:sig_usr: signal 15 received >> INFO:core:sig_usr: signal 15 received >> INFO:core:sig_usr: signal 15 received >> INFO:core:sig_usr: signal 15 received >> INFO:core:sig_usr: signal 15 received >> INFO:core:sig_usr: signal 15 received >> INFO:core:sig_usr: signal 15 received >> INFO:core:sig_usr: signal 15 received >> INFO:core:sig_usr: signal 15 received >> INFO:core:sig_usr: signal 15 received >> INFO:core:sig_usr: signal 15 received >> INFO:core:sig_usr: signal 15 received >> INFO:core:sig_usr: signal 15 received >> INFO:core:cleanup: cleanup >> NOTICE:db_cachedb:destroy: destroy module db_cachedb ... >> NOTICE:cachedb_mongodb:destroy: destroy module cachedb_mongodb ... >> >> If my secondary mongodb machine is not reachable then rather >> connection with db breaks for sometime but opensips wont crashes . But in >> case of Primary opensips crashes with above error . Is this an expected >> behavior or I am doing anything wrong . >> >> Data loss can be bearable but application cant be . So please let >> me know whats the problem . >> >> *Thanks & Regards* >> *Sasmita Panda* >> *Network Testing and Software Engineer* >> *3CLogic , ph:07827611765* >> >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Mon Jun 6 13:11:10 2016 From: liviu at opensips.org (Liviu Chircu) Date: Mon, 6 Jun 2016 14:11:10 +0300 Subject: [OpenSIPS-Users] Opensips -1.11 crash with mongodb replica set . In-Reply-To: References: <575537DA.60002@opensips.org> Message-ID: <57555A4E.8000502@opensips.org> Thank you, Sasmita! Will reply here as soon as there is progress on the matter. Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 06.06.2016 13:55, Sasmita Panda wrote: > Hi , > > I have created an issue with detailed steps . Bellow is the issue ID > . Let me know if the issue will get fixed . > > https://github.com/OpenSIPS/opensips/issues/895 > > Thanks > > */Thanks & Regards/* > /Sasmita Panda/ > /Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > On Mon, Jun 6, 2016 at 3:32 PM, Sasmita Panda > wrote: > > Thank you so much for the information . Let me try to deploy with > mongos server . > > I will post the detailed steps for opensips crash as soon as I > will be free . > > */Thanks & Regards/* > /Sasmita Panda/ > /Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > On Mon, Jun 6, 2016 at 2:14 PM, Liviu Chircu > wrote: > > Hi Sasmita! > > We have some plans to do some more testing on the module, > including your "connect directly to replica set" usage case, > and also bringing it up to date with the latest mongo driver. > However, it is hard to give you an estimation for when this > work will start. > > Currently, the fastest way to fix your problems is for you to > set up a "mongos" instance, along with 3 config server > instances (these are needed by mongos, and ensure proper write > consistency), and configure "cachedb_mongodb" to use this new > "mongos" node. > > A mongos instance is aware of all the replica sets (aka > "shards"). In your case, it will initially shard the keys to > your single replica set - thus, it will hold 100% of the data. > As your data set grows, you may partition it over to > additional replica sets, which you can configure into mongos > without modifying anything on the OpenSIPS side. This way, > failover within a replica set will also happen transparently, > again, without impacting OpenSIPS at all. > > PS: if you have a way to 100% reproduce a crash, please open a > GH ticket describing the steps, and we'll go from there! [1] > > [1]: https://github.com/OpenSIPS/opensips/issues > > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > > On 06.06.2016 11:23, Sasmita Panda wrote: >> Hi All , >> >> I am using opesips-1.11 with mongodb replica set . I >> have 3 members in the replica set , promary , secondary and >> arbitrary . >> >> Problem 1 : Sometime , If primay is not rechable , the >> secondary becomes primary , But opensips loss connection from >> mongodb . Its wont put any data in the data base . My call >> goes on but their is not data in the mongodb database . >> >> Problme 2: If the primay machine is down then >> secondary becomes primary within some millisecond time , but >> opensips crashes giving bellow error >> >> ERROR:cachedb_mongodb:mongo_con_get: Failed to run query. Err >> = 6, 0 , 0 >> CRITICAL:core:receive_fd: EOF on 10 >> INFO:core:handle_sigs: child process 5278 exited by a signal 11 >> INFO:core:handle_sigs: core was generated >> INFO:core:handle_sigs: terminating due to SIGCHLD >> INFO:core:sig_usr: signal 15 received >> INFO:core:sig_usr: signal 15 received >> INFO:core:sig_usr: signal 15 received >> INFO:core:sig_usr: signal 15 received >> INFO:core:sig_usr: signal 15 received >> INFO:core:sig_usr: signal 15 received >> INFO:core:sig_usr: signal 15 received >> INFO:core:sig_usr: signal 15 received >> INFO:core:sig_usr: signal 15 received >> INFO:core:sig_usr: signal 15 received >> INFO:core:sig_usr: signal 15 received >> INFO:core:sig_usr: signal 15 received >> INFO:core:sig_usr: signal 15 received >> INFO:core:cleanup: cleanup >> NOTICE:db_cachedb:destroy: destroy module db_cachedb ... >> NOTICE:cachedb_mongodb:destroy: destroy module >> cachedb_mongodb ... >> >> If my secondary mongodb machine is not reachable then >> rather connection with db breaks for sometime but opensips >> wont crashes . But in case of Primary opensips crashes with >> above error . Is this an expected behavior or I am doing >> anything wrong . >> Data loss can be bearable but application cant be . So >> please let me know whats the problem . >> >> */Thanks & Regards/* >> /Sasmita Panda/ >> /Network Testing and Software Engineer/ >> /3CLogic , ph:07827611765/ >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From nocbgtelcom at gmail.com Mon Jun 6 13:12:55 2016 From: nocbgtelcom at gmail.com (Hristo Donev) Date: Mon, 6 Jun 2016 14:12:55 +0300 Subject: [OpenSIPS-Users] OpenSips 2.2 - Crashed Message-ID: Hello, My OpenSips 2.2 (Stable) crashed. I see 2 core files. When make "bt full" I read follow lines: Program terminated with signal 6, Aborted. #0 0x00846402 in __kernel_vsyscall () (gdb) bt full #0 0x00846402 in __kernel_vsyscall () No symbol table info available. #1 0x005d2b10 in ?? () No symbol table info available. Regards, Hristo Donev Support office www.bg-tel.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From igor.pavlov1987 at gmail.com Mon Jun 6 13:15:04 2016 From: igor.pavlov1987 at gmail.com (Igor Pavlov) Date: Mon, 6 Jun 2016 15:15:04 +0400 Subject: [OpenSIPS-Users] opensips 1.11.7 tcp_send failed In-Reply-To: <3eed2e47-fa2a-23e7-4821-f94c4e46f515@gmail.com> References: <38329c63-8d88-0927-5717-366f5630a3fd@gmail.com> <575155E9.4060507@opensips.org> <3eed2e47-fa2a-23e7-4821-f94c4e46f515@gmail.com> Message-ID: Can anybody reproduce this issue? 03.06.2016 15:21, Igor Pavlov ?????: > Yes, it's setbflag(6) used. > > if ($si == "192.168.101.5") { > setbflag(6); > ... > } > if (isbflagset(6)) > xlog("L_INFO","BFLAG SET"); > >> Jun 3 14:11:52 vs01 /usr/sbin/opensips[27233]: BFLAG SET > > > 03.06.2016 14:03, Liviu Chircu ?????: >> Hi Igor, >> >> Are you sure you are using "setbflag(6)" and not "setflag(6)"? It's a >> very important detail, since doing a "setflag(6)" would be completely >> irrelevant, and useless. >> >> Liviu Chircu >> OpenSIPS Developer >> http://www.opensips-solutions.com >> >> On 03.06.2016 12:02, Igor Pavlov wrote: >>> Hi, all. I have a problem with TCP in opensips 1.11.7. All users are >>> registered with tcp proto. Call flow: userA -> (tcp) opensips -> >>> (udp) freeswitch -> (udp) opensips -> (tcp) userB. >>> >>> When call is coming from FS I get error: >>> >>>> Jun 3 11:53:57 ERROR:tm:msg_send: tcp_send failed >>>> Jun 3 11:53:57 ERROR:tm:t_forward_nonack: sending request failed >>> opensips.cfg contain tcp_no_new_conn_bflag = 6 and I'm setting this >>> flag when INVITE is coming from FS. >>> >>> >>> ____________ >>> Best regards, >>> Igor Pavlov >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- ____________ Best regards, Igor Pavlov From bogdan at opensips.org Mon Jun 6 13:55:44 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 6 Jun 2016 14:55:44 +0300 Subject: [OpenSIPS-Users] Too many Hops error In-Reply-To: <1465032096588-7603314.post@n2.nabble.com> References: <1464325697044-7603151.post@n2.nabble.com> <5AB35FAD44B8F4E0.F6654531-775D-455F-9487-647AE0B43214@mail.outlook.com> <1464430509795-7603171.post@n2.nabble.com> <1464954648662-7603300.post@n2.nabble.com> <02c801d1bd8e$615dadb0$24190910$@democon.be> <1465032096588-7603314.post@n2.nabble.com> Message-ID: <575564C0.4090403@opensips.org> Hi, Keep in mind you must call LB function only for the initial INVITE requests and not for anything else. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 04.06.2016 12:21, Francjos wrote: > Hello, > > Yes I configure up to 100 sessions per server, but i still have the same > error on the second server. > I'm gonna explain the scenario once again, what i'm trying to do is to put > in place redundancy with Opensips servers and do load balancing. > > I'm using sipsak to send OPTIONS to Opensips servers and load_balancer > module to do load balancing. > And i think that the error "service fuul" occurs when OPTIONS are sent, > because Opensips is supposed to sent OPTIONS to FeeSwitch servers only. > > So how can i tell Opensips server to sent OPTIONS to the other Opensips > server without generating any error? > > Thanks. > > > > > > -- > View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Too-many-Hops-error-tp7603151p7603314.html > Sent from the OpenSIPS - Users mailing list archive at Nabble.com. > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > From bogdan at opensips.org Mon Jun 6 13:57:56 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 6 Jun 2016 14:57:56 +0300 Subject: [OpenSIPS-Users] OpenSips 2.2 - Crashed In-Reply-To: References: Message-ID: <57556544.6040100@opensips.org> Hi Hristo, That does not look like a valid bracktrace. Be sure you have the debugging symbols loaded - did you install OpenSIPS from sources or packages ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 06.06.2016 14:12, Hristo Donev wrote: > Hello, > > My OpenSips 2.2 (Stable) crashed. I see 2 core files. > When make "bt full" I read follow lines: > > Program terminated with signal 6, Aborted. > #0 0x00846402 in __kernel_vsyscall () > (gdb) bt full > #0 0x00846402 in __kernel_vsyscall () > No symbol table info available. > #1 0x005d2b10 in ?? () > No symbol table info available. > > > Regards, > Hristo Donev > Support office > www.bg-tel.com > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Jun 6 14:06:15 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 6 Jun 2016 15:06:15 +0300 Subject: [OpenSIPS-Users] OpenSips 2.2 - Crashed In-Reply-To: References: <57556544.6040100@opensips.org> Message-ID: <57556737.6070600@opensips.org> And you locally compiled it ? Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 06.06.2016 15:04, Hristo Donev wrote: > Hi, > > I get Opensips from Git hub. > > Here is last messages from log: > Jun 5 20:28:23 sbc [25436]: INFO:core:handle_sigs: child process > 25448 exited by a signal 11 > Jun 5 20:28:23 sbc [25436]: INFO:core:handle_sigs: core was generated > Jun 5 20:28:23 sbc [25436]: INFO:core:handle_sigs: terminating due to > SIGCHLD > Jun 5 20:28:23 sbc [25450]: INFO:core:sig_usr: signal 15 received > Jun 5 20:28:23 sbc [25443]: INFO:core:sig_usr: signal 15 received > Jun 5 20:28:23 sbc [25446]: INFO:core:sig_usr: signal 15 received > Jun 5 20:28:23 sbc [25445]: INFO:core:sig_usr: signal 15 received > Jun 5 20:28:23 sbc [25442]: INFO:core:sig_usr: signal 15 received > Jun 5 20:28:23 sbc [25439]: INFO:core:sig_usr: signal 15 received > Jun 5 20:28:23 sbc [25440]: INFO:core:sig_usr: signal 15 received > Jun 5 20:28:23 sbc [25438]: INFO:core:sig_usr: signal 15 received > Jun 5 20:28:23 sbc [25441]: INFO:core:sig_usr: signal 15 received > Jun 5 20:28:23 sbc [25449]: INFO:core:sig_usr: signal 15 received > Jun 5 20:28:31 sbc [25436]: INFO:core:cleanup: cleanup > Jun 5 20:29:23 sbc [25436]: CRITICAL:core:sig_alarm_abort: BUG - > shutdown timeout triggered, dying... > > 2016-06-06 14:57 GMT+03:00 Bogdan-Andrei Iancu >: > > Hi Hristo, > > That does not look like a valid bracktrace. Be sure you have the > debugging symbols loaded - did you install OpenSIPS from sources > or packages ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 06.06.2016 14:12, Hristo Donev wrote: >> Hello, >> >> My OpenSips 2.2 (Stable) crashed. I see 2 core files. >> When make "bt full" I read follow lines: >> >> Program terminated with signal 6, Aborted. >> #0 0x00846402 in __kernel_vsyscall () >> (gdb) bt full >> #0 0x00846402 in __kernel_vsyscall () >> No symbol table info available. >> #1 0x005d2b10 in ?? () >> No symbol table info available. >> >> >> Regards, >> Hristo Donev >> Support office >> www.bg-tel.com >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Mon Jun 6 14:07:04 2016 From: liviu at opensips.org (Liviu Chircu) Date: Mon, 6 Jun 2016 15:07:04 +0300 Subject: [OpenSIPS-Users] opensips 1.11.7 tcp_send failed In-Reply-To: References: <38329c63-8d88-0927-5717-366f5630a3fd@gmail.com> <575155E9.4060507@opensips.org> <3eed2e47-fa2a-23e7-4821-f94c4e46f515@gmail.com> Message-ID: <57556768.7060809@opensips.org> It would help a lot if you could supply a SIP trace of both user registration and a faulty call (either .pcap or plain text). If IP privacy is a concern, please email the traces to liviu at opensips.org. Thanks! Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 06.06.2016 14:15, Igor Pavlov wrote: > Can anybody reproduce this issue? > > > 03.06.2016 15:21, Igor Pavlov ?????: >> Yes, it's setbflag(6) used. >> >> if ($si == "192.168.101.5") { >> setbflag(6); >> ... >> } >> if (isbflagset(6)) >> xlog("L_INFO","BFLAG SET"); >> >>> Jun 3 14:11:52 vs01 /usr/sbin/opensips[27233]: BFLAG SET >> >> >> 03.06.2016 14:03, Liviu Chircu ?????: >>> Hi Igor, >>> >>> Are you sure you are using "setbflag(6)" and not "setflag(6)"? It's >>> a very important detail, since doing a "setflag(6)" would be >>> completely irrelevant, and useless. >>> >>> Liviu Chircu >>> OpenSIPS Developer >>> http://www.opensips-solutions.com >>> >>> On 03.06.2016 12:02, Igor Pavlov wrote: >>>> Hi, all. I have a problem with TCP in opensips 1.11.7. All users >>>> are registered with tcp proto. Call flow: userA -> (tcp) opensips >>>> -> (udp) freeswitch -> (udp) opensips -> (tcp) userB. >>>> >>>> When call is coming from FS I get error: >>>> >>>>> Jun 3 11:53:57 ERROR:tm:msg_send: tcp_send failed >>>>> Jun 3 11:53:57 ERROR:tm:t_forward_nonack: sending request failed >>>> opensips.cfg contain tcp_no_new_conn_bflag = 6 and I'm setting this >>>> flag when INVITE is coming from FS. >>>> >>>> >>>> ____________ >>>> Best regards, >>>> Igor Pavlov >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > From igor.pavlov1987 at gmail.com Mon Jun 6 14:24:38 2016 From: igor.pavlov1987 at gmail.com (Igor Pavlov) Date: Mon, 6 Jun 2016 16:24:38 +0400 Subject: [OpenSIPS-Users] opensips 1.11.7 tcp_send failed In-Reply-To: <57556768.7060809@opensips.org> References: <38329c63-8d88-0927-5717-366f5630a3fd@gmail.com> <575155E9.4060507@opensips.org> <3eed2e47-fa2a-23e7-4821-f94c4e46f515@gmail.com> <57556768.7060809@opensips.org> Message-ID: I emailed you to liviu at opensips.org. 06.06.2016 16:07, Liviu Chircu ?????: > It would help a lot if you could supply a SIP trace of both user > registration and a faulty call (either .pcap or plain text). > > If IP privacy is a concern, please email the traces to > liviu at opensips.org. Thanks! > > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > > On 06.06.2016 14:15, Igor Pavlov wrote: >> Can anybody reproduce this issue? >> >> >> 03.06.2016 15:21, Igor Pavlov ?????: >>> Yes, it's setbflag(6) used. >>> >>> if ($si == "192.168.101.5") { >>> setbflag(6); >>> ... >>> } >>> if (isbflagset(6)) >>> xlog("L_INFO","BFLAG SET"); >>> >>>> Jun 3 14:11:52 vs01 /usr/sbin/opensips[27233]: BFLAG SET >>> >>> >>> 03.06.2016 14:03, Liviu Chircu ?????: >>>> Hi Igor, >>>> >>>> Are you sure you are using "setbflag(6)" and not "setflag(6)"? It's >>>> a very important detail, since doing a "setflag(6)" would be >>>> completely irrelevant, and useless. >>>> >>>> Liviu Chircu >>>> OpenSIPS Developer >>>> http://www.opensips-solutions.com >>>> >>>> On 03.06.2016 12:02, Igor Pavlov wrote: >>>>> Hi, all. I have a problem with TCP in opensips 1.11.7. All users >>>>> are registered with tcp proto. Call flow: userA -> (tcp) opensips >>>>> -> (udp) freeswitch -> (udp) opensips -> (tcp) userB. >>>>> >>>>> When call is coming from FS I get error: >>>>> >>>>>> Jun 3 11:53:57 ERROR:tm:msg_send: tcp_send failed >>>>>> Jun 3 11:53:57 ERROR:tm:t_forward_nonack: sending request failed >>>>> opensips.cfg contain tcp_no_new_conn_bflag = 6 and I'm setting >>>>> this flag when INVITE is coming from FS. >>>>> >>>>> >>>>> ____________ >>>>> Best regards, >>>>> Igor Pavlov >>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- ____________ Best regards, Igor Pavlov From pimenta at inatel.br Mon Jun 6 15:02:21 2016 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Mon, 6 Jun 2016 13:02:21 +0000 Subject: [OpenSIPS-Users] Does attr_avp (string) decrease the available RAM? In-Reply-To: <574FEBCD.7000003@opensips.org> References: , <574FEBCD.7000003@opensips.org> Message-ID: Thank you Liviu and Bogdan. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________ De: users-bounces at lists.opensips.org em nome de Liviu Chircu Enviado: quinta-feira, 2 de junho de 2016 05:18 Para: users at lists.opensips.org Assunto: Re: [OpenSIPS-Users] Does attr_avp (string) decrease the available RAM? AVPs eat up shared memory. For each "lookup()" you perform on a different INVITE request, a different $avp(attr) will be populated with the attributes of each contact behind the AoR you looked up. All AVPs of a transaction are then freed along with the transaction itself. Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com Home - OpenSIPS Solutions www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 01.06.2016 21:19, Rodrigo Pimenta Carvalho wrote: Hi. I'm using OpenSIPS 2.1 and the module REGISTRAR. So, in my script I have: modparam("registrar", "attr_avp", "$avp(attr)") ... if (is_method("REGISTER")) { $avp(attr) = "contact_info"; save("location"); exit; } ... lookup("location"); I would like to know whether every time such code is executed the available memory decreases. What happens? Does the avp demand more and more memory to keep its information about lots of "contact_info"? Any hint will be very helpful! Thanks alot. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Jun 6 16:07:29 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 6 Jun 2016 17:07:29 +0300 Subject: [OpenSIPS-Users] Question regarding b2b_bridge function In-Reply-To: References: <574D44F7.9090204@opensips.org> <574E8B1A.7070508@opensips.org> <574FE1C0.9000904@opensips.org> <575145C9.9080406@opensips.org> Message-ID: <575583A1.7050501@opensips.org> Hi Ping, My bad - in Call-ID and TO tag you have the B2B _entity_ ID, while you need the B2B _logic_ ID .....which is not part of the signaling at all . Now, in script, where do you need the b2b logic ID ? after creating the B2B session (after b2b_init() ) ? Or ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 06.06.2016 11:46, Ping Han wrote: > Thanks, Bogdan, > > I have done a few testing. It seems only the value of the param works > for this function. However, it does not look like the callid and the > to-tag. > Is it possible to get the param value in the Opensips config? > > Please see the test results below. > > --------------------------------- > [root at opensips-02 ~]# opensipsctl fifo b2be_list > dlg:: 37 param=237.0 state=5 last_invite_cseq=1 last_method=0 > last_reply_code=200 db_flag=2 > ruri:: sip:2401012350 at 10.216.235.38:5060 > > callid:: NjEyZWEyOTEwZmVlZDIzOTViMTc3YjJiMmJmY2RjODI. > from:: "Tropo test" uri=sip:2401012350 at 10.216.235.115:5060 > tag=1e53cf61 > to:: "2401012350" uri=sip:2401012350 at 10.216.235.115:5060 > tag=B2B.297.37 > cseq:: caller=1 callee=1 > route_set:: > caller=, > contact:: caller=sip:2401012350 at 10.203.1.196:22238 > > callee=sip:10.216.235.72:5060 > send_sock:: 10.216.235.72 > dlg:: 6715348 param=237.0 state=5 last_invite_cseq=2 last_method=4 > db_flag=2 > callid:: B2B.237.6715348 > from:: "Tropo test" uri=sip:2401012350 at 10.216.235.115:5060 > > tag=44623c403b25fd7905bfa7a7325b2b8f > to:: uri=sip:dialog at 10.216.235.38:5060 > tag=40711690 > cseq:: caller=2 callee=1 > contact:: caller=sip:10.216.235.72:5060 > callee=sip:10.216.235.38:5060 > > send_sock:: 10.216.235.72 > LEGS:: > leg:: 0 tag=40711690 cseq=2 > contact=sip:10.216.235.38:5060 > > [root at opensips-02 ~]# opensipsctl fifo b2b_bridge 237.6715348 > sip:123 at 10.203.1.196 (this is the > callid with out the "B2B" prefix) > 500 command 'b2b_bridge' failed > > [root at opensips-02 ~]# opensipsctl fifo b2b_bridge 297.37 > sip:123 at 10.203.1.196 (this is the To-tag) > 500 command 'b2b_bridge' failed > > [root at opensips-02 ~]# opensipsctl fifo b2b_bridge 237.0 > sip:123 at 10.203.1.196 (this is the > value of param, only this works) > > --------------------------------- > > Regards, > Ping > > > On Fri, Jun 3, 2016 at 6:54 PM, Bogdan-Andrei Iancu > > wrote: > > Hi Ping, > > b2b_bridge_request() is a script function: > http://www.opensips.org/html/docs/modules/2.1.x/b2b_logic.html#id294094 > > I pointed to this function as you mentioned (on my question) that > you want to do the bridging from script level. > > Indeed, the equivalent MI function is b2b_bridge: > http://www.opensips.org/html/docs/modules/2.1.x/b2b_logic.html#id294210 > > If you want to get that "1020.30", you can get it from Call-ID or > To tag, where you have B2B.1020.30 (so you have to strip that B2B > prefix). > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 03.06.2016 06:27, Ping Han wrote: >> Thanks, Bogdan, >> >> It seems the function you mentioned is the internal function >> "1.4.2 b2b_bridge_request(b2bl_key,entity_no)". >> >> Actually the function that I am trying to use is the "b2b_bridge" >> (Exported MI Functions). It is defined as below >> ---------------------------------- >> http://www.opensips.org/html/docs/modules/2.1.x/b2b_logic.html#id294210 >> >> >> 1.5.2. b2b_bridge >> >> Example: opensipsctl b2b_bridge 1020.30 sip:alice at opensips.org >> >> ---------------------------------- >> >> In the example "1020.30" is the "dialog-id". This is the >> parameter that I am not sure how to easily access in the Opensips >> config. >> >> What I am trying to do is to get the value and deliver to the >> next hop via a custom SIP header. When the next hop tries to >> transfer the call to a new destination. It can run the b2b_bridge >> command straight away with the "dialog-id" without rechieving the >> value from the Opensips database (from b2b_logic or b2b_entities >> tables). >> >> Any advice will be appreciated. >> >> Thanks, >> Ping >> >> >> On Thu, Jun 2, 2016 at 5:35 PM, Bogdan-Andrei Iancu >> > wrote: >> >> Hi Ping, >> >> In script, in a b2b route, you can look at the callid or TO >> tag (depending on the direction) to get the key : >> http://www.opensips.org/html/docs/modules/2.2.x/b2b_logic.html#id294094 >> >> The B2B uses that key as Call-ID when acting as UAC and as To >> tag when acting as UAS. You can run a SIP capture to see the >> traffic. >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> On 02.06.2016 05:13, Ping Han wrote: >>> Hi Bogdan, >>> >>> Thanks for the information. >>> >>> I need the value in the Opensips cfg. >>> >>> I understand that I can query the b2b_logic or b2b_entities >>> tables to get the value in Opensips config. Apart from that >>> could you tell me other way to easily access the value in >>> Opensips config? >>> >>> Thanks, >>> Ping >>> >>> >>> >>> On Wed, Jun 1, 2016 at 5:13 PM, Bogdan-Andrei Iancu >>> > wrote: >>> >>> Hi Ping, >>> >>> Indeed, my bad - the docs are not updated, as that param >>> was disabled long time ago (4 years ago): >>> https://sourceforge.net/p/opensips/bugs/502/ >>> >>> Still, there are available option. But the question is : >>> do you need that value in OpenSIPS cfg or outside >>> OpenSIPS ? as there are different way to get the ID. >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> OpenSIPS Founder and Developer >>> http://www.opensips-solutions.com >>> -------------- next part -------------- An HTML attachment was scrubbed... URL: From steve.woolley at me.com Tue Jun 7 03:21:31 2016 From: steve.woolley at me.com (Steve Woolley) Date: Mon, 06 Jun 2016 21:21:31 -0400 Subject: [OpenSIPS-Users] Continuous "WARNING:core:timer_ticker: timer task already scheduled" messages in log Message-ID: <352B7B4A-6668-4887-9F24-FD97771C7E3D@me.com> Running opensips on a Raspberry Pi. At some point after starting opensips (sometimes immediately, sometimes after quite a bit of time), the following message fills the log ? once a second ? continuously. ... Jun 7 01:10:41 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8258580 ms), it may overlap.. Jun 7 01:10:42 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8259570 ms), it may overlap.. Jun 7 01:10:43 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8260560 ms), it may overlap.. Jun 7 01:10:44 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8261560 ms), it may overlap.. Jun 7 01:10:45 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8262550 ms), it may overlap.. Jun 7 01:10:46 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8263550 ms), it may overlap.. Jun 7 01:10:47 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8264540 ms), it may overlap.. Jun 7 01:10:48 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8265530 ms), it may overlap.. Jun 7 01:10:49 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8266530 ms), it may overlap.. Jun 7 01:10:50 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8267520 ms), it may overlap.. root at pi1:~# opensips -V version: opensips 2.2.0 (arm6/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, QM_MALLOC, DBG_MALLOC, USE_PTHREAD_MUTEX MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. git revision: 0e1cea7 main.c compiled on 21:56:59 Jun 3 2016 with gcc 4.9.2 Anyone experiencing the same? -- Steve Woolley steve.woolley at me.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From john.nash778 at gmail.com Tue Jun 7 09:56:24 2016 From: john.nash778 at gmail.com (John Nash) Date: Tue, 7 Jun 2016 13:26:24 +0530 Subject: [OpenSIPS-Users] Register with TO Tag Message-ID: I am dealing with In-dialog requests using ------------------------------------------------------------------------------------------------------ if (has_totag() && (is_domain_local("$rd") || $Ri== "127.0.0.1") && is_method("INVITE|ACK|BYE|UPDATE")) { # sequential request within a dialog should # take the path determined by record-routing if (topology_hiding_match()) ----- --------------------------------------------------------------------------------------------------------- at the top of my script. After that I process initial requests, but I see some REGISTER messages with TO-Tag and "Route" header and they are being discarded by my script because Initial request cannot have Route header. Do i also need to pass REGISTER messages also through same block?...or i need to call loose_route after has_to_tag check. -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Tue Jun 7 12:20:09 2016 From: spanda at 3clogic.com (Sasmita Panda) Date: Tue, 7 Jun 2016 15:50:09 +0530 Subject: [OpenSIPS-Users] Opensips -1.11 crash with mongodb replica set . In-Reply-To: <57555A4E.8000502@opensips.org> References: <575537DA.60002@opensips.org> <57555A4E.8000502@opensips.org> Message-ID: Hi Liviu , I just have one question . I have implemented opensips with mongodb shard cluster . I have one replica set with three member . I have three config server for each and mongos service running . What I wanted to know is , the server in which my mongos service is running , If that machine is get down what will happen ? I need to monitor that machine so that it wont go down ? Is this show ? *Thanks & Regards* *Sasmita Panda* *Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Mon, Jun 6, 2016 at 4:41 PM, Liviu Chircu wrote: > Thank you, Sasmita! Will reply here as soon as there is progress on the > matter. > > Liviu Chircu > OpenSIPS Developerhttp://www.opensips-solutions.com > > On 06.06.2016 13:55, Sasmita Panda wrote: > > Hi , > > I have created an issue with detailed steps . Bellow is the issue ID . > Let me know if the issue will get fixed . > > https://github.com/OpenSIPS/opensips/issues/895 > > Thanks > > *Thanks & Regards* > *Sasmita Panda* > *Network Testing and Software Engineer* > *3CLogic , ph:07827611765* > > On Mon, Jun 6, 2016 at 3:32 PM, Sasmita Panda wrote: > >> Thank you so much for the information . Let me try to deploy with mongos >> server . >> >> I will post the detailed steps for opensips crash as soon as I will be >> free . >> >> *Thanks & Regards* >> *Sasmita Panda* >> *Network Testing and Software Engineer* >> *3CLogic , ph:07827611765* >> >> On Mon, Jun 6, 2016 at 2:14 PM, Liviu Chircu < >> liviu at opensips.org> wrote: >> >>> Hi Sasmita! >>> >>> We have some plans to do some more testing on the module, including your >>> "connect directly to replica set" usage case, and also bringing it up to >>> date with the latest mongo driver. However, it is hard to give you an >>> estimation for when this work will start. >>> >>> Currently, the fastest way to fix your problems is for you to set up a >>> "mongos" instance, along with 3 config server instances (these are needed >>> by mongos, and ensure proper write consistency), and configure >>> "cachedb_mongodb" to use this new "mongos" node. >>> >>> A mongos instance is aware of all the replica sets (aka "shards"). In >>> your case, it will initially shard the keys to your single replica set - >>> thus, it will hold 100% of the data. As your data set grows, you may >>> partition it over to additional replica sets, which you can configure into >>> mongos without modifying anything on the OpenSIPS side. This way, failover >>> within a replica set will also happen transparently, again, without >>> impacting OpenSIPS at all. >>> >>> PS: if you have a way to 100% reproduce a crash, please open a GH ticket >>> describing the steps, and we'll go from there! [1] >>> >>> [1]: https://github.com/OpenSIPS/opensips/issues >>> >>> Liviu Chircu >>> OpenSIPS Developerhttp://www.opensips-solutions.com >>> >>> On 06.06.2016 11:23, Sasmita Panda wrote: >>> >>> Hi All , >>> >>> I am using opesips-1.11 with mongodb replica set . I have 3 >>> members in the replica set , promary , secondary and arbitrary . >>> >>> Problem 1 : Sometime , If primay is not rechable , the secondary >>> becomes primary , But opensips loss connection from mongodb . Its wont put >>> any data in the data base . My call goes on but their is not data in the >>> mongodb database . >>> >>> Problme 2: If the primay machine is down then secondary becomes >>> primary within some millisecond time , but opensips crashes giving bellow >>> error >>> >>> ERROR:cachedb_mongodb:mongo_con_get: Failed to run query. Err = 6, >>> 0 , 0 >>> CRITICAL:core:receive_fd: EOF on 10 >>> INFO:core:handle_sigs: child process 5278 exited by a signal 11 >>> INFO:core:handle_sigs: core was generated >>> INFO:core:handle_sigs: terminating due to SIGCHLD >>> INFO:core:sig_usr: signal 15 received >>> INFO:core:sig_usr: signal 15 received >>> INFO:core:sig_usr: signal 15 received >>> INFO:core:sig_usr: signal 15 received >>> INFO:core:sig_usr: signal 15 received >>> INFO:core:sig_usr: signal 15 received >>> INFO:core:sig_usr: signal 15 received >>> INFO:core:sig_usr: signal 15 received >>> INFO:core:sig_usr: signal 15 received >>> INFO:core:sig_usr: signal 15 received >>> INFO:core:sig_usr: signal 15 received >>> INFO:core:sig_usr: signal 15 received >>> INFO:core:sig_usr: signal 15 received >>> INFO:core:cleanup: cleanup >>> NOTICE:db_cachedb:destroy: destroy module db_cachedb ... >>> NOTICE:cachedb_mongodb:destroy: destroy module cachedb_mongodb ... >>> >>> If my secondary mongodb machine is not reachable then rather >>> connection with db breaks for sometime but opensips wont crashes . But in >>> case of Primary opensips crashes with above error . Is this an expected >>> behavior or I am doing anything wrong . >>> >>> Data loss can be bearable but application cant be . So please let >>> me know whats the problem . >>> >>> *Thanks & Regards* >>> *Sasmita Panda* >>> *Network Testing and Software Engineer* >>> *3CLogic , ph:07827611765* >>> >>> >>> _______________________________________________ >>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >> > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jun 7 12:41:21 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 7 Jun 2016 13:41:21 +0300 Subject: [OpenSIPS-Users] Register with TO Tag In-Reply-To: References: Message-ID: <5756A4D1.60400@opensips.org> Hi John, Assuming you do not do REGISTER relay (but you act as a registrar), you should handle the REGISTER requests (with or without to-tag) in the same way. IF they have a Route hdr , it may be because they do pre-loaded route (the Route points to your SIP server) to be sure the REGISTER gets to the registrar server. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 07.06.2016 10:56, John Nash wrote: > I am dealing with In-dialog requests using > > ------------------------------------------------------------------------------------------------------ > if (has_totag() && (is_domain_local("$rd") || $Ri== "127.0.0.1") > && is_method("INVITE|ACK|BYE|UPDATE")) > { > # sequential request within a dialog should > # take the path determined by record-routing > if (topology_hiding_match()) > ----- > --------------------------------------------------------------------------------------------------------- > > at the top of my script. After that I process initial requests, but I > see some REGISTER messages with TO-Tag and "Route" header and they are > being discarded by my script because Initial request cannot have Route > header. > > Do i also need to pass REGISTER messages also through same block?...or > i need to call loose_route after has_to_tag check. > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jun 7 12:34:27 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 7 Jun 2016 13:34:27 +0300 Subject: [OpenSIPS-Users] Question regarding b2b_bridge function In-Reply-To: References: <574D44F7.9090204@opensips.org> <574E8B1A.7070508@opensips.org> <574FE1C0.9000904@opensips.org> <575145C9.9080406@opensips.org> <575583A1.7050501@opensips.org> Message-ID: <5756A333.1010701@opensips.org> Hi Ping, So you need the B2B Logic ID only when you create the B2B session, in order to place it in the first outgoing INVITE (as extra hdr), right ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 07.06.2016 04:59, Ping Han wrote: > Thanks, Bogdan, > > What I am trying to do is to get the b2b logic ID and deliver to the > next hop via a custom SIP header. > > I am using the b2b_init_request("top hiding") and it looks like the > only place that I can add a custom header is in the local_route. > > So in the script I need the b2b logic ID after > running b2b_init_request("top hiding") but before the new INVITE is > sent to the B party as shown below. > > I am not sure how I can do that. > Inline image 2 > > Thanks, > Ping > > On Tue, Jun 7, 2016 at 12:07 AM, Bogdan-Andrei Iancu > > wrote: > > Hi Ping, > > My bad - in Call-ID and TO tag you have the B2B _entity_ ID, while > you need the B2B _logic_ ID .....which is not part of the > signaling at all . > > Now, in script, where do you need the b2b logic ID ? after > creating the B2B session (after b2b_init() ) ? Or ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 06.06.2016 11:46, Ping Han wrote: >> Thanks, Bogdan, >> >> I have done a few testing. It seems only the value of the param >> works for this function. However, it does not look like the >> callid and the to-tag. >> Is it possible to get the param value in the Opensips config? >> >> Please see the test results below. >> >> --------------------------------- >> [root at opensips-02 ~]# opensipsctl fifo b2be_list >> dlg:: 37 param=237.0 state=5 last_invite_cseq=1 last_method=0 >> last_reply_code=200 db_flag=2 >> ruri:: sip:2401012350 at 10.216.235.38:5060 >> >> callid:: NjEyZWEyOTEwZmVlZDIzOTViMTc3YjJiMmJmY2RjODI. >> from:: "Tropo test" >> uri=sip:2401012350 at 10.216.235.115:5060 >> tag=1e53cf61 >> to:: "2401012350" uri=sip:2401012350 at 10.216.235.115:5060 >> tag=B2B.297.37 >> cseq:: caller=1 callee=1 >> route_set:: >> caller=, >> contact:: caller=sip:2401012350 at 10.203.1.196:22238 >> >> callee=sip:10.216.235.72:5060 >> send_sock:: 10.216.235.72 >> dlg:: 6715348 param=237.0 state=5 last_invite_cseq=2 >> last_method=4 db_flag=2 >> callid:: B2B.237.6715348 >> from:: "Tropo test" >> uri=sip:2401012350 at 10.216.235.115:5060 >> >> tag=44623c403b25fd7905bfa7a7325b2b8f >> to:: uri=sip:dialog at 10.216.235.38:5060 >> tag=40711690 >> cseq:: caller=2 callee=1 >> contact:: caller=sip:10.216.235.72:5060 >> callee=sip:10.216.235.38:5060 >> >> send_sock:: 10.216.235.72 >> LEGS:: >> leg:: 0 tag=40711690 cseq=2 >> contact=sip:10.216.235.38:5060 >> >> [root at opensips-02 ~]# opensipsctl fifo b2b_bridge 237.6715348 >> sip:123 at 10.203.1.196 (this is >> the callid with out the "B2B" prefix) >> 500 command 'b2b_bridge' failed >> >> [root at opensips-02 ~]# opensipsctl fifo b2b_bridge 297.37 >> sip:123 at 10.203.1.196 (this is >> the To-tag) >> 500 command 'b2b_bridge' failed >> >> [root at opensips-02 ~]# opensipsctl fifo b2b_bridge 237.0 >> sip:123 at 10.203.1.196 (this is the >> value of param, only this works) >> >> --------------------------------- >> >> Regards, >> Ping >> >> >> On Fri, Jun 3, 2016 at 6:54 PM, Bogdan-Andrei Iancu >> > wrote: >> >> Hi Ping, >> >> b2b_bridge_request() is a script function: >> http://www.opensips.org/html/docs/modules/2.1.x/b2b_logic.html#id294094 >> >> I pointed to this function as you mentioned (on my question) >> that you want to do the bridging from script level. >> >> Indeed, the equivalent MI function is b2b_bridge: >> http://www.opensips.org/html/docs/modules/2.1.x/b2b_logic.html#id294210 >> >> If you want to get that "1020.30", you can get it from >> Call-ID or To tag, where you have B2B.1020.30 (so you have to >> strip that B2B prefix). >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> On 03.06.2016 06:27, Ping Han wrote: >>> Thanks, Bogdan, >>> >>> It seems the function you mentioned is the internal function >>> "1.4.2 b2b_bridge_request(b2bl_key,entity_no)". >>> >>> Actually the function that I am trying to use is the >>> "b2b_bridge" (Exported MI Functions). It is defined as below >>> ---------------------------------- >>> http://www.opensips.org/html/docs/modules/2.1.x/b2b_logic.html#id294210 >>> >>> >>> 1.5.2. b2b_bridge >>> >>> Example: opensipsctl b2b_bridge 1020.30 >>> sip:alice at opensips.org >>> ---------------------------------- >>> >>> In the example "1020.30" is the "dialog-id". This is the >>> parameter that I am not sure how to easily access in the >>> Opensips config. >>> >>> What I am trying to do is to get the value and deliver to >>> the next hop via a custom SIP header. When the next hop >>> tries to transfer the call to a new destination. It can run >>> the b2b_bridge command straight away with the "dialog-id" >>> without rechieving the value from the Opensips database >>> (from b2b_logic or b2b_entities tables). >>> >>> Any advice will be appreciated. >>> >>> Thanks, >>> Ping >>> >>> >>> On Thu, Jun 2, 2016 at 5:35 PM, Bogdan-Andrei Iancu >>> > wrote: >>> >>> Hi Ping, >>> >>> In script, in a b2b route, you can look at the callid or >>> TO tag (depending on the direction) to get the key : >>> http://www.opensips.org/html/docs/modules/2.2.x/b2b_logic.html#id294094 >>> >>> The B2B uses that key as Call-ID when acting as UAC and >>> as To tag when acting as UAS. You can run a SIP capture >>> to see the traffic. >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> OpenSIPS Founder and Developer >>> http://www.opensips-solutions.com >>> >>> On 02.06.2016 05:13, Ping Han wrote: >>>> Hi Bogdan, >>>> >>>> Thanks for the information. >>>> >>>> I need the value in the Opensips cfg. >>>> >>>> I understand that I can query the b2b_logic or >>>> b2b_entities tables to get the value in Opensips >>>> config. Apart from that could you tell me other way to >>>> easily access the value in Opensips config? >>>> >>>> Thanks, >>>> Ping >>>> >>>> >>>> >>>> On Wed, Jun 1, 2016 at 5:13 PM, Bogdan-Andrei Iancu >>>> > wrote: >>>> >>>> Hi Ping, >>>> >>>> Indeed, my bad - the docs are not updated, as that >>>> param was disabled long time ago (4 years ago): >>>> https://sourceforge.net/p/opensips/bugs/502/ >>>> >>>> Still, there are available option. But the question >>>> is : do you need that value in OpenSIPS cfg or >>>> outside OpenSIPS ? as there are different way to >>>> get the ID. >>>> >>>> Regards, >>>> >>>> Bogdan-Andrei Iancu >>>> OpenSIPS Founder and Developer >>>> http://www.opensips-solutions.com >>>> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 16785 bytes Desc: not available URL: From bogdan at opensips.org Tue Jun 7 12:47:00 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 7 Jun 2016 13:47:00 +0300 Subject: [OpenSIPS-Users] Continuous "WARNING:core:timer_ticker: timer task already scheduled" messages in log In-Reply-To: <352B7B4A-6668-4887-9F24-FD97771C7E3D@me.com> References: <352B7B4A-6668-4887-9F24-FD97771C7E3D@me.com> Message-ID: <5756A624.7090205@opensips.org> Hello Steve, What OpenSIPS tells you is that you the TM timer routine (which gets executed once per second) takes longer than 1 second (as execution). Probably you have retransmissions or many failure routes to be executed. You can increase the level of parallelism in TM timer via the timer_partition parameter: http://www.opensips.org/html/docs/modules/2.2.x/tm.html#id294483 Try to set it to 4. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 07.06.2016 04:21, Steve Woolley wrote: > Running opensips on a Raspberry Pi. At some point after starting > opensips (sometimes immediately, sometimes after quite a bit of time), > the following message fills the log ? once a second ? continuously. > > ... > Jun 7 01:10:41 pi1 /usr/local/sbin/opensips[22396]: > WARNING:core:timer_ticker: timer task already scheduled for > 7182080 ms (now 8258580 ms), it may overlap.. > Jun 7 01:10:42 pi1 /usr/local/sbin/opensips[22396]: > WARNING:core:timer_ticker: timer task already scheduled for > 7182080 ms (now 8259570 ms), it may overlap.. > Jun 7 01:10:43 pi1 /usr/local/sbin/opensips[22396]: > WARNING:core:timer_ticker: timer task already scheduled for > 7182080 ms (now 8260560 ms), it may overlap.. > Jun 7 01:10:44 pi1 /usr/local/sbin/opensips[22396]: > WARNING:core:timer_ticker: timer task already scheduled for > 7182080 ms (now 8261560 ms), it may overlap.. > Jun 7 01:10:45 pi1 /usr/local/sbin/opensips[22396]: > WARNING:core:timer_ticker: timer task already scheduled for > 7182080 ms (now 8262550 ms), it may overlap.. > Jun 7 01:10:46 pi1 /usr/local/sbin/opensips[22396]: > WARNING:core:timer_ticker: timer task already scheduled for > 7182080 ms (now 8263550 ms), it may overlap.. > Jun 7 01:10:47 pi1 /usr/local/sbin/opensips[22396]: > WARNING:core:timer_ticker: timer task already scheduled for > 7182080 ms (now 8264540 ms), it may overlap.. > Jun 7 01:10:48 pi1 /usr/local/sbin/opensips[22396]: > WARNING:core:timer_ticker: timer task already scheduled for > 7182080 ms (now 8265530 ms), it may overlap.. > Jun 7 01:10:49 pi1 /usr/local/sbin/opensips[22396]: > WARNING:core:timer_ticker: timer task already scheduled for > 7182080 ms (now 8266530 ms), it may overlap.. > Jun 7 01:10:50 pi1 /usr/local/sbin/opensips[22396]: > WARNING:core:timer_ticker: timer task already scheduled for > 7182080 ms (now 8267520 ms), it may overlap.. > > root at pi1:~# opensips -V > version: opensips 2.2.0 (arm6/linux) > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > QM_MALLOC, DBG_MALLOC, USE_PTHREAD_MUTEX > MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, > BUF_SIZE 65535 > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. > git revision: 0e1cea7 > main.c compiled on 21:56:59 Jun 3 2016 with gcc 4.9.2 > > Anyone experiencing the same? > > > -- > Steve Woolley > steve.woolley at me.com > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jun 7 12:52:01 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 7 Jun 2016 13:52:01 +0300 Subject: [OpenSIPS-Users] Opensips CP DROUTING GATEWAY enable/disable issue In-Reply-To: References: <574D3D03.1070608@opensips.org> Message-ID: <5756A751.8030205@opensips.org> Hi Jeff, So, FIFO works ok for you, but CP (using xmlrpc fails). Can you make a capture of the XMLRPC traffic between CP and OpenSIPS and post it somewhere ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 31.05.2016 17:27, Jeff Wilkie wrote: > > OpenSIPS Control Panel version 5.0 > > > Jeff Wilkie > Chief Technology Officer > US IP Communications > 919.297.1057 > > > /"This e-mail communication and any attachments may contain > confidential and privileged information and is for use by the > designated addressee(s) named above only. Any files transmitted with > it are confidential and intended solely for the use of the individual > to whom it is addressed. Any views or opinions presented are solely > those of the author and do not necessarily represent those of USIPCOM, > LLC. If you are not the intended addressee, you are hereby notified > that you have received this communication in error and that any use or > reproduction of this email or its contents is strictly prohibited and > may be unlawful. If you have received this communication in error, > please notify us immediately by replying to this message and deleting > it from your computer. Thank you". / > > On Tue, May 31, 2016 at 3:28 AM, Bogdan-Andrei Iancu > > wrote: > > Hi Jeff, > > What OpenSIPS Control Panel version are you using ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 21.05.2016 00:15, Jeff Wilkie wrote: >> OPENSIPS 1.10.x >> I have the following set >> >> opensips.cfg >> >> #### FIFO Management Interface >> >> loadmodule "mi_fifo.so" >> >> modparam("mi_fifo", "fifo_name", "/tmp/opensips_proxy_fifo") >> >> modparam("mi_fifo", "fifo_mode", 0666) >> >> >> opensipsctlrc >> >> ## path to FIFO file >> >> OSIPS_FIFO="/tmp/opensips_proxy_fifo" >> >> >> Attempting to disable gateways via the CP gives the following errors: >> >> >> From DROUTING-Gateway interface: >> >> >> Error while disabling gateway 2 >> (the GWID is 2 for the gateway I'm attempting to disable) >> >> From the MI Commands: >> Initiating the following command: *dr_gw_status 2 0* >> 404 GW ID not found >> >> From the DROUTING-Gateway interface I am able to enable the >> interface if it is disabled >> I'm able to also enable the Gateway from the MI Commands section. >> >> I'm also able to enable and disable the Gateway using opensipsctl >> fifo commands >> >> opensipsctl fifo dr_gw_status 2 >> >> Enabled:: yes >> >> opensipsctl fifo dr_gw_status 2 0 >> >> opensipsctl fifo dr_gw_status 2 >> >> Enabled:: no >> >> opensipsctl fifo dr_gw_status 2 1 >> >> opensipsctl fifo dr_gw_status 2 >> >> Enabled:: yes >> >> >> Not sure where the problem is but I feel its somewhere in the >> syntax of how its delivered. I'm sure it's something easy I've >> overlooked. Any help on this? >> >> Thanks >> Jeff >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From john.nash778 at gmail.com Tue Jun 7 12:53:54 2016 From: john.nash778 at gmail.com (John Nash) Date: Tue, 7 Jun 2016 16:23:54 +0530 Subject: [OpenSIPS-Users] Register with TO Tag In-Reply-To: <5756A4D1.60400@opensips.org> References: <5756A4D1.60400@opensips.org> Message-ID: OK that means I should handle Register before In-dialog processing block? I also have one more doubt function mf_process_maxfwd_header should it be used before sipmsg_validate or after?...Currently mf_process_maxfwd_header is being called in my script first but in some cases with malformed packets its not even able to read max fwd header. On Tue, Jun 7, 2016 at 4:11 PM, Bogdan-Andrei Iancu wrote: > Hi John, > > Assuming you do not do REGISTER relay (but you act as a registrar), you > should handle the REGISTER requests (with or without to-tag) in the same > way. IF they have a Route hdr , it may be because they do pre-loaded route > (the Route points to your SIP server) to be sure the REGISTER gets to the > registrar server. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 07.06.2016 10:56, John Nash wrote: > > I am dealing with In-dialog requests using > > > ------------------------------------------------------------------------------------------------------ > if (has_totag() && (is_domain_local("$rd") || $Ri== "127.0.0.1") && > is_method("INVITE|ACK|BYE|UPDATE")) > { > # sequential request within a dialog should > # take the path determined by record-routing > if (topology_hiding_match()) > ----- > > --------------------------------------------------------------------------------------------------------- > > at the top of my script. After that I process initial requests, but I see > some REGISTER messages with TO-Tag and "Route" header and they are being > discarded by my script because Initial request cannot have Route header. > > Do i also need to pass REGISTER messages also through same block?...or i > need to call loose_route after has_to_tag check. > > > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jun 7 13:01:00 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 7 Jun 2016 14:01:00 +0300 Subject: [OpenSIPS-Users] Register with TO Tag In-Reply-To: References: <5756A4D1.60400@opensips.org> Message-ID: <5756A96C.8030103@opensips.org> REGISTER and INVITE requests should be handled in different ways, so split your scripting per methods. In regards, to MF, better do the validation first, to be sure that whatever you do later, at least you have a valid SIP msg. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 07.06.2016 13:53, John Nash wrote: > OK that means I should handle Register before In-dialog processing block? > > I also have one more doubt function mf_process_maxfwd_header should it > be used before sipmsg_validate or after?...Currently > mf_process_maxfwd_header is being called in my script first but in > some cases with malformed packets its not even able to read max fwd > header. > > On Tue, Jun 7, 2016 at 4:11 PM, Bogdan-Andrei Iancu > > wrote: > > Hi John, > > Assuming you do not do REGISTER relay (but you act as a > registrar), you should handle the REGISTER requests (with or > without to-tag) in the same way. IF they have a Route hdr , it may > be because they do pre-loaded route (the Route points to your SIP > server) to be sure the REGISTER gets to the registrar server. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 07.06.2016 10:56, John Nash wrote: >> I am dealing with In-dialog requests using >> >> ------------------------------------------------------------------------------------------------------ >> if (has_totag() && (is_domain_local("$rd") || $Ri== >> "127.0.0.1") && is_method("INVITE|ACK|BYE|UPDATE")) >> { >> # sequential request within a dialog should >> # take the path determined by record-routing >> if (topology_hiding_match()) >> ----- >> --------------------------------------------------------------------------------------------------------- >> >> at the top of my script. After that I process initial requests, >> but I see some REGISTER messages with TO-Tag and "Route" header >> and they are being discarded by my script because Initial request >> cannot have Route header. >> >> Do i also need to pass REGISTER messages also through same >> block?...or i need to call loose_route after has_to_tag check. >> >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Tue Jun 7 13:11:52 2016 From: liviu at opensips.org (Liviu Chircu) Date: Tue, 7 Jun 2016 14:11:52 +0300 Subject: [OpenSIPS-Users] Opensips -1.11 crash with mongodb replica set . In-Reply-To: References: <575537DA.60002@opensips.org> <57555A4E.8000502@opensips.org> Message-ID: <5756ABF8.7080508@opensips.org> To avoid those kind of problems, I would recommend putting one "mongos" instance for each OpenSIPS machine, on the same box. This way, not only do you avoid problems with the mongos machine going down, but you'll speed up query times as well, making your OpenSIPS workers more responsive! Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 07.06.2016 13:20, Sasmita Panda wrote: > Hi Liviu , > > I just have one question . I have implemented opensips with > mongodb shard cluster . I have one replica set with three member . I > have three config server for each and mongos service running . > > What I wanted to know is , the server in which my mongos service > is running , If that machine is get down what will happen ? > I need to monitor that machine so that it wont go down ? Is this show ? > > */Thanks & Regards/* > /Sasmita Panda/ > /Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > On Mon, Jun 6, 2016 at 4:41 PM, Liviu Chircu > wrote: > > Thank you, Sasmita! Will reply here as soon as there is progress > on the matter. > > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > > On 06.06.2016 13:55, Sasmita Panda wrote: >> Hi , >> >> I have created an issue with detailed steps . Bellow is the >> issue ID . Let me know if the issue will get fixed . >> >> https://github.com/OpenSIPS/opensips/issues/895 >> >> Thanks >> >> */Thanks & Regards/* >> /Sasmita Panda/ >> /Network Testing and Software Engineer/ >> /3CLogic , ph:07827611765/ >> >> On Mon, Jun 6, 2016 at 3:32 PM, Sasmita Panda > > wrote: >> >> Thank you so much for the information . Let me try to deploy >> with mongos server . >> >> I will post the detailed steps for opensips crash as soon as >> I will be free . >> >> */Thanks & Regards/* >> /Sasmita Panda/ >> /Network Testing and Software Engineer/ >> /3CLogic , ph:07827611765/ >> >> On Mon, Jun 6, 2016 at 2:14 PM, Liviu Chircu >> > wrote: >> >> Hi Sasmita! >> >> We have some plans to do some more testing on the module, >> including your "connect directly to replica set" usage >> case, and also bringing it up to date with the latest >> mongo driver. However, it is hard to give you an >> estimation for when this work will start. >> >> Currently, the fastest way to fix your problems is for >> you to set up a "mongos" instance, along with 3 config >> server instances (these are needed by mongos, and ensure >> proper write consistency), and configure >> "cachedb_mongodb" to use this new "mongos" node. >> >> A mongos instance is aware of all the replica sets (aka >> "shards"). In your case, it will initially shard the keys >> to your single replica set - thus, it will hold 100% of >> the data. As your data set grows, you may partition it >> over to additional replica sets, which you can configure >> into mongos without modifying anything on the OpenSIPS >> side. This way, failover within a replica set will also >> happen transparently, again, without impacting OpenSIPS >> at all. >> >> PS: if you have a way to 100% reproduce a crash, please >> open a GH ticket describing the steps, and we'll go from >> there! [1] >> >> [1]: https://github.com/OpenSIPS/opensips/issues >> >> Liviu Chircu >> OpenSIPS Developer >> http://www.opensips-solutions.com >> >> On 06.06.2016 11:23, Sasmita Panda wrote: >>> Hi All , >>> >>> I am using opesips-1.11 with mongodb replica set >>> . I have 3 members in the replica set , promary , >>> secondary and arbitrary . >>> >>> Problem 1 : Sometime , If primay is not rechable >>> , the secondary becomes primary , But opensips loss >>> connection from mongodb . Its wont put any data in the >>> data base . My call goes on but their is not data in the >>> mongodb database . >>> >>> Problme 2: If the primay machine is down then >>> secondary becomes primary within some millisecond time >>> , but opensips crashes giving bellow error >>> >>> ERROR:cachedb_mongodb:mongo_con_get: Failed to run >>> query. Err = 6, 0 , 0 >>> CRITICAL:core:receive_fd: EOF on 10 >>> INFO:core:handle_sigs: child process 5278 exited by a >>> signal 11 >>> INFO:core:handle_sigs: core was generated >>> INFO:core:handle_sigs: terminating due to SIGCHLD >>> INFO:core:sig_usr: signal 15 received >>> INFO:core:sig_usr: signal 15 received >>> INFO:core:sig_usr: signal 15 received >>> INFO:core:sig_usr: signal 15 received >>> INFO:core:sig_usr: signal 15 received >>> INFO:core:sig_usr: signal 15 received >>> INFO:core:sig_usr: signal 15 received >>> INFO:core:sig_usr: signal 15 received >>> INFO:core:sig_usr: signal 15 received >>> INFO:core:sig_usr: signal 15 received >>> INFO:core:sig_usr: signal 15 received >>> INFO:core:sig_usr: signal 15 received >>> INFO:core:sig_usr: signal 15 received >>> INFO:core:cleanup: cleanup >>> NOTICE:db_cachedb:destroy: destroy module db_cachedb ... >>> NOTICE:cachedb_mongodb:destroy: destroy module >>> cachedb_mongodb ... >>> >>> If my secondary mongodb machine is not reachable >>> then rather connection with db breaks for sometime but >>> opensips wont crashes . But in case of Primary opensips >>> crashes with above error . Is this an expected behavior >>> or I am doing anything wrong . >>> Data loss can be bearable but application cant be >>> . So please let me know whats the problem . >>> >>> */Thanks & Regards/* >>> /Sasmita Panda/ >>> /Network Testing and Software Engineer/ >>> /3CLogic , ph:07827611765/ >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jun 7 13:22:38 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 7 Jun 2016 14:22:38 +0300 Subject: [OpenSIPS-Users] Available OpenSIPS releases ad their lifetime Message-ID: <5756AE7E.40602@opensips.org> Hello, Following the release of OpenSIPS 2.2.0, here is the current status of the available (maintained) releases and the planned expiration date. OpenSIPS 1.11 LTS - to be maintained until 7th of May 2017 OpenSIPS 2.1 - to be maintained until 7th of November 2017 OpenSIPS 2.2 LTS - to be maintained until 27th of May 2019 Best regards, -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com From spanda at 3clogic.com Tue Jun 7 13:35:29 2016 From: spanda at 3clogic.com (Sasmita Panda) Date: Tue, 7 Jun 2016 17:05:29 +0530 Subject: [OpenSIPS-Users] Opensips -1.11 crash with mongodb replica set . In-Reply-To: <5756ABF8.7080508@opensips.org> References: <575537DA.60002@opensips.org> <57555A4E.8000502@opensips.org> <5756ABF8.7080508@opensips.org> Message-ID: Thank you so much . It will be helpful for me . *Thanks & Regards* *Sasmita Panda* *Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Tue, Jun 7, 2016 at 4:41 PM, Liviu Chircu wrote: > To avoid those kind of problems, I would recommend putting one "mongos" > instance for each OpenSIPS machine, on the same box. This way, not only do > you avoid problems with the mongos machine going down, but you'll speed up > query times as well, making your OpenSIPS workers more responsive! > > Liviu Chircu > OpenSIPS Developerhttp://www.opensips-solutions.com > > On 07.06.2016 13:20, Sasmita Panda wrote: > > Hi Liviu , > > I just have one question . I have implemented opensips with mongodb > shard cluster . I have one replica set with three member . I have three > config server for each and mongos service running . > > What I wanted to know is , the server in which my mongos service is > running , If that machine is get down what will happen ? > I need to monitor that machine so that it wont go down ? Is this show ? > > *Thanks & Regards* > *Sasmita Panda* > *Network Testing and Software Engineer* > *3CLogic , ph:07827611765* > > On Mon, Jun 6, 2016 at 4:41 PM, Liviu Chircu wrote: > >> Thank you, Sasmita! Will reply here as soon as there is progress on the >> matter. >> >> Liviu Chircu >> OpenSIPS Developerhttp://www.opensips-solutions.com >> >> On 06.06.2016 13:55, Sasmita Panda wrote: >> >> Hi , >> >> I have created an issue with detailed steps . Bellow is the issue ID . >> Let me know if the issue will get fixed . >> >> https://github.com/OpenSIPS/opensips/issues/895 >> >> Thanks >> >> *Thanks & Regards* >> *Sasmita Panda* >> *Network Testing and Software Engineer* >> *3CLogic , ph:07827611765* >> >> On Mon, Jun 6, 2016 at 3:32 PM, Sasmita Panda < >> spanda at 3clogic.com> wrote: >> >>> Thank you so much for the information . Let me try to deploy with mongos >>> server . >>> >>> I will post the detailed steps for opensips crash as soon as I will be >>> free . >>> >>> *Thanks & Regards* >>> *Sasmita Panda* >>> *Network Testing and Software Engineer* >>> *3CLogic , ph:07827611765* >>> >>> On Mon, Jun 6, 2016 at 2:14 PM, Liviu Chircu < >>> liviu at opensips.org> wrote: >>> >>>> Hi Sasmita! >>>> >>>> We have some plans to do some more testing on the module, including >>>> your "connect directly to replica set" usage case, and also bringing it up >>>> to date with the latest mongo driver. However, it is hard to give you an >>>> estimation for when this work will start. >>>> >>>> Currently, the fastest way to fix your problems is for you to set up a >>>> "mongos" instance, along with 3 config server instances (these are needed >>>> by mongos, and ensure proper write consistency), and configure >>>> "cachedb_mongodb" to use this new "mongos" node. >>>> >>>> A mongos instance is aware of all the replica sets (aka "shards"). In >>>> your case, it will initially shard the keys to your single replica set - >>>> thus, it will hold 100% of the data. As your data set grows, you may >>>> partition it over to additional replica sets, which you can configure into >>>> mongos without modifying anything on the OpenSIPS side. This way, failover >>>> within a replica set will also happen transparently, again, without >>>> impacting OpenSIPS at all. >>>> >>>> PS: if you have a way to 100% reproduce a crash, please open a GH >>>> ticket describing the steps, and we'll go from there! [1] >>>> >>>> [1]: https://github.com/OpenSIPS/opensips/issues >>>> >>>> Liviu Chircu >>>> OpenSIPS Developerhttp://www.opensips-solutions.com >>>> >>>> On 06.06.2016 11:23, Sasmita Panda wrote: >>>> >>>> Hi All , >>>> >>>> I am using opesips-1.11 with mongodb replica set . I have 3 >>>> members in the replica set , promary , secondary and arbitrary . >>>> >>>> Problem 1 : Sometime , If primay is not rechable , the secondary >>>> becomes primary , But opensips loss connection from mongodb . Its wont put >>>> any data in the data base . My call goes on but their is not data in the >>>> mongodb database . >>>> >>>> Problme 2: If the primay machine is down then secondary becomes >>>> primary within some millisecond time , but opensips crashes giving bellow >>>> error >>>> >>>> ERROR:cachedb_mongodb:mongo_con_get: Failed to run query. Err = >>>> 6, 0 , 0 >>>> CRITICAL:core:receive_fd: EOF on 10 >>>> INFO:core:handle_sigs: child process 5278 exited by a signal 11 >>>> INFO:core:handle_sigs: core was generated >>>> INFO:core:handle_sigs: terminating due to SIGCHLD >>>> INFO:core:sig_usr: signal 15 received >>>> INFO:core:sig_usr: signal 15 received >>>> INFO:core:sig_usr: signal 15 received >>>> INFO:core:sig_usr: signal 15 received >>>> INFO:core:sig_usr: signal 15 received >>>> INFO:core:sig_usr: signal 15 received >>>> INFO:core:sig_usr: signal 15 received >>>> INFO:core:sig_usr: signal 15 received >>>> INFO:core:sig_usr: signal 15 received >>>> INFO:core:sig_usr: signal 15 received >>>> INFO:core:sig_usr: signal 15 received >>>> INFO:core:sig_usr: signal 15 received >>>> INFO:core:sig_usr: signal 15 received >>>> INFO:core:cleanup: cleanup >>>> NOTICE:db_cachedb:destroy: destroy module db_cachedb ... >>>> NOTICE:cachedb_mongodb:destroy: destroy module cachedb_mongodb ... >>>> >>>> If my secondary mongodb machine is not reachable then rather >>>> connection with db breaks for sometime but opensips wont crashes . But in >>>> case of Primary opensips crashes with above error . Is this an expected >>>> behavior or I am doing anything wrong . >>>> >>>> Data loss can be bearable but application cant be . So please let >>>> me know whats the problem . >>>> >>>> *Thanks & Regards* >>>> *Sasmita Panda* >>>> *Network Testing and Software Engineer* >>>> *3CLogic , ph:07827611765* >>>> >>>> >>>> _______________________________________________ >>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>> >> >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From pimenta at inatel.br Tue Jun 7 14:04:15 2016 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Tue, 7 Jun 2016 12:04:15 +0000 Subject: [OpenSIPS-Users] Continuous "WARNING:core:timer_ticker: timer task already scheduled" messages in log In-Reply-To: <352B7B4A-6668-4887-9F24-FD97771C7E3D@me.com> References: <352B7B4A-6668-4887-9F24-FD97771C7E3D@me.com> Message-ID: Hi Steve. I had exactly this same problem! I used to see that log very frequently. So, I changed a configuration in my opensips.cfg file to be: modparam("usrloc", "db_mode", 2) modparam("usrloc", "timer_interval",3) Then, the problem almost disappeared. That is, nowadays I see this log seldom. Tell me about the solution for your case, when you have fixed it, please! I also gonna to consider the last hint from Bogdan and change my TM parameter. Best regards! RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________ De: users-bounces at lists.opensips.org em nome de Steve Woolley Enviado: segunda-feira, 6 de junho de 2016 22:21 Para: users at lists.opensips.org Assunto: [OpenSIPS-Users] Continuous "WARNING:core:timer_ticker: timer task already scheduled" messages in log Running opensips on a Raspberry Pi. At some point after starting opensips (sometimes immediately, sometimes after quite a bit of time), the following message fills the log - once a second - continuously. ... Jun 7 01:10:41 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8258580 ms), it may overlap.. Jun 7 01:10:42 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8259570 ms), it may overlap.. Jun 7 01:10:43 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8260560 ms), it may overlap.. Jun 7 01:10:44 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8261560 ms), it may overlap.. Jun 7 01:10:45 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8262550 ms), it may overlap.. Jun 7 01:10:46 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8263550 ms), it may overlap.. Jun 7 01:10:47 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8264540 ms), it may overlap.. Jun 7 01:10:48 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8265530 ms), it may overlap.. Jun 7 01:10:49 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8266530 ms), it may overlap.. Jun 7 01:10:50 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8267520 ms), it may overlap.. root at pi1:~# opensips -V version: opensips 2.2.0 (arm6/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, QM_MALLOC, DBG_MALLOC, USE_PTHREAD_MUTEX MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. git revision: 0e1cea7 main.c compiled on 21:56:59 Jun 3 2016 with gcc 4.9.2 Anyone experiencing the same? -- Steve Woolley steve.woolley at me.