From bogdan at opensips.org Thu Nov 1 07:12:01 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 1 Nov 2018 13:12:01 +0200 Subject: [OpenSIPS-Users] OpenSIPS Control Panel 8.2.4 was released Message-ID: Hi all, We are proud and happy to announce the release of the OpenSIPS Control Panel 8.2.4 . This OpenSIPS Control Panel is a provisioning (SIP users and OpenSIPS system) web interface design for and compatible with OpenSIPS 2.4 LTS - http://controlpanel.opensips.org . The OCP 8.2.4 has new awesome additions : * a new modern look by reworking the entire CSS and layout, see http://controlpanel.opensips.org/screenshots.php * a new tool for handling the RTPEngine instances via database * a new tool for provisioning TLS domains and certificates via database * a more powerful Tviewer engine to allow more complex displaying, correlation and validation of the data types and to create inter-tool web links. In the same time the OCP 8.2.4 brings great improvements under the hood: * compatibility with PHP 7, by migrating from MDB2 to PDO database support * better security of the DB operations by migrating to prepared statements, to avoid SQL injections * improved security for the WEB sessions and page access to avoid illegitimate direct access to sub-pages * extended tooltip and data validation support * code refactoring for easy further development This new OpenSIPS Control Panel release is a significant step forward for the project in terms of vision. All this happened thanks to the people and companies involved in the development of the project. Special thanks to the Voicenter team - https://www.voicenter.com - for its great contribution in re-styling and re-designing this web interface. Download and enjoy it as it's freshly baked for you, -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com From vitalik.voip at gmail.com Thu Nov 1 07:46:17 2018 From: vitalik.voip at