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jun 7 15:15:06 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 7 Jun 2016 16:15:06 +0300 Subject: [OpenSIPS-Users] IS_MYSELF() always uses 5060 if received port is 0 In-Reply-To: References: <573B3E17.6020200@opensips.org> Message-ID: <5756C8DA.5090701@opensips.org> Hi, The patch is part of the latest release on 1.11 branch, the 1.11.7 version. Here is the commits for that: https://github.com/OpenSIPS/opensips/commit/f91441d43e814d06ebf325e637206be411112879 https://github.com/OpenSIPS/opensips/commit/d39cfb73ed01b349c235949a16ccc5b559003a81 Apply them both ! Or simply use 1.11.7 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 07.06.2016 16:10, Ravitez Ravi wrote: > Hello Bogdan, > Good Morning, > I missed to updated that we are using 1.11.5 and > not sure if the patch was targeted for that. > can you please share a patch for "OpenSIPS > (1.11.5-tls (x86_64/linux))" > Thank you for the help. > > Regards, > Ravitez.D > > On Tue, May 17, 2016 at 11:51 AM, Bogdan-Andrei Iancu > > wrote: > > Hi, > > That is a great catch, thank you for finding and reporting this. > See the attach patch that should address the problem. Could you > please give it a try to see if it really solves the problem ? > > Best Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 13.05.2016 22:30, Ravitez Ravi wrote: >> Hi All, >> Good Day, >> Here's the problem i'm facing and would be a great help >> if you could comment. >> Thank you. >> >> is_myself() does not check for SIPS port if connection type is TLS >> *Configuration :* >> - Opensips V1.11.5 running in secure mode on port 5061 >> - Avaya trying to communicate with Opensips server. >> - Opensips server ip 192.168.1.11 >> - Avaya ip : 192.168.1.20 >> >> >> *Steps :* >> - Avaya sends INVITE to Opensips with route header >> Route: >> - Opensips tries to process it but fails. >> *DBG:rr:is_preloaded: is_preloaded: Yes* >> *DBG:core:grep_sock_info: checking if host==us: 14==14 && >> [192.168.1.11] == [192.168.1.11]* >> *DBG:core:grep_sock_info: checking if port 5061 matches port 5060* >> *DBG:core:check_self: host != me* >> *DBG:rr:after_loose: Topmost URI is NOT myself* >> .... >> .... >> .... >> SIP/2.0 403 Preload Route denied >> *Code Snippet :* >> /* >> * Check if URI is myself >> */ >> #ifdef ENABLE_USER_CHECK >> static inline int is_myself(str *_user, str* _host, unsigned >> short _port) >> #else >> static inline int is_myself(str* _host, unsigned short _port) >> #endif >> { >> int ret; >> >> *ret = check_self(_host, _port ? _port : SIP_PORT, 0);/* match >> all protos*/* >> if (ret < 0) return 0; >> >> *Should is_myself() check for connection type and then decide to >> either use SIP or SIPS port.* >> * >> * >> * >> * >> * >> * >> Regards, >> Ravitez.D >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From pinghan022 at gmail.com Tue Jun 7 15:15:54 2016 From: pinghan022 at gmail.com (Ping Han) Date: Tue, 7 Jun 2016 23:15:54 +1000 Subject: [OpenSIPS-Users] Question regarding b2b_bridge function In-Reply-To: <5756A333.1010701@opensips.org> References: <574D44F7.9090204@opensips.org> <574E8B1A.7070508@opensips.org> <574FE1C0.9000904@opensips.org> <575145C9.9080406@opensips.org> <575583A1.7050501@opensips.org> <5756A333.1010701@opensips.org> Message-ID: Hi Bogdan, Yes, that is exactly what I am trying to do. Regards, Ping On Tue, Jun 7, 2016 at 8:34 PM, Bogdan-Andrei Iancu wrote: > Hi Ping, > > So you need the B2B Logic ID only when you create the B2B session, in > order to place it in the first outgoing INVITE (as extra hdr), right ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 07.06.2016 04:59, Ping Han wrote: > > Thanks, Bogdan, > > What I am trying to do is to get the b2b logic ID and deliver to the next > hop via a custom SIP header. > > I am using the b2b_init_request("top hiding") and it looks like the only > place that I can add a custom header is in the local_route. > > So in the script I need the b2b logic ID after running b2b_init_request("top > hiding") but before the new INVITE is sent to the B party as shown below. > > I am not sure how I can do that. > [image: Inline image 2] > > Thanks, > Ping > > On Tue, Jun 7, 2016 at 12:07 AM, Bogdan-Andrei Iancu < > bogdan at opensips.org> wrote: > >> Hi Ping, >> >> My bad - in Call-ID and TO tag you have the B2B _entity_ ID, while you >> need the B2B _logic_ ID .....which is not part of the signaling at all . >> >> Now, in script, where do you need the b2b logic ID ? after creating the >> B2B session (after b2b_init() ) ? Or ? >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com >> >> On 06.06.2016 11:46, Ping Han wrote: >> >> Thanks, Bogdan, >> >> I have done a few testing. It seems only the value of the param works for >> this function. However, it does not look like the callid and the to-tag. >> Is it possible to get the param value in the Opensips config? >> >> Please see the test results below. >> >> --------------------------------- >> [root at opensips-02 ~]# opensipsctl fifo b2be_list >> dlg:: 37 param=237.0 state=5 last_invite_cseq=1 last_method=0 >> last_reply_code=200 db_flag=2 >> ruri:: >> sip:2401012350 at 10.216.235.38:5060 >> callid:: NjEyZWEyOTEwZmVlZDIzOTViMTc3YjJiMmJmY2RjODI. >> from:: "Tropo test" uri=sip:2401012350 at 10.216.235.115:5060 >> tag=1e53cf61 >> to:: "2401012350" uri= >> sip:2401012350 at 10.216.235.115:5060 tag=B2B.297.37 >> cseq:: caller=1 callee=1 >> route_set:: >> caller=, >> contact:: caller= >> sip:2401012350 at 10.203.1.196:22238 callee=sip:10.216.235.72:5060 >> send_sock:: 10.216.235.72 >> dlg:: 6715348 param=237.0 state=5 last_invite_cseq=2 last_method=4 >> db_flag=2 >> callid:: B2B.237.6715348 >> from:: "Tropo test" uri=sip:2401012350 at 10.216.235.115:5060 >> tag=44623c403b25fd7905bfa7a7325b2b8f >> to:: uri= >> sip:dialog at 10.216.235.38:5060 tag=40711690 >> cseq:: caller=2 callee=1 >> contact:: caller=sip:10.216.235.72:5060 callee=sip: >> 10.216.235.38:5060 >> send_sock:: 10.216.235.72 >> LEGS:: >> leg:: 0 tag=40711690 cseq=2 contact=sip: >> 10.216.235.38:5060 >> >> [root at opensips-02 ~]# opensipsctl fifo b2b_bridge 237.6715348 >> sip:123 at 10.203.1.196 (this is the callid with >> out the "B2B" prefix) >> 500 command 'b2b_bridge' failed >> >> [root at opensips-02 ~]# opensipsctl fifo b2b_bridge 297.37 >> sip:123 at 10.203.1.196 (this is the To-tag) >> 500 command 'b2b_bridge' failed >> >> [root at opensips-02 ~]# opensipsctl fifo b2b_bridge 237.0 >> sip:123 at 10.203.1.196 (this is the value of param, >> only this works) >> >> --------------------------------- >> >> Regards, >> Ping >> >> >> On Fri, Jun 3, 2016 at 6:54 PM, Bogdan-Andrei Iancu < >> bogdan at opensips.org> wrote: >> >>> Hi Ping, >>> >>> b2b_bridge_request() is a script function: >>> >>> http://www.opensips.org/html/docs/modules/2.1.x/b2b_logic.html#id294094 >>> >>> I pointed to this function as you mentioned (on my question) that you >>> want to do the bridging from script level. >>> >>> Indeed, the equivalent MI function is b2b_bridge: >>> >>> http://www.opensips.org/html/docs/modules/2.1.x/b2b_logic.html#id294210 >>> >>> If you want to get that "1020.30", you can get it from Call-ID or To >>> tag, where you have B2B.1020.30 (so you have to strip that B2B prefix). >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com >>> >>> On 03.06.2016 06:27, Ping Han wrote: >>> >>> Thanks, Bogdan, >>> >>> It seems the function you mentioned is the internal function "1.4.2 >>> b2b_bridge_request(b2bl_key,entity_no)". >>> >>> Actually the function that I am trying to use is the "b2b_bridge" >>> (Exported MI Functions). It is defined as below >>> ---------------------------------- >>> http://www.opensips.org/html/docs/modules/2.1.x/b2b_logic.html#id294210 >>> >>> >>> 1.5.2. b2b_bridge >>> >>> Example: opensipsctl b2b_bridge 1020.30 >>> sip:alice at opensips.org >>> ---------------------------------- >>> >>> In the example "1020.30" is the "dialog-id". This is the parameter that >>> I am not sure how to easily access in the Opensips config. >>> >>> What I am trying to do is to get the value and deliver to the next hop >>> via a custom SIP header. When the next hop tries to transfer the call to a >>> new destination. It can run the b2b_bridge command straight away with the >>> "dialog-id" without rechieving the value from the Opensips database (from >>> b2b_logic or b2b_entities tables). >>> >>> Any advice will be appreciated. >>> >>> Thanks, >>> Ping >>> >>> >>> On Thu, Jun 2, 2016 at 5:35 PM, Bogdan-Andrei Iancu < >>> bogdan at opensips.org> wrote: >>> >>>> Hi Ping, >>>> >>>> In script, in a b2b route, you can look at the callid or TO tag >>>> (depending on the direction) to get the key : >>>> >>>> >>>> http://www.opensips.org/html/docs/modules/2.2.x/b2b_logic.html#id294094 >>>> >>>> The B2B uses that key as Call-ID when acting as UAC and as To tag when >>>> acting as UAS. You can run a SIP capture to see the traffic. >>>> >>>> Regards, >>>> >>>> Bogdan-Andrei Iancu >>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com >>>> >>>> On 02.06.2016 05:13, Ping Han wrote: >>>> >>>> Hi Bogdan, >>>> >>>> Thanks for the information. >>>> >>>> I need the value in the Opensips cfg. >>>> >>>> I understand that I can query the b2b_logic or b2b_entities tables to >>>> get the value in Opensips config. Apart from that could you tell me other >>>> way to easily access the value in Opensips config? >>>> >>>> Thanks, >>>> Ping >>>> >>>> >>>> >>>> On Wed, Jun 1, 2016 at 5:13 PM, Bogdan-Andrei Iancu < >>>> bogdan at opensips.org> wrote: >>>> >>>>> Hi Ping, >>>>> >>>>> Indeed, my bad - the docs are not updated, as that param was disabled >>>>> long time ago (4 years ago): >>>>> >>>>> https://sourceforge.net/p/opensips/bugs/502/ >>>>> >>>>> Still, there are available option. But the question is : do you need >>>>> that value in OpenSIPS cfg or outside OpenSIPS ? as there are different way >>>>> to get the ID. >>>>> >>>>> Regards, >>>>> >>>>> Bogdan-Andrei Iancu >>>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com >>>>> >>>>> >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 16785 bytes Desc: not available URL: From spanda at 3clogic.com Wed Jun 8 09:25:16 2016 From: spanda at 3clogic.com (Sasmita Panda) Date: Wed, 8 Jun 2016 12:55:16 +0530 Subject: [OpenSIPS-Users] Opensips -1.11 crash with mongodb replica set . In-Reply-To: References: <575537DA.60002@opensips.org> <57555A4E.8000502@opensips.org> <5756ABF8.7080508@opensips.org> Message-ID: Hi Liviu , What are you suggesting for the mongodb config server ? Because confg server is the one who contains the metadata of the shard cluster . Where should I put it ? Will I put this separately or in the opensips box with mongos ? If I will put the config servers separately and each mongos instance will get connected to the same set of config server then will it be fruitful or not ? *Thanks & Regards* *Sasmita Panda* *Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Tue, Jun 7, 2016 at 5:05 PM, Sasmita Panda wrote: > Thank you so much . It will be helpful for me . > > *Thanks & Regards* > *Sasmita Panda* > *Network Testing and Software Engineer* > *3CLogic , ph:07827611765* > > On Tue, Jun 7, 2016 at 4:41 PM, Liviu Chircu wrote: > >> To avoid those kind of problems, I would recommend putting one "mongos" >> instance for each OpenSIPS machine, on the same box. This way, not only do >> you avoid problems with the mongos machine going down, but you'll speed up >> query times as well, making your OpenSIPS workers more responsive! >> >> Liviu Chircu >> OpenSIPS Developerhttp://www.opensips-solutions.com >> >> On 07.06.2016 13:20, Sasmita Panda wrote: >> >> Hi Liviu , >> >> I just have one question . I have implemented opensips with mongodb >> shard cluster . I have one replica set with three member . I have three >> config server for each and mongos service running . >> >> What I wanted to know is , the server in which my mongos service is >> running , If that machine is get down what will happen ? >> I need to monitor that machine so that it wont go down ? Is this show ? >> >> *Thanks & Regards* >> *Sasmita Panda* >> *Network Testing and Software Engineer* >> *3CLogic , ph:07827611765* >> >> On Mon, Jun 6, 2016 at 4:41 PM, Liviu Chircu wrote: >> >>> Thank you, Sasmita! Will reply here as soon as there is progress on the >>> matter. >>> >>> Liviu Chircu >>> OpenSIPS Developerhttp://www.opensips-solutions.com >>> >>> On 06.06.2016 13:55, Sasmita Panda wrote: >>> >>> Hi , >>> >>> I have created an issue with detailed steps . Bellow is the issue ID . >>> Let me know if the issue will get fixed . >>> >>> https://github.com/OpenSIPS/opensips/issues/895 >>> >>> Thanks >>> >>> *Thanks & Regards* >>> *Sasmita Panda* >>> *Network Testing and Software Engineer* >>> *3CLogic , ph:07827611765* >>> >>> On Mon, Jun 6, 2016 at 3:32 PM, Sasmita Panda < >>> spanda at 3clogic.com> wrote: >>> >>>> Thank you so much for the information . Let me try to deploy with >>>> mongos server . >>>> >>>> I will post the detailed steps for opensips crash as soon as I will be >>>> free . >>>> >>>> *Thanks & Regards* >>>> *Sasmita Panda* >>>> *Network Testing and Software Engineer* >>>> *3CLogic , ph:07827611765* >>>> >>>> On Mon, Jun 6, 2016 at 2:14 PM, Liviu Chircu < >>>> liviu at opensips.org> wrote: >>>> >>>>> Hi Sasmita! >>>>> >>>>> We have some plans to do some more testing on the module, including >>>>> your "connect directly to replica set" usage case, and also bringing it up >>>>> to date with the latest mongo driver. However, it is hard to give you an >>>>> estimation for when this work will start. >>>>> >>>>> Currently, the fastest way to fix your problems is for you to set up a >>>>> "mongos" instance, along with 3 config server instances (these are needed >>>>> by mongos, and ensure proper write consistency), and configure >>>>> "cachedb_mongodb" to use this new "mongos" node. >>>>> >>>>> A mongos instance is aware of all the replica sets (aka "shards"). In >>>>> your case, it will initially shard the keys to your single replica set - >>>>> thus, it will hold 100% of the data. As your data set grows, you may >>>>> partition it over to additional replica sets, which you can configure into >>>>> mongos without modifying anything on the OpenSIPS side. This way, failover >>>>> within a replica set will also happen transparently, again, without >>>>> impacting OpenSIPS at all. >>>>> >>>>> PS: if you have a way to 100% reproduce a crash, please open a GH >>>>> ticket describing the steps, and we'll go from there! [1] >>>>> >>>>> [1]: https://github.com/OpenSIPS/opensips/issues >>>>> >>>>> Liviu Chircu >>>>> OpenSIPS Developerhttp://www.opensips-solutions.com >>>>> >>>>> On 06.06.2016 11:23, Sasmita Panda wrote: >>>>> >>>>> Hi All , >>>>> >>>>> I am using opesips-1.11 with mongodb replica set . I have 3 >>>>> members in the replica set , promary , secondary and arbitrary . >>>>> >>>>> Problem 1 : Sometime , If primay is not rechable , the >>>>> secondary becomes primary , But opensips loss connection from mongodb . Its >>>>> wont put any data in the data base . My call goes on but their is not data >>>>> in the mongodb database . >>>>> >>>>> Problme 2: If the primay machine is down then secondary >>>>> becomes primary within some millisecond time , but opensips crashes >>>>> giving bellow error >>>>> >>>>> ERROR:cachedb_mongodb:mongo_con_get: Failed to run query. Err = >>>>> 6, 0 , 0 >>>>> CRITICAL:core:receive_fd: EOF on 10 >>>>> INFO:core:handle_sigs: child process 5278 exited by a signal 11 >>>>> INFO:core:handle_sigs: core was generated >>>>> INFO:core:handle_sigs: terminating due to SIGCHLD >>>>> INFO:core:sig_usr: signal 15 received >>>>> INFO:core:sig_usr: signal 15 received >>>>> INFO:core:sig_usr: signal 15 received >>>>> INFO:core:sig_usr: signal 15 received >>>>> INFO:core:sig_usr: signal 15 received >>>>> INFO:core:sig_usr: signal 15 received >>>>> INFO:core:sig_usr: signal 15 received >>>>> INFO:core:sig_usr: signal 15 received >>>>> INFO:core:sig_usr: signal 15 received >>>>> INFO:core:sig_usr: signal 15 received >>>>> INFO:core:sig_usr: signal 15 received >>>>> INFO:core:sig_usr: signal 15 received >>>>> INFO:core:sig_usr: signal 15 received >>>>> INFO:core:cleanup: cleanup >>>>> NOTICE:db_cachedb:destroy: destroy module db_cachedb ... >>>>> NOTICE:cachedb_mongodb:destroy: destroy module cachedb_mongodb ... >>>>> >>>>> If my secondary mongodb machine is not reachable then rather >>>>> connection with db breaks for sometime but opensips wont crashes . But in >>>>> case of Primary opensips crashes with above error . Is this an expected >>>>> behavior or I am doing anything wrong . >>>>> >>>>> Data loss can be bearable but application cant be . So please >>>>> let me know whats the problem . >>>>> >>>>> *Thanks & Regards* >>>>> *Sasmita Panda* >>>>> *Network Testing and Software Engineer* >>>>> *3CLogic , ph:07827611765* >>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> >>>> >>> >>> >>> _______________________________________________ >>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Wed Jun 8 11:45:39 2016 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 8 Jun 2016 12:45:39 +0300 Subject: [OpenSIPS-Users] Opensips -1.11 crash with mongodb replica set . In-Reply-To: References: <575537DA.60002@opensips.org> <57555A4E.8000502@opensips.org> <5756ABF8.7080508@opensips.org> Message-ID: <5757E943.4030905@opensips.org> Any way is good, since all mongos servers will connect to the same set of config servers (just to manage shard configuration and range distributions, they do not do any real-time per-query traffic with the config servers). Also keep in mind that although MongoDB recommends 3 config servers (for 100% write consistency), the cluster may run just fine even on 2, or even 1 single config server. Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 08.06.2016 10:25, Sasmita Panda wrote: > Hi Liviu , > > What are you suggesting for the mongodb config server ? Because > confg server is the one who contains the metadata of the shard cluster . > > Where should I put it ? Will I put this separately or in the > opensips box with mongos ? If I will put the config servers separately > and each mongos instance will get connected to the same set of config > server then will it be fruitful or not ? > > */Thanks & Regards/* > /Sasmita Panda/ > /Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Wed Jun 8 12:33:07 2016 From: spanda at 3clogic.com (Sasmita Panda) Date: Wed, 8 Jun 2016 16:03:07 +0530 Subject: [OpenSIPS-Users] Opensips -1.11 crash with mongodb replica set . In-Reply-To: <5757E943.4030905@opensips.org> References: <575537DA.60002@opensips.org> <57555A4E.8000502@opensips.org> <5756ABF8.7080508@opensips.org> <5757E943.4030905@opensips.org> Message-ID: Yes , I have seen this . Its running fine with 1 config server . But if the config server is in other machine rather than the opensips box , If that machine goes down mongos server wont able to do some query . That's why I have 3 config servers running in diff machines and one in the same opensips box . I think , in this case if any one of the config server wont become reachable then at least another one will be reachable at the same time . Am I right or wrong ? *Thanks & Regards* *Sasmita Panda* *Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Wed, Jun 8, 2016 at 3:15 PM, Liviu Chircu wrote: > Any way is good, since all mongos servers will connect to the same set of > config servers (just to manage shard configuration and range distributions, > they do not do any real-time per-query traffic with the config servers). > > Also keep in mind that although MongoDB recommends 3 config servers (for > 100% write consistency), the cluster may run just fine even on 2, or even 1 > single config server. > > Liviu Chircu > OpenSIPS Developerhttp://www.opensips-solutions.com > > On 08.06.2016 10:25, Sasmita Panda wrote: > > Hi Liviu , > > What are you suggesting for the mongodb config server ? Because > confg server is the one who contains the metadata of the shard cluster . > > Where should I put it ? Will I put this separately or in the opensips > box with mongos ? If I will put the config servers separately and each > mongos instance will get connected to the same set of config server then > will it be fruitful or not ? > > *Thanks & Regards* > *Sasmita Panda* > *Network Testing and Software Engineer* > *3CLogic , ph:07827611765* > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Wed Jun 8 12:41:21 2016 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 8 Jun 2016 13:41:21 +0300 Subject: [OpenSIPS-Users] Opensips -1.11 crash with mongodb replica set . In-Reply-To: References: <575537DA.60002@opensips.org> <57555A4E.8000502@opensips.org> <5756ABF8.7080508@opensips.org> <5757E943.4030905@opensips.org> Message-ID: <5757F651.3010901@opensips.org> Yes, that's how I know it works Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 08.06.2016 13:33, Sasmita Panda wrote: > Yes , I have seen this . Its running fine with 1 config > server . But if the config server is in other machine rather than the > opensips box , If that machine goes down mongos server wont able to do > some query . That's why I have 3 config servers running in diff > machines and one in the same opensips box . I think , in this case if > any one of the config server wont become reachable then at least > another one will be reachable at the same time . > > Am I right or wrong ? > > */Thanks & Regards/* > /Sasmita Panda/ > /Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > On Wed, Jun 8, 2016 at 3:15 PM, Liviu Chircu > wrote: > > Any way is good, since all mongos servers will connect to the same > set of config servers (just to manage shard configuration and > range distributions, they do not do any real-time per-query > traffic with the config servers). > > Also keep in mind that although MongoDB recommends 3 config > servers (for 100% write consistency), the cluster may run just > fine even on 2, or even 1 single config server. > > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > > On 08.06.2016 10:25, Sasmita Panda wrote: >> Hi Liviu , >> >> What are you suggesting for the mongodb config server ? >> Because confg server is the one who contains the metadata of the >> shard cluster . >> >> Where should I put it ? Will I put this separately or in the >> opensips box with mongos ? If I will put the config servers >> separately and each mongos instance will get connected to the >> same set of config server then will it be fruitful or not ? >> >> */Thanks & Regards/* >> /Sasmita Panda/ >> /Network Testing and Software Engineer/ >> /3CLogic , ph:07827611765/ > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From steve.woolley at me.com Wed Jun 8 14:12:24 2016 From: steve.woolley at me.com (Steve Woolley) Date: Wed, 08 Jun 2016 08:12:24 -0400 Subject: [OpenSIPS-Users] Continuous "WARNING:core:timer_ticker: timer task already scheduled" messages in log In-Reply-To: <5756A624.7090205@opensips.org> References: <352B7B4A-6668-4887-9F24-FD97771C7E3D@me.com> <5756A624.7090205@opensips.org> Message-ID: So i made two changes to my config: per Bogdan, changed my timer_partitions to: modparam("tm", "timer_partitions", 4) I had previously used some settings found in a number of configuration examples in my transaction module. After a little research, I commented them out. Still doing some research into whether these new (default) values may trigger other problems. My new config is as so (snippet): #### Transaction Module loadmodule "tm.so" # modparam("tm", "fr_timeout", 5) # modparam("tm", "fr_inv_timeout", 30) # modparam("tm", "restart_fr_on_each_reply", 0) # modparam("tm", "onreply_avp_mode", 1) modparam("tm", "timer_partitions", 4) Since making these changes and restarting, the messages have gone away. > On Jun 7, 2016, at 6:47 AM, Bogdan-Andrei Iancu wrote: > > Hello Steve, > > What OpenSIPS tells you is that you the TM timer routine (which gets executed once per second) takes longer than 1 second (as execution). Probably you have retransmissions or many failure routes to be executed. > You can increase the level of parallelism in TM timer via the timer_partition parameter: > http://www.opensips.org/html/docs/modules/2.2.x/tm.html#id294483 > Try to set it to 4. > > Best regards, > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > On 07.06.2016 04:21, Steve Woolley wrote: >> Running opensips on a Raspberry Pi. At some point after starting opensips (sometimes immediately, sometimes after quite a bit of time), the following message fills the log ? once a second ? continuously. >> >> ... >> Jun 7 01:10:41 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8258580 ms), it may overlap.. >> Jun 7 01:10:42 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8259570 ms), it may overlap.. >> Jun 7 01:10:43 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8260560 ms), it may overlap.. >> Jun 7 01:10:44 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8261560 ms), it may overlap.. >> Jun 7 01:10:45 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8262550 ms), it may overlap.. >> Jun 7 01:10:46 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8263550 ms), it may overlap.. >> Jun 7 01:10:47 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8264540 ms), it may overlap.. >> Jun 7 01:10:48 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8265530 ms), it may overlap.. >> Jun 7 01:10:49 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8266530 ms), it may overlap.. >> Jun 7 01:10:50 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8267520 ms), it may overlap.. >> >> root at pi1:~# opensips -V >> version: opensips 2.2.0 (arm6/linux) >> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, QM_MALLOC, DBG_MALLOC, USE_PTHREAD_MUTEX >> MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 >> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. >> git revision: 0e1cea7 >> main.c compiled on 21:56:59 Jun 3 2016 with gcc 4.9.2 >> >> Anyone experiencing the same? >> >> >> -- >> Steve Woolley >> steve.woolley at me.com >> >> >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From pimenta at inatel.br Wed Jun 8 14:20:01 2016 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Wed, 8 Jun 2016 12:20:01 +0000 Subject: [OpenSIPS-Users] Continuous "WARNING:core:timer_ticker: timer task already scheduled" messages in log In-Reply-To: References: <352B7B4A-6668-4887-9F24-FD97771C7E3D@me.com> <5756A624.7090205@opensips.org>, Message-ID: Thank you! I will compare your configuration with mine. Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________ De: Steve Woolley Enviado: quarta-feira, 8 de junho de 2016 09:12 Para: OpenSIPS users mailling list Cc: Bogdan-Andrei Iancu; Rodrigo Pimenta Carvalho Assunto: Re: [OpenSIPS-Users] Continuous "WARNING:core:timer_ticker: timer task already scheduled" messages in log So i made two changes to my config: 1. per Bogdan, changed my timer_partitions to: modparam("tm", "timer_partitions", 4) 2. I had previously used some settings found in a number of configuration examples in my transaction module. After a little research, I commented them out. Still doing some research into whether these new (default) values may trigger other problems. My new config is as so (snippet): #### Transaction Module loadmodule "tm.so" # modparam("tm", "fr_timeout", 5) # modparam("tm", "fr_inv_timeout", 30) # modparam("tm", "restart_fr_on_each_reply", 0) # modparam("tm", "onreply_avp_mode", 1) modparam("tm", "timer_partitions", 4) Since making these changes and restarting, the messages have gone away. On Jun 7, 2016, at 6:47 AM, Bogdan-Andrei Iancu > wrote: Hello Steve, What OpenSIPS tells you is that you the TM timer routine (which gets executed once per second) takes longer than 1 second (as execution). Probably you have retransmissions or many failure routes to be executed. You can increase the level of parallelism in TM timer via the timer_partition parameter: http://www.opensips.org/html/docs/modules/2.2.x/tm.html#id294483 Try to set it to 4. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 07.06.2016 04:21, Steve Woolley wrote: Running opensips on a Raspberry Pi. At some point after starting opensips (sometimes immediately, sometimes after quite a bit of time), the following message fills the log ? once a second ? continuously. ... Jun 7 01:10:41 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8258580 ms), it may overlap.. Jun 7 01:10:42 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8259570 ms), it may overlap.. Jun 7 01:10:43 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8260560 ms), it may overlap.. Jun 7 01:10:44 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8261560 ms), it may overlap.. Jun 7 01:10:45 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8262550 ms), it may overlap.. Jun 7 01:10:46 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8263550 ms), it may overlap.. Jun 7 01:10:47 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8264540 ms), it may overlap.. Jun 7 01:10:48 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8265530 ms), it may overlap.. Jun 7 01:10:49 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8266530 ms), it may overlap.. Jun 7 01:10:50 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8267520 ms), it may overlap.. root at pi1:~# opensips -V version: opensips 2.2.0 (arm6/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, QM_MALLOC, DBG_MALLOC, USE_PTHREAD_MUTEX MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. git revision: 0e1cea7 main.c compiled on 21:56:59 Jun 3 2016 with gcc 4.9.2 Anyone experiencing the same? -- Steve Woolley steve.woolley at me.com _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From pimenta at inatel.br Wed Jun 8 14:26:33 2016 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Wed, 8 Jun 2016 12:26:33 +0000 Subject: [OpenSIPS-Users] Continuous "WARNING:core:timer_ticker: timer task already scheduled" messages in log In-Reply-To: References: <352B7B4A-6668-4887-9F24-FD97771C7E3D@me.com> <5756A624.7090205@opensips.org>, Message-ID: Hi Steve. The values that you have commented is used by the "make menuconfig". That is, as long as these are the values chosen by the standard configuration, I believe that such values wouldn't be causing problems. Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________ De: Steve Woolley Enviado: quarta-feira, 8 de junho de 2016 09:12 Para: OpenSIPS users mailling list Cc: Bogdan-Andrei Iancu; Rodrigo Pimenta Carvalho Assunto: Re: [OpenSIPS-Users] Continuous "WARNING:core:timer_ticker: timer task already scheduled" messages in log So i made two changes to my config: 1. per Bogdan, changed my timer_partitions to: modparam("tm", "timer_partitions", 4) 2. I had previously used some settings found in a number of configuration examples in my transaction module. After a little research, I commented them out. Still doing some research into whether these new (default) values may trigger other problems. My new config is as so (snippet): #### Transaction Module loadmodule "tm.so" # modparam("tm", "fr_timeout", 5) # modparam("tm", "fr_inv_timeout", 30) # modparam("tm", "restart_fr_on_each_reply", 0) # modparam("tm", "onreply_avp_mode", 1) modparam("tm", "timer_partitions", 4) Since making these changes and restarting, the messages have gone away. On Jun 7, 2016, at 6:47 AM, Bogdan-Andrei Iancu > wrote: Hello Steve, What OpenSIPS tells you is that you the TM timer routine (which gets executed once per second) takes longer than 1 second (as execution). Probably you have retransmissions or many failure routes to be executed. You can increase the level of parallelism in TM timer via the timer_partition parameter: http://www.opensips.org/html/docs/modules/2.2.x/tm.html#id294483 Try to set it to 4. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 07.06.2016 04:21, Steve Woolley wrote: Running opensips on a Raspberry Pi. At some point after starting opensips (sometimes immediately, sometimes after quite a bit of time), the following message fills the log ? once a second ? continuously. ... Jun 7 01:10:41 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8258580 ms), it may overlap.. Jun 7 01:10:42 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8259570 ms), it may overlap.. Jun 7 01:10:43 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8260560 ms), it may overlap.. Jun 7 01:10:44 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8261560 ms), it may overlap.. Jun 7 01:10:45 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8262550 ms), it may overlap.. Jun 7 01:10:46 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8263550 ms), it may overlap.. Jun 7 01:10:47 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8264540 ms), it may overlap.. Jun 7 01:10:48 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8265530 ms), it may overlap.. Jun 7 01:10:49 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8266530 ms), it may overlap.. Jun 7 01:10:50 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker: timer task already scheduled for 7182080 ms (now 8267520 ms), it may overlap.. root at pi1:~# opensips -V version: opensips 2.2.0 (arm6/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, QM_MALLOC, DBG_MALLOC, USE_PTHREAD_MUTEX MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. git revision: 0e1cea7 main.c compiled on 21:56:59 Jun 3 2016 with gcc 4.9.2 Anyone experiencing the same? -- Steve Woolley steve.woolley at me.com _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From john.quick at smartvox.co.uk Wed Jun 8 16:51:01 2016 From: john.quick at smartvox.co.uk (John Quick) Date: Wed, 8 Jun 2016 15:51:01 +0100 Subject: [OpenSIPS-Users] Problems migrating from v2.1 to v2.2 Message-ID: <007801d1c195$2cca5340$865ef9c0$@smartvox.co.uk> I just wanted to report some snags I hit when migrating. Note: I had the install prefix set to / After running make+Install for v2.2, executable files in /usr/local/sbin were not updated to the v2.2 versions. The new versions were *only* in /sbin That would be okay, except that when I ran opensipsdbctl, it tried to run the old version in /usr/local/sbin When I deleted the old executable files in /usr/local/sbin, I got this error: "-bash: /usr/local/sbin/opensipsdbctl: No such file or directory" ...so I copied everything for opensips that was in /sbin to /usr/local/sbin and that seemed to fix it. Then I finally was able to run opensipsdbctl migrate and I got the following errors which I was able to skip: ERROR: failed to migrate opensips_v21.cc_flows to opensips_v22.cc_agents (ERROR 1054 (42S22) at line 1: Unknown column 'opensips_v21.cc_flows.agentid' in 'field list')!!! Skip it and continue (y/n)? y ERROR: failed to migrate opensips_v21.cc_flows to opensips_v22.cc_cdrs (ERROR 1054 (42S22) at line 1: Unknown column 'opensips_v21.cc_flows.caller' in 'field list')!!! John Quick Smartvox Limited Web: www.smartvox.co.uk From benjamin.cropley at gmail.com Wed Jun 8 17:12:51 2016 From: benjamin.cropley at gmail.com (Benjamin Cropley) Date: Wed, 8 Jun 2016 16:12:51 +0100 Subject: [OpenSIPS-Users] Opensips as proxy, config problem on INVITES In-Reply-To: <574D6256.5000606@enigmedia.es> References: <574D6256.5000606@enigmedia.es> Message-ID: There's quite a lot of trace to go through there, but this screams that you're looping to me.. I suggest you look at the routing on the OTHER boxes, see if you can spot something that might be doing that. On Tue, May 31, 2016 at 11:07 AM, Saioa Perurena < saioa.perurena at enigmedia.es> wrote: > Hi, > > We have only one proxy Opensips (version 1.11) that does all the work > (register's, invite's, tls conection...) on a DMZ behind a firewall. > > We want to move that to a new schema, with one Opensips (version 2.1) > as a frontend that handles the tls connections (with a public ip) on > the DMZ, and another Opensips (version 1.11) at the backend (with > private ip), but we are not able to complete the invite signaling > correctly. Register, message, options worked ok, but we have problems > with the invite. > The invite request arrives to the callee, and the OK answers arrives > to the caller, but the caller does not send and ACK to this OK, so the > callee keeps sending OK until it send a BYE because of timeout. > > Any idea of where is the problem or what am i doing wrong?? Any advice > will be appreciated!! > > I attach the sip_trace log and the Opensips script of the frontend server: > > Caller: sip:u1iupzg6we at jipubnx2ef.bell.enigmedia.eu > Callee: d27p6ui7q7 at jipubnx2ef.bell.enigmedia.eu > Opensips frontend ip: internal -> 192.168.3.35, external -> 192.168.1.18 > Opensips backend ip: internal -> 192.168.2.6 > > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Jun 9 10:24:30 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 9 Jun 2016 11:24:30 +0300 Subject: [OpenSIPS-Users] Problems migrating from v2.1 to v2.2 In-Reply-To: <007801d1c195$2cca5340$865ef9c0$@smartvox.co.uk> References: <007801d1c195$2cca5340$865ef9c0$@smartvox.co.uk> Message-ID: <575927BE.800@opensips.org> Hi John, Thank you for your report. I found a typo (c'n'p error) in the DB migration code - this is fixed now, so you can update from GIT and re-test. We will follow up on the issue related to the CTL scripts. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 08.06.2016 17:51, John Quick wrote: > I just wanted to report some snags I hit when migrating. Note: I had the > install prefix set to / > > After running make+Install for v2.2, executable files in /usr/local/sbin > were not updated to the v2.2 versions. The new versions were *only* in /sbin > That would be okay, except that when I ran opensipsdbctl, it tried to run > the old version in /usr/local/sbin > When I deleted the old executable files in /usr/local/sbin, I got this > error: > "-bash: /usr/local/sbin/opensipsdbctl: No such file or directory" > ...so I copied everything for opensips that was in /sbin to /usr/local/sbin > and that seemed to fix it. > > Then I finally was able to run opensipsdbctl migrate and I > got the following errors which I was able to skip: > ERROR: failed to migrate opensips_v21.cc_flows to opensips_v22.cc_agents > (ERROR 1054 (42S22) at line 1: Unknown column > 'opensips_v21.cc_flows.agentid' in 'field list')!!! > Skip it and continue (y/n)? y > ERROR: failed to migrate opensips_v21.cc_flows to opensips_v22.cc_cdrs > (ERROR 1054 (42S22) at line 1: Unknown column 'opensips_v21.cc_flows.caller' > in 'field list')!!! > > John Quick > Smartvox Limited > Web: www.smartvox.co.uk > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > From jwilkie at usipcom.com Thu Jun 9 16:38:26 2016 From: jwilkie at usipcom.com (Jeff Wilkie) Date: Thu, 9 Jun 2016 10:38:26 -0400 Subject: [OpenSIPS-Users] Opensips CP DROUTING GATEWAY enable/disable issue In-Reply-To: <5756A751.8030205@opensips.org> References: <574D3D03.1070608@opensips.org> <5756A751.8030205@opensips.org> Message-ID: Is XMLRPC required to make this function work? Currently, we are only using the fifo method as the boxes.global.inc.php only referencing this method. The documentation for CP says to use one or the other but does not mention that some functions will not work if not using XMLRPC. Thanks Jeff Jeff Wilkie Chief Technology Officer US IP Communications 919.297.1057 *"This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. Any files transmitted with it are confidential and intended solely for the use of the individual to whom it is addressed. Any views or opinions presented are solely those of the author and do not necessarily represent those of USIPCOM, LLC. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you". * On Tue, Jun 7, 2016 at 6:52 AM, Bogdan-Andrei Iancu wrote: > Hi Jeff, > > So, FIFO works ok for you, but CP (using xmlrpc fails). Can you make a > capture of the XMLRPC traffic between CP and OpenSIPS and post it somewhere > ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 31.05.2016 17:27, Jeff Wilkie wrote: > > OpenSIPS Control Panel version 5.0 > > Jeff Wilkie > Chief Technology Officer > US IP Communications > 919.297.1057 > > > *"This e-mail communication and any attachments may contain confidential > and privileged information and is for use by the designated addressee(s) > named above only. Any files transmitted with it are confidential and > intended solely for the use of the individual to whom it is addressed. Any > views or opinions presented are solely those of the author and do not > necessarily represent those of USIPCOM, LLC. If you are not the intended > addressee, you are hereby notified that you have received this > communication in error and that any use or reproduction of this email or > its contents is strictly prohibited and may be unlawful. If you have > received this communication in error, please notify us immediately by > replying to this message and deleting it from your computer. Thank you". * > > On Tue, May 31, 2016 at 3:28 AM, Bogdan-Andrei Iancu < > bogdan at opensips.org> wrote: > >> Hi Jeff, >> >> What OpenSIPS Control Panel version are you using ? >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com >> >> On 21.05.2016 00:15, Jeff Wilkie wrote: >> >> OPENSIPS 1.10.x >> I have the following set >> >> opensips.cfg >> >> #### FIFO Management Interface >> >> loadmodule "mi_fifo.so" >> >> modparam("mi_fifo", "fifo_name", "/tmp/opensips_proxy_fifo") >> >> modparam("mi_fifo", "fifo_mode", 0666) >> >> opensipsctlrc >> >> ## path to FIFO file >> >> OSIPS_FIFO="/tmp/opensips_proxy_fifo" >> >> >> Attempting to disable gateways via the CP gives the following errors: >> >> >> From DROUTING-Gateway interface: >> >> >> Error while disabling gateway 2 >> (the GWID is 2 for the gateway I'm attempting to disable) >> >> From the MI Commands: >> Initiating the following command: *dr_gw_status 2 0* >> 404 GW ID not found >> >> From the DROUTING-Gateway interface I am able to enable the interface if >> it is disabled >> I'm able to also enable the Gateway from the MI Commands section. >> >> I'm also able to enable and disable the Gateway using opensipsctl fifo >> commands >> >> opensipsctl fifo dr_gw_status 2 >> >> Enabled:: yes >> >> opensipsctl fifo dr_gw_status 2 0 >> >> opensipsctl fifo dr_gw_status 2 >> >> Enabled:: no >> >> opensipsctl fifo dr_gw_status 2 1 >> >> opensipsctl fifo dr_gw_status 2 >> >> Enabled:: yes >> >> Not sure where the problem is but I feel its somewhere in the syntax of >> how its delivered. I'm sure it's something easy I've overlooked. Any help >> on this? >> >> Thanks >> Jeff >> >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From pimenta at inatel.br Thu Jun 9 16:42:46 2016 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Thu, 9 Jun 2016 14:42:46 +0000 Subject: [OpenSIPS-Users] How to update table location, but directly on memory cache (RAM)? Message-ID: Hi. My script has the configuration: modparam("usrloc", "db_mode", 2) modparam("usrloc", "timer_interval",3) Always after receiving a new register in table location, I must to execute a code like this: avp_db_query("UPDATE location... That is, an update will complement data in the new register. However, how could I immediately update table location if data might be in memory cache (RAM) for 3 seconds. It could fail obviously. The command avp_db_query UPDATE is acting over the database on hard disc, not in obviously. So, is there a way to update table location even still in cache (RAM)? If yes, when data from RAM is recorded into the database, the register will already be updated. Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Jun 9 17:14:41 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 9 Jun 2016 18:14:41 +0300 Subject: [OpenSIPS-Users] Opensips CP DROUTING GATEWAY enable/disable issue In-Reply-To: References: <574D3D03.1070608@opensips.org> <5756A751.8030205@opensips.org> Message-ID: <575987E1.8010205@opensips.org> Jeff, To have the CP connecting to OpenSIPS you can use any of the MI backends - FIFO, XMLRPC, UDP...any will do the trick. The downside of FIFO is that the CP and OpenSIPS must be on the same server. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 09.06.2016 17:38, Jeff Wilkie wrote: > Is XMLRPC required to make this function work? Currently, we are only > using the fifo method as the boxes.global.inc.php only referencing > this method. The documentation for CP says to use one or the other > but does not mention that some functions will not work if not using > XMLRPC. > > Thanks > Jeff > > Jeff Wilkie > Chief Technology Officer > US IP Communications > 919.297.1057 > > > /"This e-mail communication and any attachments may contain > confidential and privileged information and is for use by the > designated addressee(s) named above only. Any files transmitted with > it are confidential and intended solely for the use of the individual > to whom it is addressed. Any views or opinions presented are solely > those of the author and do not necessarily represent those of USIPCOM, > LLC. If you are not the intended addressee, you are hereby notified > that you have received this communication in error and that any use or > reproduction of this email or its contents is strictly prohibited and > may be unlawful. If you have received this communication in error, > please notify us immediately by replying to this message and deleting > it from your computer. Thank you". / > > On Tue, Jun 7, 2016 at 6:52 AM, Bogdan-Andrei Iancu > > wrote: > > Hi Jeff, > > So, FIFO works ok for you, but CP (using xmlrpc fails). Can you > make a capture of the XMLRPC traffic between CP and OpenSIPS and > post it somewhere ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 31.05.2016 17:27, Jeff Wilkie wrote: >> >> OpenSIPS Control Panel version 5.0 >> >> >> Jeff Wilkie >> Chief Technology Officer >> US IP Communications >> 919.297.1057 >> >> >> /"This e-mail communication and any attachments may contain >> confidential and privileged information and is for use by the >> designated addressee(s) named above only. Any files transmitted >> with it are confidential and intended solely for the use of the >> individual to whom it is addressed. Any views or opinions >> presented are solely those of the author and do not necessarily >> represent those of USIPCOM, LLC. If you are not the intended >> addressee, you are hereby notified that you have received this >> communication in error and that any use or reproduction of this >> email or its contents is strictly prohibited and may be unlawful. >> If you have received this communication in error, please notify >> us immediately by replying to this message and deleting it from >> your computer. Thank you". / >> >> On Tue, May 31, 2016 at 3:28 AM, Bogdan-Andrei Iancu >> > wrote: >> >> Hi Jeff, >> >> What OpenSIPS Control Panel version are you using ? >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> On 21.05.2016 00:15, Jeff Wilkie wrote: >>> OPENSIPS 1.10.x >>> I have the following set >>> >>> opensips.cfg >>> >>> #### FIFO Management Interface >>> >>> loadmodule "mi_fifo.so" >>> >>> modparam("mi_fifo", "fifo_name", "/tmp/opensips_proxy_fifo") >>> >>> modparam("mi_fifo", "fifo_mode", 0666) >>> >>> >>> opensipsctlrc >>> >>> ## path to FIFO file >>> >>> OSIPS_FIFO="/tmp/opensips_proxy_fifo" >>> >>> >>> Attempting to disable gateways via the CP gives the >>> following errors: >>> >>> >>> From DROUTING-Gateway interface: >>> >>> >>> Error while disabling gateway 2 >>> (the GWID is 2 for the gateway I'm attempting to disable) >>> >>> From the MI Commands: >>> Initiating the following command: *dr_gw_status 2 0* >>> 404 GW ID not found >>> >>> From the DROUTING-Gateway interface I am able to enable the >>> interface if it is disabled >>> I'm able to also enable the Gateway from the MI Commands >>> section. >>> >>> I'm also able to enable and disable the Gateway using >>> opensipsctl fifo commands >>> >>> opensipsctl fifo dr_gw_status 2 >>> >>> Enabled:: yes >>> >>> opensipsctl fifo dr_gw_status 2 0 >>> >>> opensipsctl fifo dr_gw_status 2 >>> >>> Enabled:: no >>> >>> opensipsctl fifo dr_gw_status 2 1 >>> >>> opensipsctl fifo dr_gw_status 2 >>> >>> Enabled:: yes >>> >>> >>> Not sure where the problem is but I feel its somewhere in >>> the syntax of how its delivered. I'm sure it's something >>> easy I've overlooked. Any help on this? >>> >>> Thanks >>> Jeff >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From jwilkie at usipcom.com Thu Jun 9 20:18:38 2016 From: jwilkie at usipcom.com (Jeff Wilkie) Date: Thu, 9 Jun 2016 14:18:38 -0400 Subject: [OpenSIPS-Users] Opensips CP DROUTING GATEWAY enable/disable issue In-Reply-To: <575987E1.8010205@opensips.org> References: <574D3D03.1070608@opensips.org> <5756A751.8030205@opensips.org> <575987E1.8010205@opensips.org> Message-ID: Yes. CP and opensips are on the same server. Only the disable doesn't work. I can enable them using CP. Thanks Jeff Wilkie USIP Communications On Jun 9, 2016 11:14 AM, "Bogdan-Andrei Iancu" wrote: > Jeff, > > To have the CP connecting to OpenSIPS you can use any of the MI backends - > FIFO, XMLRPC, UDP...any will do the trick. The downside of FIFO is that the > CP and OpenSIPS must be on the same server. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 09.06.2016 17:38, Jeff Wilkie wrote: > > Is XMLRPC required to make this function work? Currently, we are only > using the fifo method as the boxes.global.inc.php only referencing this > method. The documentation for CP says to use one or the other but does not > mention that some functions will not work if not using XMLRPC. > > Thanks > Jeff > > Jeff Wilkie > Chief Technology Officer > US IP Communications > 919.297.1057 > > > *"This e-mail communication and any attachments may contain confidential > and privileged information and is for use by the designated addressee(s) > named above only. Any files transmitted with it are confidential and > intended solely for the use of the individual to whom it is addressed. Any > views or opinions presented are solely those of the author and do not > necessarily represent those of USIPCOM, LLC. If you are not the intended > addressee, you are hereby notified that you have received this > communication in error and that any use or reproduction of this email or > its contents is strictly prohibited and may be unlawful. If you have > received this communication in error, please notify us immediately by > replying to this message and deleting it from your computer. Thank you". * > > On Tue, Jun 7, 2016 at 6:52 AM, Bogdan-Andrei Iancu > wrote: > >> Hi Jeff, >> >> So, FIFO works ok for you, but CP (using xmlrpc fails). Can you make a >> capture of the XMLRPC traffic between CP and OpenSIPS and post it somewhere >> ? >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com >> >> On 31.05.2016 17:27, Jeff Wilkie wrote: >> >> OpenSIPS Control Panel version 5.0 >> >> Jeff Wilkie >> Chief Technology Officer >> US IP Communications >> 919.297.1057 >> >> >> *"This e-mail communication and any attachments may contain confidential >> and privileged information and is for use by the designated addressee(s) >> named above only. Any files transmitted with it are confidential and >> intended solely for the use of the individual to whom it is addressed. Any >> views or opinions presented are solely those of the author and do not >> necessarily represent those of USIPCOM, LLC. If you are not the intended >> addressee, you are hereby notified that you have received this >> communication in error and that any use or reproduction of this email or >> its contents is strictly prohibited and may be unlawful. If you have >> received this communication in error, please notify us immediately by >> replying to this message and deleting it from your computer. Thank you". * >> >> On Tue, May 31, 2016 at 3:28 AM, Bogdan-Andrei Iancu > > wrote: >> >>> Hi Jeff, >>> >>> What OpenSIPS Control Panel version are you using ? >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com >>> >>> On 21.05.2016 00:15, Jeff Wilkie wrote: >>> >>> OPENSIPS 1.10.x >>> I have the following set >>> >>> opensips.cfg >>> >>> #### FIFO Management Interface >>> >>> loadmodule "mi_fifo.so" >>> >>> modparam("mi_fifo", "fifo_name", "/tmp/opensips_proxy_fifo") >>> >>> modparam("mi_fifo", "fifo_mode", 0666) >>> >>> opensipsctlrc >>> >>> ## path to FIFO file >>> >>> OSIPS_FIFO="/tmp/opensips_proxy_fifo" >>> >>> >>> Attempting to disable gateways via the CP gives the following errors: >>> >>> >>> From DROUTING-Gateway interface: >>> >>> >>> Error while disabling gateway 2 >>> (the GWID is 2 for the gateway I'm attempting to disable) >>> >>> From the MI Commands: >>> Initiating the following command: *dr_gw_status 2 0* >>> 404 GW ID not found >>> >>> From the DROUTING-Gateway interface I am able to enable the interface if >>> it is disabled >>> I'm able to also enable the Gateway from the MI Commands section. >>> >>> I'm also able to enable and disable the Gateway using opensipsctl fifo >>> commands >>> >>> opensipsctl fifo dr_gw_status 2 >>> >>> Enabled:: yes >>> >>> opensipsctl fifo dr_gw_status 2 0 >>> >>> opensipsctl fifo dr_gw_status 2 >>> >>> Enabled:: no >>> >>> opensipsctl fifo dr_gw_status 2 1 >>> >>> opensipsctl fifo dr_gw_status 2 >>> >>> Enabled:: yes >>> >>> Not sure where the problem is but I feel its somewhere in the syntax of >>> how its delivered. I'm sure it's something easy I've overlooked. Any help >>> on this? >>> >>> Thanks >>> Jeff >>> >>> >>> _______________________________________________ >>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >> >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From jwilkie at usipcom.com Thu Jun 9 21:30:50 2016 From: jwilkie at usipcom.com (Jeff Wilkie) Date: Thu, 9 Jun 2016 15:30:50 -0400 Subject: [OpenSIPS-Users] Opensips CP DROUTING GATEWAY enable/disable issue In-Reply-To: <575987E1.8010205@opensips.org> References: <574D3D03.1070608@opensips.org> <5756A751.8030205@opensips.org> <575987E1.8010205@opensips.org> Message-ID: Bogdan, Based on what you have provided, I decided to change the boxes.global.inc.php file as follows: $box_id=0; // mi host:port pair || fifo_file *$boxes[$box_id]['mi']['conn']="127.0.0.1:8080 ";* //boxes[$box_id]['mi']['conn']="/tmp/opensips_proxy_fifo"; It now is able to disable and enable the gateways. I'm not sure if anything else is broken as a result but I will go through the testing. Is there a reason why it would not work using the fifo file instead? Jeff Wilkie Chief Technology Officer US IP Communications 919.297.1057 *"This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. Any files transmitted with it are confidential and intended solely for the use of the individual to whom it is addressed. Any views or opinions presented are solely those of the author and do not necessarily represent those of USIPCOM, LLC. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you". * On Thu, Jun 9, 2016 at 11:14 AM, Bogdan-Andrei Iancu wrote: > Jeff, > > To have the CP connecting to OpenSIPS you can use any of the MI backends - > FIFO, XMLRPC, UDP...any will do the trick. The downside of FIFO is that the > CP and OpenSIPS must be on the same server. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 09.06.2016 17:38, Jeff Wilkie wrote: > > Is XMLRPC required to make this function work? Currently, we are only > using the fifo method as the boxes.global.inc.php only referencing this > method. The documentation for CP says to use one or the other but does not > mention that some functions will not work if not using XMLRPC. > > Thanks > Jeff > > Jeff Wilkie > Chief Technology Officer > US IP Communications > 919.297.1057 > > > *"This e-mail communication and any attachments may contain confidential > and privileged information and is for use by the designated addressee(s) > named above only. Any files transmitted with it are confidential and > intended solely for the use of the individual to whom it is addressed. Any > views or opinions presented are solely those of the author and do not > necessarily represent those of USIPCOM, LLC. If you are not the intended > addressee, you are hereby notified that you have received this > communication in error and that any use or reproduction of this email or > its contents is strictly prohibited and may be unlawful. If you have > received this communication in error, please notify us immediately by > replying to this message and deleting it from your computer. Thank you". * > > On Tue, Jun 7, 2016 at 6:52 AM, Bogdan-Andrei Iancu < > bogdan at opensips.org> wrote: > >> Hi Jeff, >> >> So, FIFO works ok for you, but CP (using xmlrpc fails). Can you make a >> capture of the XMLRPC traffic between CP and OpenSIPS and post it somewhere >> ? >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com >> >> On 31.05.2016 17:27, Jeff Wilkie wrote: >> >> OpenSIPS Control Panel version 5.0 >> >> Jeff Wilkie >> Chief Technology Officer >> US IP Communications >> 919.297.1057 >> >> >> *"This e-mail communication and any attachments may contain confidential >> and privileged information and is for use by the designated addressee(s) >> named above only. Any files transmitted with it are confidential and >> intended solely for the use of the individual to whom it is addressed. Any >> views or opinions presented are solely those of the author and do not >> necessarily represent those of USIPCOM, LLC. If you are not the intended >> addressee, you are hereby notified that you have received this >> communication in error and that any use or reproduction of this email or >> its contents is strictly prohibited and may be unlawful. If you have >> received this communication in error, please notify us immediately by >> replying to this message and deleting it from your computer. Thank you". * >> >> On Tue, May 31, 2016 at 3:28 AM, Bogdan-Andrei Iancu < >> bogdan at opensips.org> wrote: >> >>> Hi Jeff, >>> >>> What OpenSIPS Control Panel version are you using ? >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com >>> >>> On 21.05.2016 00:15, Jeff Wilkie wrote: >>> >>> OPENSIPS 1.10.x >>> I have the following set >>> >>> opensips.cfg >>> >>> #### FIFO Management Interface >>> >>> loadmodule "mi_fifo.so" >>> >>> modparam("mi_fifo", "fifo_name", "/tmp/opensips_proxy_fifo") >>> >>> modparam("mi_fifo", "fifo_mode", 0666) >>> >>> opensipsctlrc >>> >>> ## path to FIFO file >>> >>> OSIPS_FIFO="/tmp/opensips_proxy_fifo" >>> >>> >>> Attempting to disable gateways via the CP gives the following errors: >>> >>> >>> From DROUTING-Gateway interface: >>> >>> >>> Error while disabling gateway 2 >>> (the GWID is 2 for the gateway I'm attempting to disable) >>> >>> From the MI Commands: >>> Initiating the following command: *dr_gw_status 2 0* >>> 404 GW ID not found >>> >>> From the DROUTING-Gateway interface I am able to enable the interface if >>> it is disabled >>> I'm able to also enable the Gateway from the MI Commands section. >>> >>> I'm also able to enable and disable the Gateway using opensipsctl fifo >>> commands >>> >>> opensipsctl fifo dr_gw_status 2 >>> >>> Enabled:: yes >>> >>> opensipsctl fifo dr_gw_status 2 0 >>> >>> opensipsctl fifo dr_gw_status 2 >>> >>> Enabled:: no >>> >>> opensipsctl fifo dr_gw_status 2 1 >>> >>> opensipsctl fifo dr_gw_status 2 >>> >>> Enabled:: yes >>> >>> Not sure where the problem is but I feel its somewhere in the syntax of >>> how its delivered. I'm sure it's something easy I've overlooked. Any help >>> on this? >>> >>> Thanks >>> Jeff >>> >>> >>> _______________________________________________ >>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >> >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Jun 10 12:34:46 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 10 Jun 2016 13:34:46 +0300 Subject: [OpenSIPS-Users] Opensips CP DROUTING GATEWAY enable/disable issue In-Reply-To: References: <574D3D03.1070608@opensips.org> <5756A751.8030205@opensips.org> <575987E1.8010205@opensips.org> Message-ID: <575A97C6.4060200@opensips.org> Hi Jeff, So, from CP, using the FIFO backend, the enabling works, but the disabling doesn't ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 09.06.2016 22:30, Jeff Wilkie wrote: > Bogdan, > > Based on what you have provided, I decided to change the > boxes.global.inc.php file as follows: > > $box_id=0; > > > // mi host:port pair || fifo_file > > *$boxes[$box_id]['mi']['conn']="127.0.0.1:8080 ";* > > //boxes[$box_id]['mi']['conn']="/tmp/opensips_proxy_fifo"; > > > It now is able to disable and enable the gateways. I'm not sure if > anything else is broken as a result but I will go through the > testing. Is there a reason why it would not work using the fifo file > instead? > > > Jeff Wilkie > Chief Technology Officer > US IP Communications > 919.297.1057 > > > /"This e-mail communication and any attachments may contain > confidential and privileged information and is for use by the > designated addressee(s) named above only. Any files transmitted with > it are confidential and intended solely for the use of the individual > to whom it is addressed. Any views or opinions presented are solely > those of the author and do not necessarily represent those of USIPCOM, > LLC. If you are not the intended addressee, you are hereby notified > that you have received this communication in error and that any use or > reproduction of this email or its contents is strictly prohibited and > may be unlawful. If you have received this communication in error, > please notify us immediately by replying to this message and deleting > it from your computer. Thank you". / > > On Thu, Jun 9, 2016 at 11:14 AM, Bogdan-Andrei Iancu > > wrote: > > Jeff, > > To have the CP connecting to OpenSIPS you can use any of the MI > backends - FIFO, XMLRPC, UDP...any will do the trick. The downside > of FIFO is that the CP and OpenSIPS must be on the same server. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 09.06.2016 17:38, Jeff Wilkie wrote: >> Is XMLRPC required to make this function work? Currently, we are >> only using the fifo method as the boxes.global.inc.php only >> referencing this method. The documentation for CP says to use >> one or the other but does not mention that some functions will >> not work if not using XMLRPC. >> >> Thanks >> Jeff >> >> Jeff Wilkie >> Chief Technology Officer >> US IP Communications >> 919.297.1057 >> >> >> /"This e-mail communication and any attachments may contain >> confidential and privileged information and is for use by the >> designated addressee(s) named above only. Any files transmitted >> with it are confidential and intended solely for the use of the >> individual to whom it is addressed. Any views or opinions >> presented are solely those of the author and do not necessarily >> represent those of USIPCOM, LLC. If you are not the intended >> addressee, you are hereby notified that you have received this >> communication in error and that any use or reproduction of this >> email or its contents is strictly prohibited and may be unlawful. >> If you have received this communication in error, please notify >> us immediately by replying to this message and deleting it from >> your computer. Thank you". / >> >> On Tue, Jun 7, 2016 at 6:52 AM, Bogdan-Andrei Iancu >> > wrote: >> >> Hi Jeff, >> >> So, FIFO works ok for you, but CP (using xmlrpc fails). Can >> you make a capture of the XMLRPC traffic between CP and >> OpenSIPS and post it somewhere ? >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> On 31.05.2016 17:27, Jeff Wilkie wrote: >>> >>> OpenSIPS Control Panel version 5.0 >>> >>> >>> Jeff Wilkie >>> Chief Technology Officer >>> US IP Communications >>> 919.297.1057 >>> >>> >>> /"This e-mail communication and any attachments may contain >>> confidential and privileged information and is for use by >>> the designated addressee(s) named above only. Any files >>> transmitted with it are confidential and intended solely for >>> the use of the individual to whom it is addressed. Any views >>> or opinions presented are solely those of the author and do >>> not necessarily represent those of USIPCOM, LLC. If you are >>> not the intended addressee, you are hereby notified that you >>> have received this communication in error and that any use >>> or reproduction of this email or its contents is strictly >>> prohibited and may be unlawful. If you have received this >>> communication in error, please notify us immediately by >>> replying to this message and deleting it from your computer. >>> Thank you". / >>> >>> On Tue, May 31, 2016 at 3:28 AM, Bogdan-Andrei Iancu >>> > wrote: >>> >>> Hi Jeff, >>> >>> What OpenSIPS Control Panel version are you using ? >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> OpenSIPS Founder and Developer >>> http://www.opensips-solutions.com >>> >>> On 21.05.2016 00:15, Jeff Wilkie wrote: >>>> OPENSIPS 1.10.x >>>> I have the following set >>>> >>>> opensips.cfg >>>> >>>> #### FIFO Management Interface >>>> >>>> loadmodule "mi_fifo.so" >>>> >>>> modparam("mi_fifo", "fifo_name", >>>> "/tmp/opensips_proxy_fifo") >>>> >>>> modparam("mi_fifo", "fifo_mode", 0666) >>>> >>>> >>>> opensipsctlrc >>>> >>>> ## path to FIFO file >>>> >>>> OSIPS_FIFO="/tmp/opensips_proxy_fifo" >>>> >>>> >>>> Attempting to disable gateways via the CP gives the >>>> following errors: >>>> >>>> >>>> From DROUTING-Gateway interface: >>>> >>>> >>>> Error while disabling gateway 2 >>>> (the GWID is 2 for the gateway I'm attempting to disable) >>>> >>>> From the MI Commands: >>>> Initiating the following command: *dr_gw_status 2 0* >>>> 404 GW ID not found >>>> >>>> From the DROUTING-Gateway interface I am able to enable >>>> the interface if it is disabled >>>> I'm able to also enable the Gateway from the MI >>>> Commands section. >>>> >>>> I'm also able to enable and disable the Gateway using >>>> opensipsctl fifo commands >>>> >>>> opensipsctl fifo dr_gw_status 2 >>>> >>>> Enabled:: yes >>>> >>>> opensipsctl fifo dr_gw_status 2 0 >>>> >>>> opensipsctl fifo dr_gw_status 2 >>>> >>>> Enabled:: no >>>> >>>> opensipsctl fifo dr_gw_status 2 1 >>>> >>>> opensipsctl fifo dr_gw_status 2 >>>> >>>> Enabled:: yes >>>> >>>> >>>> Not sure where the problem is but I feel its somewhere >>>> in the syntax of how its delivered. I'm sure it's >>>> something easy I've overlooked. Any help on this? >>>> >>>> Thanks >>>> Jeff >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Jun 10 12:39:09 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 10 Jun 2016 13:39:09 +0300 Subject: [OpenSIPS-Users] How to update table location, but directly on memory cache (RAM)? In-Reply-To: References: Message-ID: <575A98CD.4040201@opensips.org> Hi Rodrigo, What you try to do is not consistent. Either you use db_mode 1 to be have immediate writting in DB from usrloc module (see http://www.opensips.org/html/docs/modules/1.11.x/usrloc.html#id294459) -> it will be safe to run your script query after the save(). Either push the extra info you want to save into DB (and memory cache) via the attr AVP (see http://www.opensips.org/html/docs/modules/1.11.x/registrar.html#id293909) and opensips will do everything for you. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 09.06.2016 17:42, Rodrigo Pimenta Carvalho wrote: > > > Hi. > > > My script has the configuration: > > > modparam("usrloc", "db_mode", 2) > modparam("usrloc", "timer_interval",3) > > Always after receiving a new register in table location, I must to > execute a code like this: > > avp_db_query("UPDATE location... > > > That is, an update will complement data in the new register. > > > However, how could I immediately update table location if data might > be in memory cache (RAM) for 3 seconds. It could fail obviously. > > The command avp_db_query UPDATE is acting over the database on hard > disc, not in obviously. > > So, is there a way to update table location even still in cache (RAM)? > If yes, when data from RAM is recorded into the database, the register > will already be updated. > > > Any hint will be very helpful! > > > Best regards. > > > RODRIGO PIMENTA CARVALHO > Inatel Competence Center > Software > Ph: +55 35 3471 9200 RAMAL 979 > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From rrobson at greenlightcrm.com Fri Jun 10 15:18:23 2016 From: rrobson at greenlightcrm.com (Richard Robson) Date: Fri, 10 Jun 2016 14:18:23 +0100 Subject: [OpenSIPS-Users] problem sending HEP messages Message-ID: <9a8c116e-0979-b918-f564-8d633d1e74c1@greenlightcrm.com> I'm trying to get 2.2 to send to a Homer install, but I can't seem to get opensips to send the messages This is the opensips config loadmodule "proto_hep.so" modparam("proto_hep", "hep_port", 9060) loadmodule "siptrace.so" modparam("siptrace", "trace_on", 1) modparam("siptrace", "trace_id", "[tid]uri=hep:192.168.36.140:9060;transport=udp;version=1;") in the route: sip_trace("tid", "d"); The error messages are : Jun 10 14:13:52 gl-sip-02 /usr/sbin/opensips[13630]: ERROR:siptrace:msg_send: no sending socket found for proto 8 Jun 10 14:13:52 gl-sip-02 /usr/sbin/opensips[13630]: ERROR:siptrace:trace_send_hep_duplicate: cannot send duplicate message Jun 10 14:13:52 gl-sip-02 /usr/sbin/opensips[13630]: ERROR:siptrace:save_siptrace: Failed to duplicate with hep to <:> I'm presuming that the opensips is not understanding the trace_id somehow. I've looked at the documentation, but cant find any reason for this. -- Richard Robson Greenlight Support 01382 843843 support at greenlightcrm.com From ionutionita at opensips.org Fri Jun 10 15:28:01 2016 From: ionutionita at opensips.org (Ionita Ionut-Razvan) Date: Fri, 10 Jun 2016 16:28:01 +0300 Subject: [OpenSIPS-Users] problem sending HEP messages In-Reply-To: <9a8c116e-0979-b918-f564-8d633d1e74c1@greenlightcrm.com> References: <9a8c116e-0979-b918-f564-8d633d1e74c1@greenlightcrm.com> Message-ID: You have to define a hep listener listen="hep:ip:port"; in order to work, even though you're not receiving hep messages, only send them. Regards, Ionut Ionita OpenSIPS Developer On Jun 10, 2016, 16:18, at 16:18, Richard Robson wrote: >I'm trying to get 2.2 to send to a Homer install, but I can't seem to >get opensips to send the messages > > >This is the opensips config > >loadmodule "proto_hep.so" >modparam("proto_hep", "hep_port", 9060) >loadmodule "siptrace.so" >modparam("siptrace", "trace_on", 1) >modparam("siptrace", "trace_id", >"[tid]uri=hep:192.168.36.140:9060;transport=udp;version=1;") > > > >in the route: > > > sip_trace("tid", "d"); > > >The error messages are : > > >Jun 10 14:13:52 gl-sip-02 /usr/sbin/opensips[13630]: >ERROR:siptrace:msg_send: no sending socket found for proto 8 >Jun 10 14:13:52 gl-sip-02 /usr/sbin/opensips[13630]: >ERROR:siptrace:trace_send_hep_duplicate: cannot send duplicate message >Jun 10 14:13:52 gl-sip-02 /usr/sbin/opensips[13630]: >ERROR:siptrace:save_siptrace: Failed to duplicate with hep to <:> > > >I'm presuming that the opensips is not understanding the trace_id >somehow. I've looked at the documentation, but cant find any reason for >this. > > >-- >Richard Robson >Greenlight Support >01382 843843 >support at greenlightcrm.com > > >_______________________________________________ >Users mailing list >Users at lists.opensips.org >http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From ionutionita at opensips.org Fri Jun 10 15:30:46 2016 From: ionutionita at opensips.org (Ionita Ionut-Razvan) Date: Fri, 10 Jun 2016 16:30:46 +0300 Subject: [OpenSIPS-Users] problem sending HEP messages In-Reply-To: References: <9a8c116e-0979-b918-f564-8d633d1e74c1@greenlightcrm.com> Message-ID: <6b40b098-593b-4abf-8dc4-7ae70b04711f@typeapp.com> Sorry for the mistake. It's hep_udp:ip:port or hep_tcp:ip:port depending on the transport protocol you wish to use Ionut Ionita OpenSIPS Developer On Jun 10, 2016, 16:28, at 16:28, Ionita Ionut-Razvan wrote: >You have to define a hep listener > >listen="hep:ip:port"; > >in order to work, even though you're not receiving hep messages, only >send them. > >Regards, >Ionut Ionita >OpenSIPS Developer > > > >On Jun 10, 2016, 16:18, at 16:18, Richard Robson > wrote: >>I'm trying to get 2.2 to send to a Homer install, but I can't seem to >>get opensips to send the messages >> >> >>This is the opensips config >> >>loadmodule "proto_hep.so" >>modparam("proto_hep", "hep_port", 9060) >>loadmodule "siptrace.so" >>modparam("siptrace", "trace_on", 1) >>modparam("siptrace", "trace_id", >>"[tid]uri=hep:192.168.36.140:9060;transport=udp;version=1;") >> >> >> >>in the route: >> >> >> sip_trace("tid", "d"); >> >> >>The error messages are : >> >> >>Jun 10 14:13:52 gl-sip-02 /usr/sbin/opensips[13630]: >>ERROR:siptrace:msg_send: no sending socket found for proto 8 >>Jun 10 14:13:52 gl-sip-02 /usr/sbin/opensips[13630]: >>ERROR:siptrace:trace_send_hep_duplicate: cannot send duplicate message >>Jun 10 14:13:52 gl-sip-02 /usr/sbin/opensips[13630]: >>ERROR:siptrace:save_siptrace: Failed to duplicate with hep to <:> >> >> >>I'm presuming that the opensips is not understanding the trace_id >>somehow. I've looked at the documentation, but cant find any reason >for >>this. >> >> >>-- >>Richard Robson >>Greenlight Support >>01382 843843 >>support at greenlightcrm.com >> >> >>_______________________________________________ >>Users mailing list >>Users at lists.opensips.org >>http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > >------------------------------------------------------------------------ > >_______________________________________________ >Users mailing list >Users at lists.opensips.org >http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From pimenta at inatel.br Fri Jun 10 19:07:26 2016 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Fri, 10 Jun 2016 17:07:26 +0000 Subject: [OpenSIPS-Users] How to update table location, but directly on memory cache (RAM)? In-Reply-To: <575A98CD.4040201@opensips.org> References: , <575A98CD.4040201@opensips.org> Message-ID: Hi Bogdan-Andrei. You are right. I have been used the attr_avp, as you explained, to save a specific information in each new record for table location. It works very well and such information goes to column attr. However, I have created today a new column for such table: column callerName. And I have to save $fn in this new column for each new record too. So, what I have just tried today is something like this: modparam("registrar", "attr_avp", "$avp(attr)") modparam("registrar", "attr_avp", "$avp(callerName)") ... is_method("REGISTER")) { $avp(attr) = "my_specific_information"; $avp(callerName) = $fn; } ... But, in this case, the $fn overwrites the specific information, because it seems that attr_avp will pointer always to the same column Attr, no matter what the name I give to the AVP. Do you know how to put every information in its correct column? Is it possble to have two attr_avps related to two different columns in table location? Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________ De: users-bounces at lists.opensips.org em nome de Bogdan-Andrei Iancu Enviado: sexta-feira, 10 de junho de 2016 07:39 Para: OpenSIPS users mailling list Cc: Cleide Aparecida Ribeiro do Prado; Daniel Lopes F?ssia Assunto: Re: [OpenSIPS-Users] How to update table location, but directly on memory cache (RAM)? Hi Rodrigo, What you try to do is not consistent. Either you use db_mode 1 to be have immediate writting in DB from usrloc module (see http://www.opensips.org/html/docs/modules/1.11.x/usrloc.html#id294459) -> it will be safe to run your script query after the save(). Either push the extra info you want to save into DB (and memory cache) via the attr AVP (see http://www.opensips.org/html/docs/modules/1.11.x/registrar.html#id293909) and opensips will do everything for you. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 09.06.2016 17:42, Rodrigo Pimenta Carvalho wrote: Hi. My script has the configuration: modparam("usrloc", "db_mode", 2) modparam("usrloc", "timer_interval",3) Always after receiving a new register in table location, I must to execute a code like this: avp_db_query("UPDATE location... That is, an update will complement data in the new register. However, how could I immediately update table location if data might be in memory cache (RAM) for 3 seconds. It could fail obviously. The command avp_db_query UPDATE is acting over the database on hard disc, not in obviously. So, is there a way to update table location even still in cache (RAM)? If yes, when data from RAM is recorded into the database, the register will already be updated. Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From Agalya_Ramachandran at comcast.com Fri Jun 10 22:07:24 2016 From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor)) Date: Fri, 10 Jun 2016 20:07:24 +0000 Subject: [OpenSIPS-Users] Provisional response handling in case of forking Message-ID: <0686390e80e344cf9c2cff77c7b1dfd1@COPDCEX28.cable.comcast.com> Hi team, We are using opensips for our project and we are currently using opensips as proxy. Am forking the incoming sip call, to two destinations. It Rings in both Dest A and Dest B, as a result I get two 180 Ringing response from A and B. I want to filter only the first incoming 180 Ringing response and send to the actual caller. Is there a way to do this in opensips config file? I have seen drop() function which drops the complete provisional response. But in my case I have to forward one 180 Ringing to the caller. Can it be achieved by the changes in config file? Please guide me. Regards, Agalya -------------- next part -------------- An HTML attachment was scrubbed... URL: From jwilkie at usipcom.com Fri Jun 10 22:10:25 2016 From: jwilkie at usipcom.com (Jeff Wilkie) Date: Fri, 10 Jun 2016 16:10:25 -0400 Subject: [OpenSIPS-Users] Opensips CP DROUTING GATEWAY enable/disable issue In-Reply-To: <575A97C6.4060200@opensips.org> References: <574D3D03.1070608@opensips.org> <5756A751.8030205@opensips.org> <575987E1.8010205@opensips.org> <575A97C6.4060200@opensips.org> Message-ID: I have a more detailed message awaiting approval, but the answer is yes to the question. Although I do have to hit "Apply changes to server" before it takes place. Thanks Jeff Wilkie On Fri, Jun 10, 2016 at 6:34 AM, Bogdan-Andrei Iancu wrote: > Hi Jeff, > > So, from CP, using the FIFO backend, the enabling works, but the disabling > doesn't ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 09.06.2016 22:30, Jeff Wilkie wrote: > > Bogdan, > > Based on what you have provided, I decided to change the boxes.global.inc.php > file as follows: > > $box_id=0; > > > // mi host:port pair || fifo_file > > *$boxes[$box_id]['mi']['conn']="127.0.0.1:8080 ";* > > //boxes[$box_id]['mi']['conn']="/tmp/opensips_proxy_fifo"; > > It now is able to disable and enable the gateways. I'm not sure if > anything else is broken as a result but I will go through the testing. Is > there a reason why it would not work using the fifo file instead? > > > Jeff Wilkie > Chief Technology Officer > US IP Communications > 919.297.1057 > > > *"This e-mail communication and any attachments may contain confidential > and privileged information and is for use by the designated addressee(s) > named above only. Any files transmitted with it are confidential and > intended solely for the use of the individual to whom it is addressed. Any > views or opinions presented are solely those of the author and do not > necessarily represent those of USIPCOM, LLC. If you are not the intended > addressee, you are hereby notified that you have received this > communication in error and that any use or reproduction of this email or > its contents is strictly prohibited and may be unlawful. If you have > received this communication in error, please notify us immediately by > replying to this message and deleting it from your computer. Thank you". * > > On Thu, Jun 9, 2016 at 11:14 AM, Bogdan-Andrei Iancu < > bogdan at opensips.org> wrote: > >> Jeff, >> >> To have the CP connecting to OpenSIPS you can use any of the MI backends >> - FIFO, XMLRPC, UDP...any will do the trick. The downside of FIFO is that >> the CP and OpenSIPS must be on the same server. >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com >> >> On 09.06.2016 17:38, Jeff Wilkie wrote: >> >> Is XMLRPC required to make this function work? Currently, we are only >> using the fifo method as the boxes.global.inc.php only referencing this >> method. The documentation for CP says to use one or the other but does not >> mention that some functions will not work if not using XMLRPC. >> >> Thanks >> Jeff >> >> Jeff Wilkie >> Chief Technology Officer >> US IP Communications >> 919.297.1057 >> >> >> *"This e-mail communication and any attachments may contain confidential >> and privileged information and is for use by the designated addressee(s) >> named above only. Any files transmitted with it are confidential and >> intended solely for the use of the individual to whom it is addressed. Any >> views or opinions presented are solely those of the author and do not >> necessarily represent those of USIPCOM, LLC. If you are not the intended >> addressee, you are hereby notified that you have received this >> communication in error and that any use or reproduction of this email or >> its contents is strictly prohibited and may be unlawful. If you have >> received this communication in error, please notify us immediately by >> replying to this message and deleting it from your computer. Thank you". * >> >> On Tue, Jun 7, 2016 at 6:52 AM, Bogdan-Andrei Iancu < >> bogdan at opensips.org> wrote: >> >>> Hi Jeff, >>> >>> So, FIFO works ok for you, but CP (using xmlrpc fails). Can you make a >>> capture of the XMLRPC traffic between CP and OpenSIPS and post it somewhere >>> ? >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com >>> >>> On 31.05.2016 17:27, Jeff Wilkie wrote: >>> >>> OpenSIPS Control Panel version 5.0 >>> >>> Jeff Wilkie >>> Chief Technology Officer >>> US IP Communications >>> 919.297.1057 >>> >>> >>> *"This e-mail communication and any attachments may contain confidential >>> and privileged information and is for use by the designated addressee(s) >>> named above only. Any files transmitted with it are confidential and >>> intended solely for the use of the individual to whom it is addressed. Any >>> views or opinions presented are solely those of the author and do not >>> necessarily represent those of USIPCOM, LLC. If you are not the intended >>> addressee, you are hereby notified that you have received this >>> communication in error and that any use or reproduction of this email or >>> its contents is strictly prohibited and may be unlawful. If you have >>> received this communication in error, please notify us immediately by >>> replying to this message and deleting it from your computer. Thank you". * >>> >>> On Tue, May 31, 2016 at 3:28 AM, Bogdan-Andrei Iancu < >>> bogdan at opensips.org> wrote: >>> >>>> Hi Jeff, >>>> >>>> What OpenSIPS Control Panel version are you using ? >>>> >>>> Regards, >>>> >>>> Bogdan-Andrei Iancu >>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com >>>> >>>> On 21.05.2016 00:15, Jeff Wilkie wrote: >>>> >>>> OPENSIPS 1.10.x >>>> I have the following set >>>> >>>> opensips.cfg >>>> >>>> #### FIFO Management Interface >>>> >>>> loadmodule "mi_fifo.so" >>>> >>>> modparam("mi_fifo", "fifo_name", "/tmp/opensips_proxy_fifo") >>>> >>>> modparam("mi_fifo", "fifo_mode", 0666) >>>> >>>> opensipsctlrc >>>> >>>> ## path to FIFO file >>>> >>>> OSIPS_FIFO="/tmp/opensips_proxy_fifo" >>>> >>>> >>>> Attempting to disable gateways via the CP gives the following errors: >>>> >>>> >>>> From DROUTING-Gateway interface: >>>> >>>> >>>> Error while disabling gateway 2 >>>> (the GWID is 2 for the gateway I'm attempting to disable) >>>> >>>> From the MI Commands: >>>> Initiating the following command: *dr_gw_status 2 0* >>>> 404 GW ID not found >>>> >>>> From the DROUTING-Gateway interface I am able to enable the interface >>>> if it is disabled >>>> I'm able to also enable the Gateway from the MI Commands section. >>>> >>>> I'm also able to enable and disable the Gateway using opensipsctl fifo >>>> commands >>>> >>>> opensipsctl fifo dr_gw_status 2 >>>> >>>> Enabled:: yes >>>> >>>> opensipsctl fifo dr_gw_status 2 0 >>>> >>>> opensipsctl fifo dr_gw_status 2 >>>> >>>> Enabled:: no >>>> >>>> opensipsctl fifo dr_gw_status 2 1 >>>> >>>> opensipsctl fifo dr_gw_status 2 >>>> >>>> Enabled:: yes >>>> >>>> Not sure where the problem is but I feel its somewhere in the syntax of >>>> how its delivered. I'm sure it's something easy I've overlooked. Any help >>>> on this? >>>> >>>> Thanks >>>> Jeff >>>> >>>> >>>> _______________________________________________ >>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> >>> >>> >> >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From nabeelshikder at gmail.com Sun Jun 12 09:18:30 2016 From: nabeelshikder at gmail.com (Nabeel) Date: Sun, 12 Jun 2016 08:18:30 +0100 Subject: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions Message-ID: Hi, I will be following this tutorial to integrate OpenSIPS and Asterisk: http://www.opensips.org/Documentation/Tutorials-OpenSIPSAsteriskIntegration-1-8 The tutorial mentions the use of OpenSIPS version 1.8 and Asterisk version 1.8. I would like to know if I can use the latest versions of OpenSIPS and Asterisk instead? Have there been changes to database structure which can cause problems? Nabeel -------------- next part -------------- An HTML attachment was scrubbed... URL: From chandan.pr at webshar.org Mon Jun 13 08:27:22 2016 From: chandan.pr at webshar.org (Chandan PR) Date: Mon, 13 Jun 2016 11:57:22 +0530 Subject: [OpenSIPS-Users] OpenSips as Load Balancer - Handling Failure Scenarios In-Reply-To: <574D9747.3090608@opensips.org> References: <574D9747.3090608@opensips.org> Message-ID: Hi Bogdan-Andrei Iancu, Sorry for the late reply. This was not the issue with OpenSips. It was an edge case and fr_inv_timer configuration value was less. Works perfectly fine after updating the fr_inv_timer value. Regards, Chandan On Tue, May 31, 2016 at 7:23 PM, Bogdan-Andrei Iancu wrote: > Hi Chandan, > > If the call is rejected by callee, your opensips should receive a negative > reply and trigger the failure route - what is the reply code you get back > from callee ? Is your failure route triggered ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 31.05.2016 16:31, Chandan PR wrote: > > Hi Guys, > > I am trying to configure OpenSips as Load Balancer for our outbound > dialling. > > I am following the example from: > http://www.opensips.org/Documentation/Tutorials-LoadBalancing-1-9. > > Right now when the user rejects the call, instead of ending up in 486 we > are ending up in 408. > > This is due to the call being ended up in the failure_route after the > timeout (fr_inv_timer). > > Is there a way or configuration to send the response codes from the reply > back to client, instead of waiting for timeout? > > > Regards, > Chandan > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Jun 13 12:30:38 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 13 Jun 2016 13:30:38 +0300 Subject: [OpenSIPS-Users] Provisional response handling in case of forking In-Reply-To: <0686390e80e344cf9c2cff77c7b1dfd1@COPDCEX28.cable.comcast.com> References: <0686390e80e344cf9c2cff77c7b1dfd1@COPDCEX28.cable.comcast.com> Message-ID: <575E8B4E.2000708@opensips.org> Hi Agalya, Use the onreply route (be sure to onreply_avp_mode to be set to 1 - see http://www.opensips.org/html/docs/modules/2.2.x/tm.html#id294290) in combination with flags, to record when the an 180 reply was set. Like: if ($rs==180) { if (isflagset(FLAG_180)) drop(); setflag(FALG_180); } The onreply_avp_mode 1 will ensure that the onreply route will not overlap for 2 replies . Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 10.06.2016 23:07, Ramachandran, Agalya (Contractor) wrote: > > Hi team, > > We are using opensips for our project and we are currently using > opensips as proxy. > > Am forking the incoming sip call, to two destinations. It Rings in > both Dest A and Dest B, as a result I get two 180 Ringing response > from A and B. > > I want to filter only the first incoming 180 Ringing response and send > to the actual caller. Is there a way to do this in opensips config file? > > I have seen drop() function which drops the complete provisional > response. But in my case I have to forward one 180 Ringing to the caller. > > Can it be achieved by the changes in config file? Please guide me. > > Regards, > Agalya > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From jwilkie at usipcom.com Fri Jun 10 16:47:51 2016 From: jwilkie at usipcom.com (Jeff Wilkie) Date: Fri, 10 Jun 2016 10:47:51 -0400 Subject: [OpenSIPS-Users] Opensips CP DROUTING GATEWAY enable/disable issue In-Reply-To: <575A97C6.4060200@opensips.org> References: <574D3D03.1070608@opensips.org> <5756A751.8030205@opensips.org> <575987E1.8010205@opensips.org> <575A97C6.4060200@opensips.org> Message-ID: From Agalya_Ramachandran at comcast.com Mon Jun 13 16:27:36 2016 From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor)) Date: Mon, 13 Jun 2016 14:27:36 +0000 Subject: [OpenSIPS-Users] Provisional response handling in case of forking In-Reply-To: <575E8B4E.2000708@opensips.org> References: <0686390e80e344cf9c2cff77c7b1dfd1@COPDCEX28.cable.comcast.com> <575E8B4E.2000708@opensips.org> Message-ID: <8c8f9b61f3c94b37a7107631e2950a5c@COPDCEX28.cable.comcast.com> Hi Bogdan, Thank you for your kind response. I will try the logic you told me. But I have a question in the below logic. "FLAG_180", is this something declared and maintained by opensips and set by default on its own if Ringing response is received or we need to define this flag explicitly ? E.g for setting NAT flag, we are using 'usrloc' module and 'nat_bflag' as parameter and then value. modparam("usrloc", "nat_bflag", "NAT") Likewise if we need to define explicitly, what is the module in which we need to define the flag for FLAG_180? Regards, Agalya From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Monday, June 13, 2016 6:31 AM To: OpenSIPS users mailling list ; Ramachandran, Agalya (Contractor) Subject: Re: [OpenSIPS-Users] Provisional response handling in case of forking Hi Agalya, Use the onreply route (be sure to onreply_avp_mode to be set to 1 - see http://www.opensips.org/html/docs/modules/2.2.x/tm.html#id294290) in combination with flags, to record when the an 180 reply was set. Like: if ($rs==180) { if (isflagset(FLAG_180)) drop(); setflag(FALG_180); } The onreply_avp_mode 1 will ensure that the onreply route will not overlap for 2 replies . Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 10.06.2016 23:07, Ramachandran, Agalya (Contractor) wrote: Hi team, We are using opensips for our project and we are currently using opensips as proxy. Am forking the incoming sip call, to two destinations. It Rings in both Dest A and Dest B, as a result I get two 180 Ringing response from A and B. I want to filter only the first incoming 180 Ringing response and send to the actual caller. Is there a way to do this in opensips config file? I have seen drop() function which drops the complete provisional response. But in my case I have to forward one 180 Ringing to the caller. Can it be achieved by the changes in config file? Please guide me. Regards, Agalya _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Jun 13 18:12:57 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 13 Jun 2016 19:12:57 +0300 Subject: [OpenSIPS-Users] Provisional response handling in case of forking In-Reply-To: <8c8f9b61f3c94b37a7107631e2950a5c@COPDCEX28.cable.comcast.com> References: <0686390e80e344cf9c2cff77c7b1dfd1@COPDCEX28.cable.comcast.com> <575E8B4E.2000708@opensips.org> <8c8f9b61f3c94b37a7107631e2950a5c@COPDCEX28.cable.comcast.com> Message-ID: <575EDB89.2050200@opensips.org> Hi Agalya, It is not a predefine flag. And you do not have to define the flags in OpenSIPS. Just pickup aname and start using it in script. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 13.06.2016 17:27, Ramachandran, Agalya (Contractor) wrote: > > Hi Bogdan, > > Thank you for your kind response. I will try the logic you told me. > But I have a question in the below logic. > > ?FLAG_180?, is this something declared and maintained by opensips and > set by default on its own if Ringing response is received or we need > to define this flag explicitly ? > > E.g for setting NAT flag, we are using ?usrloc? module and ?nat_bflag? > as parameter and then value. > > modparam("usrloc", "nat_bflag", "NAT") > > Likewise if we need to define explicitly, what is the module in which > we need to define the flag for FLAG_180? > > Regards, > > Agalya > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* Monday, June 13, 2016 6:31 AM > *To:* OpenSIPS users mailling list ; > Ramachandran, Agalya (Contractor) > *Subject:* Re: [OpenSIPS-Users] Provisional response handling in case > of forking > > Hi Agalya, > > Use the onreply route (be sure to onreply_avp_mode to be set to 1 - > see http://www.opensips.org/html/docs/modules/2.2.x/tm.html#id294290) > in combination with flags, to record when the an 180 reply was set. > > Like: > > if ($rs==180) { > if (isflagset(FLAG_180)) > drop(); > setflag(FALG_180); > } > > The onreply_avp_mode 1 will ensure that the onreply route will not > overlap for 2 replies . > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 10.06.2016 23:07, Ramachandran, Agalya (Contractor) wrote: > > Hi team, > > We are using opensips for our project and we are currently using > opensips as proxy. > > Am forking the incoming sip call, to two destinations. It Rings in > both Dest A and Dest B, as a result I get two 180 Ringing response > from A and B. > > I want to filter only the first incoming 180 Ringing response and > send to the actual caller. Is there a way to do this in opensips > config file? > > I have seen drop() function which drops the complete provisional > response. But in my case I have to forward one 180 Ringing to the > caller. > > Can it be achieved by the changes in config file? Please guide me. > > Regards, > Agalya > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From Agalya_Ramachandran at comcast.com Mon Jun 13 21:48:40 2016 From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor)) Date: Mon, 13 Jun 2016 19:48:40 +0000 Subject: [OpenSIPS-Users] Provisional response handling in case of forking In-Reply-To: <575EDB89.2050200@opensips.org> References: <0686390e80e344cf9c2cff77c7b1dfd1@COPDCEX28.cable.comcast.com> <575E8B4E.2000708@opensips.org> <8c8f9b61f3c94b37a7107631e2950a5c@COPDCEX28.cable.comcast.com> <575EDB89.2050200@opensips.org> Message-ID: <4526791507af43cc93131886ee3a3cde@COPDCEX28.cable.comcast.com> Hi Bogdan, OK. Got it. One more to question to understand the snippet you gave.. I have just commented my understanding of each snippet at the same line. I don't get why we need setflag(FLAG_180); after we drop the response. if ($rs==180) { //We are checking response is equal to 180 if (isflagset(FLAG_180)) // Checking the flag is set or not. If response is 180, flag will be set drop(); //If set we are dropping the consecutive same response. setflag(FALG_180); ?? } (P.S) I tested without setting the flag after drop response. In this case it forwards both 180 response back to caller. Am curious how it controls it. Is onreply_avp_mode controls dropping the 2nd response or setflag(FLAG_180);? I want to be clear on what am working with. Hence posting you this question. Regards, Agalya From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Monday, June 13, 2016 12:13 PM To: Ramachandran, Agalya (Contractor) ; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Provisional response handling in case of forking Hi Agalya, It is not a predefine flag. And you do not have to define the flags in OpenSIPS. Just pickup a name and start using it in script. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 13.06.2016 17:27, Ramachandran, Agalya (Contractor) wrote: Hi Bogdan, Thank you for your kind response. I will try the logic you told me. But I have a question in the below logic. "FLAG_180", is this something declared and maintained by opensips and set by default on its own if Ringing response is received or we need to define this flag explicitly ? E.g for setting NAT flag, we are using 'usrloc' module and 'nat_bflag' as parameter and then value. modparam("usrloc", "nat_bflag", "NAT") Likewise if we need to define explicitly, what is the module in which we need to define the flag for FLAG_180? Regards, Agalya From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Monday, June 13, 2016 6:31 AM To: OpenSIPS users mailling list ; Ramachandran, Agalya (Contractor) Subject: Re: [OpenSIPS-Users] Provisional response handling in case of forking Hi Agalya, Use the onreply route (be sure to onreply_avp_mode to be set to 1 - see http://www.opensips.org/html/docs/modules/2.2.x/tm.html#id294290) in combination with flags, to record when the an 180 reply was set. Like: if ($rs==180) { if (isflagset(FLAG_180)) drop(); setflag(FALG_180); } The onreply_avp_mode 1 will ensure that the onreply route will not overlap for 2 replies . Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 10.06.2016 23:07, Ramachandran, Agalya (Contractor) wrote: Hi team, We are using opensips for our project and we are currently using opensips as proxy. Am forking the incoming sip call, to two destinations. It Rings in both Dest A and Dest B, as a result I get two 180 Ringing response from A and B. I want to filter only the first incoming 180 Ringing response and send to the actual caller. Is there a way to do this in opensips config file? I have seen drop() function which drops the complete provisional response. But in my case I have to forward one 180 Ringing to the caller. Can it be achieved by the changes in config file? Please guide me. Regards, Agalya _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From govoiper at gmail.com Mon Jun 13 22:11:25 2016 From: govoiper at gmail.com (SamyGo) Date: Mon, 13 Jun 2016 16:11:25 -0400 Subject: [OpenSIPS-Users] OpenSIPS 2.2 Documentation needs update ! Message-ID: Hi, It seems like mentioning of clusterer module as a dependency is missing form module docs: http://www.opensips.org/html/docs/modules/2.2.x/usrloc.html#id293640 Although it is obvious that the write refers to "valid cluster id" in function accept_replicated_contact() but I still think documentation should be updated. Adding replication modparams in opensips.cfg w/o clusterer also states the obvious: WARNING:core:solve_module_dependencies: module usrloc depends on module clusterer, but it was not loaded! ERROR:core:main: failed to solve module dependencies Regards, Sammy -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Jun 13 22:59:20 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 13 Jun 2016 23:59:20 +0300 Subject: [OpenSIPS-Users] Provisional response handling in case of forking In-Reply-To: <4526791507af43cc93131886ee3a3cde@COPDCEX28.cable.comcast.com> References: <0686390e80e344cf9c2cff77c7b1dfd1@COPDCEX28.cable.comcast.com> <575E8B4E.2000708@opensips.org> <8c8f9b61f3c94b37a7107631e2950a5c@COPDCEX28.cable.comcast.com> <575EDB89.2050200@opensips.org> <4526791507af43cc93131886ee3a3cde@COPDCEX28.cable.comcast.com> Message-ID: <575F1EA8.50907@opensips.org> Hi Agalya, For the first 180, the flag is notset (by default, the flags are off), so no drop will be doneand the execution will get to the setflag(), setting the flag. (only the drop is under the IF). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 13.06.2016 22:48, Ramachandran, Agalya (Contractor) wrote: > > Hi Bogdan, > > OK. Got it. One more to question to understand the snippet you gave.. > > I have just commented my understanding of each snippet at the same line. > > I don?t get why we need setflag(FLAG_180); after we drop the response. > > if ($rs==180) { //We are checking response is equal to 180 > if (isflagset(FLAG_180)) // Checking the flag is set or not. If > response is 180, flag will be set > drop(); //If set we are dropping the consecutive > same response. > setflag(FALG_180); ?? > } > > (P.S) I tested without setting the flag after drop response. In this > case it forwards both 180 response back to caller. > > Am curious how it controls it. > > Is onreply_avp_mode controls dropping the 2nd response or > setflag(FLAG_180);? > > I want to be clear on what am working with. Hence posting you this > question. > > Regards, > Agalya > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* Monday, June 13, 2016 12:13 PM > *To:* Ramachandran, Agalya (Contractor) > ; OpenSIPS users mailling list > > *Subject:* Re: [OpenSIPS-Users] Provisional response handling in case > of forking > > Hi Agalya, > > It is not a predefine flag. And you do not have to define the flags in > OpenSIPS. Just pickup a name and start using it in script. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 13.06.2016 17:27, Ramachandran, Agalya (Contractor) wrote: > > Hi Bogdan, > > Thank you for your kind response. I will try the logic you told > me. But I have a question in the below logic. > > ?FLAG_180?, is this something declared and maintained by opensips > and set by default on its own if Ringing response is received or > we need to define this flag explicitly ? > > E.g for setting NAT flag, we are using ?usrloc? module and > ?nat_bflag? as parameter and then value. > > modparam("usrloc", "nat_bflag", "NAT") > > Likewise if we need to define explicitly, what is the module in > which we need to define the flag for FLAG_180? > > Regards, > > Agalya > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* Monday, June 13, 2016 6:31 AM > *To:* OpenSIPS users mailling list > ; Ramachandran, Agalya > (Contractor) > > *Subject:* Re: [OpenSIPS-Users] Provisional response handling in > case of forking > > Hi Agalya, > > Use the onreply route (be sure to onreply_avp_mode to be set to 1 > - see > http://www.opensips.org/html/docs/modules/2.2.x/tm.html#id294290) > in combination with flags, to record when the an 180 reply was set. > > Like: > > if ($rs==180) { > if (isflagset(FLAG_180)) > drop(); > setflag(FALG_180); > } > > The onreply_avp_mode 1 will ensure that the onreply route will not > overlap for 2 replies . > > Regards, > > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > On 10.06.2016 23:07, Ramachandran, Agalya (Contractor) wrote: > > Hi team, > > We are using opensips for our project and we are currently > using opensips as proxy. > > Am forking the incoming sip call, to two destinations. It > Rings in both Dest A and Dest B, as a result I get two 180 > Ringing response from A and B. > > I want to filter only the first incoming 180 Ringing response > and send to the actual caller. Is there a way to do this in > opensips config file? > > I have seen drop() function which drops the complete > provisional response. But in my case I have to forward one 180 > Ringing to the caller. > > Can it be achieved by the changes in config file? Please guide me. > > Regards, > Agalya > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Jun 13 23:27:50 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 14 Jun 2016 00:27:50 +0300 Subject: [OpenSIPS-Users] OpenSIPS 2.2 Documentation needs update ! In-Reply-To: References: Message-ID: <575F2556.4070005@opensips.org> Hi Sammy, Thank you for the report, I just updated the doc in regards to the missing dependency : https://github.com/OpenSIPS/opensips/commit/4357fe858c948ded5db4b37fec5d26c7c379652a The online docs will get regenerated during the night. Thanks and regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 13.06.2016 23:11, SamyGo wrote: > Hi, > > It seems like mentioning of clusterer module as a dependency is > missing form module docs: > > http://www.opensips.org/html/docs/modules/2.2.x/usrloc.html#id293640 > > Although it is obvious that the write refers to "valid cluster id" in > function accept_replicated_contact() > but > I still think documentation should be updated. > > Adding replication modparams in opensips.cfg w/o clusterer also states > the obvious: > > WARNING:core:solve_module_dependencies: module usrloc depends on > module clusterer, but it was not loaded! > ERROR:core:main: failed to solve module dependencies > > Regards, > Sammy > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From govoiper at gmail.com Tue Jun 14 07:06:35 2016 From: govoiper at gmail.com (SamyGo) Date: Tue, 14 Jun 2016 01:06:35 -0400 Subject: [OpenSIPS-Users] OpenSIPS 2.2 Documentation needs update ! In-Reply-To: <575F2556.4070005@opensips.org> References: <575F2556.4070005@opensips.org> Message-ID: Nice, Another one: I've another one with wrong ref link.: proto_bin points to proto_hep in this list: http://opensips.org/Documentation/Modules-2-2 I had to manually edit the URL to view proto_bin page. Thanks, Sammy On Mon, Jun 13, 2016 at 5:27 PM, Bogdan-Andrei Iancu wrote: > Hi Sammy, > > Thank you for the report, I just updated the doc in regards to the missing > dependency : > > https://github.com/OpenSIPS/opensips/commit/4357fe858c948ded5db4b37fec5d26c7c379652a > > The online docs will get regenerated during the night. > > Thanks and regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 13.06.2016 23:11, SamyGo wrote: > > Hi, > > It seems like mentioning of clusterer module as a dependency is missing > form module docs: > > http://www.opensips.org/html/docs/modules/2.2.x/usrloc.html#id293640 > > Although it is obvious that the write refers to "valid cluster id" in > function accept_replicated_contact() > but > I still think documentation should be updated. > > Adding replication modparams in opensips.cfg w/o clusterer also states the > obvious: > > WARNING:core:solve_module_dependencies: module usrloc depends on module > clusterer, but it was not loaded! > ERROR:core:main: failed to solve module dependencies > > Regards, > Sammy > > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From govoiper at gmail.com Tue Jun 14 08:35:01 2016 From: govoiper at gmail.com (SamyGo) Date: Tue, 14 Jun 2016 02:35:01 -0400 Subject: [OpenSIPS-Users] Userloc, Bin Interface, and Clusterer combo Message-ID: Hi, I've a cluster setup with userloc replication enabled for 3+ servers. As soon as a user register anywhere the other servers which receive this replicated contact display these errors: ERROR:usrloc:receive_ucontact_insert: non-local socket ERROR:usrloc:receive_ucontact_insert: failed to process replication event. dom: 'location', aor: '89654 at 1X.2XX.XX.XX' ERROR:usrloc:receive_binary_packet: failed to process a binary packet! The solution(kind of) for the same error has been discussed in this thread: http://lists.opensips.org/pipermail/users/2015-February/030910.html The question here is, is setting *ip_nonlocal_bind *for this userlocation replication a solution ? or should the replication mechanism be modified to process the replications from other nodes ! Say even if this works, the next question is, what would happen to the contacts from a server which is no longer active. The Contacts disappear as soon as the clusterer node becomes inactive ? Looking for some clarity on this topic. Regards, Sammy -------------- next part -------------- An HTML attachment was scrubbed... URL: From Kevin.Stewart at m2group.co.nz Tue Jun 14 08:49:32 2016 From: Kevin.Stewart at m2group.co.nz (Kevin Stewart) Date: Tue, 14 Jun 2016 06:49:32 +0000 Subject: [OpenSIPS-Users] B2BUA not responding to reinvites Message-ID: I am trying to implement a simple topology hiding B2BUA with opensips 1.11.5 most things work except that re invites for session expiry are being ignored. I assume that I am missing something in my config. the strange thing is that I see no debug to syslog when the Invites come in even with debug set to 9. the basic config is a client on port 5061 (C) inviting the B2BUA on port 5066(B) with then invites the main server on another host on port 5060(A). I do not expect and inbount calls from A C invites B, B invites A all proceeds normally until after 15 minutes A invites B and is ignored then sends bye a number of times. none of the reinvites or byes are reported in the logs. C then hangs up sending a bye to B, B sends a bye to A and gets 481 Unknown Dialog. below is my config route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; if (is_method("OPTIONS")) { #xlog("L_NOTICE", "$ci|end|unsupported method"); sl_send_reply("404", "Not found"); exit; } if(is_method("INVITE")){ xlog("L_NOTICE","got invite"); if($sp=="5061"){ xlog("L_NOTICE","got invite 5061"); xlog("L_NOTICE","[$mi] before B2B request\n"); $ru="sip:"+$tU+"@172.22.2.140:5060"; b2b_init_request("top hiding"); xlog("L_INFO","[$mi] after B2B request\n"); }else{ xlog("L_NOTICE","got invite $sp"); xlog("L_INFO","[$mi] not from 5061\n"); } exit; }else{ xlog("L_ERR","got request method $rm from $si: $fU, $tU"); sl_send_reply("501", "Not Implemented"); exit; } } Kevin Stewart | Senior VOIP Network Engineer D: +64 9 919 6120E: Kevin.Stewart at m2group.co.nz M: +64 21 879 057W: vocus.co.nz A: PO Box 108-109, Symonds St, Auckland 1150 -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image935000.jpg Type: image/png Size: 14865 bytes Desc: image935000.jpg URL: From bogdan at opensips.org Tue Jun 14 09:17:55 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 14 Jun 2016 10:17:55 +0300 Subject: [OpenSIPS-Users] OpenSIPS 2.2 Documentation needs update ! In-Reply-To: References: <575F2556.4070005@opensips.org> Message-ID: <575FAFA3.9000202@opensips.org> Hi Sammy, Many thanks once again, I just fixed the link. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 14.06.2016 08:06, SamyGo wrote: > Nice, Another one: > I've another one with wrong ref link.: > proto_bin points to proto_hep in this list: > http://opensips.org/Documentation/Modules-2-2 > > I had to manually edit the URL to view proto_bin page. > > Thanks, > Sammy > > > On Mon, Jun 13, 2016 at 5:27 PM, Bogdan-Andrei Iancu > > wrote: > > Hi Sammy, > > Thank you for the report, I just updated the doc in regards to the > missing dependency : > https://github.com/OpenSIPS/opensips/commit/4357fe858c948ded5db4b37fec5d26c7c379652a > > The online docs will get regenerated during the night. > > Thanks and regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 13.06.2016 23:11, SamyGo wrote: >> Hi, >> >> It seems like mentioning of clusterer module as a dependency is >> missing form module docs: >> >> http://www.opensips.org/html/docs/modules/2.2.x/usrloc.html#id293640 >> >> Although it is obvious that the write refers to "valid cluster >> id" in function accept_replicated_contact() >> but >> I still think documentation should be updated. >> >> Adding replication modparams in opensips.cfg w/o clusterer also >> states the obvious: >> >> WARNING:core:solve_module_dependencies: module usrloc depends on >> module clusterer, but it was not loaded! >> ERROR:core:main: failed to solve module dependencies >> >> Regards, >> Sammy >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jun 14 11:53:53 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 14 Jun 2016 12:53:53 +0300 Subject: [OpenSIPS-Users] B2BUA not responding to reinvites In-Reply-To: References: Message-ID: <575FD431.3070708@opensips.org> Hello Kevin, Once the B2B session started, the sequential request will not land to your script as they will be captured by B2B before the script (and handled according to the XML script). Still, with debug level 6 you should see various messages from OpenSIPS when the B2B would handle the re-INVITE. Are you sure the re-INVITE is actually reaching OpenSIPS on the right IP and port ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 14.06.2016 09:49, Kevin Stewart wrote: > I am trying to implement a simple topology hiding B2BUA with opensips > 1.11.5 > > most things work except that re invites for session expiry are being > ignored. > > I assume that I am missing something in my config. > > the strange thing is that I see no debug to syslog when the Invites > come in even with debug set to 9. > > the basic config is a client on port 5061 (C) inviting the B2BUA on > port 5066(B) with then invites the main server on another host on port > 5060(A). > I do not expect and inbount calls from A > > C invites B, B invites A all proceeds normally until after 15 minutes > A invites B and is ignored then sends bye a number of times. > none of the reinvites or byes are reported in the logs. > > C then hangs up sending a bye to B, B sends a bye to A and gets 481 > Unknown Dialog. > > below is my config > > route{ > if ( !mf_process_maxfwd_header("10") ) > { > sl_send_reply("483","To Many Hops"); > drop(); > }; > > if (is_method("OPTIONS")) > { > #xlog("L_NOTICE", "$ci|end|unsupported method"); > sl_send_reply("404", "Not found"); > exit; > } > > if(is_method("INVITE")){ > xlog("L_NOTICE","got invite"); > if($sp=="5061"){ > xlog("L_NOTICE","got invite 5061"); > xlog("L_NOTICE","[$mi] before B2B request\n"); > $ru="sip:"+$tU+"@172.22.2.140:5060"; > b2b_init_request("top hiding"); > xlog("L_INFO","[$mi] after B2B request\n"); > }else{ > xlog("L_NOTICE","got invite $sp"); > xlog("L_INFO","[$mi] not from 5061\n"); > } > exit; > }else{ > xlog("L_ERR","got request method $rm from $si: $fU, $tU"); > sl_send_reply("501", "Not Implemented"); > exit; > } > } > > Kevin Stewart | Senior VOIP Network Engineer > > D: *+64 9 919 6120* E: > *Kevin.Stewart at m2group.co.nz* > > M: *+64 21 879 057* W: *vocus.co.nz* > > > A: PO Box 108-109, Symonds St, Auckland 1150 > > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 14865 bytes Desc: not available URL: From bogdan at opensips.org Tue Jun 14 12:07:12 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 14 Jun 2016 13:07:12 +0300 Subject: [OpenSIPS-Users] Userloc, Bin Interface, and Clusterer combo In-Reply-To: References: Message-ID: <575FD750.5010601@opensips.org> Hi Sammy, The registration records are usually bound to an interface (the ip and port the registration was received) - that interface will be used all the time in the communication with the registered end point. Now, when if server A receives the Registration on interface X, and this is replicated to server B, this server B may not have the X interface, but something else (an Y interface). So the registration record will loose some information (the interface it is bound to). This is the nature of the error message you see. And yes, using the ip_nonlocal_bind options is the way to go. Once replicated, the registration records are independent and they do not depend on the what is happening with the original node (where the registration was pushed from). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 14.06.2016 09:35, SamyGo wrote: > Hi, > I've a cluster setup with userloc replication enabled for 3+ servers. > As soon as a user register anywhere the other servers which receive > this replicated contact display these errors: > > ERROR:usrloc:receive_ucontact_insert: non-local socket > > ERROR:usrloc:receive_ucontact_insert: failed to process replication > event. dom: 'location', aor: '89654 at 1X.2XX.XX.XX' > ERROR:usrloc:receive_binary_packet: failed to process a binary packet! > > The solution(kind of) for the same error has been discussed in this > thread: > http://lists.opensips.org/pipermail/users/2015-February/030910.html > > The question here is, is setting /ip_nonlocal_bind /for this > userlocation replication a solution ? or should the replication > mechanism be modified to process the replications from other nodes ! > > Say even if this works, the next question is, what would happen to the > contacts from a server which is no longer active. The Contacts > disappear as soon as the clusterer node becomes inactive ? > > Looking for some clarity on this topic. > > Regards, > Sammy > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jun 14 12:08:32 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 14 Jun 2016 13:08:32 +0300 Subject: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions In-Reply-To: References: Message-ID: <575FD7A0.6000601@opensips.org> Hi Nabeel, We will update the tutorial for 2.2, but it should still match. Give it a try and if you hit issues, just let me know. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12.06.2016 10:18, Nabeel wrote: > > Hi, > > I will be following this tutorial to integrate OpenSIPS and Asterisk: > > http://www.opensips.org/Documentation/Tutorials-OpenSIPSAsteriskIntegration-1-8 > > The tutorial mentions the use of OpenSIPS version 1.8 and Asterisk > version 1.8. I would like to know if I can use the latest versions of > OpenSIPS and Asterisk instead? Have there been changes to database > structure which can cause problems? > > Nabeel > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From k.galinurov at gmail.com Tue Jun 14 12:42:06 2016 From: k.galinurov at gmail.com (Kirill Galinurov) Date: Tue, 14 Jun 2016 13:42:06 +0300 Subject: [OpenSIPS-Users] Opensips2.2 Asterisk Sub registartion Message-ID: Hi all. How i can do subregistration on asterisk like in kamaillio. http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb . I have some webrtc users/ They register on opensips. -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jun 14 13:29:33 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 14 Jun 2016 14:29:33 +0300 Subject: [OpenSIPS-Users] Opensips2.2 Asterisk Sub registartion In-Reply-To: References: Message-ID: <575FEA9D.2090101@opensips.org> Hi Kirill, I suppose you want to pass the registration to Asterisk, after is was authenticated (and eventually saved) by OpenSIPS, right ? To do this in the correct way (from SIP perspective), after you have authenticated the REGISTER on OpenSIPS, you forward it to Asterisk _WITHOUT_ saving it into OpenSIPS. When you receive the reply back from Asterisk, you do the save("location") in OpenSIPS. Why ? as in this registration chain, Asterisk is the main final registrar and it may change the registration expire value: Ex: * UAC sends register with 100 secs to OpenSIPS * OpenSIPS save() and has UAC registered for 100 secs * OpenSIPS relays/creates register to Asterisk with 100 secs * Asterisk decides to store registration only for 60 secs and returns reply with expires=60 Result ? OpenSIPS will have UAC registered for 100 and Asterisk for 60, while the UAC thinks it is registered for 100 (according to the reply from OpenSIPS) - this is totally bogus. Correct flow: * UAC sends register with 100 secs to OpenSIPS * OpenSIPS relays/creates register to Asterisk with 100 secs * Asterisk decides to store registration only for 60 secs and returns reply with expires=60 * OpenSIPS receives the reply from Asterisk and saves UAC registration for 60 secs too * OpenSIPS sends reply to UAC withe expires 60 Result ? OpenSIPS will have UAC registered for 60 and so Asterisk ; UAC thinks it is registered for 60 (according to the reply from OpenSIPS) - this is 100% correct. Conclusion - when you do have a registration chain, on the intermediary registrar you do save() on 200 OK as the next registrar may change the expire value. So, simply use the save() function in onreply_route when receiving the REGISTER reply from Asterisk: http://www.opensips.org/html/docs/modules/2.2.x/registrar.html#id294033 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 14.06.2016 13:42, Kirill Galinurov wrote: > Hi all. How i can do subregistration on asterisk like in kamaillio. > http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb. > > I have some webrtc users/ They register on opensips. > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From husnain.taseer at gmail.com Tue Jun 14 14:05:57 2016 From: husnain.taseer at gmail.com (Husnain Taseer) Date: Tue, 14 Jun 2016 15:05:57 +0300 Subject: [OpenSIPS-Users] Break in while loop Message-ID: Dear Users, I am trying to break the while loop in opensips.cfg when a particular condition meet. For that I am using 'break' statement inside a while loop. But when I type break it gives me sytax error : Jun 14 06:56:00 s81519 opensips: CRITICAL:core:yyerror: parse error in config file /usr/local/etc/opensips/opensips.cfg, line 697, column 5-10: syntax error Jun 14 06:56:00 s81519 opensips: CRITICAL:core:yyerror: parse error in config file /usr/local/etc/opensips/opensips.cfg, line 697, column 5-10: bad command!) Jun 14 06:56:00 s81519 opensips: CRITICAL:core:yyerror: parse error in config file /usr/local/etc/opensips/opensips.cfg, line 697, column 5-10: bad command!) Jun 14 06:56:00 s81519 opensips: ERROR:core:main: bad config file (3 errors) Jun 14 06:56:00 s81519 opensips: NOTICE:core:main: Exiting.... If I will use return instead of break it will return from the current route which is also not my requirement. Below is the code snippet. $var(i) = 0; while($var(i) < 10) { usleep("20000"); cache_raw_query("redis:group2","HGETALL $avp(dialed)","$avp(result)"); if ($avp(result) != NULL) { $avp(CALLER) = $(avp(result)[1]); $avp(LRN) = $(avp(result)[3]); xlog("L_NOTICE","[$Ts:$avp(cid)]: LRN Returned '$avp(LRN)'"); break; #Line number 697 } $var(i) = $var(i) + 1; } Please guide. Regards, Husnain Taseer VoIP Developer -------------- next part -------------- An HTML attachment was scrubbed... URL: From govoiper at gmail.com Tue Jun 14 14:21:18 2016 From: govoiper at gmail.com (SamyGo) Date: Tue, 14 Jun 2016 08:21:18 -0400 Subject: [OpenSIPS-Users] Userloc, Bin Interface, and Clusterer combo In-Reply-To: <575FD750.5010601@opensips.org> References: <575FD750.5010601@opensips.org> Message-ID: Thanks Bogdan, Will it be correct to say that usrloc replication is not the way to go if a mesh cluster of OpenSIPS, with every node knowing all the online subscribers anywhere, is to be created. ! Also, even if I do set ip_nonlocal_bind would each OpenSIPS getting this replicated registration start sending keepalives to that subscriber too? Do you think a shared redis cluster for storing AoRs is more suitable for the task instead of this approach! Thanks, Sammy On Jun 14, 2016 06:07, "Bogdan-Andrei Iancu" wrote: > Hi Sammy, > > The registration records are usually bound to an interface (the ip and > port the registration was received) - that interface will be used all the > time in the communication with the registered end point. > > Now, when if server A receives the Registration on interface X, and this > is replicated to server B, this server B may not have the X interface, but > something else (an Y interface). So the registration record will loose some > information (the interface it is bound to). This is the nature of the error > message you see. > And yes, using the ip_nonlocal_bind options is the way to go. > > Once replicated, the registration records are independent and they do not > depend on the what is happening with the original node (where the > registration was pushed from). > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 14.06.2016 09:35, SamyGo wrote: > > Hi, > I've a cluster setup with userloc replication enabled for 3+ servers. As > soon as a user register anywhere the other servers which receive this > replicated contact display these errors: > > ERROR:usrloc:receive_ucontact_insert: non-local socket > > ERROR:usrloc:receive_ucontact_insert: failed to process replication event. > dom: 'location', aor: '89654 at 1X.2XX.XX.XX' > ERROR:usrloc:receive_binary_packet: failed to process a binary packet! > > The solution(kind of) for the same error has been discussed in this thread: > http://lists.opensips.org/pipermail/users/2015-February/030910.html > > The question here is, is setting *ip_nonlocal_bind *for this userlocation > replication a solution ? or should the replication mechanism be modified to > process the replications from other nodes ! > > Say even if this works, the next question is, what would happen to the > contacts from a server which is no longer active. The Contacts disappear as > soon as the clusterer node becomes inactive ? > > Looking for some clarity on this topic. > > Regards, > Sammy > > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From govoiper at gmail.com Tue Jun 14 14:54:24 2016 From: govoiper at gmail.com (SamyGo) Date: Tue, 14 Jun 2016 08:54:24 -0400 Subject: [OpenSIPS-Users] Break in while loop In-Reply-To: References: Message-ID: Hi Husnain, Simple *return* should work fine in your scenario but if everything fails you can do some additional condition check in while() loop to break it ! for example: $avp(flag) = 0; while( ($var(i) < 10) && $avp(flag) != 1) { if(condition){ $avp(flag) = 1; } } I checked some of my other scripts and return was doing its job just fine. Make sure you only execute 'return' only and not 'return(1)'; Regards, Sammy On Tue, Jun 14, 2016 at 8:05 AM, Husnain Taseer wrote: > Dear Users, > I am trying to break the while loop in opensips.cfg when a particular > condition meet. For that I am using 'break' statement inside a while loop. > But when I type break it gives me sytax error : > > Jun 14 06:56:00 s81519 opensips: CRITICAL:core:yyerror: parse error in > config file /usr/local/etc/opensips/opensips.cfg, line 697, column 5-10: > syntax error > Jun 14 06:56:00 s81519 opensips: CRITICAL:core:yyerror: parse error in > config file /usr/local/etc/opensips/opensips.cfg, line 697, column 5-10: > bad command!) > Jun 14 06:56:00 s81519 opensips: CRITICAL:core:yyerror: parse error in > config file /usr/local/etc/opensips/opensips.cfg, line 697, column 5-10: > bad command!) > Jun 14 06:56:00 s81519 opensips: ERROR:core:main: bad config file (3 > errors) > Jun 14 06:56:00 s81519 opensips: NOTICE:core:main: Exiting.... > > If I will use return instead of break it will return from the current > route which is also not my requirement. Below is the code snippet. > > $var(i) = 0; > while($var(i) < 10) { > usleep("20000"); > cache_raw_query("redis:group2","HGETALL $avp(dialed)","$avp(result)"); > if ($avp(result) != NULL) { > $avp(CALLER) = $(avp(result)[1]); > $avp(LRN) = $(avp(result)[3]); > xlog("L_NOTICE","[$Ts:$avp(cid)]: LRN Returned '$avp(LRN)'"); > break; #Line number 697 > } > $var(i) = $var(i) + 1; > } > > Please guide. > > Regards, > Husnain Taseer > VoIP Developer > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From Agalya_Ramachandran at comcast.com Tue Jun 14 16:39:34 2016 From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor)) Date: Tue, 14 Jun 2016 14:39:34 +0000 Subject: [OpenSIPS-Users] Provisional response handling in case of forking In-Reply-To: <575F1EA8.50907@opensips.org> References: <0686390e80e344cf9c2cff77c7b1dfd1@COPDCEX28.cable.comcast.com> <575E8B4E.2000708@opensips.org> <8c8f9b61f3c94b37a7107631e2950a5c@COPDCEX28.cable.comcast.com> <575EDB89.2050200@opensips.org> <4526791507af43cc93131886ee3a3cde@COPDCEX28.cable.comcast.com> <575F1EA8.50907@opensips.org> Message-ID: Hi Bogdan, Thanks for your kind response. Am clear now. Regards, Agalya From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Monday, June 13, 2016 4:59 PM To: Ramachandran, Agalya (Contractor) ; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Provisional response handling in case of forking Hi Agalya, For the first 180, the flag is not set (by default, the flags are off), so no drop will be done and the execution will get to the setflag(), setting the flag. (only the drop is under the IF). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 13.06.2016 22:48, Ramachandran, Agalya (Contractor) wrote: Hi Bogdan, OK. Got it. One more to question to understand the snippet you gave.. I have just commented my understanding of each snippet at the same line. I don't get why we need setflag(FLAG_180); after we drop the response. if ($rs==180) { //We are checking response is equal to 180 if (isflagset(FLAG_180)) // Checking the flag is set or not. If response is 180, flag will be set drop(); //If set we are dropping the consecutive same response. setflag(FALG_180); ?? } (P.S) I tested without setting the flag after drop response. In this case it forwards both 180 response back to caller. Am curious how it controls it. Is onreply_avp_mode controls dropping the 2nd response or setflag(FLAG_180);? I want to be clear on what am working with. Hence posting you this question. Regards, Agalya From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Monday, June 13, 2016 12:13 PM To: Ramachandran, Agalya (Contractor) ; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Provisional response handling in case of forking Hi Agalya, It is not a predefine flag. And you do not have to define the flags in OpenSIPS. Just pickup a name and start using it in script. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 13.06.2016 17:27, Ramachandran, Agalya (Contractor) wrote: Hi Bogdan, Thank you for your kind response. I will try the logic you told me. But I have a question in the below logic. "FLAG_180", is this something declared and maintained by opensips and set by default on its own if Ringing response is received or we need to define this flag explicitly ? E.g for setting NAT flag, we are using 'usrloc' module and 'nat_bflag' as parameter and then value. modparam("usrloc", "nat_bflag", "NAT") Likewise if we need to define explicitly, what is the module in which we need to define the flag for FLAG_180? Regards, Agalya From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Monday, June 13, 2016 6:31 AM To: OpenSIPS users mailling list ; Ramachandran, Agalya (Contractor) Subject: Re: [OpenSIPS-Users] Provisional response handling in case of forking Hi Agalya, Use the onreply route (be sure to onreply_avp_mode to be set to 1 - see http://www.opensips.org/html/docs/modules/2.2.x/tm.html#id294290) in combination with flags, to record when the an 180 reply was set. Like: if ($rs==180) { if (isflagset(FLAG_180)) drop(); setflag(FALG_180); } The onreply_avp_mode 1 will ensure that the onreply route will not overlap for 2 replies . Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 10.06.2016 23:07, Ramachandran, Agalya (Contractor) wrote: Hi team, We are using opensips for our project and we are currently using opensips as proxy. Am forking the incoming sip call, to two destinations. It Rings in both Dest A and Dest B, as a result I get two 180 Ringing response from A and B. I want to filter only the first incoming 180 Ringing response and send to the actual caller. Is there a way to do this in opensips config file? I have seen drop() function which drops the complete provisional response. But in my case I have to forward one 180 Ringing to the caller. Can it be achieved by the changes in config file? Please guide me. Regards, Agalya _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From goup2010 at gmail.com Wed Jun 15 09:38:10 2016 From: goup2010 at gmail.com (Dragomir Haralambiev) Date: Wed, 15 Jun 2016 10:38:10 +0300 Subject: [OpenSIPS-Users] File name and rtpproxy_start_recording Message-ID: Hello , How to setup file name when using rtpproxy_start_recording? Best regards, Dragomir -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Jun 15 12:11:54 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 15 Jun 2016 13:11:54 +0300 Subject: [OpenSIPS-Users] Userloc, Bin Interface, and Clusterer combo In-Reply-To: References: <575FD750.5010601@opensips.org> Message-ID: <576129EA.3090909@opensips.org> Hi Sammy, Yes, you are correct . The current clustering support for usrloc covers only simple replication (usually for HA purposes). The next step is to achieve distribution and partitioning of registrations across the cluster (this is work in progress) and it will be available in the 2.3 release. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 14.06.2016 15:21, SamyGo wrote: > > Thanks Bogdan, > > Will it be correct to say that usrloc replication is not the way to go > if a mesh cluster of OpenSIPS, with every node knowing all the online > subscribers anywhere, is to be created. ! > > Also, even if I do set ip_nonlocal_bind would each OpenSIPS getting > this replicated registration start sending keepalives to that > subscriber too? > > Do you think a shared redis cluster for storing AoRs is more suitable > for the task instead of this approach! > > Thanks, > Sammy > > On Jun 14, 2016 06:07, "Bogdan-Andrei Iancu" > wrote: > > Hi Sammy, > > The registration records are usually bound to an interface (the ip > and port the registration was received) - that interface will be > used all the time in the communication with the registered end point. > > Now, when if server A receives the Registration on interface X, > and this is replicated to server B, this server B may not have the > X interface, but something else (an Y interface). So the > registration record will loose some information (the interface it > is bound to). This is the nature of the error message you see. > And yes, using the ip_nonlocal_bind options is the way to go. > > Once replicated, the registration records are independent and they > do not depend on the what is happening with the original node > (where the registration was pushed from). > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 14.06.2016 09:35, SamyGo wrote: >> Hi, >> I've a cluster setup with userloc replication enabled for 3+ >> servers. As soon as a user register anywhere the other servers >> which receive this replicated contact display these errors: >> >> ERROR:usrloc:receive_ucontact_insert: non-local socket >> >> ERROR:usrloc:receive_ucontact_insert: failed to process >> replication event. dom: 'location', aor: '89654 at 1X.2XX.XX.XX >> ' >> ERROR:usrloc:receive_binary_packet: failed to process a binary >> packet! >> >> The solution(kind of) for the same error has been discussed in >> this thread: >> http://lists.opensips.org/pipermail/users/2015-February/030910.html >> >> The question here is, is setting /ip_nonlocal_bind /for this >> userlocation replication a solution ? or should the replication >> mechanism be modified to process the replications from other nodes ! >> >> Say even if this works, the next question is, what would happen >> to the contacts from a server which is no longer active. The >> Contacts disappear as soon as the clusterer node becomes inactive ? >> >> Looking for some clarity on this topic. >> >> Regards, >> Sammy >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Wed Jun 15 14:44:10 2016 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Wed, 15 Jun 2016 15:44:10 +0300 Subject: [OpenSIPS-Users] File name and rtpproxy_start_recording In-Reply-To: References: Message-ID: <6c0e3a4c-caa2-c8e7-c609-981fe232ec1f@opensips.org> Hi, Dragomir! There is no way to set a filename of the recording. It is auto-generated by rtpproxy. Best regards, R?zvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 06/15/2016 10:38 AM, Dragomir Haralambiev wrote: > Hello , > How to setup file name when using rtpproxy_start_recording? > Best regards, > Dragomir > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From nocbgtelcom at gmail.com Wed Jun 15 15:49:32 2016 From: nocbgtelcom at gmail.com (Hristo Donev) Date: Wed, 15 Jun 2016 16:49:32 +0300 Subject: [OpenSIPS-Users] File name and rtpproxy_start_recording In-Reply-To: <6c0e3a4c-caa2-c8e7-c609-981fe232ec1f@opensips.org> References: <6c0e3a4c-caa2-c8e7-c609-981fe232ec1f@opensips.org> Message-ID: You can use Asterisk for RTP proxy. In this case you have file recording, transcoding and more other ..... 2016-06-15 15:44 GMT+03:00 R?zvan Crainea : > Hi, Dragomir! > > There is no way to set a filename of the recording. It is auto-generated > by rtpproxy. > > Best regards, > > R?zvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 06/15/2016 10:38 AM, Dragomir Haralambiev wrote: > > Hello , > How to setup file name when using rtpproxy_start_recording? > Best regards, > Dragomir > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From john.nash778 at gmail.com Wed Jun 15 18:57:57 2016 From: john.nash778 at gmail.com (John Nash) Date: Wed, 15 Jun 2016 22:27:57 +0530 Subject: [OpenSIPS-Users] codec_delete function issue Message-ID: I tried this function to delete "NSE" codec from SDP but it doesnt seem to be working. Any consideration to make it work?...I am using rtpengine module but calling it after calling codec_delete but this codec still passes to other end. -------------- next part -------------- An HTML attachment was scrubbed... URL: From pimenta at inatel.br Wed Jun 15 20:36:46 2016 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Wed, 15 Jun 2016 18:36:46 +0000 Subject: [OpenSIPS-Users] Script to get attr_avp from caller. How to do? Message-ID: Hi. In my script I use to use AVP to store specific additional information for each registration. When I have to get the attr_avp (Module Registrar) from the callee, during a call, I use to codify: branch_route[per_branch_ops] { myVariable = $(avp(attr)[$T_branch_idx] } What about the caller? I also need to get the attr_avp from caller, but I can't find a similar code in OpenSIPS documentation. How to do? P.S.: I'm using the configuration: modparam("usrloc", "db_mode", 0) Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Thu Jun 16 10:46:45 2016 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Thu, 16 Jun 2016 11:46:45 +0300 Subject: [OpenSIPS-Users] codec_delete function issue In-Reply-To: References: Message-ID: <52c1d34c-bb10-0523-95fc-2c0bbc40b7ca@opensips.org> Hi, John! So you want to remove a codec that is sent to rtpengine? If so, unfortunately you cannot do that. The only way is to first delete the codec, then loop the message to yourself (send it to yourself) and call rtpengine. Regards, R?zvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 06/15/2016 07:57 PM, John Nash wrote: > I tried this function to delete "NSE" codec from SDP but it doesnt seem > to be working. Any consideration to make it work?...I am using rtpengine > module but calling it after calling codec_delete but this codec still > passes to other end. > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > From razvan at opensips.org Thu Jun 16 10:58:05 2016 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Thu, 16 Jun 2016 11:58:05 +0300 Subject: [OpenSIPS-Users] Script to get attr_avp from caller. How to do? In-Reply-To: References: Message-ID: Hi, Rodrigo! You can do something like this: lookup("location","","$fu"); # note that this might fail, so you'll have to treat this separately $avp(caller_attr) = $avp(attr); # note that the caller might have multiple contacts/attributes, so you'll have to handle this Then continue with your logic, call lookup("location") and will load the attributes of the callee, per branch, as in your snippet. Regards, R?zvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 06/15/2016 09:36 PM, Rodrigo Pimenta Carvalho wrote: > > Hi. > > > In my script I use to use AVP to store specific additional information > for each registration. > > When I have to get the attr_avp (Module Registrar) from the callee, > during a call, I use to codify: > > > branch_route[per_branch_ops] { > > myVariable = $(avp(attr)[$T_branch_idx] > > } > > > What about the caller? I also need to get the attr_avp from caller, but > I can't find a similar code in OpenSIPS documentation. How to do? > > > P.S.: I'm using the configuration: modparam("usrloc", "db_mode", 0) > > > Any hint will be very helpful! > > Best regards. > > > RODRIGO PIMENTA CARVALHO > Inatel Competence Center > Software > Ph: +55 35 3471 9200 RAMAL 979 > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > From pimenta at inatel.br Thu Jun 16 14:10:15 2016 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Thu, 16 Jun 2016 12:10:15 +0000 Subject: [OpenSIPS-Users] Script to get attr_avp from caller. How to do? In-Reply-To: References: , Message-ID: Hi. Razvan Crainea. Thank you very much! I will try it. As I can see, I should start by understanding the function lookup and is capabilities very well. Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________ De: users-bounces at lists.opensips.org em nome de Razvan Crainea Enviado: quinta-feira, 16 de junho de 2016 05:58 Para: users at lists.opensips.org Assunto: Re: [OpenSIPS-Users] Script to get attr_avp from caller. How to do? Hi, Rodrigo! You can do something like this: lookup("location","","$fu"); # note that this might fail, so you'll have to treat this separately $avp(caller_attr) = $avp(attr); # note that the caller might have multiple contacts/attributes, so you'll have to handle this Then continue with your logic, call lookup("location") and will load the attributes of the callee, per branch, as in your snippet. Regards, Razvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com [http://www.opensips-solutions.com/imgs/slideshow/slide1.jpg] Home - OpenSIPS Solutions www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 06/15/2016 09:36 PM, Rodrigo Pimenta Carvalho wrote: > > Hi. > > > In my script I use to use AVP to store specific additional information > for each registration. > > When I have to get the attr_avp (Module Registrar) from the callee, > during a call, I use to codify: > > > branch_route[per_branch_ops] { > > myVariable = $(avp(attr)[$T_branch_idx] > > } > > > What about the caller? I also need to get the attr_avp from caller, but > I can't find a similar code in OpenSIPS documentation. How to do? > > > P.S.: I'm using the configuration: modparam("usrloc", "db_mode", 0) > > > Any hint will be very helpful! > > Best regards. > > > RODRIGO PIMENTA CARVALHO > Inatel Competence Center > Software > Ph: +55 35 3471 9200 RAMAL 979 > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From arsen.semionov at gmail.com Thu Jun 16 15:55:04 2016 From: arsen.semionov at gmail.com (Arsen) Date: Thu, 16 Jun 2016 16:55:04 +0300 Subject: [OpenSIPS-Users] Opensips 2.2 Crash Message-ID: Hi guys, We have upgraded to 2.2 but seems it's crashing occasionally. Please take a look at the bt full http://pastebin.com/HF96EYwn thanks in advance! -- Regards, Arsen. -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at inin.com Thu Jun 16 16:09:59 2016 From: Ben.Newlin at inin.com (Newlin, Ben) Date: Thu, 16 Jun 2016 14:09:59 +0000 Subject: [OpenSIPS-Users] Bad 1.11.7 tarball on open sips.org Message-ID: It looks like a new tarball for OpenSIPS 1.11.7 was uploaded earlier today, but it appears to have a bad format. I cannot extract it with any of my usual archive utilities. Can anyone else confirm? It was working fine yesterday with the previous tarball. http://opensips.org/pub/opensips/1.11.7/opensips-1.11.7.tar.gz Ben Newlin -------------- next part -------------- An HTML attachment was scrubbed... URL: From Rahul.Gupta at ipc.com Thu Jun 16 17:55:23 2016 From: Rahul.Gupta at ipc.com (Gupta, Rahul) Date: Thu, 16 Jun 2016 15:55:23 +0000 Subject: [OpenSIPS-Users] Method Name from 200OK Message-ID: <5D7DF326E497124DACCD6F9DD6A1A2A0F1D2E8E5@NWKNJEXMBX1.corp.root.ipc.com> Hi, I am using opensips as stateless proxy. What can I use to find a Reply (200 OK) is from which Request Method in onreply_route ? I just need Method Name of the reply ? Thanks Rahul Gupta ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------ DISCLAIMER: This e-mail may contain information that is confidential, privileged or otherwise protected from disclosure. If you are not an intended recipient of this e-mail, do not duplicate or redistribute it by any means. Please delete it and any attachments and notify the sender that you have received it in error. Unintended recipients are prohibited from taking action on the basis of information in this e-mail.E-mail messages may contain computer viruses or other defects, may not be accurately replicated on other systems, or may be intercepted, deleted or interfered with without the knowledge of the sender or the intended recipient. If you are not comfortable with the risks associated with e-mail messages, you may decide not to use e-mail to communicate with IPC. IPC reserves the right, to the extent and under circumstances permitted by applicable law, to retain, monitor and intercept e-mail messages to and from its systems. -------------- next part -------------- An HTML attachment was scrubbed... URL: From Rahul.Gupta at ipc.com Thu Jun 16 18:24:25 2016 From: Rahul.Gupta at ipc.com (Gupta, Rahul) Date: Thu, 16 Jun 2016 16:24:25 +0000 Subject: [OpenSIPS-Users] Method Name from 200OK In-Reply-To: <5D7DF326E497124DACCD6F9DD6A1A2A0F1D2E8E5@NWKNJEXMBX1.corp.root.ipc.com> References: <5D7DF326E497124DACCD6F9DD6A1A2A0F1D2E8E5@NWKNJEXMBX1.corp.root.ipc.com> Message-ID: <5D7DF326E497124DACCD6F9DD6A1A2A0F1D2E944@NWKNJEXMBX1.corp.root.ipc.com> Nevermind, I can just use is_method(). If there is another easy way to get the method name directly in a variable then please share. Thanks Rahul Gupta From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Gupta, Rahul Sent: Thursday, June 16, 2016 11:55 AM To: users at lists.opensips.org Subject: [OpenSIPS-Users] Method Name from 200OK Hi, I am using opensips as stateless proxy. What can I use to find a Reply (200 OK) is from which Request Method in onreply_route ? I just need Method Name of the reply ? Thanks Rahul Gupta ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------ DISCLAIMER: This e-mail may contain information that is confidential, privileged or otherwise protected from disclosure. If you are not an intended recipient of this e-mail, do not duplicate or redistribute it by any means. Please delete it and any attachments and notify the sender that you have received it in error. Unintended recipients are prohibited from taking action on the basis of information in this e-mail.E-mail messages may contain computer viruses or other defects, may not be accurately replicated on other systems, or may be intercepted, deleted or interfered with without the knowledge of the sender or the intended recipient. If you are not comfortable with the risks associated with e-mail messages, you may decide not to use e-mail to communicate with IPC. IPC reserves the right, to the extent and under circumstances permitted by applicable law, to retain, monitor and intercept e-mail messages to and from its systems. -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Thu Jun 16 18:28:33 2016 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Thu, 16 Jun 2016 19:28:33 +0300 Subject: [OpenSIPS-Users] Bad 1.11.7 tarball on open sips.org In-Reply-To: References: Message-ID: Hi, Ben! Seems to be an issue of GitHub. Please follow [1]. [1] https://github.com/OpenSIPS/opensips/issues/907 Best regards, R?zvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 06/16/2016 05:09 PM, Newlin, Ben wrote: > > It looks like a new tarball for OpenSIPS 1.11.7 was uploaded earlier > today, but it appears to have a bad format. I cannot extract it with > any of my usual archive utilities. Can anyone else confirm? > > It was working fine yesterday with the previous tarball. > > http://opensips.org/pub/opensips/1.11.7/opensips-1.11.7.tar.gz > > Ben Newlin > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From nick.altmann at gmail.com Thu Jun 16 18:57:29 2016 From: nick.altmann at gmail.com (Nick Altmann) Date: Thu, 16 Jun 2016 19:57:29 +0300 Subject: [OpenSIPS-Users] Bad 1.11.7 tarball on open sips.org In-Reply-To: References: Message-ID: There is also alternative download place you may use: http://download.opensips.org -- Nick 2016-06-16 19:28 GMT+03:00 R?zvan Crainea : > Hi, Ben! > > Seems to be an issue of GitHub. Please follow [1]. > > [1] https://github.com/OpenSIPS/opensips/issues/907 > > Best regards, > > R?zvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 06/16/2016 05:09 PM, Newlin, Ben wrote: > > It looks like a new tarball for OpenSIPS 1.11.7 was uploaded earlier > today, but it appears to have a bad format. I cannot extract it with any of > my usual archive utilities. Can anyone else confirm? > > > > It was working fine yesterday with the previous tarball. > > > > http://opensips.org/pub/opensips/1.11.7/opensips-1.11.7.tar.gz > > > > Ben Newlin > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Jun 17 10:52:42 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 17 Jun 2016 11:52:42 +0300 Subject: [OpenSIPS-Users] Opensips 2.2 Crash In-Reply-To: References: Message-ID: <5763BA5A.90302@opensips.org> Topic moved on devel list: http://lists.opensips.org/pipermail/devel/2016-June/020379.html Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 16.06.2016 16:55, Arsen wrote: > Hi guys, > > We have upgraded to 2.2 but seems it's crashing occasionally. > > Please take a look at the bt full > http://pastebin.com/HF96EYwn > > thanks in advance! > > -- > Regards, > Arsen. > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Jun 17 10:58:22 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 17 Jun 2016 11:58:22 +0300 Subject: [OpenSIPS-Users] Method Name from 200OK In-Reply-To: <5D7DF326E497124DACCD6F9DD6A1A2A0F1D2E944@NWKNJEXMBX1.corp.root.ipc.com> References: <5D7DF326E497124DACCD6F9DD6A1A2A0F1D2E8E5@NWKNJEXMBX1.corp.root.ipc.com> <5D7DF326E497124DACCD6F9DD6A1A2A0F1D2E944@NWKNJEXMBX1.corp.root.ipc.com> Message-ID: <5763BBAE.4080708@opensips.org> Hi Rahul, Even if it is a reply, use $rm - it will extract the method from the cseq hdr if reply. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 16.06.2016 19:24, Gupta, Rahul wrote: > > Nevermind, I can just use is_method(). If there is another easy way to > get the method name directly in a variable then please share. > > Thanks > > Rahul Gupta > > *From:* users-bounces at lists.opensips.org > [mailto:users-bounces at lists.opensips.org] *On Behalf Of *Gupta, Rahul > *Sent:* Thursday, June 16, 2016 11:55 AM > *To:* users at lists.opensips.org > *Subject:* [OpenSIPS-Users] Method Name from 200OK > > Hi, I am using opensips as stateless proxy. What can I use to find a > Reply (200 OK) is from which Request Method in onreply_route ? I just > need Method Name of the reply ? > > Thanks > > Rahul Gupta > > ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------ > > DISCLAIMER: This e-mail may contain information that is confidential, > privileged or otherwise protected from disclosure. If you are not an > intended recipient of this e-mail, do not duplicate or redistribute it > by any means. Please delete it and any attachments and notify the > sender that you have received it in error. Unintended recipients are > prohibited from taking action on the basis of information in this > e-mail.E-mail messages may contain computer viruses or other defects, > may not be accurately replicated on other systems, or may be > intercepted, deleted or interfered with without the knowledge of the > sender or the intended recipient. If you are not comfortable with the > risks associated with e-mail messages, you may decide not to use > e-mail to communicate with IPC. IPC reserves the right, to the extent > and under circumstances permitted by applicable law, to retain, > monitor and intercept e-mail messages to and from its systems. > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From pimenta at inatel.br Fri Jun 17 16:02:36 2016 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Fri, 17 Jun 2016 14:02:36 +0000 Subject: [OpenSIPS-Users] What is the best SQLite version to be used with OpenSIPS 2.2? We investigate a memory leak. Message-ID: Hi. People from my team is investigating a memory leak related to OpenSIPS 2.2. As I had commented in another discussion in the past, it seems that the problem comes from SQLite we are using as the Registrar for our OpenSIPS 2.2. For example, a script opensips.cfg that doesn't use SQLite didn't cause memory leak. But, a script that uses it and use another module that needs a database (EX: auth.so) causes memory leak. We are still in the beginning of the investigation. So, what is the best version of SQLite to be used with OpenSIPS 2.2? That is, what version of SQLite was very well tested with OpenSIPS 2.2 and worked without memory leaks or others issues? Any suggestion will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Fri Jun 17 16:14:17 2016 From: liviu at opensips.org (Liviu Chircu) Date: Fri, 17 Jun 2016 17:14:17 +0300 Subject: [OpenSIPS-Users] What is the best SQLite version to be used with OpenSIPS 2.2? We investigate a memory leak. In-Reply-To: References: Message-ID: <576405B9.2070605@opensips.org> Hi Rodrigo! A GitHub issue [1] regarding this leak was just reported today by Eric, so you can track the resolution process over there! You can even subscribe to that ticket if you have an account, in order to receive emails. [1]: https://github.com/OpenSIPS/opensips/issues/911 Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 17.06.2016 17:02, Rodrigo Pimenta Carvalho wrote: > > Hi. > > > People from my team is investigating a memory leak related to OpenSIPS > 2.2. > > > As I had commented in another discussion in the past, it seems that > the problem comes from SQLite we are using as the Registrar for our > OpenSIPS 2.2. > > For example, a script opensips.cfg that doesn't use SQLite didn't > cause memory leak. But, a script that uses it and use another module > that needs a database (EX: auth.so) causes memory leak. > > > We are still in the beginning of the investigation. > > So, what is the best version of SQLite to be used with OpenSIPS 2.2? > That is, what version of SQLite was very well tested with OpenSIPS 2.2 > and worked without memory leaks or others issues? > > > Any suggestion will be very helpful! > > > Best regards. > > > RODRIGO PIMENTA CARVALHO > Inatel Competence Center > Software > Ph: +55 35 3471 9200 RAMAL 979 > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From pimenta at inatel.br Fri Jun 17 16:19:31 2016 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Fri, 17 Jun 2016 14:19:31 +0000 Subject: [OpenSIPS-Users] What is the best SQLite version to be used with OpenSIPS 2.2? We investigate a memory leak. In-Reply-To: <576405B9.2070605@opensips.org> References: , <576405B9.2070605@opensips.org> Message-ID: Hi Liviu. Very good. We will see the resolution process. Thank you very much! Regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________ De: users-bounces at lists.opensips.org em nome de Liviu Chircu Enviado: sexta-feira, 17 de junho de 2016 11:14 Para: users at lists.opensips.org Assunto: Re: [OpenSIPS-Users] What is the best SQLite version to be used with OpenSIPS 2.2? We investigate a memory leak. Hi Rodrigo! A GitHub issue [1] regarding this leak was just reported today by Eric, so you can track the resolution process over there! You can even subscribe to that ticket if you have an account, in order to receive emails. [1]: https://github.com/OpenSIPS/opensips/issues/911 [https://avatars3.githubusercontent.com/u/21685?v=3&s=400] 2.2 runs out of pkg_mem because of db/db_res.c memory leak ? Issue #911 ? OpenSIPS/opensips github.com OpenSIPS 2.2 will run out of pkg_mem, i believe because of a leak in db/db_res.c Full memlog dump is available here: https://gist.github.com/etamme/7d42024ad684fe834b9fd514d2bd2412 I am using... Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 17.06.2016 17:02, Rodrigo Pimenta Carvalho wrote: Hi. People from my team is investigating a memory leak related to OpenSIPS 2.2. As I had commented in another discussion in the past, it seems that the problem comes from SQLite we are using as the Registrar for our OpenSIPS 2.2. For example, a script opensips.cfg that doesn't use SQLite didn't cause memory leak. But, a script that uses it and use another module that needs a database (EX: auth.so) causes memory leak. We are still in the beginning of the investigation. So, what is the best version of SQLite to be used with OpenSIPS 2.2? That is, what version of SQLite was very well tested with OpenSIPS 2.2 and worked without memory leaks or others issues? Any suggestion will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Jun 17 16:35:55 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 17 Jun 2016 17:35:55 +0300 Subject: [OpenSIPS-Users] OpenSIPS and FreeSWITCH Integration Training Message-ID: <57640ACB.3020101@opensips.org> Hello, As part of ClueCon, we will run an one-day hands-on Training around OpenSIPS and FreeSWITCH integration. The training follows the steps of building a complete system featuring: * OpenSIPS as a cluster front-end - http://opensips.org * FreeSWITCH PBX core system - http://freeswitch.org/ * HOMER for SIP capturing - http://sipcapture.org/ * CGRates as billing engine - http://cgrates.org/ * SIP Fraud detection The training will be held on 12th of August 2016, as part of the ClueCon event. Together with the training, we will run a new session of Design Clinics (http://www.opensips.org/Community/Clinics) Details, registration (both training and clinics) and more are available here: http://www.opensips.org/events/Training-2016ClueCon.html Best regards -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com From ionutionita at opensips.org Fri Jun 17 16:45:35 2016 From: ionutionita at opensips.org (Ionut Ionita) Date: Fri, 17 Jun 2016 17:45:35 +0300 Subject: [OpenSIPS-Users] What is the best SQLite version to be used with OpenSIPS 2.2? We investigate a memory leak. In-Reply-To: References: <576405B9.2070605@opensips.org> Message-ID: <57640D0F.5030409@opensips.org> Hi Rodrigo, Pushed a fix both into 2.2[0] and master[1] branches. If you still think sqlite leaks even with this fix, please feel free to open an issue on github. [0] https://github.com/OpenSIPS/opensips/commit/c1aa55e73f7c56ea4894997aeb7f25a03cb289cf [1] https://github.com/OpenSIPS/opensips/commit/c1aa55e73f7c56ea4894997aeb7f25a03cb289cf Regrads, Ionut Ionita OpenSIPS Developer On 06/17/2016 05:19 PM, Rodrigo Pimenta Carvalho wrote: > > Hi Liviu. > > > Very good. > > > We will see the resolution process. > > Thank you very much! > > Regards. > > > RODRIGO PIMENTA CARVALHO > Inatel Competence Center > Software > Ph: +55 35 3471 9200 RAMAL 979 > > > ------------------------------------------------------------------------ > *De:* users-bounces at lists.opensips.org > em nome de Liviu Chircu > > *Enviado:* sexta-feira, 17 de junho de 2016 11:14 > *Para:* users at lists.opensips.org > *Assunto:* Re: [OpenSIPS-Users] What is the best SQLite version to be > used with OpenSIPS 2.2? We investigate a memory leak. > Hi Rodrigo! > > A GitHub issue [1] regarding this leak was just reported today by > Eric, so you can track the resolution process over there! You can even > subscribe to that ticket if you have an account, in order to receive > emails. > > [1]: https://github.com/OpenSIPS/opensips/issues/911 > > > 2.2 runs out of pkg_mem because of db/db_res.c memory leak ? Issue > #911 ? OpenSIPS/opensips > github.com > OpenSIPS 2.2 will run out of pkg_mem, i believe because of a leak in > db/db_res.c Full memlog dump is available here: > https://gist.github.com/etamme/7d42024ad684fe834b9fd514d2bd2412 I am > using... > > > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > On 17.06.2016 17:02, Rodrigo Pimenta Carvalho wrote: >> >> Hi. >> >> >> People from my team is investigating a memory leak related to >> OpenSIPS 2.2. >> >> >> As I had commented in another discussion in the past, it seems that >> the problem comes from SQLite we are using as the Registrar for our >> OpenSIPS 2.2. >> >> For example, a script opensips.cfg that doesn't use SQLite didn't >> cause memory leak. But, a script that uses it and use another module >> that needs a database (EX: auth.so) causes memory leak. >> >> >> We are still in the beginning of the investigation. >> >> So, what is the best version of SQLite to be used with OpenSIPS 2.2? >> That is, what version of SQLite was very well tested with OpenSIPS >> 2.2 and worked without memory leaks or others issues? >> >> >> Any suggestion will be very helpful! >> >> >> Best regards. >> >> >> RODRIGO PIMENTA CARVALHO >> Inatel Competence Center >> Software >> Ph: +55 35 3471 9200 RAMAL 979 >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Fri Jun 17 17:22:41 2016 From: liviu at opensips.org (Liviu Chircu) Date: Fri, 17 Jun 2016 18:22:41 +0300 Subject: [OpenSIPS-Users] [Release] OpenSIPS 1.11.8 Message-ID: <576415C1.1020200@opensips.org> Hello all, Following several major fixes (related to TLS and natpinging) on the current OpenSIPS 1.11 version, we decided it's mandatory that we bump up the minor version, hence reaching 1.11.8. We wish to thank everyone who got involved both in testing and bug reporting! Downloads and changelogs are available at: http://opensips.org/pub/opensips/1.11.8/ Enjoy! -- Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com From pimenta at inatel.br Fri Jun 17 19:08:45 2016 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Fri, 17 Jun 2016 17:08:45 +0000 Subject: [OpenSIPS-Users] What is the best SQLite version to be used with OpenSIPS 2.2? We investigate a memory leak. In-Reply-To: <57640D0F.5030409@opensips.org> References: <576405B9.2070605@opensips.org> , <57640D0F.5030409@opensips.org> Message-ID: Thank you Ionut. We will try it so. Today morning, we noticed that OpenSIPS 2.2 while running and using SQLite, without online clients, without registers and without calls, causes a memory leak. That is, OpenSIPS even without any SIP request causes a memory leak due to the use of SQLite. After updating the SQLite to a new version, such memory leak was vanished. However, even with the newest SQLite, we still get memory leaks again if the proxy receives SIP REGISTER messages. That is, we get the issue every time some client registers. In this case we saw the memory leak in : " modparam("db_sqlite", "load_extension", "/usr/lib/sqlite3/pcre.so")" Let us try the new solution and see what happens. Best regards! RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________ De: users-bounces at lists.opensips.org em nome de Ionut Ionita Enviado: sexta-feira, 17 de junho de 2016 11:45 Para: OpenSIPS users mailling list Assunto: Re: [OpenSIPS-Users] What is the best SQLite version to be used with OpenSIPS 2.2? We investigate a memory leak. Hi Rodrigo, Pushed a fix both into 2.2[0] and master[1] branches. If you still think sqlite leaks even with this fix, please feel free to open an issue on github. [0] https://github.com/OpenSIPS/opensips/commit/c1aa55e73f7c56ea4894997aeb7f25a03cb289cf [https://avatars3.githubusercontent.com/u/7924437?v=3&s=200] [sqlite][bugfix] free column names when freeing the result ? OpenSIPS/opensips at c1aa55e github.com (cherry picked from commit cf380773cec9f91ad08e343c03261154dabc36a5) [1] https://github.com/OpenSIPS/opensips/commit/c1aa55e73f7c56ea4894997aeb7f25a03cb289cf Regrads, Ionut Ionita OpenSIPS Developer On 06/17/2016 05:19 PM, Rodrigo Pimenta Carvalho wrote: Hi Liviu. Very good. We will see the resolution process. Thank you very much! Regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________ De: users-bounces at lists.opensips.org em nome de Liviu Chircu Enviado: sexta-feira, 17 de junho de 2016 11:14 Para: users at lists.opensips.org Assunto: Re: [OpenSIPS-Users] What is the best SQLite version to be used with OpenSIPS 2.2? We investigate a memory leak. Hi Rodrigo! A GitHub issue [1] regarding this leak was just reported today by Eric, so you can track the resolution process over there! You can even subscribe to that ticket if you have an account, in order to receive emails. [1]: https://github.com/OpenSIPS/opensips/issues/911 [https://avatars3.githubusercontent.com/u/21685?v=3&s=400] 2.2 runs out of pkg_mem because of db/db_res.c memory leak ? Issue #911 ? OpenSIPS/opensips github.com OpenSIPS 2.2 will run out of pkg_mem, i believe because of a leak in db/db_res.c Full memlog dump is available here: https://gist.github.com/etamme/7d42024ad684fe834b9fd514d2bd2412 I am using... Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 17.06.2016 17:02, Rodrigo Pimenta Carvalho wrote: Hi. People from my team is investigating a memory leak related to OpenSIPS 2.2. As I had commented in another discussion in the past, it seems that the problem comes from SQLite we are using as the Registrar for our OpenSIPS 2.2. For example, a script opensips.cfg that doesn't use SQLite didn't cause memory leak. But, a script that uses it and use another module that needs a database (EX: auth.so) causes memory leak. We are still in the beginning of the investigation. So, what is the best version of SQLite to be used with OpenSIPS 2.2? That is, what version of SQLite was very well tested with OpenSIPS 2.2 and worked without memory leaks or others issues? Any suggestion will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From eric at uphreak.com Fri Jun 17 19:12:25 2016 From: eric at uphreak.com (Eric Tamme) Date: Fri, 17 Jun 2016 11:12:25 -0600 Subject: [OpenSIPS-Users] What is the best SQLite version to be used with OpenSIPS 2.2? We investigate a memory leak. In-Reply-To: References: <576405B9.2070605@opensips.org> <57640D0F.5030409@opensips.org> Message-ID: <57642F79.4070806@uphreak.com> Hey Rodrigo, Are you running https://github.com/etamme/federated-sip by chance? Your use of the PCRE module made me think you might be. I run federated-sip and I do use sqlite3 with opensips - my current sqlite version is: sqlite-3.7.17-4.el7.x86_64 I do not know that I have memory leaks outside of what I reported in the github issue. -Eric On 06/17/2016 11:08 AM, Rodrigo Pimenta Carvalho wrote: > > Thank you Ionut. > > > We will try it so. > > > Today morning, we noticed that OpenSIPS 2.2 while running and using > SQLite, without online clients, without registers and without calls, > causes a memory leak. That is, OpenSIPS even without any SIP request > causes a memory leak due to the use of SQLite. > > > After updating the SQLite to a new version, such memory leak was vanished. > > > However, even with the newest SQLite, we still get memory leaks again > if the proxy receives SIP REGISTER messages. That is, we get the issue > every time some client registers. In this case we saw the memory leak > in : " modparam("db_sqlite", "load_extension", > "/usr/lib/sqlite3/pcre.so")" > > > Let us try the new solution and see what happens. > > > Best regards! > > > RODRIGO PIMENTA CARVALHO > Inatel Competence Center > Software > Ph: +55 35 3471 9200 RAMAL 979 > > > ------------------------------------------------------------------------ > *De:* users-bounces at lists.opensips.org > em nome de Ionut Ionita > > *Enviado:* sexta-feira, 17 de junho de 2016 11:45 > *Para:* OpenSIPS users mailling list > *Assunto:* Re: [OpenSIPS-Users] What is the best SQLite version to be > used with OpenSIPS 2.2? We investigate a memory leak. > Hi Rodrigo, > > Pushed a fix both into 2.2[0] and master[1] branches. If you still > think sqlite leaks even with this fix, > please feel free to open an issue on github. > > [0] > https://github.com/OpenSIPS/opensips/commit/c1aa55e73f7c56ea4894997aeb7f25a03cb289cf > > > > [sqlite][bugfix] free column names when freeing the result ? > OpenSIPS/opensips at c1aa55e > > github.com > (cherry picked from commit cf380773cec9f91ad08e343c03261154dabc36a5) > > > [1] > https://github.com/OpenSIPS/opensips/commit/c1aa55e73f7c56ea4894997aeb7f25a03cb289cf > > Regrads, > Ionut Ionita > OpenSIPS Developer > On 06/17/2016 05:19 PM, Rodrigo Pimenta Carvalho wrote: >> >> Hi Liviu. >> >> >> Very good. >> >> >> We will see the resolution process. >> >> Thank you very much! >> >> Regards. >> >> >> RODRIGO PIMENTA CARVALHO >> Inatel Competence Center >> Software >> Ph: +55 35 3471 9200 RAMAL 979 >> >> >> ------------------------------------------------------------------------ >> *De:* users-bounces at lists.opensips.org >> em nome de Liviu Chircu >> >> *Enviado:* sexta-feira, 17 de junho de 2016 11:14 >> *Para:* users at lists.opensips.org >> *Assunto:* Re: [OpenSIPS-Users] What is the best SQLite version to be >> used with OpenSIPS 2.2? We investigate a memory leak. >> Hi Rodrigo! >> >> A GitHub issue [1] regarding this leak was just reported today by >> Eric, so you can track the resolution process over there! You can >> even subscribe to that ticket if you have an account, in order to >> receive emails. >> >> [1]: https://github.com/OpenSIPS/opensips/issues/911 >> >> >> 2.2 runs out of pkg_mem because of db/db_res.c memory leak ? Issue >> #911 ? OpenSIPS/opensips >> >> github.com >> OpenSIPS 2.2 will run out of pkg_mem, i believe because of a leak in >> db/db_res.c Full memlog dump is available here: >> https://gist.github.com/etamme/7d42024ad684fe834b9fd514d2bd2412 I am >> using... >> >> >> Liviu Chircu >> OpenSIPS Developer >> http://www.opensips-solutions.com >> On 17.06.2016 17:02, Rodrigo Pimenta Carvalho wrote: >>> >>> Hi. >>> >>> >>> People from my team is investigating a memory leak related to >>> OpenSIPS 2.2. >>> >>> >>> As I had commented in another discussion in the past, it seems that >>> the problem comes from SQLite we are using as the Registrar for our >>> OpenSIPS 2.2. >>> >>> For example, a script opensips.cfg that doesn't use SQLite didn't >>> cause memory leak. But, a script that uses it and use another module >>> that needs a database (EX: auth.so) causes memory leak. >>> >>> >>> We are still in the beginning of the investigation. >>> >>> So, what is the best version of SQLite to be used with OpenSIPS 2.2? >>> That is, what version of SQLite was very well tested with OpenSIPS >>> 2.2 and worked without memory leaks or others issues? >>> >>> >>> Any suggestion will be very helpful! >>> >>> >>> Best regards. >>> >>> >>> RODRIGO PIMENTA CARVALHO >>> Inatel Competence Center >>> Software >>> Ph: +55 35 3471 9200 RAMAL 979 >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From pimenta at inatel.br Fri Jun 17 19:23:43 2016 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Fri, 17 Jun 2016 17:23:43 +0000 Subject: [OpenSIPS-Users] What is the best SQLite version to be used with OpenSIPS 2.2? We investigate a memory leak. In-Reply-To: <57642F79.4070806@uphreak.com> References: <576405B9.2070605@opensips.org> <57640D0F.5030409@opensips.org> , <57642F79.4070806@uphreak.com> Message-ID: Hi Eric. Probably not. Because I still don't know what is a federated-sip. And I didn't have to take control of RTPs in opensips script. However, a coworker in my office will check these details and help us to conclude more things about it. Is there a quick way to check if someone is using such federated-sip? Our version 3.8.6 of SQLite presented the memory leak (when there was no SIP requests), but the version 3.13 doesn't present. P.S.: I still have to read about federated SIP and see what are its advantages. Thanks a lot. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________ De: users-bounces at lists.opensips.org em nome de Eric Tamme Enviado: sexta-feira, 17 de junho de 2016 14:12 Para: OpenSIPS users mailling list Assunto: Re: [OpenSIPS-Users] What is the best SQLite version to be used with OpenSIPS 2.2? We investigate a memory leak. Hey Rodrigo, Are you running https://github.com/etamme/federated-sip by chance? Your use of the PCRE module made me think you might be. I run federated-sip and I do use sqlite3 with opensips - my current sqlite version is: sqlite-3.7.17-4.el7.x86_64 [https://avatars3.githubusercontent.com/u/21685?v=3&s=400] GitHub - etamme/federated-sip: Federated SIP deployment github.com README.md Federated SIP server. The Federated SIP project is a set of scripts designed to run OpenSIPS + rtpengine in a way that will provide federated, open ... I do not know that I have memory leaks outside of what I reported in the github issue. -Eric On 06/17/2016 11:08 AM, Rodrigo Pimenta Carvalho wrote: Thank you Ionut. We will try it so. Today morning, we noticed that OpenSIPS 2.2 while running and using SQLite, without online clients, without registers and without calls, causes a memory leak. That is, OpenSIPS even without any SIP request causes a memory leak due to the use of SQLite. After updating the SQLite to a new version, such memory leak was vanished. However, even with the newest SQLite, we still get memory leaks again if the proxy receives SIP REGISTER messages. That is, we get the issue every time some client registers. In this case we saw the memory leak in : " modparam("db_sqlite", "load_extension", "/usr/lib/sqlite3/pcre.so")" Let us try the new solution and see what happens. Best regards! RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________ De: users-bounces at lists.opensips.org em nome de Ionut Ionita Enviado: sexta-feira, 17 de junho de 2016 11:45 Para: OpenSIPS users mailling list Assunto: Re: [OpenSIPS-Users] What is the best SQLite version to be used with OpenSIPS 2.2? We investigate a memory leak. Hi Rodrigo, Pushed a fix both into 2.2[0] and master[1] branches. If you still think sqlite leaks even with this fix, please feel free to open an issue on github. [0] https://github.com/OpenSIPS/opensips/commit/c1aa55e73f7c56ea4894997aeb7f25a03cb289cf [https://avatars3.githubusercontent.com/u/7924437?v=3&s=200] [sqlite][bugfix] free column names when freeing the result ? OpenSIPS/opensips at c1aa55e github.com (cherry picked from commit cf380773cec9f91ad08e343c03261154dabc36a5) [1] https://github.com/OpenSIPS/opensips/commit/c1aa55e73f7c56ea4894997aeb7f25a03cb289cf Regrads, Ionut Ionita OpenSIPS Developer On 06/17/2016 05:19 PM, Rodrigo Pimenta Carvalho wrote: Hi Liviu. Very good. We will see the resolution process. Thank you very much! Regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________ De: users-bounces at lists.opensips.org em nome de Liviu Chircu Enviado: sexta-feira, 17 de junho de 2016 11:14 Para: users at lists.opensips.org Assunto: Re: [OpenSIPS-Users] What is the best SQLite version to be used with OpenSIPS 2.2? We investigate a memory leak. Hi Rodrigo! A GitHub issue [1] regarding this leak was just reported today by Eric, so you can track the resolution process over there! You can even subscribe to that ticket if you have an account, in order to receive emails. [1]: https://github.com/OpenSIPS/opensips/issues/911 [https://avatars3.githubusercontent.com/u/21685?v=3&s=400] 2.2 runs out of pkg_mem because of db/db_res.c memory leak ? Issue #911 ? OpenSIPS/opensips github.com OpenSIPS 2.2 will run out of pkg_mem, i believe because of a leak in db/db_res.c Full memlog dump is available here: https://gist.github.com/etamme/7d42024ad684fe834b9fd514d2bd2412 I am using... Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 17.06.2016 17:02, Rodrigo Pimenta Carvalho wrote: Hi. People from my team is investigating a memory leak related to OpenSIPS 2.2. As I had commented in another discussion in the past, it seems that the problem comes from SQLite we are using as the Registrar for our OpenSIPS 2.2. For example, a script opensips.cfg that doesn't use SQLite didn't cause memory leak. But, a script that uses it and use another module that needs a database (EX: auth.so) causes memory leak. We are still in the beginning of the investigation. So, what is the best version of SQLite to be used with OpenSIPS 2.2? That is, what version of SQLite was very well tested with OpenSIPS 2.2 and worked without memory leaks or others issues? Any suggestion will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From pimenta at inatel.br Fri Jun 17 19:38:45 2016 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Fri, 17 Jun 2016 17:38:45 +0000 Subject: [OpenSIPS-Users] What is the best SQLite version to be used with OpenSIPS 2.2? We investigate a memory leak. In-Reply-To: References: <576405B9.2070605@opensips.org> <57640D0F.5030409@opensips.org> , <57642F79.4070806@uphreak.com>, Message-ID: Hi. We discovered another memory leak in OpenSIPS 2.2, even using newest SQLite. Now the issue doesn't relate to the data base. There is a issue related with a parser. In few minutes I will post here more details, with valgrind log. Regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________ De: users-bounces at lists.opensips.org em nome de Rodrigo Pimenta Carvalho Enviado: sexta-feira, 17 de junho de 2016 14:23 Para: OpenSIPS users mailling list Assunto: Re: [OpenSIPS-Users] What is the best SQLite version to be used with OpenSIPS 2.2? We investigate a memory leak. Hi Eric. Probably not. Because I still don't know what is a federated-sip. And I didn't have to take control of RTPs in opensips script. However, a coworker in my office will check these details and help us to conclude more things about it. Is there a quick way to check if someone is using such federated-sip? Our version 3.8.6 of SQLite presented the memory leak (when there was no SIP requests), but the version 3.13 doesn't present. P.S.: I still have to read about federated SIP and see what are its advantages. Thanks a lot. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________ De: users-bounces at lists.opensips.org em nome de Eric Tamme Enviado: sexta-feira, 17 de junho de 2016 14:12 Para: OpenSIPS users mailling list Assunto: Re: [OpenSIPS-Users] What is the best SQLite version to be used with OpenSIPS 2.2? We investigate a memory leak. Hey Rodrigo, Are you running https://github.com/etamme/federated-sip by chance? Your use of the PCRE module made me think you might be. I run federated-sip and I do use sqlite3 with opensips - my current sqlite version is: sqlite-3.7.17-4.el7.x86_64 [https://avatars3.githubusercontent.com/u/21685?v=3&s=400] GitHub - etamme/federated-sip: Federated SIP deployment github.com README.md Federated SIP server. The Federated SIP project is a set of scripts designed to run OpenSIPS + rtpengine in a way that will provide federated, open ... I do not know that I have memory leaks outside of what I reported in the github issue. -Eric On 06/17/2016 11:08 AM, Rodrigo Pimenta Carvalho wrote: Thank you Ionut. We will try it so. Today morning, we noticed that OpenSIPS 2.2 while running and using SQLite, without online clients, without registers and without calls, causes a memory leak. That is, OpenSIPS even without any SIP request causes a memory leak due to the use of SQLite. After updating the SQLite to a new version, such memory leak was vanished. However, even with the newest SQLite, we still get memory leaks again if the proxy receives SIP REGISTER messages. That is, we get the issue every time some client registers. In this case we saw the memory leak in : " modparam("db_sqlite", "load_extension", "/usr/lib/sqlite3/pcre.so")" Let us try the new solution and see what happens. Best regards! RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________ De: users-bounces at lists.opensips.org em nome de Ionut Ionita Enviado: sexta-feira, 17 de junho de 2016 11:45 Para: OpenSIPS users mailling list Assunto: Re: [OpenSIPS-Users] What is the best SQLite version to be used with OpenSIPS 2.2? We investigate a memory leak. Hi Rodrigo, Pushed a fix both into 2.2[0] and master[1] branches. If you still think sqlite leaks even with this fix, please feel free to open an issue on github. [0] https://github.com/OpenSIPS/opensips/commit/c1aa55e73f7c56ea4894997aeb7f25a03cb289cf [https://avatars3.githubusercontent.com/u/7924437?v=3&s=200] [sqlite][bugfix] free column names when freeing the result ? OpenSIPS/opensips at c1aa55e github.com (cherry picked from commit cf380773cec9f91ad08e343c03261154dabc36a5) [1] https://github.com/OpenSIPS/opensips/commit/c1aa55e73f7c56ea4894997aeb7f25a03cb289cf Regrads, Ionut Ionita OpenSIPS Developer On 06/17/2016 05:19 PM, Rodrigo Pimenta Carvalho wrote: Hi Liviu. Very good. We will see the resolution process. Thank you very much! Regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________ De: users-bounces at lists.opensips.org em nome de Liviu Chircu Enviado: sexta-feira, 17 de junho de 2016 11:14 Para: users at lists.opensips.org Assunto: Re: [OpenSIPS-Users] What is the best SQLite version to be used with OpenSIPS 2.2? We investigate a memory leak. Hi Rodrigo! A GitHub issue [1] regarding this leak was just reported today by Eric, so you can track the resolution process over there! You can even subscribe to that ticket if you have an account, in order to receive emails. [1]: https://github.com/OpenSIPS/opensips/issues/911 [https://avatars3.githubusercontent.com/u/21685?v=3&s=400] 2.2 runs out of pkg_mem because of db/db_res.c memory leak ? Issue #911 ? OpenSIPS/opensips github.com OpenSIPS 2.2 will run out of pkg_mem, i believe because of a leak in db/db_res.c Full memlog dump is available here: https://gist.github.com/etamme/7d42024ad684fe834b9fd514d2bd2412 I am using... Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 17.06.2016 17:02, Rodrigo Pimenta Carvalho wrote: Hi. People from my team is investigating a memory leak related to OpenSIPS 2.2. As I had commented in another discussion in the past, it seems that the problem comes from SQLite we are using as the Registrar for our OpenSIPS 2.2. For example, a script opensips.cfg that doesn't use SQLite didn't cause memory leak. But, a script that uses it and use another module that needs a database (EX: auth.so) causes memory leak. We are still in the beginning of the investigation. So, what is the best version of SQLite to be used with OpenSIPS 2.2? That is, what version of SQLite was very well tested with OpenSIPS 2.2 and worked without memory leaks or others issues? Any suggestion will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From pimenta at inatel.br Fri Jun 17 19:57:09 2016 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Fri, 17 Jun 2016 17:57:09 +0000 Subject: [OpenSIPS-Users] Parser memory Leak found. In-Reply-To: References: Message-ID: Hi. Thanks Daniel F?ssia, a coworker in my office, now I'm sending more details about the memory leak we saw in OpenSIPS 2.2 (newest commit from today): The following configuration doesn't causes memory leaks: modparam("dialog", "db_mode", 1) modparam("dialog", "db_url", "sqlite:///usr/local/opensips/db/sisc.sqlite") But, when db_mode is 0, it causes memory leak. The problem is that when using db_mode = 0 we can't declare db_url. That is, if db_mode is zero, we have to comment the line that declares db_url. See below the valgrind logs. One for the case without memory leak and another one with the issue. ----------------------------------------------------------------------------------------------------------------------------- ==1792== ==1792== HEAP SUMMARY: ==1792== in use at exit: 3,142,778 bytes in 2,894 blocks ==1792== total heap usage: 9,463 allocs, 6,569 frees, 4,960,116 bytes allocated ==1792== ==1792== LEAK SUMMARY: ==1792== definitely lost: 0 bytes in 0 blocks ==1792== indirectly lost: 0 bytes in 0 blocks ==1792== possibly lost: 0 bytes in 0 blocks ==1792== still reachable: 3,142,778 bytes in 2,894 blocks ==1792== suppressed: 0 bytes in 0 blocks ==1792== Reachable blocks (those to which a pointer was found) are not shown. ==1792== To see them, rerun with: --leak-check=full --show-leak-kinds=all ==1792== ------------------------------------------------------------------------------------------------------------------------------------ Now with the isse: --------------------------------------------------------------------------------------------------------------------------------------- Thank you for flying opensips ==1762== ==1762== HEAP SUMMARY: ==1762== in use at exit: 2,887,898 bytes in 2,193 blocks ==1762== total heap usage: 7,991 allocs, 5,798 frees, 4,382,036 bytes allocated ==1762== ==1762== 80 bytes in 1 blocks are definitely lost in loss record 31 of 100 ==1762== at 0x4C2745D: malloc (in /usr/lib64/valgrind/vgpreload_memcheck-amd64-linux.so) ==1762== by 0x52D50B9: strdup (strdup.c:42) ==1762== by 0x4DF87E: set_mod_param_regex (modparam.c:97) ==1762== by 0x5AD2FB: yyparse (cfg.y:1085) ==1762== by 0x4177DE: main (main.c:999) ==1762== ==1762== LEAK SUMMARY: ==1762== definitely lost: 80 bytes in 1 blocks ==1762== indirectly lost: 0 bytes in 0 blocks ==1762== possibly lost: 0 bytes in 0 blocks ==1762== still reachable: 2,887,818 bytes in 2,192 blocks ==1762== suppressed: 0 bytes in 0 blocks ==1762== Reachable blocks (those to which a pointer was found) are not shown. ==1762== To see them, rerun with: --leak-check=full --show-leak-kinds=all ==1762== ------------------------------------------------------------------------------------------------------------------------------------------------- The problem rises in the yyparser. The parser causes a memory leak whenever db_mode is zero and we still declare db_url, just in dialog module. Regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________ De: Daniel Lopes F?ssia Enviado: sexta-feira, 17 de junho de 2016 14:41 Para: Rodrigo Pimenta Carvalho Assunto: Leak no Parser Pimenta, Os logs e as configura??es est?o em anexo. Qualquer d?vida me d? um tok. Att, Daniel Fussia -------------- next part -------------- An HTML attachment was scrubbed... URL: From pimenta at inatel.br Fri Jun 17 20:09:35 2016 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Fri, 17 Jun 2016 18:09:35 +0000 Subject: [OpenSIPS-Users] Parser memory Leak found. In-Reply-To: References: , Message-ID: Hi. In addiction to my last message, the same problem also exists to the following configuration: modparam("uri", "use_uri_table", 0) modparam("uri", "db_url", "sqlite:///usr/local/opensips/db/sisc.sqlite") # CUSTOMIZE ME If use_uri_table is equal to zero, we must comment the line that declare db_url. But, what kind of side effect could I get with such decision? Regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________ De: sisc-request at listas.inatel.br em nome de Rodrigo Pimenta Carvalho Enviado: sexta-feira, 17 de junho de 2016 14:57 Para: users at lists.opensips.org Assunto: [sisc] Parser memory Leak found. Hi. Thanks Daniel F?ssia, a coworker in my office, now I'm sending more details about the memory leak we saw in OpenSIPS 2.2 (newest commit from today): The following configuration doesn't causes memory leaks: modparam("dialog", "db_mode", 1) modparam("dialog", "db_url", "sqlite:///usr/local/opensips/db/sisc.sqlite") But, when db_mode is 0, it causes memory leak. The problem is that when using db_mode = 0 we can't declare db_url. That is, if db_mode is zero, we have to comment the line that declares db_url. See below the valgrind logs. One for the case without memory leak and another one with the issue. ----------------------------------------------------------------------------------------------------------------------------- ==1792== ==1792== HEAP SUMMARY: ==1792== in use at exit: 3,142,778 bytes in 2,894 blocks ==1792== total heap usage: 9,463 allocs, 6,569 frees, 4,960,116 bytes allocated ==1792== ==1792== LEAK SUMMARY: ==1792== definitely lost: 0 bytes in 0 blocks ==1792== indirectly lost: 0 bytes in 0 blocks ==1792== possibly lost: 0 bytes in 0 blocks ==1792== still reachable: 3,142,778 bytes in 2,894 blocks ==1792== suppressed: 0 bytes in 0 blocks ==1792== Reachable blocks (those to which a pointer was found) are not shown. ==1792== To see them, rerun with: --leak-check=full --show-leak-kinds=all ==1792== ------------------------------------------------------------------------------------------------------------------------------------ Now with the isse: --------------------------------------------------------------------------------------------------------------------------------------- Thank you for flying opensips ==1762== ==1762== HEAP SUMMARY: ==1762== in use at exit: 2,887,898 bytes in 2,193 blocks ==1762== total heap usage: 7,991 allocs, 5,798 frees, 4,382,036 bytes allocated ==1762== ==1762== 80 bytes in 1 blocks are definitely lost in loss record 31 of 100 ==1762== at 0x4C2745D: malloc (in /usr/lib64/valgrind/vgpreload_memcheck-amd64-linux.so) ==1762== by 0x52D50B9: strdup (strdup.c:42) ==1762== by 0x4DF87E: set_mod_param_regex (modparam.c:97) ==1762== by 0x5AD2FB: yyparse (cfg.y:1085) ==1762== by 0x4177DE: main (main.c:999) ==1762== ==1762== LEAK SUMMARY: ==1762== definitely lost: 80 bytes in 1 blocks ==1762== indirectly lost: 0 bytes in 0 blocks ==1762== possibly lost: 0 bytes in 0 blocks ==1762== still reachable: 2,887,818 bytes in 2,192 blocks ==1762== suppressed: 0 bytes in 0 blocks ==1762== Reachable blocks (those to which a pointer was found) are not shown. ==1762== To see them, rerun with: --leak-check=full --show-leak-kinds=all ==1762== ------------------------------------------------------------------------------------------------------------------------------------------------- The problem rises in the yyparser. The parser causes a memory leak whenever db_mode is zero and we still declare db_url, just in dialog module. Regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________ De: Daniel Lopes F?ssia Enviado: sexta-feira, 17 de junho de 2016 14:41 Para: Rodrigo Pimenta Carvalho Assunto: Leak no Parser Pimenta, Os logs e as configura??es est?o em anexo. Qualquer d?vida me d? um tok. Att, Daniel Fussia -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at inin.com Fri Jun 17 23:19:48 2016 From: Ben.Newlin at inin.com (Newlin, Ben) Date: Fri, 17 Jun 2016 21:19:48 +0000 Subject: [OpenSIPS-Users] TM flag 0x01 not working in 1.11.7 Message-ID: In my script I send all ?100 Trying? responses manually* and use the 0x01 flag when calling t_relay so that it will not send its own ?100 Giving a try? response, as per the documentation [1]. Unfortunately, in 1.11.7 and 1.11.8 this flag seems to have stopped working which results in multiple 100 responses. This should be easily reproducible with a script that simply calls t_relay with flag 0x01. You will see that a ?100 Giving a try? response is still sent. [1] http://www.opensips.org/html/docs/modules/1.11.x/tm.html#id294528 * I do this because I don?t like that the default is to send the non-standard response ?100 Giving a try? instead of ?100 Trying?. I?ve always wondered why this is that way. Additionally, the automatic response to CANCEL requests is ?200 canceling? instead of the standard ?200 OK?. Unfortunately, I have yet to find a way to workaround the behavior for CANCEL. Ben Newlin -------------- next part -------------- An HTML attachment was scrubbed... URL: From denis7979 at mail.ru Sat Jun 18 15:49:08 2016 From: denis7979 at mail.ru (Denis) Date: Sat, 18 Jun 2016 16:49:08 +0300 Subject: [OpenSIPS-Users] TM flag 0x01 not working in 1.11.7 In-Reply-To: References: Message-ID: <273971259.20160618164908@ptl.ru> Hello! 2.1 has the same problem mailto:denis7979 at mail.ru In my script I send all ?100 Trying? responses manually* and use the 0x01 flag when calling t_relay so that it will not send its own ?100 Giving a try? response, as per the documentation [1]. Unfortunately, in 1.11.7 and 1.11.8 this flag seems to have stopped working which results in multiple 100 responses. This should be easily reproducible with a script that simply calls t_relay with flag 0x01. You will see that a ?100 Giving a try? response is still sent. [1] http://www.opensips.org/html/docs/modules/1.11.x/tm.html#id294528 * I do this because I don?t like that the default is to send the non-standard response ?100 Giving a try? instead of ?100 Trying?. I?ve always wondered why this is that way. Additionally, the automatic response to CANCEL requests is ?200 canceling? instead of the standard ?200 OK?. Unfortunately, I have yet to find a way to workaround the behavior for CANCEL. Ben Newlin -------------- next part -------------- An HTML attachment was scrubbed... URL: From govoiper at gmail.com Sun Jun 19 19:56:24 2016 From: govoiper at gmail.com (SamyGo) Date: Sun, 19 Jun 2016 13:56:24 -0400 Subject: [OpenSIPS-Users] Async event_route and cachdb_redis Message-ID: Hi, I'm seeing errors from cachedb_redis module when called in an event route in async mode. event_route[E_UL_CONTACT_INSERT,async] { ... cache_raw_query("redis:group1","SET ABC"); .. } OpenSIPS throws error stating that redis group1 unavailable DBG:core:cachedb_raw_query: from script [redis] - with grp [group1] ERROR:core:cachedb_raw_query: failed to get connection for grp name [group1] I tried same command in main route of reply route, all works normal. if I remove the "async" from the event_route definition it works in event route. Any logical reason why async route don't recognize the connections ? Tried with OpenSIPS 2.2 and 2.1 as well, same behavior. Regards, Sammy -------------- next part -------------- An HTML attachment was scrubbed... URL: From Agalya_Ramachandran at comcast.com Mon Jun 20 19:35:55 2016 From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor)) Date: Mon, 20 Jun 2016 17:35:55 +0000 Subject: [OpenSIPS-Users] DNS-SRV query in opensips Message-ID: <955ec37716d94d139c82be2835377bd5@COPDCEX28.cable.comcast.com> Hi team, We are using opensips for our project requirements. I have a scenario where we need DNS-SRV query and the result of this should be placed as the desturi to send request out.(In case of forking call) As far as I went through opensips documentation, there are some core parameters for dns related config such as "dns_retr_time" ,"dns_retr_time" ,"dns_servers_no" etc... According to my understanding these config variables can be declared and used in the opensips.config file to control the settings of DNS query. Is there any available function where I can use and pass the DNS server domain name, so that it fetches the IP address of the host ? Please let us know what is the best way to achieve this? Regards, Agalya -------------- next part -------------- An HTML attachment was scrubbed... URL: From Agalya_Ramachandran at comcast.com Mon Jun 20 19:45:21 2016 From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor)) Date: Mon, 20 Jun 2016 17:45:21 +0000 Subject: [OpenSIPS-Users] CURL library with respect to REST_API calls Message-ID: Hi team, I have a question regarding curl library behavior with respect to curl_easy_perform API call. Here is the snippet of the code that am using in "rest_put" API call in rest_methods.c file w_curl_easy_setopt(handle, CURLOPT_WRITEFUNCTION, write_func); w_curl_easy_setopt(handle, CURLOPT_WRITEDATA, &res_body); When curl_easy_perform API call is success, I could able to retrieve the result body from the res_body. But in the case of API call failure am not getting any details of the message. But getting only the http response code. Is there a way to get the message details as well in the case where curl_easy_perform API fails? LM_INFO(" Actual result body is %s\n", res_body.s); When I print this, in the case of success, am getting a http response message in detail. But in case of failure, the call back function write_func is not at all called. Why it is so? Please guide me if there is a way to the message details in case of failure too. Regards, Agalya -------------- next part -------------- An HTML attachment was scrubbed... URL: From pimenta at inatel.br Mon Jun 20 20:02:44 2016 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Mon, 20 Jun 2016 18:02:44 +0000 Subject: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? Message-ID: Dear OpenSIPS-users, The table location has the column attr where I use to store specific additional information for each registration. Whenever A calls B, I have to read this specific information from the A record and from the B record. That is, I need to get and handle specific information about the caller and callee. For the callee, I use to invoke the lookup("location") function that put the needed information in the attr_avp. That is good and works very well. Then, I just have to read the attr_avp to get such specific information. For the caller, I use to invoke: $var(aorChamador) = $(ct.fields(uri)); lookup("location","","$var(aorChamador)"); However it causes amazing side effect in the SIP signaling. Ex: When A calls B, B stays quiet and A rings. So A can answer A. Crazy! According to the documentation, lookup will overwritten the Request-URI. I guess that is why the SIP signaling become incoherent. How could I get the caller attr specific information without side effects? Any hint will be very helpful!! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 -------------- next part -------------- An HTML attachment was scrubbed... URL: From millennium.bug at gmail.com Tue Jun 21 09:05:20 2016 From: millennium.bug at gmail.com (Owais Ahmad) Date: Tue, 21 Jun 2016 12:05:20 +0500 Subject: [OpenSIPS-Users] Some calls fail with tcp_connect_blocking error on version 2.1.1 Message-ID: I am getting the following error on opensips version 2.1.1. The scenario is that I have a load balancer listening on TLS and it dispatches requests to backend registrar servers listening on UDP. 90% of the calls are successful, but some calls fail and the load balancer throws the following error when INVITE is being relayed to B party. DBG:core:parse_headers: flags=2000 DBG:core:check_ip_address: params 10.0.0.7, 10.0.0.7, 0 DBG:core:tcp_conn_get: 0 port 56415 DBG:core:print_ip: tcpconn_find: ip 2.4.6.8 DBG:proto_tls:proto_tls_send: no open tcp connection found, opening new one DBG:core:probe_max_sock_buff: getsockopt: snd is initially 16384 DBG:core:probe_max_sock_buff: trying : 32768 DBG:core:probe_max_sock_buff: setting snd: set=32768,verify=65536 DBG:core:probe_max_sock_buff: trying : 65536 DBG:core:probe_max_sock_buff: setting snd: set=65536,verify=131072 INFO:core:probe_max_sock_buff: using snd buffer of 128 kb INFO:core:init_sock_keepalive: TCP keepalive enabled on socket 6 ERROR:core:tcp_connect_blocking: timeout 2132 ms elapsed from 3000 s ERROR:proto_tls:tls_sync_connect: tcp_blocking_connect failed ERROR:proto_tls:proto_tls_send: connect failed ERROR:tm:msg_send: send() for proto 3 failed ERROR:tm:t_forward_nonack: sending request failed DBG:tm:t_relay_to: t_forward_nonack returned error DBG:core:parse_headers: flags=ffffffffffffffff DBG:core:check_ip_address: params 10.0.0.7, 10.0.0.7, 0 DBG:tm:reset_timer: (group 3, tl=0x7fcb6a490970) DBG:tm:reset_timer: (group 0, tl=0x7fcb6a4909a0) DBG:tm:cleanup_uac_timers: RETR/FR timers reset DBG:tm:set_timer: relative timeout is 500000 DBG:tm:insert_timer_unsafe: [4]: 0x7fcb6a490898 (609300000) DBG:tm:insert_timer_unsafe: [0]: 0x7fcb6a4908c8 (618) Any hints as to what might be going wrong? Regards, Owais -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Tue Jun 21 09:24:00 2016 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Tue, 21 Jun 2016 10:24:00 +0300 Subject: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? In-Reply-To: References: Message-ID: <81fd240b-6369-15b4-a2ee-99deed334299@opensips.org> Hi, Rodrigo! Have you tried restoring the R-URI after the caller lookup? Something like: $var(ru) = $ru; lookup("location", "", "$fu"); # this takes the caller from FROM uri, which I think is more suitable than from contact uri $ru = $var(ru); # continue your processing here .... # now do the real lookup for the callee lookup("location"); Don't do the lookups in the reversed way, because you might loose some contacts. Best regards, R?zvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 06/20/2016 09:02 PM, Rodrigo Pimenta Carvalho wrote: > > Dear OpenSIPS-users, > > > The table location has the column attr where I use to store specific > additional information for each registration. > > Whenever A calls B, I have to read this specific information from the > A record and from the B record. That is, I need to get and handle > specific information about the caller and callee. > > > For the callee, I use to invoke the lookup("location") function that > put the needed information in the attr_avp. That is good and works > very well. Then, I just have to read the attr_avp to get such specific > information. > > > For the caller, I use to invoke: > > > $var(aorChamador) = $(ct.fields(uri)); > > lookup("location","","$var(aorChamador)"); > > > However it causes amazing side effect in the SIP signaling. Ex: When A > calls B, B stays quiet and A rings. So A can answer A. Crazy! > > According to the documentation, lookup will overwritten the > Request-URI. I guess that is why the SIP signaling become incoherent. > > > How could I get the caller attr specific information without side effects? > > > Any hint will be very helpful!! > > > Best regards. > > > RODRIGO PIMENTA CARVALHO > Inatel Competence Center > Software > Ph: +55 35 3471 9200 RAMAL 979 > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jun 21 09:35:39 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 21 Jun 2016 10:35:39 +0300 Subject: [OpenSIPS-Users] TM flag 0x01 not working in 1.11.7 In-Reply-To: <273971259.20160618164908@ptl.ru> References: <273971259.20160618164908@ptl.ru> Message-ID: <5768EE4B.2040107@opensips.org> Hi Please refer to https://github.com/OpenSIPS/opensips/issues/833 . As per documentation, the 0x1 flag became obsolete, as sending the 100Trying is no longer linked to the t_relay() - the 100Trying is now sent when the transaction is created. If you want to disable the auto 100Trying, see the new TM flag auto_100trying: http://www.opensips.org/html/docs/modules/2.2.x/tm.html#id294523 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 18.06.2016 16:49, Denis wrote: > Re: [OpenSIPS-Users] TM flag 0x01 not working in 1.11.7 Hello! > > 2.1 has the same problem > > //mailto:denis7979 at mail.ru > > > In my script I send all ?100 Trying? responses manually* and use the > 0x01 flag when calling t_relay so that it will not send its own ?100 > Giving a try? response, as per the documentation [1]. > > Unfortunately, in 1.11.7 and 1.11.8 this flag seems to have stopped > working which results in multiple 100 responses. > > This should be easily reproducible with a script that simply calls > t_relay with flag 0x01. You will see that a ?100 Giving a try? > response is still sent. > > [1] http://www.opensips.org/html/docs/modules/1.11.x/tm.html#id294528 > > * I do this because I don?t like that the default is to send the > non-standard response ?100 Giving a try? instead of ?100 Trying?. I?ve > always wondered why this is that way. Additionally, the automatic > response to CANCEL requests is ?200 canceling? instead of the standard > ?200 OK?. Unfortunately, I have yet to find a way to workaround the > behavior for CANCEL. > > Ben Newlin > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Tue Jun 21 09:40:33 2016 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Tue, 21 Jun 2016 10:40:33 +0300 Subject: [OpenSIPS-Users] Async event_route and cachdb_redis In-Reply-To: References: Message-ID: <87e0609d-975d-e80f-50c7-d34c7989f15d@opensips.org> Hi, Sammy! Could you try this patch: https://gist.github.com/razvancrainea/9d239c82474bb0f1c403b6459dbdb647 Thanks, R?zvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 06/19/2016 08:56 PM, SamyGo wrote: > Hi, > I'm seeing errors from cachedb_redis module when called in an event > route in async mode. > > event_route[E_UL_CONTACT_INSERT,async] { > ... > cache_raw_query("redis:group1","SET ABC"); > .. > > } > > OpenSIPS throws error stating that redis group1 unavailable > > DBG:core:cachedb_raw_query: from script [redis] - with grp [group1] > ERROR:core:cachedb_raw_query: failed to get connection for grp name > [group1] > > I tried same command in main route of reply route, all works normal. > if I remove the "async" from the event_route definition it works in > event route. > > Any logical reason why async route don't recognize the connections ? > > Tried with OpenSIPS 2.2 and 2.1 as well, same behavior. > > > Regards, > Sammy > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jun 21 09:41:04 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 21 Jun 2016 10:41:04 +0300 Subject: [OpenSIPS-Users] DNS-SRV query in opensips In-Reply-To: <955ec37716d94d139c82be2835377bd5@COPDCEX28.cable.comcast.com> References: <955ec37716d94d139c82be2835377bd5@COPDCEX28.cable.comcast.com> Message-ID: <5768EF90.2030802@opensips.org> Hi Agalya, OpenSIPS does full flavor DNS lookup (with NATPR and SRV), but this is internal, and not accessible from script. OpenSIPS implements auto DNS-based failover : http://www.opensips.org/html/docs/modules/2.2.x/tm.html#id293694 My understanding is you want the DNS resolving to be done at script level and to have access to the results ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 20.06.2016 20:35, Ramachandran, Agalya (Contractor) wrote: > > Hi team, > > We are using opensips for our project requirements. > > I have a scenario where we need DNS-SRV query and the result of this > should be placed as the desturi to send request out.(In case of > forking call) > > As far as I went through opensips documentation, there are some core > parameters for dns related config such as ?dns_retr_time > ? > ,?dns_retr_time > ? ,?dns_servers_no > ? > etc? > > According to my understanding these config variables can be declared > and used in the opensips.config file to control the settings of DNS query. > > Is there any available function where I can use and pass the DNS > server domain name, so that it fetches the IP address of the host ? > > Please let us know what is the best way to achieve this? > > Regards, > > Agalya > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Tue Jun 21 10:16:39 2016 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Tue, 21 Jun 2016 11:16:39 +0300 Subject: [OpenSIPS-Users] Some calls fail with tcp_connect_blocking error on version 2.1.1 In-Reply-To: References: Message-ID: <07117de8-bef2-83d0-39de-e3bb849322e0@opensips.org> Hi, Owais! You should consider upgrade OpenSIPS to a newer version (at least 2.1.3), because there were some fixes done related to this issue. If you cannot upgrade right now, try to disable the auto TCP aliasing[1] (set it to 0). [1] http://www.opensips.org/Documentation/Script-CoreParameters-2-1#toc90 Best regards, R?zvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 06/21/2016 10:05 AM, Owais Ahmad wrote: > I am getting the following error on opensips version 2.1.1. The > scenario is that I have a load balancer listening on TLS and it > dispatches requests to backend registrar servers listening on UDP. > 90% of the calls are successful, but some calls fail and the load > balancer throws the following error when INVITE is being relayed to B > party. > > DBG:core:parse_headers: flags=2000 > DBG:core:check_ip_address: params 10.0.0.7, 10.0.0.7, 0 > DBG:core:tcp_conn_get: 0 port 56415 > DBG:core:print_ip: tcpconn_find: ip 2.4.6.8 > DBG:proto_tls:proto_tls_send: no open tcp connection found, opening > new one > DBG:core:probe_max_sock_buff: getsockopt: snd is initially 16384 > DBG:core:probe_max_sock_buff: trying : 32768 > DBG:core:probe_max_sock_buff: setting snd: set=32768,verify=65536 > DBG:core:probe_max_sock_buff: trying : 65536 > DBG:core:probe_max_sock_buff: setting snd: set=65536,verify=131072 > INFO:core:probe_max_sock_buff: using snd buffer of 128 kb > INFO:core:init_sock_keepalive: TCP keepalive enabled on socket 6 > ERROR:core:tcp_connect_blocking: timeout 2132 ms elapsed from 3000 s > ERROR:proto_tls:tls_sync_connect: tcp_blocking_connect failed > ERROR:proto_tls:proto_tls_send: connect failed > ERROR:tm:msg_send: send() for proto 3 failed > ERROR:tm:t_forward_nonack: sending request failed > DBG:tm:t_relay_to: t_forward_nonack returned error > DBG:core:parse_headers: flags=ffffffffffffffff > DBG:core:check_ip_address: params 10.0.0.7, 10.0.0.7, 0 > DBG:tm:reset_timer: (group 3, tl=0x7fcb6a490970) > DBG:tm:reset_timer: (group 0, tl=0x7fcb6a4909a0) > DBG:tm:cleanup_uac_timers: RETR/FR timers reset > DBG:tm:set_timer: relative timeout is 500000 > DBG:tm:insert_timer_unsafe: [4]: 0x7fcb6a490898 (609300000) > DBG:tm:insert_timer_unsafe: [0]: 0x7fcb6a4908c8 (618) > > > Any hints as to what might be going wrong? > > Regards, > Owais > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From millennium.bug at gmail.com Tue Jun 21 12:51:07 2016 From: millennium.bug at gmail.com (Owais Ahmad) Date: Tue, 21 Jun 2016 15:51:07 +0500 Subject: [OpenSIPS-Users] Some calls fail with tcp_connect_blocking error on version 2.1.1 In-Reply-To: <07117de8-bef2-83d0-39de-e3bb849322e0@opensips.org> References: <07117de8-bef2-83d0-39de-e3bb849322e0@opensips.org> Message-ID: Hi ? R?zvan, I have tested with your recommendations. Here are my findings: i) Disabling tcp_accept_aliases increases frequency of this error. ii) Upgrading to 2.1.3 does not resolve the issue. On this new version, I also tested tcp_accept_aliases=0/1. I get the exact same error I posted previously. Is there a way we can control the number of retries opensips should make if there is a UAC connect failure? Saying that because I know for sure that B party has a working network connection. I have configured the following, in case it helps you point me to the actual issue: maxbuffer=65536 tcp_children=100 tcp_accept_aliases=0 tcp_connect_timeout=3 tcp_keepalive=0 modparam("proto_tls", "tls_max_msg_chunks", 8) modparam("proto_tls", "tls_handshake_timeout", 3) modparam("proto_tls", "tls_send_timeout", 7) Regards, Owais On Tue, Jun 21, 2016 at 1:16 PM, R?zvan Crainea wrote: > Hi, Owais! > > You should consider upgrade OpenSIPS to a newer version (at least 2.1.3), > because there were some fixes done related to this issue. If you cannot > upgrade right now, try to disable the auto TCP aliasing[1] (set it to 0). > > [1] http://www.opensips.org/Documentation/Script-CoreParameters-2-1#toc90 > > Best regards, > > ?? > R?zvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From pimenta at inatel.br Tue Jun 21 14:00:38 2016 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Tue, 21 Jun 2016 12:00:38 +0000 Subject: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? In-Reply-To: <81fd240b-6369-15b4-a2ee-99deed334299@opensips.org> References: , <81fd240b-6369-15b4-a2ee-99deed334299@opensips.org> Message-ID: Hi Razvan Crainea. I didn't know about this possibility. I will try this idea now. Thank you very much!! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________ De: users-bounces at lists.opensips.org em nome de Razvan Crainea Enviado: ter?a-feira, 21 de junho de 2016 04:24 Para: users at lists.opensips.org Assunto: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? Hi, Rodrigo! Have you tried restoring the R-URI after the caller lookup? Something like: $var(ru) = $ru; lookup("location", "", "$fu"); # this takes the caller from FROM uri, which I think is more suitable than from contact uri $ru = $var(ru); # continue your processing here .... # now do the real lookup for the callee lookup("location"); Don't do the lookups in the reversed way, because you might loose some contacts. Best regards, Razvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 06/20/2016 09:02 PM, Rodrigo Pimenta Carvalho wrote: Dear OpenSIPS-users, The table location has the column attr where I use to store specific additional information for each registration. Whenever A calls B, I have to read this specific information from the A record and from the B record. That is, I need to get and handle specific information about the caller and callee. For the callee, I use to invoke the lookup("location") function that put the needed information in the attr_avp. That is good and works very well. Then, I just have to read the attr_avp to get such specific information. For the caller, I use to invoke: $var(aorChamador) = $(ct.fields(uri)); lookup("location","","$var(aorChamador)"); However it causes amazing side effect in the SIP signaling. Ex: When A calls B, B stays quiet and A rings. So A can answer A. Crazy! According to the documentation, lookup will overwritten the Request-URI. I guess that is why the SIP signaling become incoherent. How could I get the caller attr specific information without side effects? Any hint will be very helpful!! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at inin.com Tue Jun 21 14:14:16 2016 From: Ben.Newlin at inin.com (Newlin, Ben) Date: Tue, 21 Jun 2016 12:14:16 +0000 Subject: [OpenSIPS-Users] TM flag 0x01 not working in 1.11.7 In-Reply-To: <5768EE4B.2040107@opensips.org> References: <273971259.20160618164908@ptl.ru> <5768EE4B.2040107@opensips.org> Message-ID: Bogdan, That is great for versions 2.1+, but I was asking about 1.11. This sounds like a new feature to me and I don?t understand why it was ported back to break existing functionality. The new TM parameter you mention is not available in 1.11 according to the documentation, so I have no way to get the previous functionality. There is also no documentation in 1.11 or in the release notes that indicates that the 0x01 flag no longer works. Ben Newlin From: Bogdan-Andrei Iancu Date: Tuesday, June 21, 2016 at 3:35 AM To: OpenSIPS users mailling list , "Newlin, Ben" , "denis7979 at mail.ru >> Denis Putyato" Subject: Re: [OpenSIPS-Users] TM flag 0x01 not working in 1.11.7 Hi Please refer to https://github.com/OpenSIPS/opensips/issues/833 . As per documentation, the 0x1 flag became obsolete, as sending the 100Trying is no longer linked to the t_relay() - the 100Trying is now sent when the transaction is created. If you want to disable the auto 100Trying, see the new TM flag auto_100trying: http://www.opensips.org/html/docs/modules/2.2.x/tm.html#id294523 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 18.06.2016 16:49, Denis wrote: Hello! 2.1 has the same problem mailto:denis7979 at mail.ru In my script I send all ?100 Trying? responses manually* and use the 0x01 flag when calling t_relay so that it will not send its own ?100 Giving a try? response, as per the documentation [1]. Unfortunately, in 1.11.7 and 1.11.8 this flag seems to have stopped working which results in multiple 100 responses. This should be easily reproducible with a script that simply calls t_relay with flag 0x01. You will see that a ?100 Giving a try? response is still sent. [1] http://www.opensips.org/html/docs/modules/1.11.x/tm.html#id294528 * I do this because I don?t like that the default is to send the non-standard response ?100 Giving a try? instead of ?100 Trying?. I?ve always wondered why this is that way. Additionally, the automatic response to CANCEL requests is ?200 canceling? instead of the standard ?200 OK?. Unfortunately, I have yet to find a way to workaround the behavior for CANCEL. Ben Newlin _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From jim at devito.cc Tue Jun 21 15:21:00 2016 From: jim at devito.cc (Jim DeVito) Date: Tue, 21 Jun 2016 06:21:00 -0700 Subject: [OpenSIPS-Users] How do I get the latest commit in the yum repo. Message-ID: Hi All, Just submitted https://github.com/OpenSIPS/opensips/issues/914 and am planning to close it because it looks like the problem has been fixed in a commit from 20 days ago. My question is how I get the fix in the YUM release package with out running a nightly release in production. Thanks!! -- Jim DeVito From bogdan at opensips.org Tue Jun 21 15:42:50 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 21 Jun 2016 16:42:50 +0300 Subject: [OpenSIPS-Users] TM flag 0x01 not working in 1.11.7 In-Reply-To: References: <273971259.20160618164908@ptl.ru> <5768EE4B.2040107@opensips.org> Message-ID: <5769445A.9090204@opensips.org> Hi Ben, My bad, I forgot to backport this param to 1.11 too (while the change affecting the 100 Trying generation was backported). http://www.opensips.org/html/docs/modules/1.11.x/tm.html#id294521 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 21.06.2016 15:14, Newlin, Ben wrote: > Re: [OpenSIPS-Users] TM flag 0x01 not working in 1.11.7 > > Bogdan, > > That is great for versions 2.1+, but I was asking about 1.11. This > sounds like a new feature to me and I don?t understand why it was > ported back to break existing functionality. The new TM parameter you > mention is not available in 1.11 according to the documentation, so I > have no way to get the previous functionality. There is also no > documentation in 1.11 or in the release notes that indicates that the > 0x01 flag no longer works. > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > *Date: *Tuesday, June 21, 2016 at 3:35 AM > *To: *OpenSIPS users mailling list , > "Newlin, Ben" , "denis7979 at mail.ru >> Denis > Putyato" > *Subject: *Re: [OpenSIPS-Users] TM flag 0x01 not working in 1.11.7 > > Hi > > Please refer to https://github.com/OpenSIPS/opensips/issues/833 . > > As per documentation, the 0x1 flag became obsolete, as sending the > 100Trying is no longer linked to the t_relay() - the 100Trying is now > sent when the transaction is created. > > If you want to disable the auto 100Trying, see the new TM flag > auto_100trying: > http://www.opensips.org/html/docs/modules/2.2.x/tm.html#id294523 > > Regards, > > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 18.06.2016 16:49, Denis wrote: > > Hello! > > 2.1 has the same problem > > //mailto:denis7979 at mail.ru > > > > > In my script I send all ?100 Trying? responses manually* and use > the 0x01 flag when calling t_relay so that it will not send its > own ?100 Giving a try? response, as per the documentation [1]. > > Unfortunately, in 1.11.7 and 1.11.8 this flag seems to have > stopped working which results in multiple 100 responses. > > This should be easily reproducible with a script that simply calls > t_relay with flag 0x01. You will see that a ?100 Giving a try? > response is still sent. > > [1] http://www.opensips.org/html/docs/modules/1.11.x/tm.html#id294528 > > * I do this because I don?t like that the default is to send the > non-standard response ?100 Giving a try? instead of ?100 Trying?. > I?ve always wondered why this is that way. Additionally, the > automatic response to CANCEL requests is ?200 canceling? instead > of the standard ?200 OK?. Unfortunately, I have yet to find a way > to workaround the behavior for CANCEL. > > Ben Newlin > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at inin.com Tue Jun 21 15:46:15 2016 From: Ben.Newlin at inin.com (Newlin, Ben) Date: Tue, 21 Jun 2016 13:46:15 +0000 Subject: [OpenSIPS-Users] TM flag 0x01 not working in 1.11.7 In-Reply-To: <5769445A.9090204@opensips.org> References: <273971259.20160618164908@ptl.ru> <5768EE4B.2040107@opensips.org> <5769445A.9090204@opensips.org> Message-ID: <2E333A19-F6B9-4360-81C4-389579585115@inin.com> Bogdan, Thanks. I assume the parameter will be available in 1.11.9? Ben Newlin From: Bogdan-Andrei Iancu Date: Tuesday, June 21, 2016 at 9:42 AM To: OpenSIPS users mailling list , "Newlin, Ben" , "Newlin, Ben" Subject: Re: [OpenSIPS-Users] TM flag 0x01 not working in 1.11.7 Hi Ben, My bad, I forgot to backport this param to 1.11 too (while the change affecting the 100 Trying generation was backported). http://www.opensips.org/html/docs/modules/1.11.x/tm.html#id294521 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 21.06.2016 15:14, Newlin, Ben wrote: Bogdan, That is great for versions 2.1+, but I was asking about 1.11. This sounds like a new feature to me and I don?t understand why it was ported back to break existing functionality. The new TM parameter you mention is not available in 1.11 according to the documentation, so I have no way to get the previous functionality. There is also no documentation in 1.11 or in the release notes that indicates that the 0x01 flag no longer works. Ben Newlin From: Bogdan-Andrei Iancu Date: Tuesday, June 21, 2016 at 3:35 AM To: OpenSIPS users mailling list , "Newlin, Ben" , "denis7979 at mail.ru >> Denis Putyato" Subject: Re: [OpenSIPS-Users] TM flag 0x01 not working in 1.11.7 Hi Please refer to https://github.com/OpenSIPS/opensips/issues/833 . As per documentation, the 0x1 flag became obsolete, as sending the 100Trying is no longer linked to the t_relay() - the 100Trying is now sent when the transaction is created. If you want to disable the auto 100Trying, see the new TM flag auto_100trying: http://www.opensips.org/html/docs/modules/2.2.x/tm.html#id294523 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 18.06.2016 16:49, Denis wrote: Hello! 2.1 has the same problem mailto:denis7979 at mail.ru In my script I send all ?100 Trying? responses manually* and use the 0x01 flag when calling t_relay so that it will not send its own ?100 Giving a try? response, as per the documentation [1]. Unfortunately, in 1.11.7 and 1.11.8 this flag seems to have stopped working which results in multiple 100 responses. This should be easily reproducible with a script that simply calls t_relay with flag 0x01. You will see that a ?100 Giving a try? response is still sent. [1] http://www.opensips.org/html/docs/modules/1.11.x/tm.html#id294528 * I do this because I don?t like that the default is to send the non-standard response ?100 Giving a try? instead of ?100 Trying?. I?ve always wondered why this is that way. Additionally, the automatic response to CANCEL requests is ?200 canceling? instead of the standard ?200 OK?. Unfortunately, I have yet to find a way to workaround the behavior for CANCEL. Ben Newlin _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jun 21 15:57:05 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 21 Jun 2016 16:57:05 +0300 Subject: [OpenSIPS-Users] TM flag 0x01 not working in 1.11.7 In-Reply-To: <2E333A19-F6B9-4360-81C4-389579585115@inin.com> References: <273971259.20160618164908@ptl.ru> <5768EE4B.2040107@opensips.org> <5769445A.9090204@opensips.org> <2E333A19-F6B9-4360-81C4-389579585115@inin.com> Message-ID: <576947B1.9020508@opensips.org> Yes, or grab the 1.11 GIT branch Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 21.06.2016 16:46, Newlin, Ben wrote: > Re: [OpenSIPS-Users] TM flag 0x01 not working in 1.11.7 > > Bogdan, > > Thanks. I assume the parameter will be available in 1.11.9? > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > *Date: *Tuesday, June 21, 2016 at 9:42 AM > *To: *OpenSIPS users mailling list , > "Newlin, Ben" , "Newlin, Ben" > *Subject: *Re: [OpenSIPS-Users] TM flag 0x01 not working in 1.11.7 > > Hi Ben, > > My bad, I forgot to backport this param to 1.11 too (while the change > affecting the 100 Trying generation was backported). > > http://www.opensips.org/html/docs/modules/1.11.x/tm.html#id294521 > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 21.06.2016 15:14, Newlin, Ben wrote: > > Bogdan, > > That is great for versions 2.1+, but I was asking about 1.11. This > sounds like a new feature to me and I don?t understand why it was > ported back to break existing functionality. The new TM parameter > you mention is not available in 1.11 according to the > documentation, so I have no way to get the previous functionality. > There is also no documentation in 1.11 or in the release notes > that indicates that the 0x01 flag no longer works. > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > > *Date: *Tuesday, June 21, 2016 at 3:35 AM > *To: *OpenSIPS users mailling list > , "Newlin, Ben" > , > "denis7979 at mail.ru >> Denis Putyato" > > *Subject: *Re: [OpenSIPS-Users] TM flag 0x01 not working in 1.11.7 > > Hi > > Please refer to https://github.com/OpenSIPS/opensips/issues/833 . > > As per documentation, the 0x1 flag became obsolete, as sending the > 100Trying is no longer linked to the t_relay() - the 100Trying is > now sent when the transaction is created. > > If you want to disable the auto 100Trying, see the new TM flag > auto_100trying: > http://www.opensips.org/html/docs/modules/2.2.x/tm.html#id294523 > > Regards, > > > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > On 18.06.2016 16:49, Denis wrote: > > Hello! > > 2.1 has the same problem > > //mailto:denis7979 at mail.ru > > > > > In my script I send all ?100 Trying? responses manually* and > use the 0x01 flag when calling t_relay so that it will not > send its own ?100 Giving a try? response, as per the > documentation [1]. > > Unfortunately, in 1.11.7 and 1.11.8 this flag seems to have > stopped working which results in multiple 100 responses. > > This should be easily reproducible with a script that simply > calls t_relay with flag 0x01. You will see that a ?100 Giving > a try? response is still sent. > > [1] > http://www.opensips.org/html/docs/modules/1.11.x/tm.html#id294528 > > * I do this because I don?t like that the default is to send > the non-standard response ?100 Giving a try? instead of ?100 > Trying?. I?ve always wondered why this is that way. > Additionally, the automatic response to CANCEL requests is > ?200 canceling? instead of the standard ?200 OK?. > Unfortunately, I have yet to find a way to workaround the > behavior for CANCEL. > > Ben Newlin > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jun 21 16:04:54 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 21 Jun 2016 17:04:54 +0300 Subject: [OpenSIPS-Users] How do I get the latest commit in the yum repo. In-Reply-To: References: Message-ID: <57694986.9010701@opensips.org> Hi Jim, Currently we generate packages based on GIT tags only. So far there is no TAG covering this commit - the commit is on 1st of June, while 2.2.0 was released on 27th of May. Still, you can use the nightly build RPMs: |yum install http://yum.opensips.org/2.2/nightly/el/7/x86_64/opensips-yum-nightly-2.2-3.el7.noarch.rpm Regards, | Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 21.06.2016 16:21, Jim DeVito wrote: > Hi All, > > Just submitted https://github.com/OpenSIPS/opensips/issues/914 and am > planning to close it because it looks like the problem has been fixed > in a commit from 20 days ago. My question is how I get the fix in the > YUM release package with out running a nightly release in production. > > Thanks!! > -------------- next part -------------- An HTML attachment was scrubbed... URL: From jim at devito.cc Tue Jun 21 16:19:50 2016 From: jim at devito.cc (Jim DeVito) Date: Tue, 21 Jun 2016 07:19:50 -0700 Subject: [OpenSIPS-Users] How do I get the latest commit in the yum repo. In-Reply-To: <57694986.9010701@opensips.org> References: <57694986.9010701@opensips.org> Message-ID: <0bd1ad83f0f8c37865c5eb8918e1b4e4@mail.devito.cc> Super good. Thanks!! --- Jim DeVito Mobile 440.941.3860 On 2016-06-21 07:04, Bogdan-Andrei Iancu wrote: > Hi Jim, > > Currently we generate packages based on GIT tags only. So far there > is no TAG covering this commit - the commit is on 1st of June, while > 2.2.0 was released on 27th of May. > > Still, you can use the nightly build RPMs: > yum install > http://yum.opensips.org/2.2/nightly/el/7/x86_64/opensips-yum-nightly-2.2-3.el7.noarch.rpm > [3] > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com [2] > > On 21.06.2016 16:21, Jim DeVito wrote: > >> Hi All, >> >> Just submitted https://github.com/OpenSIPS/opensips/issues/914 [1] >> and am planning to close it because it looks like the problem has >> been fixed in a commit from 20 days ago. My question is how I get >> the fix in the YUM release package with out running a nightly >> release in production. >> >> Thanks!! > > > > Links: > ------ > [1] https://github.com/OpenSIPS/opensips/issues/914 > [2] http://www.opensips-solutions.com > [3] > http://yum.opensips.org/2.2/nightly/el/7/x86_64/opensips-yum-nightly-2.2-3.el7.noarch.rpm From pimenta at inatel.br Tue Jun 21 16:39:34 2016 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Tue, 21 Jun 2016 14:39:34 +0000 Subject: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? In-Reply-To: <81fd240b-6369-15b4-a2ee-99deed334299@opensips.org> References: , <81fd240b-6369-15b4-a2ee-99deed334299@opensips.org> Message-ID: Hi R?zvan. I have tried that idea. But that didn't work. The SIP INVITE message is being changed by the OpenSIPS in a wrong way, in my point of view. Do you know some way to save the entire SIP INVITE message before calling lookup() and then make the saved message take place after the lookup() execution? My original message is: INVITE sip:6001 at myDomain.com.br SIP/2.0 Via: SIP/2.0/TCP 192.168.21.40:5090;rport;branch=z9hG4bK876727215 From: ;tag=179920819 To: Call-ID: 1410250893 CSeq: 21 INVITE Contact: Proxy-Authorization: Digest username="crdphmacl_SPnuV5xqtnSX", realm="localhost", nonce="5769458c000001cc263a7c0d6995dc48d42288ec6f8e4048", uri="sip:6001 at myDomain.com.br", response="0f4c122d2a0a28dea6194c235cd77430", algorithm=MD5 Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: Linphone/3.6.1 (eXosip2/4.0.0) Subject: Phone call Content-Length: 227 This is being changed to: INVITE sip:crdphmacl_SPnuV5xqtnSX at 131.221.240.204:60672;transport=tcp;line=c6356a7d87d6f81 SIP/2.0 Record-Route: Via: SIP/2.0/TCP myDomain.com.br:5060;branch=z9hG4bKe2db.49d54587.0;i=1 Via: SIP/2.0/TCP 192.168.21.40:5090;received=xxx.yyy.240.204;rport=60672;branch=z9hG4bK716249970 From: ;tag=12586028 To: Call-ID: 1106771604 CSeq: 21 INVITE Contact: Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: Linphone/3.6.1 (eXosip2/4.0.0) Subject: Phone call Content-Length: 224 So, the caller is receiving its own SIP INVITE. That is why when A calls B, is A that rings, not B. It is becoming a bit complicated. So, I suspect I'm going to the incorrect direction.... Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________ De: users-bounces at lists.opensips.org em nome de R?zvan Crainea Enviado: ter?a-feira, 21 de junho de 2016 04:24 Para: users at lists.opensips.org Assunto: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? Hi, Rodrigo! Have you tried restoring the R-URI after the caller lookup? Something like: $var(ru) = $ru; lookup("location", "", "$fu"); # this takes the caller from FROM uri, which I think is more suitable than from contact uri $ru = $var(ru); # continue your processing here .... # now do the real lookup for the callee lookup("location"); Don't do the lookups in the reversed way, because you might loose some contacts. Best regards, R?zvan Crainea OpenSIPS Solutions www.opensips-solutions.com Home ? OpenSIPS Solutions www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 06/20/2016 09:02 PM, Rodrigo Pimenta Carvalho wrote: Dear OpenSIPS-users, The table location has the column attr where I use to store specific additional information for each registration. Whenever A calls B, I have to read this specific information from the A record and from the B record. That is, I need to get and handle specific information about the caller and callee. For the callee, I use to invoke the lookup("location") function that put the needed information in the attr_avp. That is good and works very well. Then, I just have to read the attr_avp to get such specific information. For the caller, I use to invoke: $var(aorChamador) = $(ct.fields(uri)); lookup("location","","$var(aorChamador)"); However it causes amazing side effect in the SIP signaling. Ex: When A calls B, B stays quiet and A rings. So A can answer A. Crazy! According to the documentation, lookup will overwritten the Request-URI. I guess that is why the SIP signaling become incoherent. How could I get the caller attr specific information without side effects? Any hint will be very helpful!! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From sevpal at aol.com Tue Jun 21 17:20:57 2016 From: sevpal at aol.com (sevpal) Date: Tue, 21 Jun 2016 11:20:57 -0400 Subject: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? In-Reply-To: References: , <81fd240b-6369-15b4-a2ee-99deed334299@opensips.org> Message-ID: Hi, have you tried/considered running a simple query on the database and parsing for the information you need? From: Rodrigo Pimenta Carvalho Sent: Tuesday, June 21, 2016 10:39 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? Hi R?zvan. I have tried that idea. But that didn't work. The SIP INVITE message is being changed by the OpenSIPS in a wrong way, in my point of view. Do you know some way to save the entire SIP INVITE message before calling lookup() and then make the saved message take place after the lookup() execution? My original message is: INVITE sip:6001 at myDomain.com.br SIP/2.0 Via: SIP/2.0/TCP 192.168.21.40:5090;rport;branch=z9hG4bK876727215 From: ;tag=179920819 To: Call-ID: 1410250893 CSeq: 21 INVITE Contact: Proxy-Authorization: Digest username="crdphmacl_SPnuV5xqtnSX", realm="localhost", nonce="5769458c000001cc263a7c0d6995dc48d42288ec6f8e4048", uri="sip:6001 at myDomain.com.br", response="0f4c122d2a0a28dea6194c235cd77430", algorithm=MD5 Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: Linphone/3.6.1 (eXosip2/4.0.0) Subject: Phone call Content-Length: 227 This is being changed to: INVITE sip:crdphmacl_SPnuV5xqtnSX at 131.221.240.204:60672;transport=tcp;line=c6356a7d87d6f81 SIP/2.0 Record-Route: Via: SIP/2.0/TCP myDomain.com.br:5060;branch=z9hG4bKe2db.49d54587.0;i=1 Via: SIP/2.0/TCP 192.168.21.40:5090;received=xxx.yyy.240.204;rport=60672;branch=z9hG4bK716249970 From: ;tag=12586028 To: Call-ID: 1106771604 CSeq: 21 INVITE Contact: Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: Linphone/3.6.1 (eXosip2/4.0.0) Subject: Phone call Content-Length: 224 So, the caller is receiving its own SIP INVITE. That is why when A calls B, is A that rings, not B. It is becoming a bit complicated. So, I suspect I'm going to the incorrect direction.... Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 -------------------------------------------------------------------------------- De: users-bounces at lists.opensips.org em nome de R?zvan Crainea Enviado: ter?a-feira, 21 de junho de 2016 04:24 Para: users at lists.opensips.org Assunto: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? Hi, Rodrigo! Have you tried restoring the R-URI after the caller lookup? Something like: $var(ru) = $ru; lookup("location", "", "$fu"); # this takes the caller from FROM uri, which I think is more suitable than from contact uri $ru = $var(ru); # continue your processing here .... # now do the real lookup for the callee lookup("location"); Don't do the lookups in the reversed way, because you might loose some contacts. Best regards, R?zvan Crainea OpenSIPS Solutions www.opensips-solutions.comHome ? OpenSIPS Solutions www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 06/20/2016 09:02 PM, Rodrigo Pimenta Carvalho wrote: Dear OpenSIPS-users, The table location has the column attr where I use to store specific additional information for each registration. Whenever A calls B, I have to read this specific information from the A record and from the B record. That is, I need to get and handle specific information about the caller and callee. For the callee, I use to invoke the lookup("location") function that put the needed information in the attr_avp. That is good and works very well. Then, I just have to read the attr_avp to get such specific information. For the caller, I use to invoke: $var(aorChamador) = $(ct.fields(uri)); lookup("location","","$var(aorChamador)"); However it causes amazing side effect in the SIP signaling. Ex: When A calls B, B stays quiet and A rings. So A can answer A. Crazy! According to the documentation, lookup will overwritten the Request-URI. I guess that is why the SIP signaling become incoherent. How could I get the caller attr specific information without side effects? Any hint will be very helpful!! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------------------------------------------------------------------------- _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From john.nash778 at gmail.com Tue Jun 21 18:38:22 2016 From: john.nash778 at gmail.com (John Nash) Date: Tue, 21 Jun 2016 22:08:22 +0530 Subject: [OpenSIPS-Users] web sockets (wss) error Message-ID: I downloaded opensips 2.2 stable tar file and upgraded my existing opensips.cfg to use wss as per document http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 but when I start opensips I get error "cannot handle protocol " right at the line where I have listener wss:127.0.0.1:443 Do I need to clone current git version in order to test wss? -------------- next part -------------- An HTML attachment was scrubbed... URL: From pimenta at inatel.br Tue Jun 21 18:39:16 2016 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Tue, 21 Jun 2016 16:39:16 +0000 Subject: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? In-Reply-To: References: , <81fd240b-6369-15b4-a2ee-99deed334299@opensips.org> , Message-ID: Hi Sevpal. Yes. That is what I was doing. It worked very well. But, nowadays I'm using db_mode = 0 for usrloc. So, the information is always only in RAM. In this case, the query will return no result. That is why I'm trying to read the attr column from table location, from RAM, and get specific information for the caller. For the callee, everything is all right. Regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________ De: users-bounces at lists.opensips.org em nome de sevpal Enviado: ter?a-feira, 21 de junho de 2016 12:20 Para: OpenSIPS users mailling list Assunto: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? Hi, have you tried/considered running a simple query on the database and parsing for the information you need? From: Rodrigo Pimenta Carvalho Sent: Tuesday, June 21, 2016 10:39 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? Hi R?zvan. I have tried that idea. But that didn't work. The SIP INVITE message is being changed by the OpenSIPS in a wrong way, in my point of view. Do you know some way to save the entire SIP INVITE message before calling lookup() and then make the saved message take place after the lookup() execution? My original message is: INVITE sip:6001 at myDomain.com.br SIP/2.0 Via: SIP/2.0/TCP 192.168.21.40:5090;rport;branch=z9hG4bK876727215 From: ;tag=179920819 To: Call-ID: 1410250893 CSeq: 21 INVITE Contact: Proxy-Authorization: Digest username="crdphmacl_SPnuV5xqtnSX", realm="localhost", nonce="5769458c000001cc263a7c0d6995dc48d42288ec6f8e4048", uri="sip:6001 at myDomain.com.br", response="0f4c122d2a0a28dea6194c235cd77430", algorithm=MD5 Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: Linphone/3.6.1 (eXosip2/4.0.0) Subject: Phone call Content-Length: 227 This is being changed to: INVITE sip:crdphmacl_SPnuV5xqtnSX at 131.221.240.204:60672;transport=tcp;line=c6356a7d87d6f81 SIP/2.0 Record-Route: Via: SIP/2.0/TCP myDomain.com.br:5060;branch=z9hG4bKe2db.49d54587.0;i=1 Via: SIP/2.0/TCP 192.168.21.40:5090;received=xxx.yyy.240.204;rport=60672;branch=z9hG4bK716249970 From: ;tag=12586028 To: Call-ID: 1106771604 CSeq: 21 INVITE Contact: Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: Linphone/3.6.1 (eXosip2/4.0.0) Subject: Phone call Content-Length: 224 So, the caller is receiving its own SIP INVITE. That is why when A calls B, is A that rings, not B. It is becoming a bit complicated. So, I suspect I'm going to the incorrect direction.... Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________ De: users-bounces at lists.opensips.org em nome de R?zvan Crainea Enviado: ter?a-feira, 21 de junho de 2016 04:24 Para: users at lists.opensips.org Assunto: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? Hi, Rodrigo! Have you tried restoring the R-URI after the caller lookup? Something like: $var(ru) = $ru; lookup("location", "", "$fu"); # this takes the caller from FROM uri, which I think is more suitable than from contact uri $ru = $var(ru); # continue your processing here .... # now do the real lookup for the callee lookup("location"); Don't do the lookups in the reversed way, because you might loose some contacts. Best regards, R?zvan Crainea OpenSIPS Solutions www.opensips-solutions.com Home - OpenSIPS Solutions www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 06/20/2016 09:02 PM, Rodrigo Pimenta Carvalho wrote: Dear OpenSIPS-users, The table location has the column attr where I use to store specific additional information for each registration. Whenever A calls B, I have to read this specific information from the A record and from the B record. That is, I need to get and handle specific information about the caller and callee. For the callee, I use to invoke the lookup("location") function that put the needed information in the attr_avp. That is good and works very well. Then, I just have to read the attr_avp to get such specific information. For the caller, I use to invoke: $var(aorChamador) = $(ct.fields(uri)); lookup("location","","$var(aorChamador)"); However it causes amazing side effect in the SIP signaling. Ex: When A calls B, B stays quiet and A rings. So A can answer A. Crazy! According to the documentation, lookup will overwritten the Request-URI. I guess that is why the SIP signaling become incoherent. How could I get the caller attr specific information without side effects? Any hint will be very helpful!! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ________________________________ _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From Agalya_Ramachandran at comcast.com Tue Jun 21 20:08:40 2016 From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor)) Date: Tue, 21 Jun 2016 18:08:40 +0000 Subject: [OpenSIPS-Users] DNS-SRV query in opensips References: <955ec37716d94d139c82be2835377bd5@COPDCEX28.cable.comcast.com> <5768EF90.2030802@opensips.org> Message-ID: Hi Bogdan, I have a question regarding seturi and setdsturi function calls. As far as my understanding, when append_branch() is called, seturi () is called to set the URI where to fork the call. I tried by calling only seturi () function call, after append_branch it was working same behavior as when I used setdsturi() as well. My question is do we really need setdsturi or when is the case when setdsturi() is used.? Regards, Agalya From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Tuesday, June 21, 2016 3:41 AM To: OpenSIPS users mailling list >; Ramachandran, Agalya (Contractor) >; Ramachandran, Agalya (Contractor) > Subject: Re: [OpenSIPS-Users] DNS-SRV query in opensips Hi Agalya, OpenSIPS does full flavor DNS lookup (with NATPR and SRV), but this is internal, and not accessible from script. OpenSIPS implements auto DNS-based failover : http://www.opensips.org/html/docs/modules/2.2.x/tm.html#id293694 My understanding is you want the DNS resolving to be done at script level and to have access to the results ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 20.06.2016 20:35, Ramachandran, Agalya (Contractor) wrote: Hi team, We are using opensips for our project requirements. I have a scenario where we need DNS-SRV query and the result of this should be placed as the desturi to send request out.(In case of forking call) As far as I went through opensips documentation, there are some core parameters for dns related config such as "dns_retr_time" ,"dns_retr_time" ,"dns_servers_no" etc... According to my understanding these config variables can be declared and used in the opensips.config file to control the settings of DNS query. Is there any available function where I can use and pass the DNS server domain name, so that it fetches the IP address of the host ? Please let us know what is the best way to achieve this? Regards, Agalya _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From pimenta at inatel.br Tue Jun 21 21:38:00 2016 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Tue, 21 Jun 2016 19:38:00 +0000 Subject: [OpenSIPS-Users] Leak AVPOS + SQLITE In-Reply-To: References: Message-ID: Hi. Does someone here is getting/handling memory leaks with OpenSIPS 2.2 and last version of SQLite? I'm using newest commit from OpenSIPS 2.2 and newest version of SQLite. My query is : avp_db_query("select Value from GeneralConfigurations where Attribute = 'CONFIGURATION_INTERCOM_A_NAME'"); Valgrind shows: ==16087== ERROR SUMMARY: 0 errors from 0 contexts (suppressed: 2 from 2) ==16088== Searching for pointers to 296,489 not-freed blocks ==16088== Checked 103,297,688 bytes ==16088== ==16088== 1,024 bytes in 1 blocks are possibly lost in loss record 184 of 246 ==16088== at 0x4C2745D: malloc (in /usr/lib64/valgrind/vgpreload_memcheck-amd64-linux.so) ==16088== by 0x8F8B05F: sqlite3MemMalloc (sqlite3.c:20167) ==16088== by 0x8F701C7: sqlite3Malloc (sqlite3.c:23846) ==16088== by 0x8F75459: pcache1Alloc (sqlite3.c:44312) ==16088== by 0x8F8019F: sqlite3BtreeCursor (sqlite3.c:44455) ==16088== by 0x8FD0FDD: sqlite3VdbeExec (sqlite3.c:80098) ==16088== by 0x8FDB89F: sqlite3_step (sqlite3.c:75131) ==16088== by 0x8FDC9A1: sqlite3_exec (sqlite3.c:108122) ==16088== by 0x8D20736: db_sqlite_raw_query (dbase.c:358) ==16088== by 0x9464DB8: db_query_avp (avpops_db.c:436) ==16088== by 0x946943E: ops_dbquery_avps (avpops_impl.c:840) ==16088== by 0x9459A61: w_dbquery_avps (avpops.c:1395) ==16088== ==16088== LEAK SUMMARY: ==16088== definitely lost: 0 bytes in 0 blocks ==16088== indirectly lost: 0 bytes in 0 blocks ==16088== possibly lost: 1,024 bytes in 1 blocks ==16088== still reachable: 47,457,573 bytes in 296,488 blocks ==16088== suppressed: 0 bytes in 0 blocks ==16088== Reachable blocks (those to which a pointer was found) are not shown. ==16088== To see them, rerun with: --leak-check=full --show-leak-kinds=all After some time running that query, I can see, via command top, that the available memory is decreasing. In fact, the memory is not freed even after stop running the query for a time. Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 -------------- next part -------------- An HTML attachment was scrubbed... URL: From govoiper at gmail.com Tue Jun 21 21:57:29 2016 From: govoiper at gmail.com (SamyGo) Date: Tue, 21 Jun 2016 15:57:29 -0400 Subject: [OpenSIPS-Users] Async event_route and cachdb_redis In-Reply-To: <87e0609d-975d-e80f-50c7-d34c7989f15d@opensips.org> References: <87e0609d-975d-e80f-50c7-d34c7989f15d@opensips.org> Message-ID: Hi , After recompiling , when I start opensips it gives this error: ERROR:event_route:event_route_handler: invalid receive sock info The two event routes I have are these: event_route[E_UL_CONTACT_INSERT,async] { fetch_event_params("aor=$avp(aor);address=$avp(address);received=$avp(received)"); .... cache_raw_query("redis:group1","HSET GLOBAL_USER_LOCATION $avp(aor) $var(my_value1)"); } event_route[E_UL_AOR_DELETE,async] { fetch_event_params("aor=$avp(aor)"); ... cache_raw_query("redis:group1","DEL GLOBAL_USER_LOCATION $avp(aor)"); } Some Xlog lines in both of these routes, nothing seems to be printed now, no error , no cache data modifications executing.. I'll see in further detail what is happening and if I find anything abnormal will reply. Regards. Sammy On Tue, Jun 21, 2016 at 3:40 AM, R?zvan Crainea wrote: > Hi, Sammy! > > Could you try this patch: > > https://gist.github.com/razvancrainea/9d239c82474bb0f1c403b6459dbdb647 > > Thanks, > > R?zvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 06/19/2016 08:56 PM, SamyGo wrote: > > Hi, > I'm seeing errors from cachedb_redis module when called in an event route > in async mode. > > event_route[E_UL_CONTACT_INSERT,async] { > ... > cache_raw_query("redis:group1","SET ABC"); > .. > > } > > OpenSIPS throws error stating that redis group1 unavailable > > DBG:core:cachedb_raw_query: from script [redis] - with grp [group1] > ERROR:core:cachedb_raw_query: failed to get connection for grp name > [group1] > > I tried same command in main route of reply route, all works normal. if I > remove the "async" from the event_route definition it works in event route. > > Any logical reason why async route don't recognize the connections ? > > Tried with OpenSIPS 2.2 and 2.1 as well, same behavior. > > > Regards, > Sammy > > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From john.nash778 at gmail.com Wed Jun 22 05:31:27 2016 From: john.nash778 at gmail.com (John Nash) Date: Wed, 22 Jun 2016 09:01:27 +0530 Subject: [OpenSIPS-Users] web sockets (wss) error In-Reply-To: References: Message-ID: Git version (2.2) is OK. May be in tar download latest files are not there no biggi. On Tue, Jun 21, 2016 at 10:08 PM, John Nash wrote: > I downloaded opensips 2.2 stable tar file and upgraded my existing > opensips.cfg to use wss as per document > http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 > > but when I start opensips I get error "cannot handle protocol " right > at the line where I have listener wss:127.0.0.1:443 > > Do I need to clone current git version in order to test wss? > -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Wed Jun 22 08:57:26 2016 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Wed, 22 Jun 2016 09:57:26 +0300 Subject: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? In-Reply-To: References: <81fd240b-6369-15b4-a2ee-99deed334299@opensips.org> Message-ID: Hi, Rodrigo! Can you print the $ru variable before and after each lookup() query? Something like: $var(ru) = $ru; xlog("R-URI before caller lookup: $ru\n"); lookup("location", "", "$fu"); # this takes the caller from FROM uri, which I think is more suitable than from contact uri $ru = $var(ru); # continue your processing here xlog("R-URI after caller lookup: $ru\n"); ... # now do the real lookup for the callee xlog("R-URI before callee lookup: $ru\n"); lookup("location"); xlog("R-URI after callee lookup: $ru\n"); Make sure they are all correct, or if they are not, send me these logs. Thanks, R?zvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 06/21/2016 07:39 PM, Rodrigo Pimenta Carvalho wrote: > > Hi Sevpal. > > > Yes. That is what I was doing. It worked very well. > > But, nowadays I'm using db_mode = 0 for usrloc. So, the information is > always only in RAM. In this case, the query will return no result. > That is why I'm trying to read the attr column from table location, > from RAM, and get specific information for the caller. > > > For the callee, everything is all right. > > > Regards. > > > RODRIGO PIMENTA CARVALHO > Inatel Competence Center > Software > Ph: +55 35 3471 9200 RAMAL 979 > > > ------------------------------------------------------------------------ > *De:* users-bounces at lists.opensips.org > em nome de sevpal > *Enviado:* ter?a-feira, 21 de junho de 2016 12:20 > *Para:* OpenSIPS users mailling list > *Assunto:* Re: [OpenSIPS-Users] How to invok lookup() and get attr > from the caller, without side effects? > Hi, have you tried/considered running a simple query on the database > and parsing for the information you need? > *From:* Rodrigo Pimenta Carvalho > *Sent:* Tuesday, June 21, 2016 10:39 AM > *To:* OpenSIPS users mailling list > *Subject:* Re: [OpenSIPS-Users] How to invok lookup() and get attr > from the caller, without side effects? > > Hi R?zvan. > > I have tried that idea. But that didn't work. The SIP INVITE message > is being changed by the OpenSIPS in a wrong way, in my point of view. > > Do you know some way to save the entire SIP INVITE message before > calling lookup() and then make the saved message take place after the > lookup() execution? > > My original message is: > > INVITE sip:6001 at myDomain.com.br SIP/2.0 > Via: SIP/2.0/TCP 192.168.21.40:5090;rport;branch=z9hG4bK876727215 > From: ;tag=179920819 > To: > Call-ID: 1410250893 > CSeq: 21 INVITE > Contact: > Proxy-Authorization: Digest username="crdphmacl_SPnuV5xqtnSX", > realm="localhost", > nonce="5769458c000001cc263a7c0d6995dc48d42288ec6f8e4048", > uri="sip:6001 at myDomain.com.br", > response="0f4c122d2a0a28dea6194c235cd77430", algorithm=MD5 > Content-Type: application/sdp > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, > SUBSCRIBE, INFO > Max-Forwards: 70 > User-Agent: Linphone/3.6.1 (eXosip2/4.0.0) > Subject: Phone call > Content-Length: 227 > > This is being changed to: > > INVITE > sip:crdphmacl_SPnuV5xqtnSX at 131.221.240.204:60672;transport=tcp;line=c6356a7d87d6f81 > SIP/2.0 > Record-Route: > > Via: SIP/2.0/TCP myDomain.com.br:5060;branch=z9hG4bKe2db.49d54587.0;i=1 > Via: SIP/2.0/TCP > 192.168.21.40:5090;received=xxx.yyy.240.204;rport=60672;branch=z9hG4bK716249970 > From: ;tag=12586028 > To: > Call-ID: 1106771604 > CSeq: 21 INVITE > Contact: > Content-Type: application/sdp > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, > SUBSCRIBE, INFO > Max-Forwards: 70 > User-Agent: Linphone/3.6.1 (eXosip2/4.0.0) > Subject: Phone call > Content-Length: 224 > > So, the caller is receiving its own SIP INVITE. > > That is why when A calls B, is A that rings, not B. > > It is becoming a bit complicated. So, I suspect I'm going to the > incorrect direction.... > > Best regards. > > RODRIGO PIMENTA CARVALHO > Inatel Competence Center > Software > Ph: +55 35 3471 9200 RAMAL 979 > > > ------------------------------------------------------------------------ > *De:* users-bounces at lists.opensips.org > em nome de R?zvan Crainea > > *Enviado:* ter?a-feira, 21 de junho de 2016 04:24 > *Para:* users at lists.opensips.org > *Assunto:* Re: [OpenSIPS-Users] How to invok lookup() and get attr > from the caller, without side effects? > > Hi, Rodrigo! > > > Have you tried restoring the R-URI after the caller lookup? Something > like: > > > $var(ru) = $ru; > > lookup("location", "", "$fu"); # this takes the caller from FROM uri, > which I think is more suitable than from contact uri > > $ru = $var(ru); > > # continue your processing here > > .... > > # now do the real lookup for the callee > > lookup("location"); > > > Don't do the lookups in the reversed way, because you might loose some > contacts. > > > Best regards, > > R?zvan Crainea > OpenSIPS Solutions > www.opensips-solutions.com > Home ? OpenSIPS Solutions > www.opensips-solutions.com > OpenSIPS is a mature Open Source implementation of a SIP server. > OpenSIPS is more than a SIP proxy/router as it includes > application-level functionalities. > > On 06/20/2016 09:02 PM, Rodrigo Pimenta Carvalho wrote: >> >> Dear OpenSIPS-users, >> >> The table location has the column attr where I use to store specific >> additional information for each registration. >> >> Whenever A calls B, I have to read this specific information from the >> A record and from the B record. That is, I need to get and handle >> specific information about the caller and callee. >> >> For the callee, I use to invoke the lookup("location") function that >> put the needed information in the attr_avp. That is good and works >> very well. Then, I just have to read the attr_avp to get such >> specific information. >> >> For the caller, I use to invoke: >> >> $var(aorChamador) = $(ct.fields(uri)); >> >> lookup("location","","$var(aorChamador)"); >> >> However it causes amazing side effect in the SIP signaling. Ex: When >> A calls B, B stays quiet and A rings. So A can answer A. Crazy! >> >> According to the documentation, lookup will overwritten the >> Request-URI. I guess that is why the SIP signaling become incoherent. >> >> >> How could I get the caller attr specific information without side >> effects? >> >> >> Any hint will be very helpful!! >> >> >> Best regards. >> >> RODRIGO PIMENTA CARVALHO >> Inatel Competence Center >> Software >> Ph: +55 35 3471 9200 RAMAL 979 >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > ------------------------------------------------------------------------ > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Wed Jun 22 09:04:16 2016 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Wed, 22 Jun 2016 10:04:16 +0300 Subject: [OpenSIPS-Users] Leak AVPOS + SQLITE In-Reply-To: References: Message-ID: Hi, Rodrigo! Valgrind may report some memory allocated, and not freed, but that is not necessarily a memory leak. There is a single block of 1024 bytes not freed during runtime, so I think that is peanuts. The memory used by OpenSIPS is not allocated with malloc, so cannot be traced by valgrind. Regarding the system memory, it is normal to decrease as OpenSIPS uses that memory during runtime. However, after some time, this should stabilize. Anyhow, sometimes the system memory might generate false alarms, so if you are tracing any memory leaks, you should check OpenSIPS's internal statistics. Best regards, R?zvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 06/21/2016 10:38 PM, Rodrigo Pimenta Carvalho wrote: > > Hi. > > > Does someone here is getting/handling memory leaks with OpenSIPS 2.2 > and last version of SQLite? > > I'm using newest commit from OpenSIPS 2.2 and newest version of SQLite. > > > My query is : > > > avp_db_query("select Value from GeneralConfigurations where Attribute > = 'CONFIGURATION_INTERCOM_A_NAME'"); > > > Valgrind shows: > > > ==16087== ERROR SUMMARY: 0 errors from 0 contexts (suppressed: 2 from 2) > > ==16088== Searching for pointers to 296,489 not-freed blocks > > ==16088== Checked 103,297,688 bytes > > ==16088== > > ==16088== 1,024 bytes in 1 blocks are possibly lost in loss record 184 > of 246 > > ==16088== at 0x4C2745D: malloc (in > /usr/lib64/valgrind/vgpreload_memcheck-amd64-linux.so) > > ==16088== by 0x8F8B05F: sqlite3MemMalloc (sqlite3.c:20167) > > ==16088== by 0x8F701C7: sqlite3Malloc (sqlite3.c:23846) > > ==16088== by 0x8F75459: pcache1Alloc (sqlite3.c:44312) > > ==16088== by 0x8F8019F: sqlite3BtreeCursor (sqlite3.c:44455) > > ==16088== by 0x8FD0FDD: sqlite3VdbeExec (sqlite3.c:80098) > > ==16088== by 0x8FDB89F: sqlite3_step (sqlite3.c:75131) > > ==16088== by 0x8FDC9A1: sqlite3_exec (sqlite3.c:108122) > > ==16088== by 0x8D20736: db_sqlite_raw_query (dbase.c:358) > > ==16088== by 0x9464DB8: db_query_avp (avpops_db.c:436) > > ==16088== by 0x946943E: ops_dbquery_avps (avpops_impl.c:840) > > ==16088== by 0x9459A61: w_dbquery_avps (avpops.c:1395) > > ==16088== > > ==16088== LEAK SUMMARY: > > ==16088== definitely lost: 0 bytes in 0 blocks > > ==16088== indirectly lost: 0 bytes in 0 blocks > > ==16088== possibly lost: 1,024 bytes in 1 blocks > > ==16088== still reachable: 47,457,573 bytes in 296,488 blocks > > ==16088== suppressed: 0 bytes in 0 blocks > > ==16088== Reachable blocks (those to which a pointer was found) are > not shown. > > ==16088== To see them, rerun with: --leak-check=full --show-leak-kinds=all > > > > After some time running that query, I can see, via command top, that > the available memory is decreasing. > > In fact, the memory is not freed even after stop running the query for > a time. > > > Best regards. > > > RODRIGO PIMENTA CARVALHO > Inatel Competence Center > Software > Ph: +55 35 3471 9200 RAMAL 979 > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From john.nash778 at gmail.com Wed Jun 22 09:07:21 2016 From: john.nash778 at gmail.com (John Nash) Date: Wed, 22 Jun 2016 12:37:21 +0530 Subject: [OpenSIPS-Users] webrtc native client for opensips Message-ID: Apart from sipml5 is there any native webrtc client also which I can explore to work with opensips? The examples I find for webrtc native seem to be using jingle protocol but in case to make it work with opensips, It has to use SIP/SDP at client end right?..Any examples? -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Wed Jun 22 09:10:07 2016 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Wed, 22 Jun 2016 10:10:07 +0300 Subject: [OpenSIPS-Users] Async event_route and cachdb_redis In-Reply-To: References: <87e0609d-975d-e80f-50c7-d34c7989f15d@opensips.org> Message-ID: Hi, Sammy! Does this happen only at startime, or happens during runtime too? Regards, R?zvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 06/21/2016 10:57 PM, SamyGo wrote: > Hi , > > After recompiling , when I start opensips it gives this error: > > ERROR:event_route:event_route_handler: invalid receive sock info > > The two event routes I have are these: > > event_route[E_UL_CONTACT_INSERT,async] { > fetch_event_params("aor=$avp(aor);address=$avp(address);received=$avp(received)"); > .... > cache_raw_query("redis:group1","HSET GLOBAL_USER_LOCATION $avp(aor) > $var(my_value1)"); > > } > > event_route[E_UL_AOR_DELETE,async] { > fetch_event_params("aor=$avp(aor)"); > ... > cache_raw_query("redis:group1","DEL GLOBAL_USER_LOCATION $avp(aor)"); > > } > > > Some Xlog lines in both of these routes, nothing seems to be printed > now, no error , no cache data modifications executing.. > > I'll see in further detail what is happening and if I find anything > abnormal will reply. > > > Regards. > Sammy > > > > > On Tue, Jun 21, 2016 at 3:40 AM, R?zvan Crainea > wrote: > > Hi, Sammy! > > Could you try this patch: > > https://gist.github.com/razvancrainea/9d239c82474bb0f1c403b6459dbdb647 > > Thanks, > > R?zvan Crainea > OpenSIPS Solutions > www.opensips-solutions.com > > On 06/19/2016 08:56 PM, SamyGo wrote: >> Hi, >> I'm seeing errors from cachedb_redis module when called in an >> event route in async mode. >> >> event_route[E_UL_CONTACT_INSERT,async] { >> ... >> cache_raw_query("redis:group1","SET ABC"); >> .. >> >> } >> >> OpenSIPS throws error stating that redis group1 unavailable >> >> DBG:core:cachedb_raw_query: from script [redis] - with grp [group1] >> ERROR:core:cachedb_raw_query: failed to get connection for grp >> name [group1] >> >> I tried same command in main route of reply route, all works >> normal. if I remove the "async" from the event_route definition >> it works in event route. >> >> Any logical reason why async route don't recognize the connections ? >> >> Tried with OpenSIPS 2.2 and 2.1 as well, same behavior. >> >> >> Regards, >> Sammy >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Jun 22 09:15:17 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 22 Jun 2016 10:15:17 +0300 Subject: [OpenSIPS-Users] DNS-SRV query in opensips In-Reply-To: References: <955ec37716d94d139c82be2835377bd5@COPDCEX28.cable.comcast.com> <5768EF90.2030802@opensips.org> Message-ID: <576A3B05.8030702@opensips.org> Hi Agalya, seturi() and setdsturi() set the RURI / DestinationURI for the current message / branch. When youdo an append_branch() a new branch is stored for serial/parallel forking (note that the current branch does not changes - this is branch number 1). So, append_branch() will make a copy of the current branch (RURI, DURI, PATH, Forced Socket, etc) and store a new branch for later forking. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 21.06.2016 21:08, Ramachandran, Agalya (Contractor) wrote: > > Hi Bogdan, > > I have a question regarding /seturi/and /setdsturi///function calls. > > As far as my understanding, when append_branch() is called, seturi () > is called to set the URI where to fork the call. > > I tried by calling only /seturi ()/function call, after append_branch > it was working same behavior as when I used /setdsturi()/as well. > > My question is do we really need /setdsturi /or when is the case when > /setdsturi()/is used.? > > Regards, > Agalya > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* Tuesday, June 21, 2016 3:41 AM > *To:* OpenSIPS users mailling list >; Ramachandran, Agalya (Contractor) > >; Ramachandran, Agalya > (Contractor) > > *Subject:* Re: [OpenSIPS-Users] DNS-SRV query in opensips > > Hi Agalya, > > OpenSIPS does full flavor DNS lookup (with NATPR and SRV), but this is > internal, and not accessible from script. OpenSIPS implements auto > DNS-based failover : > http://www.opensips.org/html/docs/modules/2.2.x/tm.html#id293694 > > My understanding is you want the DNS resolving to be done at script > level and to have access to the results ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 20.06.2016 20:35, Ramachandran, Agalya (Contractor) wrote: > > Hi team, > > We are using opensips for our project requirements. > > I have a scenario where we need DNS-SRV query and the result of > this should be placed as the desturi to send request out.(In case > of forking call) > > As far as I went through opensips documentation, there are some > core parameters for dns related config such as ?dns_retr_time > ? > ,?dns_retr_time > ? ,?dns_servers_no > ? > etc? > > According to my understanding these config variables can be > declared and used in the opensips.config file to control the > settings of DNS query. > > Is there any available function where I can use and pass the DNS > server domain name, so that it fetches the IP address of the host ? > > Please let us know what is the best way to achieve this? > > Regards, > > Agalya > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From Kevin.Stewart at m2group.co.nz Wed Jun 15 03:26:36 2016 From: Kevin.Stewart at m2group.co.nz (Kevin Stewart) Date: Wed, 15 Jun 2016 01:26:36 +0000 Subject: [OpenSIPS-Users] B2BUA not responding to reinvites In-Reply-To: <575FD431.3070708@opensips.org> References: , <575FD431.3070708@opensips.org> Message-ID: After some stracing of opensips I found that the reinvites where not making it to opensips then tracked the issue down to fail2ban :/ solution: aptitude remove ufw fail2ban I am going back to hand crafted iptables rules [?] thanks for the help. Kevin ________________________________ From: users-bounces at lists.opensips.org on behalf of Bogdan-Andrei Iancu Sent: Tuesday, 14 June 2016 9:53:53 p.m. To: OpenSIPS users mailling list; Kevin Stewart Subject: Re: [OpenSIPS-Users] B2BUA not responding to reinvites Hello Kevin, Once the B2B session started, the sequential request will not land to your script as they will be captured by B2B before the script (and handled according to the XML script). Still, with debug level 6 you should see various messages from OpenSIPS when the B2B would handle the re-INVITE. Are you sure the re-INVITE is actually reaching OpenSIPS on the right IP and port ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 14.06.2016 09:49, Kevin Stewart wrote: I am trying to implement a simple topology hiding B2BUA with opensips 1.11.5 most things work except that re invites for session expiry are being ignored. I assume that I am missing something in my config. the strange thing is that I see no debug to syslog when the Invites come in even with debug set to 9. the basic config is a client on port 5061 (C) inviting the B2BUA on port 5066(B) with then invites the main server on another host on port 5060(A). I do not expect and inbount calls from A C invites B, B invites A all proceeds normally until after 15 minutes A invites B and is ignored then sends bye a number of times. none of the reinvites or byes are reported in the logs. C then hangs up sending a bye to B, B sends a bye to A and gets 481 Unknown Dialog. below is my config route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; if (is_method("OPTIONS")) { #xlog("L_NOTICE", "$ci|end|unsupported method"); sl_send_reply("404", "Not found"); exit; } if(is_method("INVITE")){ xlog("L_NOTICE","got invite"); if($sp=="5061"){ xlog("L_NOTICE","got invite 5061"); xlog("L_NOTICE","[$mi] before B2B request\n"); $ru="sip:"+$tU+"@172.22.2.140:5060"; b2b_init_request("top hiding"); xlog("L_INFO","[$mi] after B2B request\n"); }else{ xlog("L_NOTICE","got invite $sp"); xlog("L_INFO","[$mi] not from 5061\n"); } exit; }else{ xlog("L_ERR","got request method $rm from $si: $fU, $tU"); sl_send_reply("501", "Not Implemented"); exit; } } Kevin Stewart | Senior VOIP Network Engineer D: +64 9 919 6120 E: Kevin.Stewart at m2group.co.nz M: +64 21 879 057 W: vocus.co.nz A: PO Box 108-109, Symonds St, Auckland 1150 [cid:part5.09070902.08000600 at opensips.org] Kevin Stewart | Senior VOIP Network Engineer D: +64 9 919 6120E: Kevin.Stewart at m2group.co.nz M: +64 21 879 057W: vocus.co.nz A: PO Box 108-109, Symonds St, Auckland 1150 _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: ATT00001.png Type: image/png Size: 14865 bytes Desc: ATT00001.png URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: OutlookEmoji-?.png Type: image/png Size: 488 bytes Desc: OutlookEmoji-?.png URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image989000.jpg Type: image/png Size: 14865 bytes Desc: image989000.jpg URL: From Ben.Newlin at inin.com Tue Jun 21 17:23:45 2016 From: Ben.Newlin at inin.com (Newlin, Ben) Date: Tue, 21 Jun 2016 15:23:45 +0000 Subject: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? In-Reply-To: References: <81fd240b-6369-15b4-a2ee-99deed334299@opensips.org> Message-ID: <4B53E496-10F8-4824-B4A3-4BF57FEDC242@inin.com> It also seems like AVPOPS module [1] may be a good solution here as it has functions to pull data from a database into AVPs based by user. [1] http://www.opensips.org/html/docs/modules/2.2.x/avpops.html Ben Newlin From: on behalf of sevpal Reply-To: sevpal , OpenSIPS users mailling list Date: Tuesday, June 21, 2016 at 11:20 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? Hi, have you tried/considered running a simple query on the database and parsing for the information you need? From: Rodrigo Pimenta Carvalho Sent: Tuesday, June 21, 2016 10:39 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? Hi R?zvan. I have tried that idea. But that didn't work. The SIP INVITE message is being changed by the OpenSIPS in a wrong way, in my point of view. Do you know some way to save the entire SIP INVITE message before calling lookup() and then make the saved message take place after the lookup() execution? My original message is: INVITE sip:6001 at myDomain.com.br SIP/2.0 Via: SIP/2.0/TCP 192.168.21.40:5090;rport;branch=z9hG4bK876727215 From: ;tag=179920819 To: Call-ID: 1410250893 CSeq: 21 INVITE Contact: Proxy-Authorization: Digest username="crdphmacl_SPnuV5xqtnSX", realm="localhost", nonce="5769458c000001cc263a7c0d6995dc48d42288ec6f8e4048", uri="sip:6001 at myDomain.com.br", response="0f4c122d2a0a28dea6194c235cd77430", algorithm=MD5 Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: Linphone/3.6.1 (eXosip2/4.0.0) Subject: Phone call Content-Length: 227 This is being changed to: INVITE sip:crdphmacl_SPnuV5xqtnSX at 131.221.240.204:60672;transport=tcp;line=c6356a7d87d6f81 SIP/2.0 Record-Route: Via: SIP/2.0/TCP myDomain.com.br:5060;branch=z9hG4bKe2db.49d54587.0;i=1 Via: SIP/2.0/TCP 192.168.21.40:5090;received=xxx.yyy.240.204;rport=60672;branch=z9hG4bK716249970 From: ;tag=12586028 To: Call-ID: 1106771604 CSeq: 21 INVITE Contact: Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: Linphone/3.6.1 (eXosip2/4.0.0) Subject: Phone call Content-Length: 224 So, the caller is receiving its own SIP INVITE. That is why when A calls B, is A that rings, not B. It is becoming a bit complicated. So, I suspect I'm going to the incorrect direction.... Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________ De: users-bounces at lists.opensips.org em nome de R?zvan Crainea Enviado: ter?a-feira, 21 de junho de 2016 04:24 Para: users at lists.opensips.org Assunto: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? Hi, Rodrigo! Have you tried restoring the R-URI after the caller lookup? Something like: $var(ru) = $ru; lookup("location", "", "$fu"); # this takes the caller from FROM uri, which I think is more suitable than from contact uri $ru = $var(ru); # continue your processing here .... # now do the real lookup for the callee lookup("location"); Don't do the lookups in the reversed way, because you might loose some contacts. Best regards, R?zvan Crainea OpenSIPS Solutions www.opensips-solutions.com Home ? OpenSIPS Solutions www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 06/20/2016 09:02 PM, Rodrigo Pimenta Carvalho wrote: Dear OpenSIPS-users, The table location has the column attr where I use to store specific additional information for each registration. Whenever A calls B, I have to read this specific information from the A record and from the B record. That is, I need to get and handle specific information about the caller and callee. For the callee, I use to invoke the lookup("location") function that put the needed information in the attr_avp. That is good and works very well. Then, I just have to read the attr_avp to get such specific information. For the caller, I use to invoke: $var(aorChamador) = $(ct.fields(uri)); lookup("location","","$var(aorChamador)"); However it causes amazing side effect in the SIP signaling. Ex: When A calls B, B stays quiet and A rings. So A can answer A. Crazy! According to the documentation, lookup will overwritten the Request-URI. I guess that is why the SIP signaling become incoherent. How could I get the caller attr specific information without side effects? Any hint will be very helpful!! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ________________________________ _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From tito at xsvoce.com Wed Jun 22 11:25:53 2016 From: tito at xsvoce.com (Tito Cumpen) Date: Wed, 22 Jun 2016 05:25:53 -0400 Subject: [OpenSIPS-Users] webrtc native client for opensips In-Reply-To: References: Message-ID: John, You can utilize sipjs and jssip on account that they utilize sip over websocket. Take into consideration that chrome will only allow getusermedia if you are using wss and https . On Jun 22, 2016 3:07 AM, "John Nash" wrote: > Apart from sipml5 is there any native webrtc client also which I can > explore to work with opensips? > > The examples I find for webrtc native seem to be using jingle protocol but > in case to make it work with opensips, It has to use SIP/SDP at client end > right?..Any examples? > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From john.nash778 at gmail.com Wed Jun 22 11:34:11 2016 From: john.nash778 at gmail.com (John Nash) Date: Wed, 22 Jun 2016 15:04:11 +0530 Subject: [OpenSIPS-Users] webrtc native client for opensips In-Reply-To: References: Message-ID: My objective is to make a native webrtc application which can use SIP over wss for signalling and for media also I do not want to be dependent on chrome as in future I wish to incorporate more codecs into it. Any pointers for me? On Wed, Jun 22, 2016 at 2:55 PM, Tito Cumpen wrote: > John, > > You can utilize sipjs and jssip on account that they utilize sip over > websocket. Take into consideration that chrome will only allow getusermedia > if you are using wss and https . > On Jun 22, 2016 3:07 AM, "John Nash" wrote: > >> Apart from sipml5 is there any native webrtc client also which I can >> explore to work with opensips? >> >> The examples I find for webrtc native seem to be using jingle protocol >> but in case to make it work with opensips, It has to use SIP/SDP at client >> end right?..Any examples? >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From tito at xsvoce.com Wed Jun 22 11:53:10 2016 From: tito at xsvoce.com (Tito Cumpen) Date: Wed, 22 Jun 2016 05:53:10 -0400 Subject: [OpenSIPS-Users] webrtc native client for opensips In-Reply-To: References: Message-ID: If you intend on running on android and iOS restcomm appears to have native clients that support sip over websocket . I've tested the iOS app they with OpenSIPS and baseline functionality was there . https://github.com/RestComm/restcomm-android-sdk On Jun 22, 2016 5:34 AM, "John Nash" wrote: > My objective is to make a native webrtc application which can use SIP over > wss for signalling and for media also I do not want to be dependent on > chrome as in future I wish to incorporate more codecs into it. > > Any pointers for me? > > On Wed, Jun 22, 2016 at 2:55 PM, Tito Cumpen wrote: > >> John, >> >> You can utilize sipjs and jssip on account that they utilize sip over >> websocket. Take into consideration that chrome will only allow getusermedia >> if you are using wss and https . >> On Jun 22, 2016 3:07 AM, "John Nash" wrote: >> >>> Apart from sipml5 is there any native webrtc client also which I can >>> explore to work with opensips? >>> >>> The examples I find for webrtc native seem to be using jingle protocol >>> but in case to make it work with opensips, It has to use SIP/SDP at client >>> end right?..Any examples? >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From john.nash778 at gmail.com Wed Jun 22 12:23:18 2016 From: john.nash778 at gmail.com (John Nash) Date: Wed, 22 Jun 2016 15:53:18 +0530 Subject: [OpenSIPS-Users] webrtc native client for opensips In-Reply-To: References: Message-ID: OK but in this i will not have control over webrtc codecs. What I am looking for is ... 1- Native application can send SIP messages over wss. 2- For media instead of using chrome functions , use my custom webrtc ( https://webrtc.org/native-code/ios/) which I use as library I am not even sure at this point if I am making sense. On Wed, Jun 22, 2016 at 3:23 PM, Tito Cumpen wrote: > If you intend on running on android and iOS restcomm appears to have > native clients that support sip over websocket . I've tested the iOS app > they with OpenSIPS and baseline functionality was there . > https://github.com/RestComm/restcomm-android-sdk > On Jun 22, 2016 5:34 AM, "John Nash" wrote: > >> My objective is to make a native webrtc application which can use SIP >> over wss for signalling and for media also I do not want to be dependent on >> chrome as in future I wish to incorporate more codecs into it. >> >> Any pointers for me? >> >> On Wed, Jun 22, 2016 at 2:55 PM, Tito Cumpen wrote: >> >>> John, >>> >>> You can utilize sipjs and jssip on account that they utilize sip over >>> websocket. Take into consideration that chrome will only allow getusermedia >>> if you are using wss and https . >>> On Jun 22, 2016 3:07 AM, "John Nash" wrote: >>> >>>> Apart from sipml5 is there any native webrtc client also which I can >>>> explore to work with opensips? >>>> >>>> The examples I find for webrtc native seem to be using jingle protocol >>>> but in case to make it work with opensips, It has to use SIP/SDP at client >>>> end right?..Any examples? >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From johan at democon.be Wed Jun 22 12:35:24 2016 From: johan at democon.be (johan de clercq) Date: Wed, 22 Jun 2016 12:35:24 +0200 Subject: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? In-Reply-To: <4B53E496-10F8-4824-B4A3-4BF57FEDC242@inin.com> References: <81fd240b-6369-15b4-a2ee-99deed334299@opensips.org> <4B53E496-10F8-4824-B4A3-4BF57FEDC242@inin.com> Message-ID: <019701d1cc71$ca9cf5f0$5fd6e1d0$@democon.be> Ben is correct. In my opinion, a very easy solution. From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Newlin, Ben Sent: Tuesday, June 21, 2016 5:24 PM To: sevpal ; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? It also seems like AVPOPS module [1] may be a good solution here as it has functions to pull data from a database into AVPs based by user. [1] http://www.opensips.org/html/docs/modules/2.2.x/avpops.html Ben Newlin From: > on behalf of sevpal > Reply-To: sevpal >, OpenSIPS users mailling list > Date: Tuesday, June 21, 2016 at 11:20 AM To: OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? Hi, have you tried/considered running a simple query on the database and parsing for the information you need? From: Rodrigo Pimenta Carvalho Sent: Tuesday, June 21, 2016 10:39 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? Hi R?zvan. I have tried that idea. But that didn't work. The SIP INVITE message is being changed by the OpenSIPS in a wrong way, in my point of view. Do you know some way to save the entire SIP INVITE message before calling lookup() and then make the saved message take place after the lookup() execution? My original message is: INVITE sip:6001 at myDomain.com.br SIP/2.0 Via: SIP/2.0/TCP 192.168.21.40:5090;rport;branch=z9hG4bK876727215 From: ;tag=179920819 To: Call-ID: 1410250893 CSeq: 21 INVITE Contact: Proxy-Authorization: Digest username="crdphmacl_SPnuV5xqtnSX", realm="localhost", nonce="5769458c000001cc263a7c0d6995dc48d42288ec6f8e4048", uri="sip:6001 at myDomain.com.br", response="0f4c122d2a0a28dea6194c235cd77430", algorithm=MD5 Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: Linphone/3.6.1 (eXosip2/4.0.0) Subject: Phone call Content-Length: 227 This is being changed to: INVITE sip:crdphmacl_SPnuV5xqtnSX at 131.221.240.204:60672;transport=tcp;line=c6356a7d87d6f81 SIP/2.0 Record-Route: Via: SIP/2.0/TCP myDomain.com.br:5060;branch=z9hG4bKe2db.49d54587.0;i=1 Via: SIP/2.0/TCP 192.168.21.40:5090;received=xxx.yyy.240.204;rport=60672;branch=z9hG4bK716249970 From: ;tag=12586028 To: Call-ID: 1106771604 CSeq: 21 INVITE Contact: Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: Linphone/3.6.1 (eXosip2/4.0.0) Subject: Phone call Content-Length: 224 So, the caller is receiving its own SIP INVITE. That is why when A calls B, is A that rings, not B. It is becoming a bit complicated. So, I suspect I'm going to the incorrect direction.... Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 _____ De: users-bounces at lists.opensips.org > em nome de R?zvan Crainea > Enviado: ter?a-feira, 21 de junho de 2016 04:24 Para: users at lists.opensips.org Assunto: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? Hi, Rodrigo! Have you tried restoring the R-URI after the caller lookup? Something like: $var(ru) = $ru; lookup("location", "", "$fu"); # this takes the caller from FROM uri, which I think is more suitable than from contact uri $ru = $var(ru); # continue your processing here .... # now do the real lookup for the callee lookup("location"); Don't do the lookups in the reversed way, because you might loose some contacts. Best regards, R?zvan Crainea OpenSIPS Solutions www.opensips-solutions.com Home ? OpenSIPS Solutions www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 06/20/2016 09:02 PM, Rodrigo Pimenta Carvalho wrote: Dear OpenSIPS-users, The table location has the column attr where I use to store specific additional information for each registration. Whenever A calls B, I have to read this specific information from the A record and from the B record. That is, I need to get and handle specific information about the caller and callee. For the callee, I use to invoke the lookup("location") function that put the needed information in the attr_avp. That is good and works very well. Then, I just have to read the attr_avp to get such specific information. For the caller, I use to invoke: $var(aorChamador) = $(ct.fields(uri)); lookup("location","","$var(aorChamador)"); However it causes amazing side effect in the SIP signaling. Ex: When A calls B, B stays quiet and A rings. So A can answer A. Crazy! According to the documentation, lookup will overwritten the Request-URI. I guess that is why the SIP signaling become incoherent. How could I get the caller attr specific information without side effects? Any hint will be very helpful!! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _____ _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From pimenta at inatel.br Wed Jun 22 14:16:40 2016 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Wed, 22 Jun 2016 12:16:40 +0000 Subject: [OpenSIPS-Users] Leak AVPOS + SQLITE In-Reply-To: References: , Message-ID: Ok. Thank you very much! Regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________ De: users-bounces at lists.opensips.org em nome de Razvan Crainea Enviado: quarta-feira, 22 de junho de 2016 04:04 Para: users at lists.opensips.org Assunto: Re: [OpenSIPS-Users] Leak AVPOS + SQLITE Hi, Rodrigo! Valgrind may report some memory allocated, and not freed, but that is not necessarily a memory leak. There is a single block of 1024 bytes not freed during runtime, so I think that is peanuts. The memory used by OpenSIPS is not allocated with malloc, so cannot be traced by valgrind. Regarding the system memory, it is normal to decrease as OpenSIPS uses that memory during runtime. However, after some time, this should stabilize. Anyhow, sometimes the system memory might generate false alarms, so if you are tracing any memory leaks, you should check OpenSIPS's internal statistics. Best regards, Razvan Crainea OpenSIPS Solutions www.opensips-solutions.com Home - OpenSIPS Solutions www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 06/21/2016 10:38 PM, Rodrigo Pimenta Carvalho wrote: Hi. Does someone here is getting/handling memory leaks with OpenSIPS 2.2 and last version of SQLite? I'm using newest commit from OpenSIPS 2.2 and newest version of SQLite. My query is : avp_db_query("select Value from GeneralConfigurations where Attribute = 'CONFIGURATION_INTERCOM_A_NAME'"); Valgrind shows: ==16087== ERROR SUMMARY: 0 errors from 0 contexts (suppressed: 2 from 2) ==16088== Searching for pointers to 296,489 not-freed blocks ==16088== Checked 103,297,688 bytes ==16088== ==16088== 1,024 bytes in 1 blocks are possibly lost in loss record 184 of 246 ==16088== at 0x4C2745D: malloc (in /usr/lib64/valgrind/vgpreload_memcheck-amd64-linux.so) ==16088== by 0x8F8B05F: sqlite3MemMalloc (sqlite3.c:20167) ==16088== by 0x8F701C7: sqlite3Malloc (sqlite3.c:23846) ==16088== by 0x8F75459: pcache1Alloc (sqlite3.c:44312) ==16088== by 0x8F8019F: sqlite3BtreeCursor (sqlite3.c:44455) ==16088== by 0x8FD0FDD: sqlite3VdbeExec (sqlite3.c:80098) ==16088== by 0x8FDB89F: sqlite3_step (sqlite3.c:75131) ==16088== by 0x8FDC9A1: sqlite3_exec (sqlite3.c:108122) ==16088== by 0x8D20736: db_sqlite_raw_query (dbase.c:358) ==16088== by 0x9464DB8: db_query_avp (avpops_db.c:436) ==16088== by 0x946943E: ops_dbquery_avps (avpops_impl.c:840) ==16088== by 0x9459A61: w_dbquery_avps (avpops.c:1395) ==16088== ==16088== LEAK SUMMARY: ==16088== definitely lost: 0 bytes in 0 blocks ==16088== indirectly lost: 0 bytes in 0 blocks ==16088== possibly lost: 1,024 bytes in 1 blocks ==16088== still reachable: 47,457,573 bytes in 296,488 blocks ==16088== suppressed: 0 bytes in 0 blocks ==16088== Reachable blocks (those to which a pointer was found) are not shown. ==16088== To see them, rerun with: --leak-check=full --show-leak-kinds=all After some time running that query, I can see, via command top, that the available memory is decreasing. In fact, the memory is not freed even after stop running the query for a time. Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From pimenta at inatel.br Wed Jun 22 14:22:21 2016 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Wed, 22 Jun 2016 12:22:21 +0000 Subject: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? In-Reply-To: References: <81fd240b-6369-15b4-a2ee-99deed334299@opensips.org> , Message-ID: Hi R?zvan Crainea. Thank you very much for trying to help me. Yesterday my boss asked me to work in another part of our project. So, I will have to pause this verification for a while. When I return to it, I will check the log. Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________ De: users-bounces at lists.opensips.org em nome de R?zvan Crainea Enviado: quarta-feira, 22 de junho de 2016 03:57 Para: users at lists.opensips.org Assunto: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? Hi, Rodrigo! Can you print the $ru variable before and after each lookup() query? Something like: $var(ru) = $ru; xlog("R-URI before caller lookup: $ru\n"); lookup("location", "", "$fu"); # this takes the caller from FROM uri, which I think is more suitable than from contact uri $ru = $var(ru); # continue your processing here xlog("R-URI after caller lookup: $ru\n"); ... # now do the real lookup for the callee xlog("R-URI before callee lookup: $ru\n"); lookup("location"); xlog("R-URI after callee lookup: $ru\n"); Make sure they are all correct, or if they are not, send me these logs. Thanks, R?zvan Crainea OpenSIPS Solutions www.opensips-solutions.com Home ? OpenSIPS Solutions www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 06/21/2016 07:39 PM, Rodrigo Pimenta Carvalho wrote: Hi Sevpal. Yes. That is what I was doing. It worked very well. But, nowadays I'm using db_mode = 0 for usrloc. So, the information is always only in RAM. In this case, the query will return no result. That is why I'm trying to read the attr column from table location, from RAM, and get specific information for the caller. For the callee, everything is all right. Regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________ De: users-bounces at lists.opensips.org em nome de sevpal Enviado: ter?a-feira, 21 de junho de 2016 12:20 Para: OpenSIPS users mailling list Assunto: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? Hi, have you tried/considered running a simple query on the database and parsing for the information you need? From: Rodrigo Pimenta Carvalho Sent: Tuesday, June 21, 2016 10:39 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? Hi R?zvan. I have tried that idea. But that didn't work. The SIP INVITE message is being changed by the OpenSIPS in a wrong way, in my point of view. Do you know some way to save the entire SIP INVITE message before calling lookup() and then make the saved message take place after the lookup() execution? My original message is: INVITE sip:6001 at myDomain.com.br SIP/2.0 Via: SIP/2.0/TCP 192.168.21.40:5090;rport;branch=z9hG4bK876727215 From: ;tag=179920819 To: Call-ID: 1410250893 CSeq: 21 INVITE Contact: Proxy-Authorization: Digest username="crdphmacl_SPnuV5xqtnSX", realm="localhost", nonce="5769458c000001cc263a7c0d6995dc48d42288ec6f8e4048", uri="sip:6001 at myDomain.com.br", response="0f4c122d2a0a28dea6194c235cd77430", algorithm=MD5 Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: Linphone/3.6.1 (eXosip2/4.0.0) Subject: Phone call Content-Length: 227 This is being changed to: INVITE sip:crdphmacl_SPnuV5xqtnSX at 131.221.240.204:60672;transport=tcp;line=c6356a7d87d6f81 SIP/2.0 Record-Route: Via: SIP/2.0/TCP myDomain.com.br:5060;branch=z9hG4bKe2db.49d54587.0;i=1 Via: SIP/2.0/TCP 192.168.21.40:5090;received=xxx.yyy.240.204;rport=60672;branch=z9hG4bK716249970 From: ;tag=12586028 To: Call-ID: 1106771604 CSeq: 21 INVITE Contact: Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: Linphone/3.6.1 (eXosip2/4.0.0) Subject: Phone call Content-Length: 224 So, the caller is receiving its own SIP INVITE. That is why when A calls B, is A that rings, not B. It is becoming a bit complicated. So, I suspect I'm going to the incorrect direction.... Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________ De: users-bounces at lists.opensips.org em nome de R?zvan Crainea Enviado: ter?a-feira, 21 de junho de 2016 04:24 Para: users at lists.opensips.org Assunto: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? Hi, Rodrigo! Have you tried restoring the R-URI after the caller lookup? Something like: $var(ru) = $ru; lookup("location", "", "$fu"); # this takes the caller from FROM uri, which I think is more suitable than from contact uri $ru = $var(ru); # continue your processing here .... # now do the real lookup for the callee lookup("location"); Don't do the lookups in the reversed way, because you might loose some contacts. Best regards, R?zvan Crainea OpenSIPS Solutions www.opensips-solutions.com Home ? OpenSIPS Solutions www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 06/20/2016 09:02 PM, Rodrigo Pimenta Carvalho wrote: Dear OpenSIPS-users, The table location has the column attr where I use to store specific additional information for each registration. Whenever A calls B, I have to read this specific information from the A record and from the B record. That is, I need to get and handle specific information about the caller and callee. For the callee, I use to invoke the lookup("location") function that put the needed information in the attr_avp. That is good and works very well. Then, I just have to read the attr_avp to get such specific information. For the caller, I use to invoke: $var(aorChamador) = $(ct.fields(uri)); lookup("location","","$var(aorChamador)"); However it causes amazing side effect in the SIP signaling. Ex: When A calls B, B stays quiet and A rings. So A can answer A. Crazy! According to the documentation, lookup will overwritten the Request-URI. I guess that is why the SIP signaling become incoherent. How could I get the caller attr specific information without side effects? Any hint will be very helpful!! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ________________________________ _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Jun 22 17:31:37 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 22 Jun 2016 18:31:37 +0300 Subject: [OpenSIPS-Users] DNS-SRV query in opensips In-Reply-To: <45ada32d906e44559fb82fe37d6cb1c3@COPDCEX28.cable.comcast.com> References: <955ec37716d94d139c82be2835377bd5@COPDCEX28.cable.comcast.com> <5768EF90.2030802@opensips.org> <576A3B05.8030702@opensips.org> <45ada32d906e44559fb82fe37d6cb1c3@COPDCEX28.cable.comcast.com> Message-ID: <576AAF59.8090305@opensips.org> Hi, They are different. RURI is the part in request's first line, while the DURI is a an outbound proxy used just to finding the destination at network level (it will not be present in the SIP request). by using the set functions you do not create a new branch, you are just changing the RURI and DURI from the default branch. To create a new branch you have to use append_branch(). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 22.06.2016 17:13, Ramachandran, Agalya (Contractor) wrote: > > Hi Bogdan, > > Request URI and destination URI are one and the same right? > > Basically what I understand from the below mail is , we can use both > seturi and setdsturi to set a new branch. > > Am I right? > > > Regards, > Agalya > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* Wednesday, June 22, 2016 3:15 AM > *To:* Ramachandran, Agalya (Contractor) > *Cc:* OpenSIPS users mailling list > *Subject:* Re: [OpenSIPS-Users] DNS-SRV query in opensips > > Hi Agalya, > > seturi() and setdsturi() set the RURI / DestinationURI for the current > message / branch. > > When you do an append_branch() a new branch is stored for > serial/parallel forking (note that the current branch does not changes > - this is branch number 1). > > So, append_branch() will make a copy of the current branch (RURI, > DURI, PATH, Forced Socket, etc) and store a new branch for later forking. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 21.06.2016 21:08, Ramachandran, Agalya (Contractor) wrote: > > Hi Bogdan, > > I have a question regarding /seturi/and /setdsturi///function calls. > > As far as my understanding, when append_branch() is called, seturi > () is called to set the URI where to fork the call. > > I tried by calling only /seturi ()/function call, after > append_branch it was working same behavior as when I used > /setdsturi()/as well. > > My question is do we really need /setdsturi /or when is the case > when /setdsturi()/is used.? > > Regards, > Agalya > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* Tuesday, June 21, 2016 3:41 AM > *To:* OpenSIPS users mailling list >; Ramachandran, Agalya > (Contractor) >; Ramachandran, Agalya > (Contractor) > > *Subject:* Re: [OpenSIPS-Users] DNS-SRV query in opensips > > Hi Agalya, > > OpenSIPS does full flavor DNS lookup (with NATPR and SRV), but > this is internal, and not accessible from script. OpenSIPS > implements auto DNS-based failover : > http://www.opensips.org/html/docs/modules/2.2.x/tm.html#id293694 > > My understanding is you want the DNS resolving to be done at > script level and to have access to the results ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > On 20.06.2016 20:35, Ramachandran, Agalya (Contractor) wrote: > > Hi team, > > We are using opensips for our project requirements. > > I have a scenario where we need DNS-SRV query and the result > of this should be placed as the desturi to send request > out.(In case of forking call) > > As far as I went through opensips documentation, there are > some core parameters for dns related config such as > ?dns_retr_time > ? > ,?dns_retr_time > ? ,?dns_servers_no > ? > etc? > > According to my understanding these config variables can be > declared and used in the opensips.config file to control the > settings of DNS query. > > Is there any available function where I can use and pass the > DNS server domain name, so that it fetches the IP address of > the host ? > > Please let us know what is the best way to achieve this? > > Regards, > > Agalya > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From govoiper at gmail.com Wed Jun 22 18:07:23 2016 From: govoiper at gmail.com (SamyGo) Date: Wed, 22 Jun 2016 12:07:23 -0400 Subject: [OpenSIPS-Users] Async event_route and cachdb_redis In-Reply-To: References: <87e0609d-975d-e80f-50c7-d34c7989f15d@opensips.org> Message-ID: Yeah it only happens at startup. If I start opensips in debug_mode=yes then the error prints for infinite time. With your patch; putting "async" doesn't even call the event route. If I remove async attribute then it works just like before the patch. Regards, Sammy On Wed, Jun 22, 2016 at 3:10 AM, R?zvan Crainea wrote: > Hi, Sammy! > > Does this happen only at startime, or happens during runtime too? > > Regards, > > R?zvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 06/21/2016 10:57 PM, SamyGo wrote: > > Hi , > > After recompiling , when I start opensips it gives this error: > > ERROR:event_route:event_route_handler: invalid receive sock info > > The two event routes I have are these: > > event_route[E_UL_CONTACT_INSERT,async] { > > fetch_event_params("aor=$avp(aor);address=$avp(address);received=$avp(received)"); > .... > cache_raw_query("redis:group1","HSET GLOBAL_USER_LOCATION > $avp(aor) $var(my_value1)"); > > } > > event_route[E_UL_AOR_DELETE,async] { > fetch_event_params("aor=$avp(aor)"); > ... > cache_raw_query("redis:group1","DEL GLOBAL_USER_LOCATION > $avp(aor)"); > > } > > > Some Xlog lines in both of these routes, nothing seems to be printed now, > no error , no cache data modifications executing.. > > I'll see in further detail what is happening and if I find anything > abnormal will reply. > > > Regards. > Sammy > > > > > On Tue, Jun 21, 2016 at 3:40 AM, R?zvan Crainea > wrote: > >> Hi, Sammy! >> >> Could you try this patch: >> >> https://gist.github.com/razvancrainea/9d239c82474bb0f1c403b6459dbdb647 >> >> Thanks, >> >> R?zvan Crainea >> OpenSIPS Solutionswww.opensips-solutions.com >> >> On 06/19/2016 08:56 PM, SamyGo wrote: >> >> Hi, >> I'm seeing errors from cachedb_redis module when called in an event route >> in async mode. >> >> event_route[E_UL_CONTACT_INSERT,async] { >> ... >> cache_raw_query("redis:group1","SET ABC"); >> .. >> >> } >> >> OpenSIPS throws error stating that redis group1 unavailable >> >> DBG:core:cachedb_raw_query: from script [redis] - with grp [group1] >> ERROR:core:cachedb_raw_query: failed to get connection for grp name >> [group1] >> >> I tried same command in main route of reply route, all works normal. if I >> remove the "async" from the event_route definition it works in event route. >> >> Any logical reason why async route don't recognize the connections ? >> >> Tried with OpenSIPS 2.2 and 2.1 as well, same behavior. >> >> >> Regards, >> Sammy >> >> >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Wed Jun 22 18:36:55 2016 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Wed, 22 Jun 2016 19:36:55 +0300 Subject: [OpenSIPS-Users] Async event_route and cachdb_redis In-Reply-To: References: <87e0609d-975d-e80f-50c7-d34c7989f15d@opensips.org> Message-ID: <951907b6-949e-60d6-3670-e6b3078e643f@opensips.org> So the patch doesn't do anything but stops triggering the event? Regards, R?zvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 06/22/2016 07:07 PM, SamyGo wrote: > Yeah it only happens at startup. If I start opensips in debug_mode=yes > then the error prints for infinite time. > > With your patch; putting "async" doesn't even call the event route. If > I remove async attribute then it works just like before the patch. > > Regards, > Sammy > > > On Wed, Jun 22, 2016 at 3:10 AM, R?zvan Crainea > wrote: > > Hi, Sammy! > > Does this happen only at startime, or happens during runtime too? > > Regards, > > R?zvan Crainea > OpenSIPS Solutions > www.opensips-solutions.com > > On 06/21/2016 10:57 PM, SamyGo wrote: >> Hi , >> >> After recompiling , when I start opensips it gives this error: >> >> ERROR:event_route:event_route_handler: invalid receive sock info >> >> The two event routes I have are these: >> >> event_route[E_UL_CONTACT_INSERT,async] { >> fetch_event_params("aor=$avp(aor);address=$avp(address);received=$avp(received)"); >> .... >> cache_raw_query("redis:group1","HSET GLOBAL_USER_LOCATION >> $avp(aor) $var(my_value1)"); >> >> } >> >> event_route[E_UL_AOR_DELETE,async] { >> fetch_event_params("aor=$avp(aor)"); >> ... >> cache_raw_query("redis:group1","DEL GLOBAL_USER_LOCATION >> $avp(aor)"); >> >> } >> >> >> Some Xlog lines in both of these routes, nothing seems to be >> printed now, no error , no cache data modifications executing.. >> >> I'll see in further detail what is happening and if I find >> anything abnormal will reply. >> >> >> Regards. >> Sammy >> >> >> >> >> On Tue, Jun 21, 2016 at 3:40 AM, R?zvan Crainea >> > wrote: >> >> Hi, Sammy! >> >> Could you try this patch: >> >> https://gist.github.com/razvancrainea/9d239c82474bb0f1c403b6459dbdb647 >> >> Thanks, >> >> R?zvan Crainea >> OpenSIPS Solutions >> www.opensips-solutions.com >> >> On 06/19/2016 08:56 PM, SamyGo wrote: >>> Hi, >>> I'm seeing errors from cachedb_redis module when called in >>> an event route in async mode. >>> >>> event_route[E_UL_CONTACT_INSERT,async] { >>> ... >>> cache_raw_query("redis:group1","SET ABC"); >>> .. >>> >>> } >>> >>> OpenSIPS throws error stating that redis group1 unavailable >>> >>> DBG:core:cachedb_raw_query: from script [redis] - with grp >>> [group1] >>> ERROR:core:cachedb_raw_query: failed to get connection for >>> grp name [group1] >>> >>> I tried same command in main route of reply route, all works >>> normal. if I remove the "async" from the event_route >>> definition it works in event route. >>> >>> Any logical reason why async route don't recognize the >>> connections ? >>> >>> Tried with OpenSIPS 2.2 and 2.1 as well, same behavior. >>> >>> >>> Regards, >>> Sammy >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From govoiper at gmail.com Wed Jun 22 18:39:58 2016 From: govoiper at gmail.com (SamyGo) Date: Wed, 22 Jun 2016 12:39:58 -0400 Subject: [OpenSIPS-Users] Async event_route and cachdb_redis In-Reply-To: <951907b6-949e-60d6-3670-e6b3078e643f@opensips.org> References: <87e0609d-975d-e80f-50c7-d34c7989f15d@opensips.org> <951907b6-949e-60d6-3670-e6b3078e643f@opensips.org> Message-ID: Yes correct. Async event route even stops to be executed. On Jun 22, 2016 12:37, "R?zvan Crainea" wrote: > So the patch doesn't do anything but stops triggering the event? > > Regards, > > R?zvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 06/22/2016 07:07 PM, SamyGo wrote: > > Yeah it only happens at startup. If I start opensips in debug_mode=yes > then the error prints for infinite time. > > With your patch; putting "async" doesn't even call the event route. If I > remove async attribute then it works just like before the patch. > > Regards, > Sammy > > > On Wed, Jun 22, 2016 at 3:10 AM, R?zvan Crainea > wrote: > >> Hi, Sammy! >> >> Does this happen only at startime, or happens during runtime too? >> >> Regards, >> >> R?zvan Crainea >> OpenSIPS Solutionswww.opensips-solutions.com >> >> On 06/21/2016 10:57 PM, SamyGo wrote: >> >> Hi , >> >> After recompiling , when I start opensips it gives this error: >> >> ERROR:event_route:event_route_handler: invalid receive sock info >> >> The two event routes I have are these: >> >> event_route[E_UL_CONTACT_INSERT,async] { >> >> fetch_event_params("aor=$avp(aor);address=$avp(address);received=$avp(received)"); >> .... >> cache_raw_query("redis:group1","HSET >> GLOBAL_USER_LOCATION $avp(aor) $var(my_value1)"); >> >> } >> >> event_route[E_UL_AOR_DELETE,async] { >> fetch_event_params("aor=$avp(aor)"); >> ... >> cache_raw_query("redis:group1","DEL GLOBAL_USER_LOCATION >> $avp(aor)"); >> >> } >> >> >> Some Xlog lines in both of these routes, nothing seems to be printed now, >> no error , no cache data modifications executing.. >> >> I'll see in further detail what is happening and if I find anything >> abnormal will reply. >> >> >> Regards. >> Sammy >> >> >> >> >> On Tue, Jun 21, 2016 at 3:40 AM, R?zvan Crainea >> wrote: >> >>> Hi, Sammy! >>> >>> Could you try this patch: >>> >>> https://gist.github.com/razvancrainea/9d239c82474bb0f1c403b6459dbdb647 >>> >>> Thanks, >>> >>> R?zvan Crainea >>> OpenSIPS Solutionswww.opensips-solutions.com >>> >>> On 06/19/2016 08:56 PM, SamyGo wrote: >>> >>> Hi, >>> I'm seeing errors from cachedb_redis module when called in an event >>> route in async mode. >>> >>> event_route[E_UL_CONTACT_INSERT,async] { >>> ... >>> cache_raw_query("redis:group1","SET ABC"); >>> .. >>> >>> } >>> >>> OpenSIPS throws error stating that redis group1 unavailable >>> >>> DBG:core:cachedb_raw_query: from script [redis] - with grp [group1] >>> ERROR:core:cachedb_raw_query: failed to get connection for grp name >>> [group1] >>> >>> I tried same command in main route of reply route, all works normal. if >>> I remove the "async" from the event_route definition it works in event >>> route. >>> >>> Any logical reason why async route don't recognize the connections ? >>> >>> Tried with OpenSIPS 2.2 and 2.1 as well, same behavior. >>> >>> >>> Regards, >>> Sammy >>> >>> >>> >>> _______________________________________________ >>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Wed Jun 22 18:48:00 2016 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Wed, 22 Jun 2016 19:48:00 +0300 Subject: [OpenSIPS-Users] Async event_route and cachdb_redis In-Reply-To: References: <87e0609d-975d-e80f-50c7-d34c7989f15d@opensips.org> <951907b6-949e-60d6-3670-e6b3078e643f@opensips.org> Message-ID: Could you run the 'opensipsctl trap' command and paste the output on pastebin. Best regards, R?zvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 06/22/2016 07:39 PM, SamyGo wrote: > > Yes correct. Async event route even stops to be executed. > > On Jun 22, 2016 12:37, "R?zvan Crainea" > wrote: > > So the patch doesn't do anything but stops triggering the event? > > Regards, > > R?zvan Crainea > OpenSIPS Solutions > www.opensips-solutions.com > > On 06/22/2016 07:07 PM, SamyGo wrote: >> Yeah it only happens at startup. If I start opensips in >> debug_mode=yes then the error prints for infinite time. >> >> With your patch; putting "async" doesn't even call the event >> route. If I remove async attribute then it works just like before >> the patch. >> >> Regards, >> Sammy >> >> >> On Wed, Jun 22, 2016 at 3:10 AM, R?zvan Crainea >> > wrote: >> >> Hi, Sammy! >> >> Does this happen only at startime, or happens during runtime too? >> >> Regards, >> >> R?zvan Crainea >> OpenSIPS Solutions >> www.opensips-solutions.com >> >> On 06/21/2016 10:57 PM, SamyGo wrote: >>> Hi , >>> >>> After recompiling , when I start opensips it gives this error: >>> >>> ERROR:event_route:event_route_handler: invalid receive sock info >>> >>> The two event routes I have are these: >>> >>> event_route[E_UL_CONTACT_INSERT,async] { >>> fetch_event_params("aor=$avp(aor);address=$avp(address);received=$avp(received)"); >>> .... >>> cache_raw_query("redis:group1","HSET GLOBAL_USER_LOCATION >>> $avp(aor) $var(my_value1)"); >>> >>> } >>> >>> event_route[E_UL_AOR_DELETE,async] { >>> fetch_event_params("aor=$avp(aor)"); >>> ... >>> cache_raw_query("redis:group1","DEL GLOBAL_USER_LOCATION >>> $avp(aor)"); >>> >>> } >>> >>> >>> Some Xlog lines in both of these routes, nothing seems to be >>> printed now, no error , no cache data modifications executing.. >>> >>> I'll see in further detail what is happening and if I find >>> anything abnormal will reply. >>> >>> >>> Regards. >>> Sammy >>> >>> >>> >>> >>> On Tue, Jun 21, 2016 at 3:40 AM, R?zvan Crainea >>> > wrote: >>> >>> Hi, Sammy! >>> >>> Could you try this patch: >>> >>> https://gist.github.com/razvancrainea/9d239c82474bb0f1c403b6459dbdb647 >>> >>> Thanks, >>> >>> R?zvan Crainea >>> OpenSIPS Solutions >>> www.opensips-solutions.com >>> >>> >>> On 06/19/2016 08:56 PM, SamyGo wrote: >>>> Hi, >>>> I'm seeing errors from cachedb_redis module when called >>>> in an event route in async mode. >>>> >>>> event_route[E_UL_CONTACT_INSERT,async] { >>>> ... >>>> cache_raw_query("redis:group1","SET ABC"); >>>> .. >>>> >>>> } >>>> >>>> OpenSIPS throws error stating that redis group1 unavailable >>>> >>>> DBG:core:cachedb_raw_query: from script [redis] - with >>>> grp [group1] >>>> ERROR:core:cachedb_raw_query: failed to get connection >>>> for grp name [group1] >>>> >>>> I tried same command in main route of reply route, all >>>> works normal. if I remove the "async" from the >>>> event_route definition it works in event route. >>>> >>>> Any logical reason why async route don't recognize the >>>> connections ? >>>> >>>> Tried with OpenSIPS 2.2 and 2.1 as well, same behavior. >>>> >>>> >>>> Regards, >>>> Sammy >>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From Agalya_Ramachandran at comcast.com Wed Jun 22 19:19:00 2016 From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor)) Date: Wed, 22 Jun 2016 17:19:00 +0000 Subject: [OpenSIPS-Users] CURL library with respect to REST_API calls Message-ID: <1dc72c09449b413a84275d01bcff3548@COPDCEX28.cable.comcast.com> Hi team, Any one has any clue on the below topic? Regards, Agalya From: Ramachandran, Agalya (Contractor) Sent: Monday, June 20, 2016 1:45 PM To: OpenSIPS users mailling list Subject: CURL library with respect to REST_API calls Hi team, I have a question regarding curl library behavior with respect to curl_easy_perform API call. Here is the snippet of the code that am using in "rest_put" API call in rest_methods.c file w_curl_easy_setopt(handle, CURLOPT_WRITEFUNCTION, write_func); w_curl_easy_setopt(handle, CURLOPT_WRITEDATA, &res_body); When curl_easy_perform API call is success, I could able to retrieve the result body from the res_body. But in the case of API call failure am not getting any details of the message. But getting only the http response code. Is there a way to get the message details as well in the case where curl_easy_perform API fails? LM_INFO(" Actual result body is %s\n", res_body.s); When I print this, in the case of success, am getting a http response message in detail. But in case of failure, the call back function write_func is not at all called. Why it is so? Please guide me if there is a way to the message details in case of failure too. Regards, Agalya -------------- next part -------------- An HTML attachment was scrubbed... URL: From govoiper at gmail.com Wed Jun 22 20:45:56 2016 From: govoiper at gmail.com (SamyGo) Date: Wed, 22 Jun 2016 14:45:56 -0400 Subject: [OpenSIPS-Users] Async event_route and cachdb_redis In-Reply-To: References: <87e0609d-975d-e80f-50c7-d34c7989f15d@opensips.org> <951907b6-949e-60d6-3670-e6b3078e643f@opensips.org> Message-ID: Thanks Razvan for dedicating time for me. You can find the output from the given command here: http://pastebin.com/fh11mkXS On Wed, Jun 22, 2016 at 12:48 PM, R?zvan Crainea wrote: > Could you run the 'opensipsctl trap' command and paste the output on > pastebin. > > Best regards, > > R?zvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 06/22/2016 07:39 PM, SamyGo wrote: > > Yes correct. Async event route even stops to be executed. > On Jun 22, 2016 12:37, "R?zvan Crainea" wrote: > >> So the patch doesn't do anything but stops triggering the event? >> >> Regards, >> >> R?zvan Crainea >> OpenSIPS Solutionswww.opensips-solutions.com >> >> On 06/22/2016 07:07 PM, SamyGo wrote: >> >> Yeah it only happens at startup. If I start opensips in debug_mode=yes >> then the error prints for infinite time. >> >> With your patch; putting "async" doesn't even call the event route. If I >> remove async attribute then it works just like before the patch. >> >> Regards, >> Sammy >> >> >> On Wed, Jun 22, 2016 at 3:10 AM, R?zvan Crainea < >> razvan at opensips.org> wrote: >> >>> Hi, Sammy! >>> >>> Does this happen only at startime, or happens during runtime too? >>> >>> Regards, >>> >>> R?zvan Crainea >>> OpenSIPS Solutionswww.opensips-solutions.com >>> >>> On 06/21/2016 10:57 PM, SamyGo wrote: >>> >>> Hi , >>> >>> After recompiling , when I start opensips it gives this error: >>> >>> ERROR:event_route:event_route_handler: invalid receive sock info >>> >>> The two event routes I have are these: >>> >>> event_route[E_UL_CONTACT_INSERT,async] { >>> >>> fetch_event_params("aor=$avp(aor);address=$avp(address);received=$avp(received)"); >>> .... >>> cache_raw_query("redis:group1","HSET >>> GLOBAL_USER_LOCATION $avp(aor) $var(my_value1)"); >>> >>> } >>> >>> event_route[E_UL_AOR_DELETE,async] { >>> fetch_event_params("aor=$avp(aor)"); >>> ... >>> cache_raw_query("redis:group1","DEL GLOBAL_USER_LOCATION >>> $avp(aor)"); >>> >>> } >>> >>> >>> Some Xlog lines in both of these routes, nothing seems to be printed >>> now, no error , no cache data modifications executing.. >>> >>> I'll see in further detail what is happening and if I find anything >>> abnormal will reply. >>> >>> >>> Regards. >>> Sammy >>> >>> >>> >>> >>> On Tue, Jun 21, 2016 at 3:40 AM, R?zvan Crainea < >>> razvan at opensips.org> wrote: >>> >>>> Hi, Sammy! >>>> >>>> Could you try this patch: >>>> >>>> https://gist.github.com/razvancrainea/9d239c82474bb0f1c403b6459dbdb647 >>>> >>>> Thanks, >>>> >>>> R?zvan Crainea >>>> OpenSIPS Solutionswww.opensips-solutions.com >>>> >>>> On 06/19/2016 08:56 PM, SamyGo wrote: >>>> >>>> Hi, >>>> I'm seeing errors from cachedb_redis module when called in an event >>>> route in async mode. >>>> >>>> event_route[E_UL_CONTACT_INSERT,async] { >>>> ... >>>> cache_raw_query("redis:group1","SET ABC"); >>>> .. >>>> >>>> } >>>> >>>> OpenSIPS throws error stating that redis group1 unavailable >>>> >>>> DBG:core:cachedb_raw_query: from script [redis] - with grp [group1] >>>> ERROR:core:cachedb_raw_query: failed to get connection for grp name >>>> [group1] >>>> >>>> I tried same command in main route of reply route, all works normal. if >>>> I remove the "async" from the event_route definition it works in event >>>> route. >>>> >>>> Any logical reason why async route don't recognize the connections ? >>>> >>>> Tried with OpenSIPS 2.2 and 2.1 as well, same behavior. >>>> >>>> >>>> Regards, >>>> Sammy >>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>> >>> >>> _______________________________________________ >>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Thu Jun 23 11:51:31 2016 From: liviu at opensips.org (Liviu Chircu) Date: Thu, 23 Jun 2016 12:51:31 +0300 Subject: [OpenSIPS-Users] CURL library with respect to REST_API calls In-Reply-To: <1dc72c09449b413a84275d01bcff3548@COPDCEX28.cable.comcast.com> References: <1dc72c09449b413a84275d01bcff3548@COPDCEX28.cable.comcast.com> Message-ID: <576BB123.7020102@opensips.org> Hi, Ramachandran! I read the "CURLOPT_WRITEFUNCTION" section [1] one more time, and it really looks like it should pass the body of the reply to the "write_func" callback we register before sending the HTTP PUT, even if we got an error code (3XX or higher). Did you change anything in the "write_func()"? Notice how their docs say that if a proper "len" is not returned, the transfer will be aborted. Apart from that, I have no other ideas for now but to try and fetch the body myself on an error HTTP ret code of a CUROPT_PUT operation, see how (or if) it works for me, and give you more feedback. [1]: https://curl.haxx.se/libcurl/c/CURLOPT_WRITEFUNCTION.html All the best, Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 22.06.2016 20:19, Ramachandran, Agalya (Contractor) wrote: > > Hi team, > > Any one has any clue on the below topic? > > Regards, > Agalya > > *From:* Ramachandran, Agalya (Contractor) > *Sent:* Monday, June 20, 2016 1:45 PM > *To:* OpenSIPS users mailling list > *Subject:* CURL library with respect to REST_API calls > > Hi team, > > I have a question regarding curl library behavior with respect to > curl_easy_perform API call. > > Here is the snippet of the code that am using in ?rest_put? API call > in rest_methods.c file > > /w_curl_easy_setopt(handle, CURLOPT_WRITEFUNCTION, write_func);/ > > / w_curl_easy_setopt(handle, CURLOPT_WRITEDATA, &res_body);/ > > // > > When curl_easy_perform API call is success, I could able to retrieve > the result body from the *res_body*. > > But in the case of API call failure am not getting any details of the > message. But getting only the http response code. > > Is there a way to get the message details as well in the case where > curl_easy_perform API fails? > > LM_INFO(" Actual result body is %s\n", res_body.s); > > When I print this, in the case of success, am getting a http response > message in detail. > > But in case of failure, the call back function /write_func /is not at > all called. Why it is so? > > Please guide me if there is a way to the message details in case of > failure too. > > Regards, > Agalya > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From john.nash778 at gmail.com Thu Jun 23 16:20:08 2016 From: john.nash778 at gmail.com (John Nash) Date: Thu, 23 Jun 2016 19:50:08 +0530 Subject: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc Message-ID: I am following http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 and trying to test a call sipml5 ----------->Opensips + rtpengine --------> SIP end point (Freeswitch) But I do not have any audio on both sides. I see this error at rtpengine log "SRTP output wanted, but no crypto suite was negotiated" Anyone tested this scenario positive? -------------- next part -------------- An HTML attachment was scrubbed... URL: From eric at uphreak.com Thu Jun 23 16:28:27 2016 From: eric at uphreak.com (Eric Tamme) Date: Thu, 23 Jun 2016 08:28:27 -0600 Subject: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc In-Reply-To: References: Message-ID: <576BF20B.3060301@uphreak.com> 1. I would suggest using SIP.js - https://github.com/onsip/SIP.js it is a much more active project that sipml5. 2. Im guessing that you are not properly passing flags to RTPEngine. If you want to have DTLS-SRTP between the browser, and plain RTP/AVP between RTPEngine and freeswitch, you need to "offer" rtp/avp to freeswitch, and "answer" dtls-srtp back up to the browser. the offer to freeswitch would be: $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove"; and the answer back up to the browswer would be: $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force"; -Eric On 06/23/2016 08:20 AM, John Nash wrote: > I am following > http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 and > trying to test a call > > sipml5 ----------->Opensips + rtpengine --------> SIP end point > (Freeswitch) > > But I do not have any audio on both sides. I see this error at > rtpengine log "SRTP output wanted, but no crypto suite was negotiated" > > Anyone tested this scenario positive? > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From jim at devito.cc Thu Jun 23 16:39:25 2016 From: jim at devito.cc (Jim DeVito) Date: Thu, 23 Jun 2016 07:39:25 -0700 Subject: [OpenSIPS-Users] Dialog DB backend only updates on clean shutdown not crash Message-ID: Hi All. 2.2.0.rc2.20160617.c1aa55e-1.el7 modparam("dialog", "db_url", "text:///var/lib/opensips/dbtext") modparam("dialog", "db_mode", 1) modparam("dialog", "table_name", "dialog") db_mode 1 should be realtime correct? The DB is only being synced from memory on clean shutdown. If I kill -9 the process the dialog info is lost. Thoughts? Thanks!! -- Jim DeVito From john.nash778 at gmail.com Thu Jun 23 16:54:15 2016 From: john.nash778 at gmail.com (John Nash) Date: Thu, 23 Jun 2016 20:24:15 +0530 Subject: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc In-Reply-To: <576BF20B.3060301@uphreak.com> References: <576BF20B.3060301@uphreak.com> Message-ID: Thank you Eric, I will give it a try. On Thu, Jun 23, 2016 at 7:58 PM, Eric Tamme wrote: > 1. I would suggest using SIP.js - https://github.com/onsip/SIP.js it is a > much more active project that sipml5. > > 2. Im guessing that you are not properly passing flags to RTPEngine. If > you want to have DTLS-SRTP between the browser, and plain RTP/AVP between > RTPEngine and freeswitch, you need to "offer" rtp/avp to freeswitch, and > "answer" dtls-srtp back up to the browser. > > the offer to freeswitch would be: > > $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove"; > > > and the answer back up to the browswer would be: > > $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force"; > > > -Eric > > > > On 06/23/2016 08:20 AM, John Nash wrote: > > I am following > http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 and trying > to test a call > > sipml5 ----------->Opensips + rtpengine --------> SIP end point > (Freeswitch) > > But I do not have any audio on both sides. I see this error at rtpengine > log "SRTP output wanted, but no crypto suite was negotiated" > > Anyone tested this scenario positive? > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From john.nash778 at gmail.com Thu Jun 23 18:56:27 2016 From: john.nash778 at gmail.com (John Nash) Date: Thu, 23 Jun 2016 22:26:27 +0530 Subject: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc In-Reply-To: <576BF20B.3060301@uphreak.com> References: <576BF20B.3060301@uphreak.com> Message-ID: I double checked my rtpengine offer answer calls and now using https://github.com/onsip/sipjs-examples/tree/master/demo-phone but I face same issue (no audio either side) and error "SRTP output wanted, but no crypto suite was negotiated" Rtpengine also I updated to the latest now. Am I using correct sip.js example? I copied it to my server and accessing it using https: (used letsencrypt) On Thu, Jun 23, 2016 at 7:58 PM, Eric Tamme wrote: > 1. I would suggest using SIP.js - https://github.com/onsip/SIP.js it is a > much more active project that sipml5. > > 2. Im guessing that you are not properly passing flags to RTPEngine. If > you want to have DTLS-SRTP between the browser, and plain RTP/AVP between > RTPEngine and freeswitch, you need to "offer" rtp/avp to freeswitch, and > "answer" dtls-srtp back up to the browser. > > the offer to freeswitch would be: > > $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove"; > > > and the answer back up to the browswer would be: > > $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force"; > > > -Eric > > > > On 06/23/2016 08:20 AM, John Nash wrote: > > I am following > http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 and trying > to test a call > > sipml5 ----------->Opensips + rtpengine --------> SIP end point > (Freeswitch) > > But I do not have any audio on both sides. I see this error at rtpengine > log "SRTP output wanted, but no crypto suite was negotiated" > > Anyone tested this scenario positive? > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From eric at uphreak.com Thu Jun 23 19:02:16 2016 From: eric at uphreak.com (Eric Tamme) Date: Thu, 23 Jun 2016 11:02:16 -0600 Subject: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc In-Reply-To: References: <576BF20B.3060301@uphreak.com> Message-ID: <576C1618.5010300@uphreak.com> Hey John, Please paste a full UNALTERED sip trace into a gist (gist.github.com) from the proxy servers perspective and provide a link so that we can see what comes in, and what goes out from both sides. EG: ngrep -qtd any -W byline port 5060 This will show us the traffic that is leaving the proxy destined for the Freeswitch box, and what the freeswitch box sends back. Also - you can look in your browsers console log and provide the SIP trace from there in a seperate gist, so that we can see what opensips sends back up to your browser. -Eric > Am I using correct sip.js example? I copied it to my server and > accessing it using https: (used letsencrypt) > > On Thu, Jun 23, 2016 at 7:58 PM, Eric Tamme > wrote: > > 1. I would suggest using SIP.js - https://github.com/onsip/SIP.js > it is a much more active project that sipml5. > > 2. Im guessing that you are not properly passing flags to > RTPEngine. If you want to have DTLS-SRTP between the browser, and > plain RTP/AVP between RTPEngine and freeswitch, you need to > "offer" rtp/avp to freeswitch, and "answer" dtls-srtp back up to > the browser. > > the offer to freeswitch would be: > > $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove"; > > and the answer back up to the browswer would be: > > $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force"; > > > -Eric > > > > On 06/23/2016 08:20 AM, John Nash wrote: >> I am following >> http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 and >> trying to test a call >> >> sipml5 ----------->Opensips + rtpengine --------> SIP end point >> (Freeswitch) >> >> But I do not have any audio on both sides. I see this error at >> rtpengine log "SRTP output wanted, but no crypto suite was >> negotiated" >> >> Anyone tested this scenario positive? >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From john.nash778 at gmail.com Thu Jun 23 20:23:16 2016 From: john.nash778 at gmail.com (John Nash) Date: Thu, 23 Jun 2016 23:53:16 +0530 Subject: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc In-Reply-To: <576C1618.5010300@uphreak.com> References: <576BF20B.3060301@uphreak.com> <576C1618.5010300@uphreak.com> Message-ID: Ok i am ready with logs. About gist may I use private option as traces have our IPs, user On Thu, Jun 23, 2016 at 10:32 PM, Eric Tamme wrote: > Hey John, > > Please paste a full UNALTERED sip trace into a gist (gist.github.com) > from the proxy servers perspective and provide a link so that we can see > what comes in, and what goes out from both sides. > > EG: ngrep -qtd any -W byline port 5060 > > This will show us the traffic that is leaving the proxy destined for the > Freeswitch box, and what the freeswitch box sends back. > > Also - you can look in your browsers console log and provide the SIP trace > from there in a seperate gist, so that we can see what opensips sends back > up to your browser. > > -Eric > > > Am I using correct sip.js example? I copied it to my server and accessing > it using https: (used letsencrypt) > > On Thu, Jun 23, 2016 at 7:58 PM, Eric Tamme wrote: > >> 1. I would suggest using SIP.js - https://github.com/onsip/SIP.js it is >> a much more active project that sipml5. >> >> 2. Im guessing that you are not properly passing flags to RTPEngine. If >> you want to have DTLS-SRTP between the browser, and plain RTP/AVP between >> RTPEngine and freeswitch, you need to "offer" rtp/avp to freeswitch, and >> "answer" dtls-srtp back up to the browser. >> >> the offer to freeswitch would be: >> >> $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove"; >> >> >> and the answer back up to the browswer would be: >> >> $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force"; >> >> >> -Eric >> >> >> >> On 06/23/2016 08:20 AM, John Nash wrote: >> >> I am following >> >> http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 and trying >> to test a call >> >> sipml5 ----------->Opensips + rtpengine --------> SIP end point >> (Freeswitch) >> >> But I do not have any audio on both sides. I see this error at rtpengine >> log "SRTP output wanted, but no crypto suite was negotiated" >> >> Anyone tested this scenario positive? >> >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From eric at uphreak.com Thu Jun 23 20:31:04 2016 From: eric at uphreak.com (Eric Tamme) Date: Thu, 23 Jun 2016 12:31:04 -0600 Subject: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc In-Reply-To: References: <576BF20B.3060301@uphreak.com> <576C1618.5010300@uphreak.com> Message-ID: <576C2AE8.2050604@uphreak.com> I mean you can use a private gist, but you will be publishing the link in a public email list. In general I personally dont believe revealing ip addresses etc. is any problem - to put my money where my mouth is here is a gist link to an unaltered SIP trace on my server :) https://gist.github.com/etamme/b864010448a29007b7e0457682e81d52 -Eric On 06/23/2016 12:23 PM, John Nash wrote: > Ok i am ready with logs. About gist may I use private option as traces > have our IPs, user > > On Thu, Jun 23, 2016 at 10:32 PM, Eric Tamme > wrote: > > Hey John, > > Please paste a full UNALTERED sip trace into a gist > (gist.github.com ) from the proxy servers > perspective and provide a link so that we can see what comes in, > and what goes out from both sides. > > EG: ngrep -qtd any -W byline port 5060 > > This will show us the traffic that is leaving the proxy destined > for the Freeswitch box, and what the freeswitch box sends back. > > Also - you can look in your browsers console log and provide the > SIP trace from there in a seperate gist, so that we can see what > opensips sends back up to your browser. > > -Eric > > >> Am I using correct sip.js example? I copied it to my server and >> accessing it using https: (used letsencrypt) >> >> On Thu, Jun 23, 2016 at 7:58 PM, Eric Tamme > > wrote: >> >> 1. I would suggest using SIP.js - >> https://github.com/onsip/SIP.js it is a much more active >> project that sipml5. >> >> 2. Im guessing that you are not properly passing flags to >> RTPEngine. If you want to have DTLS-SRTP between the >> browser, and plain RTP/AVP between RTPEngine and freeswitch, >> you need to "offer" rtp/avp to freeswitch, and "answer" >> dtls-srtp back up to the browser. >> >> the offer to freeswitch would be: >> >> $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove"; >> >> and the answer back up to the browswer would be: >> >> $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force"; >> >> >> -Eric >> >> >> >> On 06/23/2016 08:20 AM, John Nash wrote: >>> I am following >>> http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 and >>> trying to test a call >>> >>> sipml5 ----------->Opensips + rtpengine --------> SIP end >>> point (Freeswitch) >>> >>> But I do not have any audio on both sides. I see this error >>> at rtpengine log "SRTP output wanted, but no crypto suite >>> was negotiated" >>> >>> Anyone tested this scenario positive? >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From pwakano at gmail.com Thu Jun 23 21:06:33 2016 From: pwakano at gmail.com (Patrick Wakano) Date: Thu, 23 Jun 2016 16:06:33 -0300 Subject: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc In-Reply-To: <576C2AE8.2050604@uphreak.com> References: <576BF20B.3060301@uphreak.com> <576C1618.5010300@uphreak.com> <576C2AE8.2050604@uphreak.com> Message-ID: my paranoic side would recommend to hide/change private informations, specially any authentication line that might appear... this is certainly a sort of social engineering threat we should worry... better be safe than sorry.... On Thu, Jun 23, 2016 at 3:31 PM, Eric Tamme wrote: > I mean you can use a private gist, but you will be publishing the link in > a public email list. In general I personally dont believe revealing ip > addresses etc. is any problem - to put my money where my mouth is here is a > gist link to an unaltered SIP trace on my server :) > > https://gist.github.com/etamme/b864010448a29007b7e0457682e81d52 > > -Eric > > > On 06/23/2016 12:23 PM, John Nash wrote: > > Ok i am ready with logs. About gist may I use private option as traces > have our IPs, user > > On Thu, Jun 23, 2016 at 10:32 PM, Eric Tamme wrote: > >> Hey John, >> >> Please paste a full UNALTERED sip trace into a gist (gist.github.com) >> from the proxy servers perspective and provide a link so that we can see >> what comes in, and what goes out from both sides. >> >> EG: ngrep -qtd any -W byline port 5060 >> >> This will show us the traffic that is leaving the proxy destined for the >> Freeswitch box, and what the freeswitch box sends back. >> >> Also - you can look in your browsers console log and provide the SIP >> trace from there in a seperate gist, so that we can see what opensips sends >> back up to your browser. >> >> -Eric >> >> >> Am I using correct sip.js example? I copied it to my server and accessing >> it using https: (used letsencrypt) >> >> On Thu, Jun 23, 2016 at 7:58 PM, Eric Tamme < >> eric at uphreak.com> wrote: >> >>> 1. I would suggest using SIP.js - >>> https://github.com/onsip/SIP.js it is a much more active project that >>> sipml5. >>> >>> 2. Im guessing that you are not properly passing flags to RTPEngine. If >>> you want to have DTLS-SRTP between the browser, and plain RTP/AVP between >>> RTPEngine and freeswitch, you need to "offer" rtp/avp to freeswitch, and >>> "answer" dtls-srtp back up to the browser. >>> >>> the offer to freeswitch would be: >>> >>> $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove"; >>> >>> >>> and the answer back up to the browswer would be: >>> >>> $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force"; >>> >>> >>> -Eric >>> >>> >>> >>> On 06/23/2016 08:20 AM, John Nash wrote: >>> >>> I am following >>> >>> http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 and >>> trying to test a call >>> >>> sipml5 ----------->Opensips + rtpengine --------> SIP end point >>> (Freeswitch) >>> >>> But I do not have any audio on both sides. I see this error at rtpengine >>> log "SRTP output wanted, but no crypto suite was negotiated" >>> >>> Anyone tested this scenario positive? >>> >>> >>> _______________________________________________ >>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From eric at uphreak.com Thu Jun 23 21:13:00 2016 From: eric at uphreak.com (Eric Tamme) Date: Thu, 23 Jun 2016 13:13:00 -0600 Subject: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc In-Reply-To: References: <576BF20B.3060301@uphreak.com> <576C1618.5010300@uphreak.com> <576C2AE8.2050604@uphreak.com> Message-ID: <576C34BC.4060403@uphreak.com> No - it's annoying to look at a trace that's had information removed and try and piece together whats happening. Your paranoid side is wrong, sorry. -Eric On 06/23/2016 01:06 PM, Patrick Wakano wrote: > my paranoic side would recommend to hide/change private informations, > specially any authentication line that might appear... this is > certainly a sort of social engineering threat we should worry... > better be safe than sorry.... > > > On Thu, Jun 23, 2016 at 3:31 PM, Eric Tamme > wrote: > > I mean you can use a private gist, but you will be publishing the > link in a public email list. In general I personally dont believe > revealing ip addresses etc. is any problem - to put my money where > my mouth is here is a gist link to an unaltered SIP trace on my > server :) > > https://gist.github.com/etamme/b864010448a29007b7e0457682e81d52 > > -Eric > > > On 06/23/2016 12:23 PM, John Nash wrote: >> Ok i am ready with logs. About gist may I use private option as >> traces have our IPs, user >> >> On Thu, Jun 23, 2016 at 10:32 PM, Eric Tamme > > wrote: >> >> Hey John, >> >> Please paste a full UNALTERED sip trace into a gist >> (gist.github.com ) from the proxy >> servers perspective and provide a link so that we can see >> what comes in, and what goes out from both sides. >> >> EG: ngrep -qtd any -W byline port 5060 >> >> This will show us the traffic that is leaving the proxy >> destined for the Freeswitch box, and what the freeswitch box >> sends back. >> >> Also - you can look in your browsers console log and provide >> the SIP trace from there in a seperate gist, so that we can >> see what opensips sends back up to your browser. >> >> -Eric >> >> >>> Am I using correct sip.js example? I copied it to my server >>> and accessing it using https: (used letsencrypt) >>> >>> On Thu, Jun 23, 2016 at 7:58 PM, Eric Tamme >>> > wrote: >>> >>> 1. I would suggest using SIP.js - >>> https://github.com/onsip/SIP.js it is a much more active >>> project that sipml5. >>> >>> 2. Im guessing that you are not properly passing flags >>> to RTPEngine. If you want to have DTLS-SRTP between the >>> browser, and plain RTP/AVP between RTPEngine and >>> freeswitch, you need to "offer" rtp/avp to freeswitch, >>> and "answer" dtls-srtp back up to the browser. >>> >>> the offer to freeswitch would be: >>> >>> $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove"; >>> >>> and the answer back up to the browswer would be: >>> >>> $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force"; >>> >>> >>> -Eric >>> >>> >>> >>> On 06/23/2016 08:20 AM, John Nash wrote: >>>> I am following >>>> http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 >>>> and trying to test a call >>>> >>>> sipml5 ----------->Opensips + rtpengine --------> SIP >>>> end point (Freeswitch) >>>> >>>> But I do not have any audio on both sides. I see this >>>> error at rtpengine log "SRTP output wanted, but no >>>> crypto suite was negotiated" >>>> >>>> Anyone tested this scenario positive? >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From john.nash778 at gmail.com Thu Jun 23 21:24:42 2016 From: john.nash778 at gmail.com (John Nash) Date: Fri, 24 Jun 2016 00:54:42 +0530 Subject: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc In-Reply-To: <576C34BC.4060403@uphreak.com> References: <576BF20B.3060301@uphreak.com> <576C1618.5010300@uphreak.com> <576C2AE8.2050604@uphreak.com> <576C34BC.4060403@uphreak.com> Message-ID: OK here is the log https://gist.github.com/johnnash13/0d2cb5238f3551cd3a8c6b4e638dd744 Sorry took me a while to convert wireshark trace to text file. My freeswitch is running on private IP (127.0.0.1) and opensips I run on both public and private so that for outside world opensips is the only public IP they see. In proxy log I pasted Opensips ===> Freeswitch logs and back. On Fri, Jun 24, 2016 at 12:43 AM, Eric Tamme wrote: > No - it's annoying to look at a trace that's had information removed and > try and piece together whats happening. Your paranoid side is wrong, sorry. > > -Eric > > > On 06/23/2016 01:06 PM, Patrick Wakano wrote: > > my paranoic side would recommend to hide/change private informations, > specially any authentication line that might appear... this is certainly a > sort of social engineering threat we should worry... > better be safe than sorry.... > > > On Thu, Jun 23, 2016 at 3:31 PM, Eric Tamme wrote: > >> I mean you can use a private gist, but you will be publishing the link in >> a public email list. In general I personally dont believe revealing ip >> addresses etc. is any problem - to put my money where my mouth is here is a >> gist link to an unaltered SIP trace on my server :) >> >> https://gist.github.com/etamme/b864010448a29007b7e0457682e81d52 >> >> -Eric >> >> >> On 06/23/2016 12:23 PM, John Nash wrote: >> >> Ok i am ready with logs. About gist may I use private option as traces >> have our IPs, user >> >> On Thu, Jun 23, 2016 at 10:32 PM, Eric Tamme < >> eric at uphreak.com> wrote: >> >>> Hey John, >>> >>> Please paste a full UNALTERED sip trace into a gist (gist.github.com) >>> from the proxy servers perspective and provide a link so that we can see >>> what comes in, and what goes out from both sides. >>> >>> EG: ngrep -qtd any -W byline port 5060 >>> >>> This will show us the traffic that is leaving the proxy destined for the >>> Freeswitch box, and what the freeswitch box sends back. >>> >>> Also - you can look in your browsers console log and provide the SIP >>> trace from there in a seperate gist, so that we can see what opensips sends >>> back up to your browser. >>> >>> -Eric >>> >>> >>> Am I using correct sip.js example? I copied it to my server and >>> accessing it using https: (used letsencrypt) >>> >>> On Thu, Jun 23, 2016 at 7:58 PM, Eric Tamme < >>> eric at uphreak.com> wrote: >>> >>>> 1. I would suggest using SIP.js - >>>> https://github.com/onsip/SIP.js it is a much more active project that >>>> sipml5. >>>> >>>> 2. Im guessing that you are not properly passing flags to RTPEngine. >>>> If you want to have DTLS-SRTP between the browser, and plain RTP/AVP >>>> between RTPEngine and freeswitch, you need to "offer" rtp/avp to >>>> freeswitch, and "answer" dtls-srtp back up to the browser. >>>> >>>> the offer to freeswitch would be: >>>> >>>> $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove"; >>>> >>>> >>>> and the answer back up to the browswer would be: >>>> >>>> $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force"; >>>> >>>> >>>> -Eric >>>> >>>> >>>> >>>> On 06/23/2016 08:20 AM, John Nash wrote: >>>> >>>> I am following >>>> >>>> http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 and >>>> trying to test a call >>>> >>>> sipml5 ----------->Opensips + rtpengine --------> SIP end point >>>> (Freeswitch) >>>> >>>> But I do not have any audio on both sides. I see this error at >>>> rtpengine log "SRTP output wanted, but no crypto suite was negotiated" >>>> >>>> Anyone tested this scenario positive? >>>> >>>> >>>> _______________________________________________ >>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>> >>> >>> _______________________________________________ >>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From eric at uphreak.com Thu Jun 23 21:35:01 2016 From: eric at uphreak.com (Eric Tamme) Date: Thu, 23 Jun 2016 13:35:01 -0600 Subject: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc In-Reply-To: References: <576BF20B.3060301@uphreak.com> <576C1618.5010300@uphreak.com> <576C2AE8.2050604@uphreak.com> <576C34BC.4060403@uphreak.com> Message-ID: <576C39E5.8090607@uphreak.com> So - i dont see a problem here - Chrome is getting UDP/TLS/RTP/SAVPF and Freeswitch is getting RTP/AVP. Freeswitch responded to the offer in the invite with an answer in the 183, and in the 200. What is the failure you are seeing, and where is it happening (in freeswitch? in the browser?) The only thing that looks bad is that you are retransmitting the ACK which FS either ... doesnt like, or is never getting, because it keeps retransmitting the 200, which is why you get a 481 when you send BYE. -Eric On 06/23/2016 01:24 PM, John Nash wrote: > OK here is the log > https://gist.github.com/johnnash13/0d2cb5238f3551cd3a8c6b4e638dd744 > > Sorry took me a while to convert wireshark trace to text file. > > My freeswitch is running on private IP (127.0.0.1) and opensips I run > on both public and private so that for outside world opensips is the > only public IP they see. In proxy log I pasted Opensips ===> > Freeswitch logs and back. > > > > > > > On Fri, Jun 24, 2016 at 12:43 AM, Eric Tamme > wrote: > > No - it's annoying to look at a trace that's had information > removed and try and piece together whats happening. Your paranoid > side is wrong, sorry. > > -Eric > > > On 06/23/2016 01:06 PM, Patrick Wakano wrote: >> my paranoic side would recommend to hide/change private >> informations, specially any authentication line that might >> appear... this is certainly a sort of social engineering threat >> we should worry... >> better be safe than sorry.... >> >> >> On Thu, Jun 23, 2016 at 3:31 PM, Eric Tamme > > wrote: >> >> I mean you can use a private gist, but you will be publishing >> the link in a public email list. In general I personally dont >> believe revealing ip addresses etc. is any problem - to put >> my money where my mouth is here is a gist link to an >> unaltered SIP trace on my server :) >> >> https://gist.github.com/etamme/b864010448a29007b7e0457682e81d52 >> >> -Eric >> >> >> On 06/23/2016 12:23 PM, John Nash wrote: >>> Ok i am ready with logs. About gist may I use private option >>> as traces have our IPs, user >>> >>> On Thu, Jun 23, 2016 at 10:32 PM, Eric Tamme >>> > wrote: >>> >>> Hey John, >>> >>> Please paste a full UNALTERED sip trace into a gist >>> (gist.github.com ) from the >>> proxy servers perspective and provide a link so that we >>> can see what comes in, and what goes out from both sides. >>> >>> EG: ngrep -qtd any -W byline port 5060 >>> >>> This will show us the traffic that is leaving the proxy >>> destined for the Freeswitch box, and what the freeswitch >>> box sends back. >>> >>> Also - you can look in your browsers console log and >>> provide the SIP trace from there in a seperate gist, so >>> that we can see what opensips sends back up to your browser. >>> >>> -Eric >>> >>> >>>> Am I using correct sip.js example? I copied it to my >>>> server and accessing it using https: (used letsencrypt) >>>> >>>> On Thu, Jun 23, 2016 at 7:58 PM, Eric Tamme >>>> > wrote: >>>> >>>> 1. I would suggest using SIP.js - >>>> https://github.com/onsip/SIP.js it is a much more >>>> active project that sipml5. >>>> >>>> 2. Im guessing that you are not properly passing >>>> flags to RTPEngine. If you want to have DTLS-SRTP >>>> between the browser, and plain RTP/AVP between >>>> RTPEngine and freeswitch, you need to "offer" >>>> rtp/avp to freeswitch, and "answer" dtls-srtp back >>>> up to the browser. >>>> >>>> the offer to freeswitch would be: >>>> >>>> $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove"; >>>> >>>> and the answer back up to the browswer would be: >>>> >>>> $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force"; >>>> >>>> >>>> -Eric >>>> >>>> >>>> >>>> On 06/23/2016 08:20 AM, John Nash wrote: >>>>> I am following >>>>> http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 >>>>> and trying to test a call >>>>> >>>>> sipml5 ----------->Opensips + rtpengine --------> >>>>> SIP end point (Freeswitch) >>>>> >>>>> But I do not have any audio on both sides. I see >>>>> this error at rtpengine log "SRTP output wanted, >>>>> but no crypto suite was negotiated" >>>>> >>>>> Anyone tested this scenario positive? >>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From john.nash778 at gmail.com Thu Jun 23 21:42:14 2016 From: john.nash778 at gmail.com (John Nash) Date: Fri, 24 Jun 2016 01:12:14 +0530 Subject: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc In-Reply-To: <576C39E5.8090607@uphreak.com> References: <576BF20B.3060301@uphreak.com> <576C1618.5010300@uphreak.com> <576C2AE8.2050604@uphreak.com> <576C34BC.4060403@uphreak.com> <576C39E5.8090607@uphreak.com> Message-ID: Actually the issue is i hear no audio on either side and just after session progress (I guess when media starts coming from remote media server) i see error "SRTP output wanted, but no crypto suite was negotiated" I had also checked media logs i could see RTP packets being sent from freeswitch to RTPengine IP but there was no packet at all just after that. Ideally after RTP packet from freeswitch to rtpengine, Rtpengine should send that packet to browser using wss? On Fri, Jun 24, 2016 at 1:05 AM, Eric Tamme wrote: > So - i dont see a problem here - Chrome is getting UDP/TLS/RTP/SAVPF and > Freeswitch is getting RTP/AVP. Freeswitch responded to the offer in the > invite with an answer in the 183, and in the 200. What is the failure you > are seeing, and where is it happening (in freeswitch? in the browser?) > > The only thing that looks bad is that you are retransmitting the ACK which > FS either ... doesnt like, or is never getting, because it keeps > retransmitting the 200, which is why you get a 481 when you send BYE. > > -Eric > > > On 06/23/2016 01:24 PM, John Nash wrote: > > OK here is the log > https://gist.github.com/johnnash13/0d2cb5238f3551cd3a8c6b4e638dd744 > > Sorry took me a while to convert wireshark trace to text file. > > My freeswitch is running on private IP (127.0.0.1) and opensips I run on > both public and private so that for outside world opensips is the only > public IP they see. In proxy log I pasted Opensips ===> Freeswitch logs and > back. > > > > > > > On Fri, Jun 24, 2016 at 12:43 AM, Eric Tamme wrote: > >> No - it's annoying to look at a trace that's had information removed and >> try and piece together whats happening. Your paranoid side is wrong, sorry. >> >> -Eric >> >> >> On 06/23/2016 01:06 PM, Patrick Wakano wrote: >> >> my paranoic side would recommend to hide/change private informations, >> specially any authentication line that might appear... this is certainly a >> sort of social engineering threat we should worry... >> better be safe than sorry.... >> >> >> On Thu, Jun 23, 2016 at 3:31 PM, Eric Tamme < >> eric at uphreak.com> wrote: >> >>> I mean you can use a private gist, but you will be publishing the link >>> in a public email list. In general I personally dont believe revealing ip >>> addresses etc. is any problem - to put my money where my mouth is here is a >>> gist link to an unaltered SIP trace on my server :) >>> >>> https://gist.github.com/etamme/b864010448a29007b7e0457682e81d52 >>> >>> -Eric >>> >>> >>> On 06/23/2016 12:23 PM, John Nash wrote: >>> >>> Ok i am ready with logs. About gist may I use private option as traces >>> have our IPs, user >>> >>> On Thu, Jun 23, 2016 at 10:32 PM, Eric Tamme < >>> eric at uphreak.com> wrote: >>> >>>> Hey John, >>>> >>>> Please paste a full UNALTERED sip trace into a gist (gist.github.com) >>>> from the proxy servers perspective and provide a link so that we can see >>>> what comes in, and what goes out from both sides. >>>> >>>> EG: ngrep -qtd any -W byline port 5060 >>>> >>>> This will show us the traffic that is leaving the proxy destined for >>>> the Freeswitch box, and what the freeswitch box sends back. >>>> >>>> Also - you can look in your browsers console log and provide the SIP >>>> trace from there in a seperate gist, so that we can see what opensips sends >>>> back up to your browser. >>>> >>>> -Eric >>>> >>>> >>>> Am I using correct sip.js example? I copied it to my server and >>>> accessing it using https: (used letsencrypt) >>>> >>>> On Thu, Jun 23, 2016 at 7:58 PM, Eric Tamme < >>>> eric at uphreak.com> wrote: >>>> >>>>> 1. I would suggest using SIP.js - >>>>> https://github.com/onsip/SIP.js it is a much more active project that >>>>> sipml5. >>>>> >>>>> 2. Im guessing that you are not properly passing flags to RTPEngine. >>>>> If you want to have DTLS-SRTP between the browser, and plain RTP/AVP >>>>> between RTPEngine and freeswitch, you need to "offer" rtp/avp to >>>>> freeswitch, and "answer" dtls-srtp back up to the browser. >>>>> >>>>> the offer to freeswitch would be: >>>>> >>>>> $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove"; >>>>> >>>>> >>>>> and the answer back up to the browswer would be: >>>>> >>>>> $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force"; >>>>> >>>>> >>>>> -Eric >>>>> >>>>> >>>>> >>>>> On 06/23/2016 08:20 AM, John Nash wrote: >>>>> >>>>> I am following >>>>> >>>>> http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 and >>>>> trying to test a call >>>>> >>>>> sipml5 ----------->Opensips + rtpengine --------> SIP end point >>>>> (Freeswitch) >>>>> >>>>> But I do not have any audio on both sides. I see this error at >>>>> rtpengine log "SRTP output wanted, but no crypto suite was negotiated" >>>>> >>>>> Anyone tested this scenario positive? >>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>> >>> >>> _______________________________________________ >>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From sevpal at aol.com Thu Jun 23 23:20:30 2016 From: sevpal at aol.com (sevpal) Date: Thu, 23 Jun 2016 17:20:30 -0400 Subject: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc In-Reply-To: References: <576BF20B.3060301@uphreak.com><576C1618.5010300@uphreak.com><576C2AE8.2050604@uphreak.com><576C34BC.4060403@uphreak.com><576C39E5.8090607@uphreak.com> Message-ID: <6F0C92D572884D2AAAFDBC6EE354CC53@LenovoPC> Hi, the rtpengine cannot negotiate SRTP between the two points, both must support the same cryptography and protocol. eg; SRTP to SRTP , DTLS/SRTP to DTLS/SRTP cipher 128 to 128 and 256 to 256. You can print the request body ($rb) on the INVITE with ?application/sdp? and visually compare the exchange, do this on offer and answer. From: John Nash Sent: Thursday, June 23, 2016 3:42 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc Actually the issue is i hear no audio on either side and just after session progress (I guess when media starts coming from remote media server) i see error "SRTP output wanted, but no crypto suite was negotiated" I had also checked media logs i could see RTP packets being sent from freeswitch to RTPengine IP but there was no packet at all just after that. Ideally after RTP packet from freeswitch to rtpengine, Rtpengine should send that packet to browser using wss? On Fri, Jun 24, 2016 at 1:05 AM, Eric Tamme wrote: So - i dont see a problem here - Chrome is getting UDP/TLS/RTP/SAVPF and Freeswitch is getting RTP/AVP. Freeswitch responded to the offer in the invite with an answer in the 183, and in the 200. What is the failure you are seeing, and where is it happening (in freeswitch? in the browser?) The only thing that looks bad is that you are retransmitting the ACK which FS either ... doesnt like, or is never getting, because it keeps retransmitting the 200, which is why you get a 481 when you send BYE. -Eric On 06/23/2016 01:24 PM, John Nash wrote: OK here is the log https://gist.github.com/johnnash13/0d2cb5238f3551cd3a8c6b4e638dd744 Sorry took me a while to convert wireshark trace to text file. My freeswitch is running on private IP (127.0.0.1) and opensips I run on both public and private so that for outside world opensips is the only public IP they see. In proxy log I pasted Opensips ===> Freeswitch logs and back. On Fri, Jun 24, 2016 at 12:43 AM, Eric Tamme wrote: No - it's annoying to look at a trace that's had information removed and try and piece together whats happening. Your paranoid side is wrong, sorry. -Eric On 06/23/2016 01:06 PM, Patrick Wakano wrote: my paranoic side would recommend to hide/change private informations, specially any authentication line that might appear... this is certainly a sort of social engineering threat we should worry... better be safe than sorry.... On Thu, Jun 23, 2016 at 3:31 PM, Eric Tamme wrote: I mean you can use a private gist, but you will be publishing the link in a public email list. In general I personally dont believe revealing ip addresses etc. is any problem - to put my money where my mouth is here is a gist link to an unaltered SIP trace on my server :) https://gist.github.com/etamme/b864010448a29007b7e0457682e81d52 -Eric On 06/23/2016 12:23 PM, John Nash wrote: Ok i am ready with logs. About gist may I use private option as traces have our IPs, user On Thu, Jun 23, 2016 at 10:32 PM, Eric Tamme wrote: Hey John, Please paste a full UNALTERED sip trace into a gist (gist.github.com) from the proxy servers perspective and provide a link so that we can see what comes in, and what goes out from both sides. EG: ngrep -qtd any -W byline port 5060 This will show us the traffic that is leaving the proxy destined for the Freeswitch box, and what the freeswitch box sends back. Also - you can look in your browsers console log and provide the SIP trace from there in a seperate gist, so that we can see what opensips sends back up to your browser. -Eric Am I using correct sip.js example? I copied it to my server and accessing it using https: (used letsencrypt) On Thu, Jun 23, 2016 at 7:58 PM, Eric Tamme wrote: 1. I would suggest using SIP.js - https://github.com/onsip/SIP.js it is a much more active project that sipml5. 2. Im guessing that you are not properly passing flags to RTPEngine. If you want to have DTLS-SRTP between the browser, and plain RTP/AVP between RTPEngine and freeswitch, you need to "offer" rtp/avp to freeswitch, and "answer" dtls-srtp back up to the browser. the offer to freeswitch would be: $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove"; and the answer back up to the browswer would be: $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force"; -Eric On 06/23/2016 08:20 AM, John Nash wrote: I am following http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 and trying to test a call sipml5 ----------->Opensips + rtpengine --------> SIP end point (Freeswitch) But I do not have any audio on both sides. I see this error at rtpengine log "SRTP output wanted, but no crypto suite was negotiated" Anyone tested this scenario positive? _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------------------------------------------------------------------------- _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From sevpal at aol.com Thu Jun 23 23:30:05 2016 From: sevpal at aol.com (sevpal) Date: Thu, 23 Jun 2016 17:30:05 -0400 Subject: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc In-Reply-To: References: <576BF20B.3060301@uphreak.com><576C1618.5010300@uphreak.com><576C2AE8.2050604@uphreak.com><576C34BC.4060403@uphreak.com><576C39E5.8090607@uphreak.com> Message-ID: <7C2F3A12B1504B40BB32EE2A92895B76@LenovoPC> Take a look at the ?fingerprint:? line. From: John Nash Sent: Thursday, June 23, 2016 3:42 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc Actually the issue is i hear no audio on either side and just after session progress (I guess when media starts coming from remote media server) i see error "SRTP output wanted, but no crypto suite was negotiated" I had also checked media logs i could see RTP packets being sent from freeswitch to RTPengine IP but there was no packet at all just after that. Ideally after RTP packet from freeswitch to rtpengine, Rtpengine should send that packet to browser using wss? On Fri, Jun 24, 2016 at 1:05 AM, Eric Tamme wrote: So - i dont see a problem here - Chrome is getting UDP/TLS/RTP/SAVPF and Freeswitch is getting RTP/AVP. Freeswitch responded to the offer in the invite with an answer in the 183, and in the 200. What is the failure you are seeing, and where is it happening (in freeswitch? in the browser?) The only thing that looks bad is that you are retransmitting the ACK which FS either ... doesnt like, or is never getting, because it keeps retransmitting the 200, which is why you get a 481 when you send BYE. -Eric On 06/23/2016 01:24 PM, John Nash wrote: OK here is the log https://gist.github.com/johnnash13/0d2cb5238f3551cd3a8c6b4e638dd744 Sorry took me a while to convert wireshark trace to text file. My freeswitch is running on private IP (127.0.0.1) and opensips I run on both public and private so that for outside world opensips is the only public IP they see. In proxy log I pasted Opensips ===> Freeswitch logs and back. On Fri, Jun 24, 2016 at 12:43 AM, Eric Tamme wrote: No - it's annoying to look at a trace that's had information removed and try and piece together whats happening. Your paranoid side is wrong, sorry. -Eric On 06/23/2016 01:06 PM, Patrick Wakano wrote: my paranoic side would recommend to hide/change private informations, specially any authentication line that might appear... this is certainly a sort of social engineering threat we should worry... better be safe than sorry.... On Thu, Jun 23, 2016 at 3:31 PM, Eric Tamme wrote: I mean you can use a private gist, but you will be publishing the link in a public email list. In general I personally dont believe revealing ip addresses etc. is any problem - to put my money where my mouth is here is a gist link to an unaltered SIP trace on my server :) https://gist.github.com/etamme/b864010448a29007b7e0457682e81d52 -Eric On 06/23/2016 12:23 PM, John Nash wrote: Ok i am ready with logs. About gist may I use private option as traces have our IPs, user On Thu, Jun 23, 2016 at 10:32 PM, Eric Tamme wrote: Hey John, Please paste a full UNALTERED sip trace into a gist (gist.github.com) from the proxy servers perspective and provide a link so that we can see what comes in, and what goes out from both sides. EG: ngrep -qtd any -W byline port 5060 This will show us the traffic that is leaving the proxy destined for the Freeswitch box, and what the freeswitch box sends back. Also - you can look in your browsers console log and provide the SIP trace from there in a seperate gist, so that we can see what opensips sends back up to your browser. -Eric Am I using correct sip.js example? I copied it to my server and accessing it using https: (used letsencrypt) On Thu, Jun 23, 2016 at 7:58 PM, Eric Tamme wrote: 1. I would suggest using SIP.js - https://github.com/onsip/SIP.js it is a much more active project that sipml5. 2. Im guessing that you are not properly passing flags to RTPEngine. If you want to have DTLS-SRTP between the browser, and plain RTP/AVP between RTPEngine and freeswitch, you need to "offer" rtp/avp to freeswitch, and "answer" dtls-srtp back up to the browser. the offer to freeswitch would be: $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove"; and the answer back up to the browswer would be: $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force"; -Eric On 06/23/2016 08:20 AM, John Nash wrote: I am following http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 and trying to test a call sipml5 ----------->Opensips + rtpengine --------> SIP end point (Freeswitch) But I do not have any audio on both sides. I see this error at rtpengine log "SRTP output wanted, but no crypto suite was negotiated" Anyone tested this scenario positive? _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------------------------------------------------------------------------- _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From ozzn at 3flabs.com Fri Jun 24 00:32:13 2016 From: ozzn at 3flabs.com (Ozz Nixon) Date: Thu, 23 Jun 2016 18:32:13 -0400 Subject: [OpenSIPS-Users] n00b SIP question Message-ID: <048501d1cd9f$168871f0$439955d0$@3flabs.com> Hello, I have read this mail list for months, everything is way above me ? but, I am ready to jump in. I have a twilio account, I have static COMCAST business class (Deluxe), not in bridged mode. What are the steps/products I should install on a Linux server to do my own VoIP? Design goal, 1 SIP phone in the same 10.1.0.x network, 1 SIP phone in Georgia, and 1 SIP phone in Philadelphia. Goal is to get an understanding, and migrate my 8 or 9 DIDs to inhouse, have a nice DELL (16GB RAM, 3TB of disk space), dual FAST-E NICs. * I gave I a whirl a couple weeks ago, with one of those ?all-in-one? projects, like ?Sip on a stick?. Migrated 2 DIDs without issue adding 3rd, and 4th brought the whole environment down where none of the DIDs worked. I am interested in just doing it module by module, product by product until I have this understood (like when do I need STUN or will I ever, etc). Thanks guys! Ozz -------------- next part -------------- An HTML attachment was scrubbed... URL: From jim at devito.cc Fri Jun 24 01:33:03 2016 From: jim at devito.cc (Jim DeVito) Date: Thu, 23 Jun 2016 19:33:03 -0400 Subject: [OpenSIPS-Users] n00b SIP question In-Reply-To: <048501d1cd9f$168871f0$439955d0$@3flabs.com> Message-ID: <4263fc76-3797-476c-8deb-f5a22f432c55@email.android.com> An HTML attachment was scrubbed... URL: From john.nash778 at gmail.com Fri Jun 24 08:25:48 2016 From: john.nash778 at gmail.com (John Nash) Date: Fri, 24 Jun 2016 11:55:48 +0530 Subject: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc In-Reply-To: <6F0C92D572884D2AAAFDBC6EE354CC53@LenovoPC> References: <576BF20B.3060301@uphreak.com> <576C1618.5010300@uphreak.com> <576C2AE8.2050604@uphreak.com> <576C34BC.4060403@uphreak.com> <576C39E5.8090607@uphreak.com> <6F0C92D572884D2AAAFDBC6EE354CC53@LenovoPC> Message-ID: Sorry sevpal somehow your message went to spam. I am not sure I get what you are trying to say as I was under the impression rtpengine is supposed to bridge protocols. Better I explain my test setup properly to you .. 1- On linux server I installed certificates from letsencrypt for a domain (). 2- I have opensips (wss listner is there as well as udp), Rtpengine and freeswitch (udp only and it terminate calls to SIP network) 3- On web server I copied sipml5 code which I access on chrome browser using https://:443. In sipml5 I give wss url of the opensips wss listener (wss://:4431 along with SIP credentials 4- My call flow is Chrome(sipml5) ==wss==>Opensips===udp==>Freeswitch. Before sending Invite to freeswitch Rtpengine call is made as per http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2. Same being done when session progress or 200 OK comes from freeswitch. Now in this setup how can I make sure same crypto is used? On Fri, Jun 24, 2016 at 2:50 AM, sevpal wrote: > Hi, the rtpengine cannot negotiate SRTP between the two points, both must > support the same cryptography and protocol. eg; SRTP to SRTP , DTLS/SRTP to > DTLS/SRTP cipher 128 to 128 and 256 to 256. > > You can print the request body ($rb) on the INVITE with ?application/sdp? > and visually compare the exchange, do this on offer and answer. > > *From:* John Nash > *Sent:* Thursday, June 23, 2016 3:42 PM > *To:* OpenSIPS users mailling list > *Subject:* Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc > > Actually the issue is i hear no audio on either side and just after > session progress (I guess when media starts coming from remote media > server) i see error "SRTP output wanted, but no crypto suite was > negotiated" > > I had also checked media logs i could see RTP packets being sent from > freeswitch to RTPengine IP but there was no packet at all just after that. > Ideally after RTP packet from freeswitch to rtpengine, Rtpengine should > send that packet to browser using wss? > > On Fri, Jun 24, 2016 at 1:05 AM, Eric Tamme wrote: > >> So - i dont see a problem here - Chrome is getting UDP/TLS/RTP/SAVPF and >> Freeswitch is getting RTP/AVP. Freeswitch responded to the offer in the >> invite with an answer in the 183, and in the 200. What is the failure you >> are seeing, and where is it happening (in freeswitch? in the browser?) >> >> The only thing that looks bad is that you are retransmitting the ACK >> which FS either ... doesnt like, or is never getting, because it keeps >> retransmitting the 200, which is why you get a 481 when you send BYE. >> >> -Eric >> >> >> On 06/23/2016 01:24 PM, John Nash wrote: >> >> OK here is the log >> https://gist.github.com/johnnash13/0d2cb5238f3551cd3a8c6b4e638dd744 >> >> Sorry took me a while to convert wireshark trace to text file. >> >> My freeswitch is running on private IP (127.0.0.1) and opensips I run on >> both public and private so that for outside world opensips is the only >> public IP they see. In proxy log I pasted Opensips ===> Freeswitch logs and >> back. >> >> >> >> >> >> >> On Fri, Jun 24, 2016 at 12:43 AM, Eric Tamme wrote: >> >>> No - it's annoying to look at a trace that's had information removed and >>> try and piece together whats happening. Your paranoid side is wrong, sorry. >>> >>> -Eric >>> >>> >>> On 06/23/2016 01:06 PM, Patrick Wakano wrote: >>> >>> my paranoic side would recommend to hide/change private informations, >>> specially any authentication line that might appear... this is certainly a >>> sort of social engineering threat we should worry... >>> better be safe than sorry.... >>> >>> >>> On Thu, Jun 23, 2016 at 3:31 PM, Eric Tamme wrote: >>> >>>> I mean you can use a private gist, but you will be publishing the link >>>> in a public email list. In general I personally dont believe revealing ip >>>> addresses etc. is any problem - to put my money where my mouth is here is a >>>> gist link to an unaltered SIP trace on my server :) >>>> >>>> https://gist.github.com/etamme/b864010448a29007b7e0457682e81d52 >>>> >>>> -Eric >>>> >>>> >>>> On 06/23/2016 12:23 PM, John Nash wrote: >>>> >>>> Ok i am ready with logs. About gist may I use private option as traces >>>> have our IPs, user >>>> >>>> On Thu, Jun 23, 2016 at 10:32 PM, Eric Tamme wrote: >>>> >>>>> Hey John, >>>>> >>>>> Please paste a full UNALTERED sip trace into a gist (gist.github.com) >>>>> from the proxy servers perspective and provide a link so that we can see >>>>> what comes in, and what goes out from both sides. >>>>> >>>>> EG: ngrep -qtd any -W byline port 5060 >>>>> >>>>> This will show us the traffic that is leaving the proxy destined for >>>>> the Freeswitch box, and what the freeswitch box sends back. >>>>> >>>>> Also - you can look in your browsers console log and provide the SIP >>>>> trace from there in a seperate gist, so that we can see what opensips sends >>>>> back up to your browser. >>>>> >>>>> -Eric >>>>> >>>>> >>>>> Am I using correct sip.js example? I copied it to my server and >>>>> accessing it using https: (used letsencrypt) >>>>> >>>>> On Thu, Jun 23, 2016 at 7:58 PM, Eric Tamme wrote: >>>>> >>>>>> 1. I would suggest using SIP.js - https://github.com/onsip/SIP.js it >>>>>> is a much more active project that sipml5. >>>>>> >>>>>> 2. Im guessing that you are not properly passing flags to RTPEngine. >>>>>> If you want to have DTLS-SRTP between the browser, and plain RTP/AVP >>>>>> between RTPEngine and freeswitch, you need to "offer" rtp/avp to >>>>>> freeswitch, and "answer" dtls-srtp back up to the browser. >>>>>> >>>>>> the offer to freeswitch would be: >>>>>> >>>>>> $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove"; >>>>>> >>>>>> >>>>>> and the answer back up to the browswer would be: >>>>>> >>>>>> $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force"; >>>>>> >>>>>> >>>>>> -Eric >>>>>> >>>>>> >>>>>> >>>>>> On 06/23/2016 08:20 AM, John Nash wrote: >>>>>> >>>>>> I am following >>>>>> http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 and >>>>>> trying to test a call >>>>>> >>>>>> sipml5 ----------->Opensips + rtpengine --------> SIP end point >>>>>> (Freeswitch) >>>>>> >>>>>> But I do not have any audio on both sides. I see this error at >>>>>> rtpengine log "SRTP output wanted, but no crypto suite was negotiated" >>>>>> >>>>>> Anyone tested this scenario positive? >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>>> >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Users mailing list >>>>>> Users at lists.opensips.org >>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>>> >>>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>> >>> >>> _______________________________________________ >>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > ------------------------------ > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Fri Jun 24 09:00:06 2016 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Fri, 24 Jun 2016 10:00:06 +0300 Subject: [OpenSIPS-Users] Dialog DB backend only updates on clean shutdown not crash In-Reply-To: References: Message-ID: <1d15ec82-1e1a-952d-b2af-bc566574fdc6@opensips.org> Hi, Jim! The behavior you are describing is normal: if you kill opensips with SIGKILL, OpenSIPS closes instantly, without being able to run the code that dumps the data into the file. If you want to prevent this, you should always dump the db data into the file (note that this is not very good in terms of performance). To do this, you should turn off the db_text caching mode[1]. [1] http://www.opensips.org/html/docs/modules/2.2.x/db_text#id293561 Best regards, R?zvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 06/23/2016 05:39 PM, Jim DeVito wrote: > Hi All. > > 2.2.0.rc2.20160617.c1aa55e-1.el7 > > modparam("dialog", "db_url", "text:///var/lib/opensips/dbtext") > modparam("dialog", "db_mode", 1) > modparam("dialog", "table_name", "dialog") > > db_mode 1 should be realtime correct? The DB is only being synced from > memory on clean shutdown. If I kill -9 the process the dialog info is > lost. Thoughts? > > Thanks!! > From jim at devito.cc Fri Jun 24 13:08:36 2016 From: jim at devito.cc (Jim DeVito) Date: Fri, 24 Jun 2016 07:08:36 -0400 Subject: [OpenSIPS-Users] Dialog DB backend only updates on clean shutdown not crash In-Reply-To: <1d15ec82-1e1a-952d-b2af-bc566574fdc6@opensips.org> Message-ID: <0ee37775-c991-4c93-9a85-b47b8c770cd5@email.android.com> An HTML attachment was scrubbed... URL: From john.nash778 at gmail.com Fri Jun 24 14:56:26 2016 From: john.nash778 at gmail.com (John Nash) Date: Fri, 24 Jun 2016 18:26:26 +0530 Subject: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc In-Reply-To: <7C2F3A12B1504B40BB32EE2A92895B76@LenovoPC> References: <576BF20B.3060301@uphreak.com> <576C1618.5010300@uphreak.com> <576C2AE8.2050604@uphreak.com> <576C34BC.4060403@uphreak.com> <576C39E5.8090607@uphreak.com> <7C2F3A12B1504B40BB32EE2A92895B76@LenovoPC> Message-ID: Yes fingerprints are different in Invite and session progress. On Fri, Jun 24, 2016 at 3:00 AM, sevpal wrote: > Take a look at the ?fingerprint:? line. > > *From:* John Nash > *Sent:* Thursday, June 23, 2016 3:42 PM > *To:* OpenSIPS users mailling list > *Subject:* Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc > > Actually the issue is i hear no audio on either side and just after > session progress (I guess when media starts coming from remote media > server) i see error "SRTP output wanted, but no crypto suite was > negotiated" > > I had also checked media logs i could see RTP packets being sent from > freeswitch to RTPengine IP but there was no packet at all just after that. > Ideally after RTP packet from freeswitch to rtpengine, Rtpengine should > send that packet to browser using wss? > > On Fri, Jun 24, 2016 at 1:05 AM, Eric Tamme wrote: > >> So - i dont see a problem here - Chrome is getting UDP/TLS/RTP/SAVPF and >> Freeswitch is getting RTP/AVP. Freeswitch responded to the offer in the >> invite with an answer in the 183, and in the 200. What is the failure you >> are seeing, and where is it happening (in freeswitch? in the browser?) >> >> The only thing that looks bad is that you are retransmitting the ACK >> which FS either ... doesnt like, or is never getting, because it keeps >> retransmitting the 200, which is why you get a 481 when you send BYE. >> >> -Eric >> >> >> On 06/23/2016 01:24 PM, John Nash wrote: >> >> OK here is the log >> https://gist.github.com/johnnash13/0d2cb5238f3551cd3a8c6b4e638dd744 >> >> Sorry took me a while to convert wireshark trace to text file. >> >> My freeswitch is running on private IP (127.0.0.1) and opensips I run on >> both public and private so that for outside world opensips is the only >> public IP they see. In proxy log I pasted Opensips ===> Freeswitch logs and >> back. >> >> >> >> >> >> >> On Fri, Jun 24, 2016 at 12:43 AM, Eric Tamme wrote: >> >>> No - it's annoying to look at a trace that's had information removed and >>> try and piece together whats happening. Your paranoid side is wrong, sorry. >>> >>> -Eric >>> >>> >>> On 06/23/2016 01:06 PM, Patrick Wakano wrote: >>> >>> my paranoic side would recommend to hide/change private informations, >>> specially any authentication line that might appear... this is certainly a >>> sort of social engineering threat we should worry... >>> better be safe than sorry.... >>> >>> >>> On Thu, Jun 23, 2016 at 3:31 PM, Eric Tamme wrote: >>> >>>> I mean you can use a private gist, but you will be publishing the link >>>> in a public email list. In general I personally dont believe revealing ip >>>> addresses etc. is any problem - to put my money where my mouth is here is a >>>> gist link to an unaltered SIP trace on my server :) >>>> >>>> https://gist.github.com/etamme/b864010448a29007b7e0457682e81d52 >>>> >>>> -Eric >>>> >>>> >>>> On 06/23/2016 12:23 PM, John Nash wrote: >>>> >>>> Ok i am ready with logs. About gist may I use private option as traces >>>> have our IPs, user >>>> >>>> On Thu, Jun 23, 2016 at 10:32 PM, Eric Tamme wrote: >>>> >>>>> Hey John, >>>>> >>>>> Please paste a full UNALTERED sip trace into a gist (gist.github.com) >>>>> from the proxy servers perspective and provide a link so that we can see >>>>> what comes in, and what goes out from both sides. >>>>> >>>>> EG: ngrep -qtd any -W byline port 5060 >>>>> >>>>> This will show us the traffic that is leaving the proxy destined for >>>>> the Freeswitch box, and what the freeswitch box sends back. >>>>> >>>>> Also - you can look in your browsers console log and provide the SIP >>>>> trace from there in a seperate gist, so that we can see what opensips sends >>>>> back up to your browser. >>>>> >>>>> -Eric >>>>> >>>>> >>>>> Am I using correct sip.js example? I copied it to my server and >>>>> accessing it using https: (used letsencrypt) >>>>> >>>>> On Thu, Jun 23, 2016 at 7:58 PM, Eric Tamme wrote: >>>>> >>>>>> 1. I would suggest using SIP.js - https://github.com/onsip/SIP.js it >>>>>> is a much more active project that sipml5. >>>>>> >>>>>> 2. Im guessing that you are not properly passing flags to RTPEngine. >>>>>> If you want to have DTLS-SRTP between the browser, and plain RTP/AVP >>>>>> between RTPEngine and freeswitch, you need to "offer" rtp/avp to >>>>>> freeswitch, and "answer" dtls-srtp back up to the browser. >>>>>> >>>>>> the offer to freeswitch would be: >>>>>> >>>>>> $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove"; >>>>>> >>>>>> >>>>>> and the answer back up to the browswer would be: >>>>>> >>>>>> $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force"; >>>>>> >>>>>> >>>>>> -Eric >>>>>> >>>>>> >>>>>> >>>>>> On 06/23/2016 08:20 AM, John Nash wrote: >>>>>> >>>>>> I am following >>>>>> http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 and >>>>>> trying to test a call >>>>>> >>>>>> sipml5 ----------->Opensips + rtpengine --------> SIP end point >>>>>> (Freeswitch) >>>>>> >>>>>> But I do not have any audio on both sides. I see this error at >>>>>> rtpengine log "SRTP output wanted, but no crypto suite was negotiated" >>>>>> >>>>>> Anyone tested this scenario positive? >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>>> >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Users mailing list >>>>>> Users at lists.opensips.org >>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>>> >>>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>> >>> >>> _______________________________________________ >>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > ------------------------------ > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From Agalya_Ramachandran at comcast.com Fri Jun 24 16:00:15 2016 From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor)) Date: Fri, 24 Jun 2016 14:00:15 +0000 Subject: [OpenSIPS-Users] DNS-SRV query in opensips In-Reply-To: <576AAF59.8090305@opensips.org> References: <955ec37716d94d139c82be2835377bd5@COPDCEX28.cable.comcast.com> <5768EF90.2030802@opensips.org> <576A3B05.8030702@opensips.org> <45ada32d906e44559fb82fe37d6cb1c3@COPDCEX28.cable.comcast.com> <576AAF59.8090305@opensips.org> Message-ID: <73ebf60ebe214a129c9d4245d5b70629@COPDCEX28.cable.comcast.com> Thank a lot Bogdan. I got it. Regards, Agalya From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Wednesday, June 22, 2016 11:32 AM To: Ramachandran, Agalya (Contractor) ; users at lists.opensips.org Subject: Re: [OpenSIPS-Users] DNS-SRV query in opensips Hi, They are different. RURI is the part in request's first line, while the DURI is a an outbound proxy used just to finding the destination at network level (it will not be present in the SIP request). by using the set functions you do not create a new branch, you are just changing the RURI and DURI from the default branch. To create a new branch you have to use append_branch(). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 22.06.2016 17:13, Ramachandran, Agalya (Contractor) wrote: Hi Bogdan, Request URI and destination URI are one and the same right? Basically what I understand from the below mail is , we can use both seturi and setdsturi to set a new branch. Am I right? Regards, Agalya From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Wednesday, June 22, 2016 3:15 AM To: Ramachandran, Agalya (Contractor) Cc: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] DNS-SRV query in opensips Hi Agalya, seturi() and setdsturi() set the RURI / DestinationURI for the current message / branch. When you do an append_branch() a new branch is stored for serial/parallel forking (note that the current branch does not changes - this is branch number 1). So, append_branch() will make a copy of the current branch (RURI, DURI, PATH, Forced Socket, etc) and store a new branch for later forking. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 21.06.2016 21:08, Ramachandran, Agalya (Contractor) wrote: Hi Bogdan, I have a question regarding seturi and setdsturi function calls. As far as my understanding, when append_branch() is called, seturi () is called to set the URI where to fork the call. I tried by calling only seturi () function call, after append_branch it was working same behavior as when I used setdsturi() as well. My question is do we really need setdsturi or when is the case when setdsturi() is used.? Regards, Agalya From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Tuesday, June 21, 2016 3:41 AM To: OpenSIPS users mailling list >; Ramachandran, Agalya (Contractor) >; Ramachandran, Agalya (Contractor) > Subject: Re: [OpenSIPS-Users] DNS-SRV query in opensips Hi Agalya, OpenSIPS does full flavor DNS lookup (with NATPR and SRV), but this is internal, and not accessible from script. OpenSIPS implements auto DNS-based failover : http://www.opensips.org/html/docs/modules/2.2.x/tm.html#id293694 My understanding is you want the DNS resolving to be done at script level and to have access to the results ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 20.06.2016 20:35, Ramachandran, Agalya (Contractor) wrote: Hi team, We are using opensips for our project requirements. I have a scenario where we need DNS-SRV query and the result of this should be placed as the desturi to send request out.(In case of forking call) As far as I went through opensips documentation, there are some core parameters for dns related config such as "dns_retr_time" ,"dns_retr_time" ,"dns_servers_no" etc... According to my understanding these config variables can be declared and used in the opensips.config file to control the settings of DNS query. Is there any available function where I can use and pass the DNS server domain name, so that it fetches the IP address of the host ? Please let us know what is the best way to achieve this? Regards, Agalya _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From Agalya_Ramachandran at comcast.com Fri Jun 24 23:02:53 2016 From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor)) Date: Fri, 24 Jun 2016 21:02:53 +0000 Subject: [OpenSIPS-Users] CURL library with respect to REST_API calls In-Reply-To: <576BB123.7020102@opensips.org> References: <1dc72c09449b413a84275d01bcff3548@COPDCEX28.cable.comcast.com> <576BB123.7020102@opensips.org> Message-ID: Hi Liviu, I have not changed anything in the write_func(). Also I tried by adding some debug statements. I observed that only in the case of success write_func() is being called and I get the debug statements in my logs. In the case of failure, the callback function is never called. This is what I observe. If you try for failure case, let me know how it worked for you. Regards, Agalya From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Liviu Chircu Sent: Thursday, June 23, 2016 5:52 AM To: users at lists.opensips.org Subject: Re: [OpenSIPS-Users] CURL library with respect to REST_API calls Hi, Ramachandran! I read the "CURLOPT_WRITEFUNCTION" section [1] one more time, and it really looks like it should pass the body of the reply to the "write_func" callback we register before sending the HTTP PUT, even if we got an error code (3XX or higher). Did you change anything in the "write_func()"? Notice how their docs say that if a proper "len" is not returned, the transfer will be aborted. Apart from that, I have no other ideas for now but to try and fetch the body myself on an error HTTP ret code of a CUROPT_PUT operation, see how (or if) it works for me, and give you more feedback. [1]: https://curl.haxx.se/libcurl/c/CURLOPT_WRITEFUNCTION.html All the best, Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 22.06.2016 20:19, Ramachandran, Agalya (Contractor) wrote: Hi team, Any one has any clue on the below topic? Regards, Agalya From: Ramachandran, Agalya (Contractor) Sent: Monday, June 20, 2016 1:45 PM To: OpenSIPS users mailling list Subject: CURL library with respect to REST_API calls Hi team, I have a question regarding curl library behavior with respect to curl_easy_perform API call. Here is the snippet of the code that am using in "rest_put" API call in rest_methods.c file w_curl_easy_setopt(handle, CURLOPT_WRITEFUNCTION, write_func); w_curl_easy_setopt(handle, CURLOPT_WRITEDATA, &res_body); When curl_easy_perform API call is success, I could able to retrieve the result body from the res_body. But in the case of API call failure am not getting any details of the message. But getting only the http response code. Is there a way to get the message details as well in the case where curl_easy_perform API fails? LM_INFO(" Actual result body is %s\n", res_body.s); When I print this, in the case of success, am getting a http response message in detail. But in case of failure, the call back function write_func is not at all called. Why it is so? Please guide me if there is a way to the message details in case of failure too. Regards, Agalya _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at inin.com Sat Jun 25 02:41:34 2016 From: Ben.Newlin at inin.com (Newlin, Ben) Date: Sat, 25 Jun 2016 00:41:34 +0000 Subject: [OpenSIPS-Users] ACK after set_advertised_address contains wrong address in VIA header Message-ID: I have run into the same problem that was described in this previous post [1], however it doesn?t appear it was ever solved at the time. I am using the dispatcher module to route calls to external carriers and I am using set_advertised_address to set the outgoing public address prior to sending the request. If the first destination returns failure, the ACK is sent correctly. Then I select a different destination and set a different public address using set_advertised_address. If this second call also fails, the ACK that is sent out uses the first advertised address, not the current on for the request. Has anyone figured this out? I am using 1.11.6. [1] http://lists.opensips.org/pipermail/users/2014-August/029779.html Ben Newlin -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Jun 27 11:37:51 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 27 Jun 2016 12:37:51 +0300 Subject: [OpenSIPS-Users] ACK after set_advertised_address contains wrong address in VIA header In-Reply-To: References: Message-ID: <5770F3EF.2040201@opensips.org> Hi Ben, Where in the script do you do the first advertise_address ? In the request route or in a branch route ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 25.06.2016 03:41, Newlin, Ben wrote: > > I have run into the same problem that was described in this previous > post [1], however it doesn?t appear it was ever solved at the time. > > I am using the dispatcher module to route calls to external carriers > and I am using set_advertised_address to set the outgoing public > address prior to sending the request. If the first destination returns > failure, the ACK is sent correctly. Then I select a different > destination and set a different public address using > set_advertised_address. If this second call also fails, the ACK that > is sent out uses the first advertised address, not the current on for > the request. > > Has anyone figured this out? I am using 1.11.6. > > [1] http://lists.opensips.org/pipermail/users/2014-August/029779.html > > Ben Newlin > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at inin.com Mon Jun 27 14:45:15 2016 From: Ben.Newlin at inin.com (Newlin, Ben) Date: Mon, 27 Jun 2016 12:45:15 +0000 Subject: [OpenSIPS-Users] ACK after set_advertised_address contains wrong address in VIA header In-Reply-To: <5770F3EF.2040201@opensips.org> References: <5770F3EF.2040201@opensips.org> Message-ID: <739103AC-2405-47B1-A37F-EB034BD62598@inin.com> I always set the advertised address in request route. Also as the original issue noted the second INVITE does go out with the correct advertised address in the VIA. It is only the local ACK for the failed second request that contains the wrong address in the VIA. So set_advertised_address appears to be working, but the local generated ACK is not using that address. Ben Newlin From: Bogdan-Andrei Iancu Date: Monday, June 27, 2016 at 5:37 AM To: "users at lists.opensips.org" , "Newlin, Ben" Subject: Re: [OpenSIPS-Users] ACK after set_advertised_address contains wrong address in VIA header Hi Ben, Where in the script do you do the first advertise_address ? In the request route or in a branch route ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 25.06.2016 03:41, Newlin, Ben wrote: I have run into the same problem that was described in this previous post [1], however it doesn?t appear it was ever solved at the time. I am using the dispatcher module to route calls to external carriers and I am using set_advertised_address to set the outgoing public address prior to sending the request. If the first destination returns failure, the ACK is sent correctly. Then I select a different destination and set a different public address using set_advertised_address. If this second call also fails, the ACK that is sent out uses the first advertised address, not the current on for the request. Has anyone figured this out? I am using 1.11.6. [1] http://lists.opensips.org/pipermail/users/2014-August/029779.html Ben Newlin _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From pimenta at inatel.br Mon Jun 27 15:45:05 2016 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Mon, 27 Jun 2016 13:45:05 +0000 Subject: [OpenSIPS-Users] How to configure opensips to show passwords in column password from table subscriber? Message-ID: Hi. When I execute opensipsctl add user password, the column password in table subscriber remains empty. Ha1 has a kind of encrypted password. How to configure opensips to show passwords in column password from table subscriber? I have tried changing some parameters in module auth_db, but it didn't take effect. So, what is the correct configuration? Any hint will be very helpful! Regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Jun 27 16:41:08 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 27 Jun 2016 17:41:08 +0300 Subject: [OpenSIPS-Users] ACK after set_advertised_address contains wrong address in VIA header In-Reply-To: <739103AC-2405-47B1-A37F-EB034BD62598@inin.com> References: <5770F3EF.2040201@opensips.org> <739103AC-2405-47B1-A37F-EB034BD62598@inin.com> Message-ID: <57713B04.9040204@opensips.org> Hi Ben, If you set the advertised host / port in branch route, it will have impact over the entire transaction (all branches). So, any local replies (CANCEL and ACK) that are constructed by OpenSIPS (for any branch) will use the same set of advertised values. Which is of course wrong. Let us come up with the fix (as idea and code). Could you open a bug report on the GITHUB tracker, please ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 27.06.2016 15:45, Newlin, Ben wrote: > > I always set the advertised address in request route. > > Also as the original issue noted the second INVITE does go out with > the correct advertised address in the VIA. It is only the local ACK > for the failed second request that contains the wrong address in the > VIA. So set_advertised_address appears to be working, but the local > generated ACK is not using that address. > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > *Date: *Monday, June 27, 2016 at 5:37 AM > *To: *"users at lists.opensips.org" , "Newlin, > Ben" > *Subject: *Re: [OpenSIPS-Users] ACK after set_advertised_address > contains wrong address in VIA header > > Hi Ben, > > Where in the script do you do the first advertise_address ? In the > request route or in a branch route ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 25.06.2016 03:41, Newlin, Ben wrote: > > I have run into the same problem that was described in this > previous post [1], however it doesn?t appear it was ever solved at > the time. > > I am using the dispatcher module to route calls to external > carriers and I am using set_advertised_address to set the outgoing > public address prior to sending the request. If the first > destination returns failure, the ACK is sent correctly. Then I > select a different destination and set a different public address > using set_advertised_address. If this second call also fails, the > ACK that is sent out uses the first advertised address, not the > current on for the request. > > Has anyone figured this out? I am using 1.11.6. > > [1] http://lists.opensips.org/pipermail/users/2014-August/029779.html > > Ben Newlin > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at inin.com Mon Jun 27 17:06:42 2016 From: Ben.Newlin at inin.com (Newlin, Ben) Date: Mon, 27 Jun 2016 15:06:42 +0000 Subject: [OpenSIPS-Users] ACK after set_advertised_address contains wrong address in VIA header In-Reply-To: <57713B04.9040204@opensips.org> References: <5770F3EF.2040201@opensips.org> <739103AC-2405-47B1-A37F-EB034BD62598@inin.com> <57713B04.9040204@opensips.org> Message-ID: Did you mean to say if you set it in request route? I should clarify that when I am setting the advertised address the second time it is of course happening in failure_route as the first request has failed at that point. Perhaps that is the issue? I will open a bug. Thanks. Ben Newlin From: Bogdan-Andrei Iancu Date: Monday, June 27, 2016 at 10:41 AM To: "Newlin, Ben" , "users at lists.opensips.org" Subject: Re: [OpenSIPS-Users] ACK after set_advertised_address contains wrong address in VIA header Hi Ben, If you set the advertised host / port in branch route, it will have impact over the entire transaction (all branches). So, any local replies (CANCEL and ACK) that are constructed by OpenSIPS (for any branch) will use the same set of advertised values. Which is of course wrong. Let us come up with the fix (as idea and code). Could you open a bug report on the GITHUB tracker, please ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 27.06.2016 15:45, Newlin, Ben wrote: I always set the advertised address in request route. Also as the original issue noted the second INVITE does go out with the correct advertised address in the VIA. It is only the local ACK for the failed second request that contains the wrong address in the VIA. So set_advertised_address appears to be working, but the local generated ACK is not using that address. Ben Newlin From: Bogdan-Andrei Iancu Date: Monday, June 27, 2016 at 5:37 AM To: "users at lists.opensips.org" , "Newlin, Ben" Subject: Re: [OpenSIPS-Users] ACK after set_advertised_address contains wrong address in VIA header Hi Ben, Where in the script do you do the first advertise_address ? In the request route or in a branch route ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 25.06.2016 03:41, Newlin, Ben wrote: I have run into the same problem that was described in this previous post [1], however it doesn?t appear it was ever solved at the time. I am using the dispatcher module to route calls to external carriers and I am using set_advertised_address to set the outgoing public address prior to sending the request. If the first destination returns failure, the ACK is sent correctly. Then I select a different destination and set a different public address using set_advertised_address. If this second call also fails, the ACK that is sent out uses the first advertised address, not the current on for the request. Has anyone figured this out? I am using 1.11.6. [1] http://lists.opensips.org/pipermail/users/2014-August/029779.html Ben Newlin _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at inin.com Mon Jun 27 19:02:40 2016 From: Ben.Newlin at inin.com (Newlin, Ben) Date: Mon, 27 Jun 2016 17:02:40 +0000 Subject: [OpenSIPS-Users] ACK after set_advertised_address contains wrong address in VIA header In-Reply-To: References: <5770F3EF.2040201@opensips.org> <739103AC-2405-47B1-A37F-EB034BD62598@inin.com> <57713B04.9040204@opensips.org> Message-ID: <4E5A77D0-3095-4A78-93A4-C922A5B5862B@inin.com> I have opened issue #917 on Github [1]. I should mention that I have worked around this problem by deciding to use Record-Routes instead of set_advertised_address. It is much cleaner, even though it does expose my private IPs. However, I have kept this configuration in case you need me to perform any tests or get tracing/logs. Thanks! [1] https://github.com/OpenSIPS/opensips/issues/917 Ben Newlin From: "Newlin, Ben" Date: Monday, June 27, 2016 at 11:06 AM To: Bogdan-Andrei Iancu , "users at lists.opensips.org" Subject: Re: [OpenSIPS-Users] ACK after set_advertised_address contains wrong address in VIA header Did you mean to say if you set it in request route? I should clarify that when I am setting the advertised address the second time it is of course happening in failure_route as the first request has failed at that point. Perhaps that is the issue? I will open a bug. Thanks. Ben Newlin From: Bogdan-Andrei Iancu Date: Monday, June 27, 2016 at 10:41 AM To: "Newlin, Ben" , "users at lists.opensips.org" Subject: Re: [OpenSIPS-Users] ACK after set_advertised_address contains wrong address in VIA header Hi Ben, If you set the advertised host / port in branch route, it will have impact over the entire transaction (all branches). So, any local replies (CANCEL and ACK) that are constructed by OpenSIPS (for any branch) will use the same set of advertised values. Which is of course wrong. Let us come up with the fix (as idea and code). Could you open a bug report on the GITHUB tracker, please ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 27.06.2016 15:45, Newlin, Ben wrote: I always set the advertised address in request route. Also as the original issue noted the second INVITE does go out with the correct advertised address in the VIA. It is only the local ACK for the failed second request that contains the wrong address in the VIA. So set_advertised_address appears to be working, but the local generated ACK is not using that address. Ben Newlin From: Bogdan-Andrei Iancu Date: Monday, June 27, 2016 at 5:37 AM To: "users at lists.opensips.org" , "Newlin, Ben" Subject: Re: [OpenSIPS-Users] ACK after set_advertised_address contains wrong address in VIA header Hi Ben, Where in the script do you do the first advertise_address ? In the request route or in a branch route ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 25.06.2016 03:41, Newlin, Ben wrote: I have run into the same problem that was described in this previous post [1], however it doesn?t appear it was ever solved at the time. I am using the dispatcher module to route calls to external carriers and I am using set_advertised_address to set the outgoing public address prior to sending the request. If the first destination returns failure, the ACK is sent correctly. Then I select a different destination and set a different public address using set_advertised_address. If this second call also fails, the ACK that is sent out uses the first advertised address, not the current on for the request. Has anyone figured this out? I am using 1.11.6. [1] http://lists.opensips.org/pipermail/users/2014-August/029779.html Ben Newlin _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From pimenta at inatel.br Mon Jun 27 20:05:32 2016 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Mon, 27 Jun 2016 18:05:32 +0000 Subject: [OpenSIPS-Users] I have a Patch that fixes memory leak on OpenSIPS. How to apply this path via github? In-Reply-To: References: Message-ID: Dear OpenSIPS users, Daniel F?ssia, from Inatel Competence Center (www.inatel.br) has discovered some issues related to the code in OpenSIPS 2.2 that handles some transactions in SQLite. He also has proposed the solution for such issues and his work is attached on this message. How could I resquet to the OpenSIPS development team to apply this fix? That is, can someone here give me the instructions on how to use github and request that fix? I? very new on github. Any hint will be very helpful! Thanks alot! RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 Brazil ________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: 0001-Fix-memory-leak-after-sqlite-prepare-was-deleted-stm.patch Type: application/octet-stream Size: 6009 bytes Desc: 0001-Fix-memory-leak-after-sqlite-prepare-was-deleted-stm.patch URL: From tito at xsvoce.com Mon Jun 27 20:20:21 2016 From: tito at xsvoce.com (Tito Cumpen) Date: Mon, 27 Jun 2016 14:20:21 -0400 Subject: [OpenSIPS-Users] running sip tls on 443 Message-ID: Group, I am experiencing strange behavior when configuring sip tls on port 443. At time opensips crashes or stops accepting new connections. Here are the tcp configs I am using: #disable_tcp=no tcp_connection_lifetime=3600 tcp_connect_timeout=3 tcp_keepidle = 30 tcp_keepinterval = 5 tcp_keepalive = 1 tcp_keepcount = 5 tcp_max_msg_time = 8 tcp_children=10 Any idea what would case this? I am assuming there are probes out in the internet that eventually make opensips crash? Thanks, Tito -------------- next part -------------- An HTML attachment was scrubbed... URL: From tito at xsvoce.com Mon Jun 27 20:29:44 2016 From: tito at xsvoce.com (Tito Cumpen) Date: Mon, 27 Jun 2016 14:29:44 -0400 Subject: [OpenSIPS-Users] opensips 2.3 crash In-Reply-To: <5728DBD9.7070403@opensips.org> References: <5728DBD9.7070403@opensips.org> Message-ID: Bogdan, The QM_malloc option specified in this document does not exist.[image: Inline image 1] Was it replaced with On Tue, May 3, 2016 at 1:11 PM, Bogdan-Andrei Iancu wrote: > Hi Tito, > > You should have reported it sooner:) > Could you compile in the memory debugger (as the backtrace you posted here > points to a memory corruption). Use DBG_MALLOC flag in combination with F_ > / HP_ / QM_ memeory managers : > http://www.opensips.org/Documentation/TroubleShooting-OutOfMem > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 03.05.2016 20:00, Tito Cumpen wrote: > > Group, > > > I get about two nightly crashes in the latest opensips head this has been > happening consistently for about a week and half now. All crashes are > identical to: > > Using host libthread_db library "/lib64/libthread_db.so.1". > > Core was generated by `/sbin/opensips -P /var/run/opensips.pid -u opensips > -g opensips -M 1024 -f /etc'. > > Program terminated with signal 11, Segmentation fault. > > #0 0x0000000000512512 in fm_remove_free (n=0x7f384a06b680, > qm=0x7f3849fc8010) at mem/f_malloc.c:187 > > 187 *pf=n->u.nxt_free; > > Missing separate debuginfos, use: debuginfo-install > cyrus-sasl-lib-2.1.26-20.el7_2.x86_64 glibc-2.17-106.el7_2.4.x86_64 > gmp-6.0.0-12.el7_1.x86_64 gnutls-3.3.8-14.el7_2.x86_64 > keyutils-libs-1.5.8-3.el7.x86_64 krb5-libs-1.13.2-12.el7_2.x86_64 > libcom_err-1.42.9-7.el7.x86_64 libcurl-7.29.0-25.el7.centos.x86_64 > libffi-3.0.13-16.el7.x86_64 libgcc-4.8.5-4.el7.x86_64 > libgcrypt-1.5.3-12.el7_1.1.x86_64 libgpg-error-1.12-3.el7.x86_64 > libidn-1.28-4.el7.x86_64 libmicrohttpd-0.9.33-2.el7.x86_64 > librabbitmq-0.5.2-1.el7.x86_64 libselinux-2.2.2-6.el7.x86_64 > libssh2-1.4.3-10.el7_2.1.x86_64 libstdc++-4.8.5-4.el7.x86_64 > libtasn1-3.8-2.el7.x86_64 nettle-2.7.1-4.el7.x86_64 > nspr-4.11.0-1.el7_2.x86_64 nss-3.21.0-9.el7_2.x86_64 > nss-softokn-freebl-3.16.2.3-14.2.el7_2.x86_64 > nss-util-3.21.0-2.2.el7_2.x86_64 openldap-2.4.40-9.el7_2.x86_64 > openssl-libs-1.0.1e-51.el7_2.4.x86_64 p11-kit-0.20.7-3.el7.x86_64 > pcre-8.32-15.el7.x86_64 trousers-0.3.13-1.el7.x86_64 > xz-libs-5.1.2-12alpha.el7.x86_64 zlib-1.2.7-15.el7.x86_64 > > (gdb) bt full > > #0 0x0000000000512512 in fm_remove_free (n=0x7f384a06b680, > qm=0x7f3849fc8010) at mem/f_malloc.c:187 > > pf = 0x0 > > hash = 2051 > > #1 fm_malloc (qm=0x7f3849fc8010, size=size at entry=65592) at > mem/f_malloc.c:415 > > frag = 0x7f384a06b680 > > n = > > hash = 2051 > > __FUNCTION__ = "fm_malloc" > > #2 0x00007f383e018652 in tcp_handle_req (_max_msg_chunks=, > con=, req=) at > ../../net/proto_tcp/tcp_common.h:459 > > local_rcv = {src_ip = {af = 760673648, len = 32766, u = {addrl = > {16, 20}, addr32 = {16, 0, 20, 0}, addr16 = {16, 0, 0, 0, 20, 0, 0, 0}, > addr = "\020\000\000\000\000\000\000\000\024\000\000\000\000\000\000"}}, > dst_ip = {af = 1, len = 1, u = {addrl = { > > 139878494896234, 5869476}, addr32 = {106, 32568, 5869476, > 0}, addr16 = {106, 0, 32568, 0, 36772, 89, 0, 0}, addr = > "j\000\000\000\070\177\000\000\244\217Y\000\000\000\000"}}, src_port = 0, > dst_port = 0, proto = 0, proto_reserved1 = 0, > > proto_reserved2 = 0, src_su = {s = {sa_family = 62576, sa_data = > "V-\376\177\000\000\001\000\000\000\000\000\000"}, sin = {sin_family = > 62576, sin_port = 11606, sin_addr = {s_addr = 32766}, sin_zero = > "\001\000\000\000\000\000\000"}, sin6 = { > > sin6_family = 62576, sin6_port = 11606, sin6_flowinfo = > 32766, sin6_addr = {__in6_u = {__u6_addr8 = > "\001\000\000\000\000\000\000\000\200\364V-\376\177\000", __u6_addr16 = {1, > 0, 0, 0, 62592, 11606, 32766, 0}, __u6_addr32 = {1, 0, 760673408, 32766}}}, > > sin6_scope_id = 24}}, bind_address = 0x4a03e63000000001} > > msg_buf = > > msg_len = > > c = > > size = > > #3 tls_read_req (con=0x7f38481398c0, bytes_read=0x7ffe2d56f550) at > proto_tls.c:441 > > bytes = > > total_bytes = 0 > > req = > > __FUNCTION__ = "tls_read_req" > > #4 0x000000000059c478 in handle_io (fm=fm at entry=0x7f384a03e630, > idx=idx at entry=3, event_type=event_type at entry=1) at net/net_tcp_proc.c:205 > > ret = 0 > > n = > > con = 0x7f38481398c0 > > s = 106 > > rw = > > resp = > > response = {139879704139968, 1} > > __FUNCTION__ = "handle_io" > > #5 0x000000000059daee in io_wait_loop_epoll (h=, > t=, repeat=) at net/../io_wait_loop.h:221 > > ret = > > e = > > n = 1 > > r = 4 > > #6 tcp_worker_proc (unix_sock=) at net/net_tcp_proc.c:312 > > __FUNCTION__ = "tcp_worker_proc" > > #7 0x00000000005a80c3 in tcp_start_processes (chd_rank=chd_rank at entry=0x841360 > , startup_done=startup_done at entry=0x7f3848136888) at > net/net_tcp.c:1758 > > r = 2 > > reader_fd = {98, 100} > > pid = 0 > > load_p = 0x7f38481374e8 > > __FUNCTION__ = "tcp_start_processes" > > #8 0x0000000000419f05 in main_loop () at main.c:677 > > startup_done = 0x7f3848136888 > > chd_rank = 15 > > #9 main (argc=, argv=) at main.c:1258 > > cfg_stream = > > c = > > r = > > tmp = 0x7ffe2d56ff65 "" > > tmp_len = > > port = > > proto = > > protos_no = > > options = 0x5daf00 "f:cCm:M:b:l:n:N:rRvdDFETSVhw:t:u:g:P:G:W:o:" > > ret = -1 > > seed = 3936233749 > > rfd = > > __FUNCTION__ = "main" > > > > Please advise if anything else is needed. > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: Screen Shot 2016-06-27 at 2.26.11 PM.png Type: image/png Size: 11509 bytes Desc: not available URL: From razvan at opensips.org Tue Jun 28 09:19:51 2016 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Tue, 28 Jun 2016 10:19:51 +0300 Subject: [OpenSIPS-Users] I have a Patch that fixes memory leak on OpenSIPS. How to apply this path via github? In-Reply-To: References: Message-ID: Hi, Rodrigo! Please fork opensips and open a pull request on Github[1]. The idea is simple: 1. fork the repository[2] 2. Apply the patch, commit it and push it in your fork 3. Open a pull request[3] [1] https://github.com/OpenSIPS/opensips/pulls [2] https://help.github.com/articles/fork-a-repo/ [3] https://help.github.com/articles/using-pull-requests/#initiating-the-pull-request PS: it is not a good idea to attach a file on a mailing list. Use gist.github.com, or pastebin.com next time :). Thanks and regards, R?zvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 06/27/2016 09:05 PM, Rodrigo Pimenta Carvalho wrote: > > > Dear OpenSIPS users, > > > Daniel F?ssia, from Inatel Competence Center (www.inatel.br > ) has discovered some issues related to the code > in OpenSIPS 2.2 that handles some transactions in SQLite. He also has > proposed the solution for such issues and his work is attached on this > message. > > > How could I resquet to the OpenSIPS development team to apply this > fix? That is, can someone here give me the instructions on how to use > github and request that fix? I? very new on github. > > > Any hint will be very helpful! > > > Thanks alot! > > > RODRIGO PIMENTA CARVALHO > Inatel Competence Center > Software > Ph: +55 35 3471 9200 RAMAL 979 > Brazil > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From pimenta at inatel.br Wed Jun 22 14:13:57 2016 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Wed, 22 Jun 2016 12:13:57 +0000 Subject: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? In-Reply-To: <019701d1cc71$ca9cf5f0$5fd6e1d0$@democon.be> References: <81fd240b-6369-15b4-a2ee-99deed334299@opensips.org> <4B53E496-10F8-4824-B4A3-4BF57FEDC242@inin.com>, <019701d1cc71$ca9cf5f0$5fd6e1d0$@democon.be> Message-ID: Hi Johan and Ben. Yes. AVPops is a easy solution. However, it is easy for dada stored in DB. What about data stored in RAM? I'm using db_mode = 0 for module usrloc (so user location is always in RAM). So, if AVPops could extract data from the RAM too, as it does with queries and DB, it would be very easy. I have looked for a solution using avp_db_query, but it works only over DB, not over RAM. That is why I started trying to use function lookup() and avp attr, to get caller specific information. Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________ De: users-bounces at lists.opensips.org em nome de johan de clercq Enviado: quarta-feira, 22 de junho de 2016 07:35 Para: 'OpenSIPS users mailling list'; 'sevpal' Assunto: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? Ben is correct. In my opinion, a very easy solution. From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Newlin, Ben Sent: Tuesday, June 21, 2016 5:24 PM To: sevpal ; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? It also seems like AVPOPS module [1] may be a good solution here as it has functions to pull data from a database into AVPs based by user. [1] http://www.opensips.org/html/docs/modules/2.2.x/avpops.html Ben Newlin From: > on behalf of sevpal > Reply-To: sevpal >, OpenSIPS users mailling list > Date: Tuesday, June 21, 2016 at 11:20 AM To: OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? Hi, have you tried/considered running a simple query on the database and parsing for the information you need? From: Rodrigo Pimenta Carvalho Sent: Tuesday, June 21, 2016 10:39 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? Hi R?zvan. I have tried that idea. But that didn't work. The SIP INVITE message is being changed by the OpenSIPS in a wrong way, in my point of view. Do you know some way to save the entire SIP INVITE message before calling lookup() and then make the saved message take place after the lookup() execution? My original message is: INVITE sip:6001 at myDomain.com.br SIP/2.0 Via: SIP/2.0/TCP 192.168.21.40:5090;rport;branch=z9hG4bK876727215 From: ;tag=179920819 To: Call-ID: 1410250893 CSeq: 21 INVITE Contact: Proxy-Authorization: Digest username="crdphmacl_SPnuV5xqtnSX", realm="localhost", nonce="5769458c000001cc263a7c0d6995dc48d42288ec6f8e4048", uri="sip:6001 at myDomain.com.br", response="0f4c122d2a0a28dea6194c235cd77430", algorithm=MD5 Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: Linphone/3.6.1 (eXosip2/4.0.0) Subject: Phone call Content-Length: 227 This is being changed to: INVITE sip:crdphmacl_SPnuV5xqtnSX at 131.221.240.204:60672;transport=tcp;line=c6356a7d87d6f81 SIP/2.0 Record-Route: Via: SIP/2.0/TCP myDomain.com.br:5060;branch=z9hG4bKe2db.49d54587.0;i=1 Via: SIP/2.0/TCP 192.168.21.40:5090;received=xxx.yyy.240.204;rport=60672;branch=z9hG4bK716249970 From: ;tag=12586028 To: Call-ID: 1106771604 CSeq: 21 INVITE Contact: Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: Linphone/3.6.1 (eXosip2/4.0.0) Subject: Phone call Content-Length: 224 So, the caller is receiving its own SIP INVITE. That is why when A calls B, is A that rings, not B. It is becoming a bit complicated. So, I suspect I'm going to the incorrect direction.... Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________ De: users-bounces at lists.opensips.org > em nome de R?zvan Crainea > Enviado: ter?a-feira, 21 de junho de 2016 04:24 Para: users at lists.opensips.org Assunto: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? Hi, Rodrigo! Have you tried restoring the R-URI after the caller lookup? Something like: $var(ru) = $ru; lookup("location", "", "$fu"); # this takes the caller from FROM uri, which I think is more suitable than from contact uri $ru = $var(ru); # continue your processing here .... # now do the real lookup for the callee lookup("location"); Don't do the lookups in the reversed way, because you might loose some contacts. Best regards, R?zvan Crainea OpenSIPS Solutions www.opensips-solutions.com Home - OpenSIPS Solutions www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 06/20/2016 09:02 PM, Rodrigo Pimenta Carvalho wrote: Dear OpenSIPS-users, The table location has the column attr where I use to store specific additional information for each registration. Whenever A calls B, I have to read this specific information from the A record and from the B record. That is, I need to get and handle specific information about the caller and callee. For the callee, I use to invoke the lookup("location") function that put the needed information in the attr_avp. That is good and works very well. Then, I just have to read the attr_avp to get such specific information. For the caller, I use to invoke: $var(aorChamador) = $(ct.fields(uri)); lookup("location","","$var(aorChamador)"); However it causes amazing side effect in the SIP signaling. Ex: When A calls B, B stays quiet and A rings. So A can answer A. Crazy! According to the documentation, lookup will overwritten the Request-URI. I guess that is why the SIP signaling become incoherent. How could I get the caller attr specific information without side effects? Any hint will be very helpful!! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ________________________________ _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From eric at uphreak.com Thu Jun 23 21:54:49 2016 From: eric at uphreak.com (Eric Tamme) Date: Thu, 23 Jun 2016 13:54:49 -0600 Subject: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc In-Reply-To: References: <576BF20B.3060301@uphreak.com> <576C1618.5010300@uphreak.com> <576C2AE8.2050604@uphreak.com> <576C34BC.4060403@uphreak.com> <576C39E5.8090607@uphreak.com> Message-ID: <576C3E89.3050902@uphreak.com> And are you forcing RTPengine to act as an ice light client? It looks like you are gettin a single ICE candidate in the answer back from freeswitch which would indicate that you are. I'd check your chrome webrtc statistics to see if tis failed to do do ice/stun negotiation on the 183. In general the signalling looks good. I think you may have an error on your Freeswitch side - some thing that is trying to force it to use SRTP all the time, even though the signalling has requested plain RTP (to freeswitch). I think you should ask in #freeswitch on freenode at this point. -Eric On 06/23/2016 01:42 PM, John Nash wrote: > Actually the issue is i hear no audio on either side and just after > session progress (I guess when media starts coming from remote media > server) i see error "SRTP output wanted, but no crypto suite was > negotiated" > > I had also checked media logs i could see RTP packets being sent from > freeswitch to RTPengine IP but there was no packet at all just after > that. Ideally after RTP packet from freeswitch to rtpengine, Rtpengine > should send that packet to browser using wss? > > On Fri, Jun 24, 2016 at 1:05 AM, Eric Tamme > wrote: > > So - i dont see a problem here - Chrome is getting > UDP/TLS/RTP/SAVPF and Freeswitch is getting RTP/AVP. Freeswitch > responded to the offer in the invite with an answer in the 183, > and in the 200. What is the failure you are seeing, and where is > it happening (in freeswitch? in the browser?) > > The only thing that looks bad is that you are retransmitting the > ACK which FS either ... doesnt like, or is never getting, because > it keeps retransmitting the 200, which is why you get a 481 when > you send BYE. > > -Eric > > > On 06/23/2016 01:24 PM, John Nash wrote: >> OK here is the log >> https://gist.github.com/johnnash13/0d2cb5238f3551cd3a8c6b4e638dd744 >> >> Sorry took me a while to convert wireshark trace to text file. >> >> My freeswitch is running on private IP (127.0.0.1) and opensips I >> run on both public and private so that for outside world opensips >> is the only public IP they see. In proxy log I pasted Opensips >> ===> Freeswitch logs and back. >> >> >> >> >> >> >> On Fri, Jun 24, 2016 at 12:43 AM, Eric Tamme > > wrote: >> >> No - it's annoying to look at a trace that's had information >> removed and try and piece together whats happening. Your >> paranoid side is wrong, sorry. >> >> -Eric >> >> >> On 06/23/2016 01:06 PM, Patrick Wakano wrote: >>> my paranoic side would recommend to hide/change private >>> informations, specially any authentication line that might >>> appear... this is certainly a sort of social engineering >>> threat we should worry... >>> better be safe than sorry.... >>> >>> >>> On Thu, Jun 23, 2016 at 3:31 PM, Eric Tamme >>> > wrote: >>> >>> I mean you can use a private gist, but you will be >>> publishing the link in a public email list. In general I >>> personally dont believe revealing ip addresses etc. is >>> any problem - to put my money where my mouth is here is >>> a gist link to an unaltered SIP trace on my server :) >>> >>> https://gist.github.com/etamme/b864010448a29007b7e0457682e81d52 >>> >>> -Eric >>> >>> >>> On 06/23/2016 12:23 PM, John Nash wrote: >>>> Ok i am ready with logs. About gist may I use private >>>> option as traces have our IPs, user >>>> >>>> On Thu, Jun 23, 2016 at 10:32 PM, Eric Tamme >>>> > wrote: >>>> >>>> Hey John, >>>> >>>> Please paste a full UNALTERED sip trace into a gist >>>> (gist.github.com ) from the >>>> proxy servers perspective and provide a link so >>>> that we can see what comes in, and what goes out >>>> from both sides. >>>> >>>> EG: ngrep -qtd any -W byline port 5060 >>>> >>>> This will show us the traffic that is leaving the >>>> proxy destined for the Freeswitch box, and what the >>>> freeswitch box sends back. >>>> >>>> Also - you can look in your browsers console log >>>> and provide the SIP trace from there in a seperate >>>> gist, so that we can see what opensips sends back >>>> up to your browser. >>>> >>>> -Eric >>>> >>>> >>>>> Am I using correct sip.js example? I copied it to >>>>> my server and accessing it using https: (used >>>>> letsencrypt) >>>>> >>>>> On Thu, Jun 23, 2016 at 7:58 PM, Eric Tamme >>>>> > wrote: >>>>> >>>>> 1. I would suggest using SIP.js - >>>>> https://github.com/onsip/SIP.js it is a much >>>>> more active project that sipml5. >>>>> >>>>> 2. Im guessing that you are not properly >>>>> passing flags to RTPEngine. If you want to >>>>> have DTLS-SRTP between the browser, and plain >>>>> RTP/AVP between RTPEngine and freeswitch, you >>>>> need to "offer" rtp/avp to freeswitch, and >>>>> "answer" dtls-srtp back up to the browser. >>>>> >>>>> the offer to freeswitch would be: >>>>> >>>>> $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove"; >>>>> >>>>> and the answer back up to the browswer would be: >>>>> >>>>> $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force"; >>>>> >>>>> >>>>> -Eric >>>>> >>>>> >>>>> >>>>> On 06/23/2016 08:20 AM, John Nash wrote: >>>>>> I am following >>>>>> http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 >>>>>> and trying to test a call >>>>>> >>>>>> sipml5 ----------->Opensips + rtpengine >>>>>> --------> SIP end point (Freeswitch) >>>>>> >>>>>> But I do not have any audio on both sides. I >>>>>> see this error at rtpengine log "SRTP output >>>>>> wanted, but no crypto suite was negotiated" >>>>>> >>>>>> Anyone tested this scenario positive? >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Users mailing list >>>>>> Users at lists.opensips.org >>>>>> >>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From alain.bieuzent at free.fr Tue Jun 28 11:14:35 2016 From: alain.bieuzent at free.fr (Alain Bieuzent) Date: Tue, 28 Jun 2016 11:14:35 +0200 Subject: [OpenSIPS-Users] Testing Reason Header Message-ID: Hi all, I need to test if some particular value exist in Reason header on a 404 reply. I receive : SIP/2.0 404 Not Found Via: SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK72222454 From: "+33111111111" ;tag=as005364a4 To: ;tag=SDufib599-019e0074000099e0 Call-ID: 323ca6b15f8404cd3d05d53014336311 at 1.2.3.4:5060 CSeq: 102 INVITE Content-Length: 0 Reason: Q.850;cause=001 I?m trying using this code, but is_present_hf() function return always false. if ( t_check_status("404") ) { ??????????????? if (is_present_hf("Reason:")) xlog("L_WARN","Reason exist"); ??? ????????????????????xlog("L_WARN","Reason not exist"); } Any Idea ? -------------- next part -------------- An HTML attachment was scrubbed... URL: From nick.altmann at gmail.com Tue Jun 28 11:18:53 2016 From: nick.altmann at gmail.com (Nick Altmann) Date: Tue, 28 Jun 2016 12:18:53 +0300 Subject: [OpenSIPS-Users] Testing Reason Header In-Reply-To: References: Message-ID: Use is_present_hf("Reason"), but not is_present_hf("Reason:") -- Nick 2016-06-28 12:14 GMT+03:00 Alain Bieuzent : > Hi all, > > > > I need to test if some particular value exist in Reason header on a 404 > reply. > > > > I receive : > > > > SIP/2.0 404 Not Found > > Via: SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK72222454 > > From: "+33111111111" ;tag=as005364a4 > > To: ;tag=SDufib599-019e0074000099e0 > > Call-ID: 323ca6b15f8404cd3d05d53014336311 at 1.2.3.4:5060 > > CSeq: 102 INVITE > > Content-Length: 0 > > Reason: Q.850;cause=001 > > > > I?m trying using this code, but is_present_hf() function return always > false. > > > > if ( t_check_status("404") ) { > > if (is_present_hf("Reason:")) xlog("L_WARN","Reason > exist"); > > xlog("L_WARN","Reason not exist"); > > } > > > > Any Idea ? > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From alain.bieuzent at free.fr Tue Jun 28 11:41:27 2016 From: alain.bieuzent at free.fr (Alain Bieuzent) Date: Tue, 28 Jun 2016 11:41:27 +0200 Subject: [OpenSIPS-Users] Testing Reason Header In-Reply-To: References: Message-ID: Hi Nick, thanks for response. Allready test without ? : ?, doesn?t work also thanks De : au nom de Nick Altmann R?pondre ? : OpenSIPS users mailling list Date : mardi 28 juin 2016 11:18 ? : OpenSIPS users mailling list Objet : Re: [OpenSIPS-Users] Testing Reason Header Use is_present_hf("Reason"), but not is_present_hf("Reason:") -- Nick 2016-06-28 12:14 GMT+03:00 Alain Bieuzent : Hi all, I need to test if some particular value exist in Reason header on a 404 reply. I receive : SIP/2.0 404 Not Found Via: SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK72222454 From: "+33111111111" ;tag=as005364a4 To: ;tag=SDufib599-019e0074000099e0 Call-ID: 323ca6b15f8404cd3d05d53014336311 at 1.2.3.4:5060 CSeq: 102 INVITE Content-Length: 0 Reason: Q.850;cause=001 I?m trying using this code, but is_present_hf() function return always false. if ( t_check_status("404") ) { if (is_present_hf("Reason:")) xlog("L_WARN","Reason exist"); xlog("L_WARN","Reason not exist"); } Any Idea ? _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jun 28 11:43:52 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 28 Jun 2016 12:43:52 +0300 Subject: [OpenSIPS-Users] ACK after set_advertised_address contains wrong address in VIA header In-Reply-To: References: <5770F3EF.2040201@opensips.org> <739103AC-2405-47B1-A37F-EB034BD62598@inin.com> <57713B04.9040204@opensips.org> Message-ID: <577246D8.4010209@opensips.org> Hi Ben, Thanks for the clarification. I labeled this as a bug, as whatever you do in the script, you cannot get the desired (which is valid) behavior. I got the ticker, we will try to get it fixed shortly. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 27.06.2016 18:06, Newlin, Ben wrote: > > Did you mean to say if you set it in request route? > > I should clarify that when I am setting the advertised address the > second time it is of course happening in failure_route as the first > request has failed at that point. Perhaps that is the issue? > > I will open a bug. Thanks. > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > *Date: *Monday, June 27, 2016 at 10:41 AM > *To: *"Newlin, Ben" , "users at lists.opensips.org" > > *Subject: *Re: [OpenSIPS-Users] ACK after set_advertised_address > contains wrong address in VIA header > > Hi Ben, > > If you set the advertised host / port in branch route, it will have > impact over the entire transaction (all branches). So, any local > replies (CANCEL and ACK) that are constructed by OpenSIPS (for any > branch) will use the same set of advertised values. Which is of course > wrong. Let us come up with the fix (as idea and code). > > Could you open a bug report on the GITHUB tracker, please ? > > Regards, > > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 27.06.2016 15:45, Newlin, Ben wrote: > > I always set the advertised address in request route. > > Also as the original issue noted the second INVITE does go out > with the correct advertised address in the VIA. It is only the > local ACK for the failed second request that contains the wrong > address in the VIA. So set_advertised_address appears to be > working, but the local generated ACK is not using that address. > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > > *Date: *Monday, June 27, 2016 at 5:37 AM > *To: *"users at lists.opensips.org" > , > "Newlin, Ben" > *Subject: *Re: [OpenSIPS-Users] ACK after set_advertised_address > contains wrong address in VIA header > > Hi Ben, > > Where in the script do you do the first advertise_address ? In the > request route or in a branch route ? > > Regards, > > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > On 25.06.2016 03:41, Newlin, Ben wrote: > > I have run into the same problem that was described in this > previous post [1], however it doesn?t appear it was ever > solved at the time. > > I am using the dispatcher module to route calls to external > carriers and I am using set_advertised_address to set the > outgoing public address prior to sending the request. If the > first destination returns failure, the ACK is sent correctly. > Then I select a different destination and set a different > public address using set_advertised_address. If this second > call also fails, the ACK that is sent out uses the first > advertised address, not the current on for the request. > > Has anyone figured this out? I am using 1.11.6. > > [1] > http://lists.opensips.org/pipermail/users/2014-August/029779.html > > Ben Newlin > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jun 28 11:45:47 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 28 Jun 2016 12:45:47 +0300 Subject: [OpenSIPS-Users] ACK after set_advertised_address contains wrong address in VIA header In-Reply-To: <4E5A77D0-3095-4A78-93A4-C922A5B5862B@inin.com> References: <5770F3EF.2040201@opensips.org> <739103AC-2405-47B1-A37F-EB034BD62598@inin.com> <57713B04.9040204@opensips.org> <4E5A77D0-3095-4A78-93A4-C922A5B5862B@inin.com> Message-ID: <5772474B.4090207@opensips.org> Hi Ben, You mean you used the record_route_preset ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 27.06.2016 20:02, Newlin, Ben wrote: > > I have opened issue #917 on Github [1]. > > I should mention that I have worked around this problem by deciding to > use Record-Routes instead of set_advertised_address. It is much > cleaner, even though it does expose my private IPs. However, I have > kept this configuration in case you need me to perform any tests or > get tracing/logs. Thanks! > > [1] https://github.com/OpenSIPS/opensips/issues/917 > > Ben Newlin > > *From: *"Newlin, Ben" > *Date: *Monday, June 27, 2016 at 11:06 AM > *To: *Bogdan-Andrei Iancu , > "users at lists.opensips.org" > *Subject: *Re: [OpenSIPS-Users] ACK after set_advertised_address > contains wrong address in VIA header > > Did you mean to say if you set it in request route? > > I should clarify that when I am setting the advertised address the > second time it is of course happening in failure_route as the first > request has failed at that point. Perhaps that is the issue? > > I will open a bug. Thanks. > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > *Date: *Monday, June 27, 2016 at 10:41 AM > *To: *"Newlin, Ben" , "users at lists.opensips.org" > > *Subject: *Re: [OpenSIPS-Users] ACK after set_advertised_address > contains wrong address in VIA header > > Hi Ben, > > If you set the advertised host / port in branch route, it will have > impact over the entire transaction (all branches). So, any local > replies (CANCEL and ACK) that are constructed by OpenSIPS (for any > branch) will use the same set of advertised values. Which is of course > wrong. Let us come up with the fix (as idea and code). > > Could you open a bug report on the GITHUB tracker, please ? > > Regards, > > > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 27.06.2016 15:45, Newlin, Ben wrote: > > I always set the advertised address in request route. > > Also as the original issue noted the second INVITE does go out > with the correct advertised address in the VIA. It is only the > local ACK for the failed second request that contains the wrong > address in the VIA. So set_advertised_address appears to be > working, but the local generated ACK is not using that address. > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > > *Date: *Monday, June 27, 2016 at 5:37 AM > *To: *"users at lists.opensips.org" > , > "Newlin, Ben" > *Subject: *Re: [OpenSIPS-Users] ACK after set_advertised_address > contains wrong address in VIA header > > Hi Ben, > > Where in the script do you do the first advertise_address ? In the > request route or in a branch route ? > > Regards, > > > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > On 25.06.2016 03:41, Newlin, Ben wrote: > > I have run into the same problem that was described in this > previous post [1], however it doesn?t appear it was ever > solved at the time. > > I am using the dispatcher module to route calls to external > carriers and I am using set_advertised_address to set the > outgoing public address prior to sending the request. If the > first destination returns failure, the ACK is sent correctly. > Then I select a different destination and set a different > public address using set_advertised_address. If this second > call also fails, the ACK that is sent out uses the first > advertised address, not the current on for the request. > > Has anyone figured this out? I am using 1.11.6. > > [1] > http://lists.opensips.org/pipermail/users/2014-August/029779.html > > Ben Newlin > > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jun 28 11:53:03 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 28 Jun 2016 12:53:03 +0300 Subject: [OpenSIPS-Users] opensips 2.3 crash In-Reply-To: References: <5728DBD9.7070403@opensips.org> Message-ID: <577248FF.3020605@opensips.org> Hi Tito, Are you sure there is no error in your local installation ? Starting with 2.2 there is no DBG_QM_MALLOC option as per your snapshot. Be sure you do not have mixture between different OpenSIPS versions. Also, in this link : http://www.opensips.org/Documentation/TroubleShooting-OutOfMem, in "How to handle it" section, you have special paragraph for >= "2.2" . Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 27.06.2016 21:29, Tito Cumpen wrote: > Bogdan, > > > The QM_malloc option specified in this document does not exist.Inline > image 1 > Was it replaced with > > > On Tue, May 3, 2016 at 1:11 PM, Bogdan-Andrei Iancu > > wrote: > > Hi Tito, > > You should have reported it sooner:) > Could you compile in the memory debugger (as the backtrace you > posted here points to a memory corruption). Use DBG_MALLOC flag in > combination with F_ / HP_ / QM_ memeory managers : > http://www.opensips.org/Documentation/TroubleShooting-OutOfMem > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 03.05.2016 20:00, Tito Cumpen wrote: >> Group, >> >> >> I get about two nightly crashes in the latest opensips head this >> has been happening consistently for about a week and half now. >> All crashes are identical to: >> >> Using host libthread_db library "/lib64/libthread_db.so.1". >> >> Core was generated by `/sbin/opensips -P /var/run/opensips.pid -u >> opensips -g opensips -M 1024 -f /etc'. >> >> Program terminated with signal 11, Segmentation fault. >> >> #0 0x0000000000512512 in fm_remove_free (n=0x7f384a06b680, >> qm=0x7f3849fc8010) at mem/f_malloc.c:187 >> >> 187*pf=n->u.nxt_free; >> >> Missing separate debuginfos, use: debuginfo-install >> cyrus-sasl-lib-2.1.26-20.el7_2.x86_64 >> glibc-2.17-106.el7_2.4.x86_64 gmp-6.0.0-12.el7_1.x86_64 >> gnutls-3.3.8-14.el7_2.x86_64 keyutils-libs-1.5.8-3.el7.x86_64 >> krb5-libs-1.13.2-12.el7_2.x86_64 libcom_err-1.42.9-7.el7.x86_64 >> libcurl-7.29.0-25.el7.centos.x86_64 libffi-3.0.13-16.el7.x86_64 >> libgcc-4.8.5-4.el7.x86_64 libgcrypt-1.5.3-12.el7_1.1.x86_64 >> libgpg-error-1.12-3.el7.x86_64 libidn-1.28-4.el7.x86_64 >> libmicrohttpd-0.9.33-2.el7.x86_64 librabbitmq-0.5.2-1.el7.x86_64 >> libselinux-2.2.2-6.el7.x86_64 libssh2-1.4.3-10.el7_2.1.x86_64 >> libstdc++-4.8.5-4.el7.x86_64 libtasn1-3.8-2.el7.x86_64 >> nettle-2.7.1-4.el7.x86_64 nspr-4.11.0-1.el7_2.x86_64 >> nss-3.21.0-9.el7_2.x86_64 >> nss-softokn-freebl-3.16.2.3-14.2.el7_2.x86_64 >> nss-util-3.21.0-2.2.el7_2.x86_64 openldap-2.4.40-9.el7_2.x86_64 >> openssl-libs-1.0.1e-51.el7_2.4.x86_64 p11-kit-0.20.7-3.el7.x86_64 >> pcre-8.32-15.el7.x86_64 trousers-0.3.13-1.el7.x86_64 >> xz-libs-5.1.2-12alpha.el7.x86_64 zlib-1.2.7-15.el7.x86_64 >> >> (gdb) bt full >> >> #0 0x0000000000512512 in fm_remove_free (n=0x7f384a06b680, >> qm=0x7f3849fc8010) at mem/f_malloc.c:187 >> >> pf = 0x0 >> >> hash = 2051 >> >> #1 fm_malloc (qm=0x7f3849fc8010, size=size at entry=65592) at >> mem/f_malloc.c:415 >> >> frag = 0x7f384a06b680 >> >> n = >> >> hash = 2051 >> >> __FUNCTION__ = "fm_malloc" >> >> #2 0x00007f383e018652 in tcp_handle_req >> (_max_msg_chunks=, con=, >> req=) at ../../net/proto_tcp/tcp_common.h:459 >> >> local_rcv = {src_ip = {af = 760673648, len = 32766, u = >> {addrl = {16, 20}, addr32 = {16, 0, 20, 0}, addr16 = {16, 0, 0, >> 0, 20, 0, 0, 0}, addr = >> "\020\000\000\000\000\000\000\000\024\000\000\000\000\000\000"}}, >> dst_ip = {af = 1, len = 1, u = {addrl = { >> >> 139878494896234, 5869476}, addr32 = {106, 32568, >> 5869476, 0}, addr16 = {106, 0, 32568, 0, 36772, 89, 0, 0}, addr = >> "j\000\000\000\070\177\000\000\244\217Y\000\000\000\000"}}, >> src_port = 0, dst_port = 0, proto = 0, proto_reserved1 = 0, >> >> proto_reserved2 = 0, src_su = {s = {sa_family = 62576, >> sa_data = "V-\376\177\000\000\001\000\000\000\000\000\000"}, sin >> = {sin_family = 62576, sin_port = 11606, sin_addr = {s_addr = >> 32766}, sin_zero = "\001\000\000\000\000\000\000"}, sin6 = { >> >> sin6_family = 62576, sin6_port = 11606, >> sin6_flowinfo = 32766, sin6_addr = {__in6_u = {__u6_addr8 = >> "\001\000\000\000\000\000\000\000\200\364V-\376\177\000", >> __u6_addr16 = {1, 0, 0, 0, 62592, 11606, 32766, 0}, __u6_addr32 = >> {1, 0, 760673408, 32766}}}, >> >> sin6_scope_id = 24}}, bind_address = >> 0x4a03e63000000001} >> >> msg_buf = >> >> msg_len = >> >> c = >> >> size = >> >> #3 tls_read_req (con=0x7f38481398c0, bytes_read=0x7ffe2d56f550) >> at proto_tls.c:441 >> >> bytes = >> >> total_bytes = 0 >> >> req = >> >> __FUNCTION__ = "tls_read_req" >> >> #4 0x000000000059c478 in handle_io (fm=fm at entry=0x7f384a03e630, >> idx=idx at entry=3, event_type=event_type at entry=1) at >> net/net_tcp_proc.c:205 >> >> ret = 0 >> >> n = >> >> con = 0x7f38481398c0 >> >> s = 106 >> >> rw = >> >> resp = >> >> response = {139879704139968, 1} >> >> __FUNCTION__ = "handle_io" >> >> #5 0x000000000059daee in io_wait_loop_epoll (h=, >> t=, repeat=) at >> net/../io_wait_loop.h:221 >> >> ret = >> >> e = >> >> n = 1 >> >> r = 4 >> >> #6 tcp_worker_proc (unix_sock=) at >> net/net_tcp_proc.c:312 >> >> __FUNCTION__ = "tcp_worker_proc" >> >> #7 0x00000000005a80c3 in tcp_start_processes >> (chd_rank=chd_rank at entry=0x841360 , >> startup_done=startup_done at entry=0x7f3848136888) at net/net_tcp.c:1758 >> >> r = 2 >> >> reader_fd = {98, 100} >> >> pid = 0 >> >> load_p = 0x7f38481374e8 >> >> __FUNCTION__ = "tcp_start_processes" >> >> #8 0x0000000000419f05 in main_loop () at main.c:677 >> >> startup_done = 0x7f3848136888 >> >> chd_rank = 15 >> >> #9 main (argc=, argv=) at main.c:1258 >> >> cfg_stream = >> >> c = >> >> r = >> >> tmp = 0x7ffe2d56ff65 "" >> >> tmp_len = >> >> port = >> >> proto = >> >> protos_no = >> >> options = 0x5daf00 >> "f:cCm:M:b:l:n:N:rRvdDFETSVhw:t:u:g:P:G:W:o:" >> >> ret = -1 >> >> seed = 3936233749 >> >> rfd = >> >> __FUNCTION__ = "main" >> >> >> >> Please advise if anything else is needed. >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 11509 bytes Desc: not available URL: From bogdan at opensips.org Tue Jun 28 12:03:51 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 28 Jun 2016 13:03:51 +0300 Subject: [OpenSIPS-Users] Testing Reason Header In-Reply-To: References: Message-ID: <57724B87.305@opensips.org> Hi Alain, In failure route, the processed SIP message is the original INVITE - see http://www.opensips.org/Documentation/Script-Routes-2-2#toc3 And you want to get the "Reason" from the reply message , so you should do $(hdr(Reason)) - see http://www.opensips.org/Documentation/Script-CoreVar-2-2 . Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 28.06.2016 12:41, Alain Bieuzent wrote: > > Hi Nick, thanks for response. > > Allready test without ? : ?, doesn?t work also > > thanks > > *De : * au nom de Nick Altmann > > *R?pondre ? : *OpenSIPS users mailling list > *Date : *mardi 28 juin 2016 11:18 > *? : *OpenSIPS users mailling list > *Objet : *Re: [OpenSIPS-Users] Testing Reason Header > > Use is_present_hf("Reason"), but not is_present_hf("Reason:") > > > -- > Nick > > 2016-06-28 12:14 GMT+03:00 Alain Bieuzent >: > > Hi all, > > I need to test if some particular value exist in Reason header on > a 404 reply. > > I receive : > > SIP/2.0 404 Not Found > > Via: SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK72222454 > > From: "+33111111111" >;tag=as005364a4 > > To: >;tag=SDufib599-019e0074000099e0 > > Call-ID: 323ca6b15f8404cd3d05d53014336311 at 1.2.3.4:5060 > > > CSeq: 102 INVITE > > Content-Length: 0 > > Reason: Q.850;cause=001 > > I?m trying using this code, but is_present_hf() function return > always false. > > if ( t_check_status("404") ) { > > if (is_present_hf("Reason:")) > xlog("L_WARN","Reason exist"); > > xlog("L_WARN","Reason not exist"); > > } > > Any Idea ? > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From pimenta at inatel.br Tue Jun 28 14:17:04 2016 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Tue, 28 Jun 2016 12:17:04 +0000 Subject: [OpenSIPS-Users] I have a Patch that fixes memory leak on OpenSIPS. How to apply this path via github? In-Reply-To: References: , Message-ID: Hi Razvan Thank you very much for the instructions and the alert! I will fork and pull that. Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________ De: users-bounces at lists.opensips.org em nome de Razvan Crainea Enviado: ter?a-feira, 28 de junho de 2016 04:19 Para: users at lists.opensips.org Assunto: Re: [OpenSIPS-Users] I have a Patch that fixes memory leak on OpenSIPS. How to apply this path via github? Hi, Rodrigo! Please fork opensips and open a pull request on Github[1]. The idea is simple: 1. fork the repository[2] 2. Apply the patch, commit it and push it in your fork 3. Open a pull request[3] [1] https://github.com/OpenSIPS/opensips/pulls [2] https://help.github.com/articles/fork-a-repo/ [3] https://help.github.com/articles/using-pull-requests/#initiating-the-pull-request PS: it is not a good idea to attach a file on a mailing list. Use gist.github.com, or pastebin.com next time :). Thanks and regards, Razvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 06/27/2016 09:05 PM, Rodrigo Pimenta Carvalho wrote: Dear OpenSIPS users, Daniel F?ssia, from Inatel Competence Center (www.inatel.br) has discovered some issues related to the code in OpenSIPS 2.2 that handles some transactions in SQLite. He also has proposed the solution for such issues and his work is attached on this message. How could I resquet to the OpenSIPS development team to apply this fix? That is, can someone here give me the instructions on how to use github and request that fix? I? very new on github. Any hint will be very helpful! Thanks alot! RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 Brazil ________________________________ _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jun 28 14:25:51 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 28 Jun 2016 15:25:51 +0300 Subject: [OpenSIPS-Users] How to configure opensips to show passwords in column password from table subscriber? In-Reply-To: References: Message-ID: <57726CCF.4090706@opensips.org> Hi Rodrigo, IF you want to use plain text passwords, see http://www.opensips.org/Documentation/TipsFAQ#toc2 - in opensipsctl set the STORE_PLAINTEXT_PW to 1 (to store the text pwd too). On the OpenSIPS side, to use the plain text pwd, enable the calculate_ha1 param in the auth module: http://www.opensips.org/html/docs/modules/2.2.x/auth.html#id293572 Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 27.06.2016 16:45, Rodrigo Pimenta Carvalho wrote: > > Hi. > > > > When I execute /opensipsctl add user password/, the column password in > table subscriber remains empty. Ha1 has a kind of encrypted password. > > > How to configure opensips to show passwords in column password from > table subscriber? > > > I have tried changing some parameters in module auth_db, but it didn't > take effect. So, what is the correct configuration? > > > Any hint will be very helpful! > > > Regards. > > > RODRIGO PIMENTA CARVALHO > Inatel Competence Center > Software > Ph: +55 35 3471 9200 RAMAL 979 > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jun 28 14:26:57 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 28 Jun 2016 15:26:57 +0300 Subject: [OpenSIPS-Users] running sip tls on 443 In-Reply-To: References: Message-ID: <57726D11.5010904@opensips.org> Hi Tito, If opensips crashes, were you able to extract a backtrace from the core file(s) ? Thanks and regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 27.06.2016 21:20, Tito Cumpen wrote: > Group, > > > I am experiencing strange behavior when configuring sip tls on port > 443. At time opensips crashes or stops accepting new connections. Here > are the tcp configs I am using: > > #disable_tcp=no > > tcp_connection_lifetime=3600 > > tcp_connect_timeout=3 > > tcp_keepidle = 30 > > tcp_keepinterval = 5 > > tcp_keepalive = 1 > > tcp_keepcount = 5 > > tcp_max_msg_time = 8 > > tcp_children=10 > > > Any idea what would case this? I am assuming there are probes out in > the internet that eventually make opensips crash? > > Thanks, > Tito > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at inin.com Tue Jun 28 15:19:35 2016 From: Ben.Newlin at inin.com (Newlin, Ben) Date: Tue, 28 Jun 2016 13:19:35 +0000 Subject: [OpenSIPS-Users] ACK after set_advertised_address contains wrong address in VIA header In-Reply-To: <5772474B.4090207@opensips.org> References: <5770F3EF.2040201@opensips.org> <739103AC-2405-47B1-A37F-EB034BD62598@inin.com> <57713B04.9040204@opensips.org> <4E5A77D0-3095-4A78-93A4-C922A5B5862B@inin.com> <5772474B.4090207@opensips.org> Message-ID: No. I am using topology_hiding on my private nodes, so the intent of using set_advertised_address was to have the Contact and other headers contain the public node?s IP after topology_hiding. Instead I removed the set_advertised_address on the private nodes and I record_route on the public nodes. This makes the calls route properly as before, but it exposes the private node?s IP in Contact and other headers. It was not desired, but it?s not a big deal especially since the logic and routing is much cleaner this way. Ben Newlin From: Bogdan-Andrei Iancu Date: Tuesday, June 28, 2016 at 5:45 AM To: "Newlin, Ben" , "users at lists.opensips.org" Subject: Re: [OpenSIPS-Users] ACK after set_advertised_address contains wrong address in VIA header Hi Ben, You mean you used the record_route_preset ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 27.06.2016 20:02, Newlin, Ben wrote: I have opened issue #917 on Github [1]. I should mention that I have worked around this problem by deciding to use Record-Routes instead of set_advertised_address. It is much cleaner, even though it does expose my private IPs. However, I have kept this configuration in case you need me to perform any tests or get tracing/logs. Thanks! [1] https://github.com/OpenSIPS/opensips/issues/917 Ben Newlin From: "Newlin, Ben" Date: Monday, June 27, 2016 at 11:06 AM To: Bogdan-Andrei Iancu , "users at lists.opensips.org" Subject: Re: [OpenSIPS-Users] ACK after set_advertised_address contains wrong address in VIA header Did you mean to say if you set it in request route? I should clarify that when I am setting the advertised address the second time it is of course happening in failure_route as the first request has failed at that point. Perhaps that is the issue? I will open a bug. Thanks. Ben Newlin From: Bogdan-Andrei Iancu Date: Monday, June 27, 2016 at 10:41 AM To: "Newlin, Ben" , "users at lists.opensips.org" Subject: Re: [OpenSIPS-Users] ACK after set_advertised_address contains wrong address in VIA header Hi Ben, If you set the advertised host / port in branch route, it will have impact over the entire transaction (all branches). So, any local replies (CANCEL and ACK) that are constructed by OpenSIPS (for any branch) will use the same set of advertised values. Which is of course wrong. Let us come up with the fix (as idea and code). Could you open a bug report on the GITHUB tracker, please ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 27.06.2016 15:45, Newlin, Ben wrote: I always set the advertised address in request route. Also as the original issue noted the second INVITE does go out with the correct advertised address in the VIA. It is only the local ACK for the failed second request that contains the wrong address in the VIA. So set_advertised_address appears to be working, but the local generated ACK is not using that address. Ben Newlin From: Bogdan-Andrei Iancu Date: Monday, June 27, 2016 at 5:37 AM To: "users at lists.opensips.org" , "Newlin, Ben" Subject: Re: [OpenSIPS-Users] ACK after set_advertised_address contains wrong address in VIA header Hi Ben, Where in the script do you do the first advertise_address ? In the request route or in a branch route ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 25.06.2016 03:41, Newlin, Ben wrote: I have run into the same problem that was described in this previous post [1], however it doesn?t appear it was ever solved at the time. I am using the dispatcher module to route calls to external carriers and I am using set_advertised_address to set the outgoing public address prior to sending the request. If the first destination returns failure, the ACK is sent correctly. Then I select a different destination and set a different public address using set_advertised_address. If this second call also fails, the ACK that is sent out uses the first advertised address, not the current on for the request. Has anyone figured this out? I am using 1.11.6. [1] http://lists.opensips.org/pipermail/users/2014-August/029779.html Ben Newlin _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From millennium.bug at gmail.com Tue Jun 28 17:02:20 2016 From: millennium.bug at gmail.com (Owais Ahmad) Date: Tue, 28 Jun 2016 20:02:20 +0500 Subject: [OpenSIPS-Users] Preventing invalid packet parsing Message-ID: Hello, I am getting UDP (non-SIP) packets on opensips listening port. How can I drop those packets to ensure it only consumes the least opensips resources. I do have mechanisms to prevent this on the firewall but just want to be sure that if such a network packet does reach the server, opensips doesn't end up parsing it assuming its SIP. I want to do this without using: sipmsg_validate Regards, Owais -------------- next part -------------- An HTML attachment was scrubbed... URL: From abalashov at evaristesys.com Tue Jun 28 17:04:38 2016 From: abalashov at evaristesys.com (Alex Balashov) Date: Tue, 28 Jun 2016 11:04:38 -0400 Subject: [OpenSIPS-Users] Preventing invalid packet parsing In-Reply-To: References: Message-ID: <57729206.4010000@evaristesys.com> Are you under the impression that these consume a significant amount of OpenSIPS resources? If so, how did you arrive at that conclusion? On 06/28/2016 11:02 AM, Owais Ahmad wrote: > Hello, > > I am getting UDP (non-SIP) packets on opensips listening port. How can I > drop those packets to ensure it only consumes the least opensips resources. > > I do have mechanisms to prevent this on the firewall but just want to be > sure that if such a network packet does reach the server, opensips > doesn't end up parsing it assuming its SIP. > > I want to do this without using: sipmsg_validate > > > > Regards, > Owais > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ From millennium.bug at gmail.com Tue Jun 28 17:06:27 2016 From: millennium.bug at gmail.com (Owais Ahmad) Date: Tue, 28 Jun 2016 20:06:27 +0500 Subject: [OpenSIPS-Users] Preventing invalid packet parsing In-Reply-To: <57729206.4010000@evaristesys.com> References: <57729206.4010000@evaristesys.com> Message-ID: Thats not the case Alex. But I am expecting a large number of UDP messages arriving on the same port as my udp listening socket. Just want to be sure there is no wasteful work done parsing such packets. On Tue, Jun 28, 2016 at 8:04 PM, Alex Balashov wrote: > Are you under the impression that these consume a significant amount of > OpenSIPS resources? If so, how did you arrive at that conclusion? > > On 06/28/2016 11:02 AM, Owais Ahmad wrote: > >> Hello, >> >> I am getting UDP (non-SIP) packets on opensips listening port. How can I >> drop those packets to ensure it only consumes the least opensips >> resources. >> >> I do have mechanisms to prevent this on the firewall but just want to be >> sure that if such a network packet does reach the server, opensips >> doesn't end up parsing it assuming its SIP. >> >> I want to do this without using: sipmsg_validate >> >> >> >> Regards, >> Owais >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > -- > Alex Balashov | Principal | Evariste Systems LLC > 1447 Peachtree Street NE, Suite 700 > Atlanta, GA 30309 > United States > > Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) > Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From abalashov at evaristesys.com Tue Jun 28 17:24:58 2016 From: abalashov at evaristesys.com (Alex Balashov) Date: Tue, 28 Jun 2016 11:24:58 -0400 Subject: [OpenSIPS-Users] Preventing invalid packet parsing In-Reply-To: References: <57729206.4010000@evaristesys.com> Message-ID: <577296CA.6020006@evaristesys.com> On 06/28/2016 11:06 AM, Owais Ahmad wrote: > Thats not the case Alex. But I am expecting a large number of UDP > messages arriving on the same port as my udp listening socket. > Just want to be sure there is no wasteful work done parsing such packets. You can be reasonably sure that OpenSIPS "fingerprints" such packets quite efficiently and, if it doesn't type as a SIP packet, it won't be parsed. For more details, see: https://github.com/OpenSIPS/opensips/blob/master/parser/msg_parser.c#L543 As you can see, it parses the first line of the message first, and doesn't do any unnecessary work to parse the whole message if the first line doesn't suggest it's SIP. I wouldn't worry about this "premature optimisation". :-) -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ From bogdan at opensips.org Tue Jun 28 17:33:02 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 28 Jun 2016 18:33:02 +0300 Subject: [OpenSIPS-Users] Preventing invalid packet parsing In-Reply-To: References: <57729206.4010000@evaristesys.com> Message-ID: <577298AE.9020301@opensips.org> Hi, Well, if the "message" got to the script level, it means is "looks" like a SIP message - to reach the script, the message must have a SIP valid first line and Top most VIA. So, if your "bogus" messages do not have anything to do with SIP, considering actions at script level is a non sense as it is too late. What you can do is to rely on the size of the package in order to drop it (by default, OpenSIPS drops any UDP packet shorter than 20 bytes - see MIN_UDP_PACKET in config.h). You can change that value to match your needs. And recompile. IF the incoming traffic looks like SIP packages (and it lands in your script), you should use the sipmsg_validate() to get ready of it asap. It is not about processing, it is more about avoid polluting the logs with errors :) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 28.06.2016 18:06, Owais Ahmad wrote: > Thats not the case Alex. But I am expecting a large number of UDP > messages arriving on the same port as my udp listening socket. > Just want to be sure there is no wasteful work done parsing such packets. > > On Tue, Jun 28, 2016 at 8:04 PM, Alex Balashov > > wrote: > > Are you under the impression that these consume a significant > amount of OpenSIPS resources? If so, how did you arrive at that > conclusion? > > On 06/28/2016 11:02 AM, Owais Ahmad wrote: > > Hello, > > I am getting UDP (non-SIP) packets on opensips listening port. > How can I > drop those packets to ensure it only consumes the least > opensips resources. > > I do have mechanisms to prevent this on the firewall but just > want to be > sure that if such a network packet does reach the server, opensips > doesn't end up parsing it assuming its SIP. > > I want to do this without using: sipmsg_validate > > > > Regards, > Owais > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > Alex Balashov | Principal | Evariste Systems LLC > 1447 Peachtree Street NE, Suite 700 > Atlanta, GA 30309 > United States > > Tel: +1-800-250-5920 (toll-free) / > +1-678-954-0671 (direct) > Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jun 28 19:06:31 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 28 Jun 2016 20:06:31 +0300 Subject: [OpenSIPS-Users] OpenSIPS @ ClueCon weekly call Message-ID: <5772AE97.7030003@opensips.org> Hi, Join us tomorrow, 29th of June 2016, 12:00 CST to the ClueCon weekly call where OpenSIPS is the main guest. We will have discussion about the latest news/plans of OpenSIPS, about the integration with FreeSwitch and the joined training at ClueCon. Where? https://conference.freeswitch.org/vc/ extension 888 See you tomorrow !!! Regards, -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From govoiper at gmail.com Tue Jun 28 21:59:29 2016 From: govoiper at gmail.com (SamyGo) Date: Tue, 28 Jun 2016 15:59:29 -0400 Subject: [OpenSIPS-Users] OpenSIPS 2.2 SIP ping not triggering Message-ID: Hi All, I've OpenSIPS 2.2 and doing some tests with it I can't seem to find any OPTIONS triggering for the registered users. I've following settings for usrloc module and nathelper loadmodule "usrloc.so" modparam("usrloc", "nat_bflag", "NAT") modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://root:Q4FRX at localhost/opensips_2_2") loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 10) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", "SIP_PING_FLAG") modparam("nathelper", "sipping_from", "sip:pinger at siptest.saevolgo.ca") modparam("nathelper", "received_avp", "$avp(received)") modparam("nathelper", "natping_tcp", 1) modparam("nathelper", "ping_threshold", 10) modparam("nathelper", "max_pings_lost", 5) I force set NAT flag for all SIP packets either INVITE or REGISTER route{ route(SIP_CHECK); # Handles and adjusts packets for NAT'e clients force_rport(); if (nat_uac_test("23")) { if (is_method("REGISTER")) { fix_nated_register(); setbflag(NAT); } else { fix_nated_contact(); setflag(NAT); } } setflag(NAT); and set SIP_PING_FLAG before save() function: if ( proto==TCP || proto==TLS || 0 ) setflag(TCP_PERSISTENT); if (isflagset(NAT)) { setbflag(SIP_PING_FLAG); } if (!save("location")) sl_reply_error(); If I look at online users the flags are seen applied to the extensions: root at test1:/etc/opensips# opensipsctl ul show Domain:: location table=512 records=1 AOR:: 1009 at siptest.saevolgo.ca Contact:: sip:1009 at 64.231.959.69:30514;rinstance=38d9f5f00287404f;transport=udp Q= Expires:: 3093 Callid:: 1955a010ad311f31OGZjZmJjZWZhZTQxMjNjOGRhOTZiYjUxNWQ4Y2JiODg. Cseq:: 2 User-agent:: eyeBeam release 1003s stamp 31159 State:: CS_SYNC Flags:: 0 Cflags:: SIP_PING_FLAG Socket:: udp:X.X.X.X:5060 Methods:: 5951 I waited for like 15 minutes but no OPTIONS triggered from opensips ! Any guidance/help will be appreciated. Regards. Sammy -------------- next part -------------- An HTML attachment was scrubbed... URL: From pimenta at inatel.br Tue Jun 28 22:52:03 2016 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Tue, 28 Jun 2016 20:52:03 +0000 Subject: [OpenSIPS-Users] Does OpenSIPS use SQLite configuration to hold on queries when database is locked? Message-ID: Dear OpenSIPS users, In my software, programmed in QT (framework for C++) and handling data with a SQLite database, I have used this: pDb.setConnectOptions("QSQLITE_BUSY_TIMEOUT=6000"); That is, if SQLite complains that "database is locked" sometime when my software tries to register some datum there, such database keeps the query paused (hold on), and then after 6 seconds let the query execute. This mechanism is transparent for my software and certify that the query will be tried every 6 seconds until it complete successfully. Is there something similar to it in OpenSIPS ? That is, does OpenSIPS uses some kind of configuration provided by SQLite? If not, why the developers team decided not to use such mechanism? Any comment will be very helpful! Thanks a lot! RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Jun 29 12:23:47 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 29 Jun 2016 13:23:47 +0300 Subject: [OpenSIPS-Users] OpenSIPS 2.2 SIP ping not triggering In-Reply-To: References: Message-ID: <5773A1B3.7020507@opensips.org> Hello Sammy, What OpenSIPS version you have (opensips -V) and where did you get it ? There is a very similar bug which was already fixed. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 28.06.2016 22:59, SamyGo wrote: > Hi All, > > I've OpenSIPS 2.2 and doing some tests with it I can't seem to find > any OPTIONS triggering for the registered users. > > I've following settings for usrloc module and nathelper > > loadmodule "usrloc.so" > modparam("usrloc", "nat_bflag", "NAT") > modparam("usrloc", "db_mode", 2) > modparam("usrloc", "db_url", > "mysql://root:Q4FRX at localhost/opensips_2_2") > > > loadmodule "nathelper.so" > modparam("nathelper", "natping_interval", 10) > modparam("nathelper", "ping_nated_only", 1) > modparam("nathelper", "sipping_bflag", "SIP_PING_FLAG") > modparam("nathelper", "sipping_from", "sip:pinger at siptest.saevolgo.ca > ") > modparam("nathelper", "received_avp", "$avp(received)") > modparam("nathelper", "natping_tcp", 1) > modparam("nathelper", "ping_threshold", 10) > modparam("nathelper", "max_pings_lost", 5) > > I force set NAT flag for all SIP packets either INVITE or REGISTER > > route{ > route(SIP_CHECK); > # Handles and adjusts packets for NAT'e clients > force_rport(); > if (nat_uac_test("23")) { > if (is_method("REGISTER")) { > fix_nated_register(); > setbflag(NAT); > } else { > fix_nated_contact(); > setflag(NAT); > } > } > setflag(NAT); > > and set SIP_PING_FLAG before save() function: > > > if ( proto==TCP || proto==TLS || 0 ) setflag(TCP_PERSISTENT); > > if (isflagset(NAT)) { > setbflag(SIP_PING_FLAG); > } > > if (!save("location")) > sl_reply_error(); > > If I look at online users the flags are seen applied to the extensions: > > root at test1:/etc/opensips# opensipsctl ul show > Domain:: location table=512 records=1 > AOR:: 1009 at siptest.saevolgo.ca > Contact:: > sip:1009 at 64.231.959.69:30514;rinstance=38d9f5f00287404f;transport=udp Q= > Expires:: 3093 > Callid:: 1955a010ad311f31OGZjZmJjZWZhZTQxMjNjOGRhOTZiYjUxNWQ4Y2JiODg. > Cseq:: 2 > User-agent:: eyeBeam release 1003s stamp 31159 > State:: CS_SYNC > Flags:: 0 > Cflags:: SIP_PING_FLAG > Socket:: udp:X.X.X.X:5060 > Methods:: 5951 > > > I waited for like 15 minutes but no OPTIONS triggered from opensips ! > > Any guidance/help will be appreciated. > > > Regards. > Sammy > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From ionutionita at opensips.org Wed Jun 29 16:02:53 2016 From: ionutionita at opensips.org (Ionut Ionita) Date: Wed, 29 Jun 2016 17:02:53 +0300 Subject: [OpenSIPS-Users] Does OpenSIPS use SQLite configuration to hold on queries when database is locked? In-Reply-To: References: Message-ID: <5773D50D.5010105@opensips.org> Hi Rodrigo, I am the one who implemented SQLITE module. First of all I wasn't aware of the BUSY_TIMEOUT option, but still if I was I wouldn't use it and I'll explain why. Setting such a parameter will cause the OpenSIPS processes, except the one that has the lock on the database, to sleep for X ms. After those X ms the database may be locked again and you have to sleep again for that amout of seconds, even though at some point in that sleeping interval you may have had the change to get the lock. In the current implementation, all processes keep trying to get the lock, avoiding dead times, when they could have had the lock but they were sleeping. We can't do anything else while the database is locked since we need to process the current message in order to get to the next one. SQLITE has limitations based on the fact that for each query the whole database is locked, and to explain that I would like to quote official documentation: /"However, client/server database engines (such as PostgreSQL, MySQL, or Oracle) usually support a higher level of concurrency and allow multiple processes to be writing to the same database at the same time. This is possible in a client/server database because there is always a single well-controlled server process available to coordinate access. If your application has a need for a lot of concurrency, then you should consider using a client/server database. But experience suggests that most applications need much less concurrency than their designers i//magin//e./*//*/"[0] / We would be glad to implement such a mechanism if our software would benefit from it, but in my humble opinion it would bring nothing useful for our module. / /[0] http://www.sqlite.org/faq.html#q5 Regards, Ionut Ionita OpenSIPS Developer On 06/28/2016 11:52 PM, Rodrigo Pimenta Carvalho wrote: > > > Dear OpenSIPS users, > > > In my software, programmed in QT (framework for C++) and handling data > with a SQLite database, I have used this: > > > pDb.setConnectOptions("QSQLITE_BUSY_TIMEOUT=6000"); > > > That is, if SQLite complains that "database is locked" sometime when > my software tries to register some datum there, such database keeps > the query paused (hold on), and then after 6 seconds let the query > execute. This mechanism is transparent for my software and certify > that the query will be tried every 6 seconds until it complete > successfully. > > > Is there something similar to it in OpenSIPS ? That is, does OpenSIPS > uses some kind of configuration provided by SQLite? > > If not, why the developers team decided not to use such mechanism? > > > Any comment will be very helpful! > > > Thanks a lot! > > > > RODRIGO PIMENTA CARVALHO > Inatel Competence Center > Software > Ph: +55 35 3471 9200 RAMAL 979 > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From pimenta at inatel.br Wed Jun 29 16:49:09 2016 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Wed, 29 Jun 2016 14:49:09 +0000 Subject: [OpenSIPS-Users] Does OpenSIPS use SQLite configuration to hold on queries when database is locked? In-Reply-To: <5773D50D.5010105@opensips.org> References: , <5773D50D.5010105@opensips.org> Message-ID: Dear Ionut. Thank you very much for the precise explanation! I got the point. Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________ De: users-bounces at lists.opensips.org em nome de Ionut Ionita Enviado: quarta-feira, 29 de junho de 2016 11:02 Para: OpenSIPS users mailling list Assunto: Re: [OpenSIPS-Users] Does OpenSIPS use SQLite configuration to hold on queries when database is locked? Hi Rodrigo, I am the one who implemented SQLITE module. First of all I wasn't aware of the BUSY_TIMEOUT option, but still if I was I wouldn't use it and I'll explain why. Setting such a parameter will cause the OpenSIPS processes, except the one that has the lock on the database, to sleep for X ms. After those X ms the database may be locked again and you have to sleep again for that amout of seconds, even though at some point in that sleeping interval you may have had the change to get the lock. In the current implementation, all processes keep trying to get the lock, avoiding dead times, when they could have had the lock but they were sleeping. We can't do anything else while the database is locked since we need to process the current message in order to get to the next one. SQLITE has limitations based on the fact that for each query the whole database is locked, and to explain that I would like to quote official documentation: "However, client/server database engines (such as PostgreSQL, MySQL, or Oracle) usually support a higher level of concurrency and allow multiple processes to be writing to the same database at the same time. This is possible in a client/server database because there is always a single well-controlled server process available to coordinate access. If your application has a need for a lot of concurrency, then you should consider using a client/server database. But experience suggests that most applications need much less concurrency than their designers imagine."[0] We would be glad to implement such a mechanism if our software would benefit from it, but in my humble opinion it would bring nothing useful for our module. [0] http://www.sqlite.org/faq.html#q5 Regards, Ionut Ionita OpenSIPS Developer On 06/28/2016 11:52 PM, Rodrigo Pimenta Carvalho wrote: Dear OpenSIPS users, In my software, programmed in QT (framework for C++) and handling data with a SQLite database, I have used this: pDb.setConnectOptions("QSQLITE_BUSY_TIMEOUT=6000"); That is, if SQLite complains that "database is locked" sometime when my software tries to register some datum there, such database keeps the query paused (hold on), and then after 6 seconds let the query execute. This mechanism is transparent for my software and certify that the query will be tried every 6 seconds until it complete successfully. Is there something similar to it in OpenSIPS ? That is, does OpenSIPS uses some kind of configuration provided by SQLite? If not, why the developers team decided not to use such mechanism? Any comment will be very helpful! Thanks a lot! RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From govoiper at gmail.com Wed Jun 29 17:04:49 2016 From: govoiper at gmail.com (SamyGo) Date: Wed, 29 Jun 2016 11:04:49 -0400 Subject: [OpenSIPS-Users] OpenSIPS 2.2 SIP ping not triggering In-Reply-To: <5773A1B3.7020507@opensips.org> References: <5773A1B3.7020507@opensips.org> Message-ID: Thanks Bogdan for looking into this. This is my version; version: opensips 2.2.0 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. git revision: 9512363 main.c compiled on 15:54:46 Jun 22 2016 with gcc 4.8 Best Regards, Sammy On Wed, Jun 29, 2016 at 6:23 AM, Bogdan-Andrei Iancu wrote: > Hello Sammy, > > What OpenSIPS version you have (opensips -V) and where did you get it ? > There is a very similar bug which was already fixed. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 28.06.2016 22:59, SamyGo wrote: > > Hi All, > > I've OpenSIPS 2.2 and doing some tests with it I can't seem to find any > OPTIONS triggering for the registered users. > > I've following settings for usrloc module and nathelper > > loadmodule "usrloc.so" > modparam("usrloc", "nat_bflag", "NAT") > modparam("usrloc", "db_mode", 2) > modparam("usrloc", "db_url", > "mysql://root:Q4FRX at localhost/opensips_2_2") > > > loadmodule "nathelper.so" > modparam("nathelper", "natping_interval", 10) > modparam("nathelper", "ping_nated_only", 1) > modparam("nathelper", "sipping_bflag", "SIP_PING_FLAG") > modparam("nathelper", "sipping_from", "sip:pinger at siptest.saevolgo.ca") > modparam("nathelper", "received_avp", "$avp(received)") > modparam("nathelper", "natping_tcp", 1) > modparam("nathelper", "ping_threshold", 10) > modparam("nathelper", "max_pings_lost", 5) > > I force set NAT flag for all SIP packets either INVITE or REGISTER > > route{ > > route(SIP_CHECK); > # Handles and adjusts packets for NAT'e clients > force_rport(); > if (nat_uac_test("23")) { > if (is_method("REGISTER")) { > fix_nated_register(); > setbflag(NAT); > } else { > fix_nated_contact(); > setflag(NAT); > } > } > setflag(NAT); > > and set SIP_PING_FLAG before save() function: > > > if ( proto==TCP || proto==TLS || 0 ) setflag(TCP_PERSISTENT); > > if (isflagset(NAT)) { > setbflag(SIP_PING_FLAG); > } > > if (!save("location")) > sl_reply_error(); > > If I look at online users the flags are seen applied to the extensions: > > root at test1:/etc/opensips# opensipsctl ul show > Domain:: location table=512 records=1 > AOR:: <1009 at siptest.saevolgo.ca>1009 at siptest.saevolgo.ca > Contact:: > sip:1009 at 64.231.959.69:30514;rinstance=38d9f5f00287404f;transport=udp Q= > Expires:: 3093 > Callid:: > 1955a010ad311f31OGZjZmJjZWZhZTQxMjNjOGRhOTZiYjUxNWQ4Y2JiODg. > Cseq:: 2 > User-agent:: eyeBeam release 1003s stamp 31159 > State:: CS_SYNC > Flags:: 0 > Cflags:: SIP_PING_FLAG > Socket:: udp:X.X.X.X:5060 > Methods:: 5951 > > > I waited for like 15 minutes but no OPTIONS triggered from opensips ! > > Any guidance/help will be appreciated. > > > Regards. > Sammy > > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Jun 29 17:07:41 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 29 Jun 2016 18:07:41 +0300 Subject: [OpenSIPS-Users] OpenSIPS 2.2 SIP ping not triggering In-Reply-To: References: <5773A1B3.7020507@opensips.org> Message-ID: <5773E43D.5080209@opensips.org> Sammy, Please update your sources with the latest 2.2 (either from git branch 2.2, either nightly tarbal for 2.2 from download.opensips.org) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 29.06.2016 18:04, SamyGo wrote: > Thanks Bogdan for looking into this. > > This is my version; > > version: opensips 2.2.0 (x86_64/linux) > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > MAX_URI_SIZE 1024, BUF_SIZE 65535 > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. > git revision: 9512363 > main.c compiled on 15:54:46 Jun 22 2016 with gcc 4.8 > > Best Regards, > Sammy > > > On Wed, Jun 29, 2016 at 6:23 AM, Bogdan-Andrei Iancu > > wrote: > > Hello Sammy, > > What OpenSIPS version you have (opensips -V) and where did you get > it ? There is a very similar bug which was already fixed. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 28.06.2016 22:59, SamyGo wrote: >> Hi All, >> >> I've OpenSIPS 2.2 and doing some tests with it I can't seem to >> find any OPTIONS triggering for the registered users. >> >> I've following settings for usrloc module and nathelper >> >> loadmodule "usrloc.so" >> modparam("usrloc", "nat_bflag", "NAT") >> modparam("usrloc", "db_mode", 2) >> modparam("usrloc", "db_url", >> "mysql://root:Q4FRX at localhost/opensips_2_2") >> >> >> loadmodule "nathelper.so" >> modparam("nathelper", "natping_interval", 10) >> modparam("nathelper", "ping_nated_only", 1) >> modparam("nathelper", "sipping_bflag", "SIP_PING_FLAG") >> modparam("nathelper", "sipping_from", >> "sip:pinger at siptest.saevolgo.ca >> ") >> modparam("nathelper", "received_avp", "$avp(received)") >> modparam("nathelper", "natping_tcp", 1) >> modparam("nathelper", "ping_threshold", 10) >> modparam("nathelper", "max_pings_lost", 5) >> >> I force set NAT flag for all SIP packets either INVITE or REGISTER >> >> route{ >> route(SIP_CHECK); >> # Handles and adjusts packets for NAT'e clients >> force_rport(); >> if (nat_uac_test("23")) { >> if (is_method("REGISTER")) { >> fix_nated_register(); >> setbflag(NAT); >> } else { >> fix_nated_contact(); >> setflag(NAT); >> } >> } >> setflag(NAT); >> >> and set SIP_PING_FLAG before save() function: >> >> >> if ( proto==TCP || proto==TLS || 0 ) setflag(TCP_PERSISTENT); >> >> if (isflagset(NAT)) { >> setbflag(SIP_PING_FLAG); >> } >> >> if (!save("location")) >> sl_reply_error(); >> >> If I look at online users the flags are seen applied to the >> extensions: >> >> root at test1:/etc/opensips# opensipsctl ul show >> Domain:: location table=512 records=1 >> AOR:: 1009 at siptest.saevolgo.ca >> Contact:: >> sip:1009 at 64.231.959.69:30514;rinstance=38d9f5f00287404f;transport=udp >> >> Q= >> Expires:: 3093 >> Callid:: >> 1955a010ad311f31OGZjZmJjZWZhZTQxMjNjOGRhOTZiYjUxNWQ4Y2JiODg. >> Cseq:: 2 >> User-agent:: eyeBeam release 1003s stamp 31159 >> State:: CS_SYNC >> Flags:: 0 >> Cflags:: SIP_PING_FLAG >> Socket:: udp:X.X.X.X:5060 >> Methods:: 5951 >> >> >> I waited for like 15 minutes but no OPTIONS triggered from opensips ! >> >> Any guidance/help will be appreciated. >> >> >> Regards. >> Sammy >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From pimenta at inatel.br Wed Jun 29 20:50:19 2016 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Wed, 29 Jun 2016 18:50:19 +0000 Subject: [OpenSIPS-Users] OpensSIPS 2.2 memory leak issue fixed today. Message-ID: Dear OpenSIPS users; Daniel F?ssia has just pulled a request in GitHub. See the link: https://github.com/OpenSIPS/opensips/pull/919 It has one commit with 4 files, fixing some issues related to memory leaks and SQLite. This is for OpenSIPS 2.2 HEAD. We will appreciate comments about it. Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 -------------- next part -------------- An HTML attachment was scrubbed... URL: From nabeelshikder at gmail.com Thu Jun 30 06:19:33 2016 From: nabeelshikder at gmail.com (Nabeel) Date: Thu, 30 Jun 2016 05:19:33 +0100 Subject: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions In-Reply-To: <575FD7A0.6000601@opensips.org> References: <575FD7A0.6000601@opensips.org> Message-ID: Hi Bogdan, I was able to install the latest versions of Asterisk (13.1) and Opensips (2.3) according to the tutorial, but when attempting to leave a voicemail I get the following errors: > [Jun 30 01:07:53] NOTICE[17067][C-00000000] chan_sip.c: Call from > '+447867958678' (162.249.6.206:12221) to extension 'VMR_+447479189410' > rejected because extension not found in context 'default'. > [Jun 30 01:07:53] WARNING[17112] res_odbc.c: SetConnectAttr (Txn > isolation) returned an error: HY000: [MySQL][ODBC 5.2(w) Driver]You have an > error in your SQL syntax; check the manual that corresponds to your MariaDB > server version for the right syntax to use near '7' at line 1 > [Jun 30 01:07:53] WARNING[17112] res_config_odbc.c: SQL Prepare > failed![SELECT * FROM sipusers WHERE name = ? AND host = ?] > [Jun 30 01:07:53] WARNING[17112] res_odbc.c: Connection is down attempting > to reconnect... > Also I had to change 'nat=yes' to 'nat=force_rport,comedia' as it is deprecated. Nabeel On 14 June 2016 at 11:08, Bogdan-Andrei Iancu wrote: > Hi Nabeel, > > We will update the tutorial for 2.2, but it should still match. Give it a > try and if you hit issues, just let me know. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 12.06.2016 10:18, Nabeel wrote: > > Hi, > > I will be following this tutorial to integrate OpenSIPS and Asterisk: > > > http://www.opensips.org/Documentation/Tutorials-OpenSIPSAsteriskIntegration-1-8 > > The tutorial mentions the use of OpenSIPS version 1.8 and Asterisk version > 1.8. I would like to know if I can use the latest versions of OpenSIPS and > Asterisk instead? Have there been changes to database structure which can > cause problems? > > Nabeel > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Jun 30 11:18:37 2016 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 30 Jun 2016 12:18:37 +0300 Subject: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions In-Reply-To: References: <575FD7A0.6000601@opensips.org> Message-ID: <5774E3ED.6020301@opensips.org> Hi Nabeel, The "sipusers" mysql view (as per http://www.opensips.org/Documentation/Tutorials-OpenSIPSAsteriskIntegration-1-8#toc7 ) has both the name and host fields - not sure why that query may fail..... Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 30.06.2016 07:19, Nabeel wrote: > Hi Bogdan, > > I was able to install the latest versions of Asterisk (13.1) and > Opensips (2.3) according to the tutorial, but when attempting to leave > a voicemail I get the following errors: > > [Jun 30 01:07:53] NOTICE[17067][C-00000000] chan_sip.c: Call from > '+447867958678' (162.249.6.206:12221 ) > to extension 'VMR_+447479189410' rejected because extension not > found in context 'default'. > [Jun 30 01:07:53] WARNING[17112] res_odbc.c: SetConnectAttr (Txn > isolation) returned an error: HY000: [MySQL][ODBC 5.2(w) > Driver]You have an error in your SQL syntax; check the manual that > corresponds to your MariaDB server version for the right syntax to > use near '7' at line 1 > [Jun 30 01:07:53] WARNING[17112] res_config_odbc.c: SQL Prepare > failed![SELECT * FROM sipusers WHERE name = ? AND host = ?] > [Jun 30 01:07:53] WARNING[17112] res_odbc.c: Connection is down > attempting to reconnect... > > > > Also I had to change 'nat=yes' to 'nat=force_rport,comedia' as it is > deprecated. > > Nabeel > > > On 14 June 2016 at 11:08, Bogdan-Andrei Iancu > wrote: > > Hi Nabeel, > > We will update the tutorial for 2.2, but it should still match. > Give it a try and if you hit issues, just let me know. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 12.06.2016 10:18, Nabeel wrote: >> >> Hi, >> >> I will be following this tutorial to integrate OpenSIPS and Asterisk: >> >> http://www.opensips.org/Documentation/Tutorials-OpenSIPSAsteriskIntegration-1-8 >> >> The tutorial mentions the use of OpenSIPS version 1.8 and >> Asterisk version 1.8. I would like to know if I can use the >> latest versions of OpenSIPS and Asterisk instead? Have there been >> changes to database structure which can cause problems? >> >> Nabeel >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From alain.bieuzent at free.fr Thu Jun 30 11:49:55 2016 From: alain.bieuzent at free.fr (Alain Bieuzent) Date: Thu, 30 Jun 2016 11:49:55 +0200 Subject: [OpenSIPS-Users] Migrating from 1.11 to 2.1 load balancer Probing problem Message-ID: Hi? all, I?m currently trying to migrate my opensips V1.11.5 to? V2.1.3. My server has 3 interfaces: eth0 with public IP eth0:0 with public IP provide via a keepalived script eth1 with a private IP Opensips listen on IP of eth0:0 and on eth1 Mhomed=no Since i migrate to V2.1.3, load balancer probing is UP and down. After checking traffic, i find for each gateway used by load balancer table, there is 2 SIP OPTIONS sent. - First is sent with source IP of eth0:0 - Second is sent with source IP of eth1 Any idea ? Thanls -------------- next part -------------- An HTML attachment was scrubbed... URL: From ionutionita at opensips.org Thu Jun 30 13:06:48 2016 From: ionutionita at opensips.org (Ionut Ionita) Date: Thu, 30 Jun 2016 14:06:48 +0300 Subject: [OpenSIPS-Users] OpensSIPS 2.2 memory leak issue fixed today. In-Reply-To: References: Message-ID: <5774FD48.8060400@opensips.org> Merged Ionut Ionita OpenSIPS Developer On 06/29/2016 09:50 PM, Rodrigo Pimenta Carvalho wrote: > > Dear OpenSIPS users; > > > Daniel F?ssia has just pulled a request in GitHub. See the link: > > > https://github.com/OpenSIPS/opensips/pull/919 > > > > > It has one commit with 4 files, fixing some issues related to memory > leaks and SQLite. This is for OpenSIPS 2.2 HEAD. > > > We will appreciate comments about it. > > > Best regards. > > > RODRIGO PIMENTA CARVALHO > Inatel Competence Center > Software > Ph: +55 35 3471 9200 RAMAL 979 > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From miha at softnet.si Thu Jun 30 14:06:10 2016 From: miha at softnet.si (Miha) Date: Thu, 30 Jun 2016 14:06:10 +0200 Subject: [OpenSIPS-Users] Issue with ACK and rtpproxy, setID Message-ID: HI I have two RTPproxies and doing spiral so that I can put them in chain. Before t_relay() I am setting avps (setID) so that I can do rtpproxy_answer latter if there is SDP in ACK. The issue is that avp is all the time null for ACK (for Initial invite avp was set). In TM module i set onreply_avp_mode to 1. Is there anything else I must do or do you suggest some other approche? tnx miha From Ben.Newlin at inin.com Thu Jun 30 15:17:26 2016 From: Ben.Newlin at inin.com (Newlin, Ben) Date: Thu, 30 Jun 2016 13:17:26 +0000 Subject: [OpenSIPS-Users] Issue with ACK and rtpproxy, setID In-Reply-To: References: Message-ID: <5783BB89-736B-4AFA-9B12-9BD997DDBC50@inin.com> AVPs are tied to a transaction, so the transaction must be matched before they will be available. You should use t_check_trans() to do this. However, I think this will not work for you because ACKs are their own transactions and I don?t believe they will have access to the AVPs from the INVITE transaction. If you need to store state information across multiple transactions, you will need to use the dialog module and the $dlg_val variables. These persist across the entire SIP call. Ben Newlin On 6/30/16, 8:06 AM, "users-bounces at lists.opensips.org on behalf of Miha" wrote: HI I have two RTPproxies and doing spiral so that I can put them in chain. Before t_relay() I am setting avps (setID) so that I can do rtpproxy_answer latter if there is SDP in ACK. The issue is that avp is all the time null for ACK (for Initial invite avp was set). In TM module i set onreply_avp_mode to 1. Is there anything else I must do or do you suggest some other approche? tnx miha _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users From achalkov at ya.ru Thu Jun 30 18:32:35 2016 From: achalkov at ya.ru (=?utf-8?B?0KfQsNC70LrQvtCyINCQ0YDRgtGR0Lw=?=) Date: Thu, 30 Jun 2016 19:32:35 +0300 Subject: [OpenSIPS-Users] change_reply_status if failure route Message-ID: <669771467304355@web18g.yandex.ru> An HTML attachment was scrubbed... URL: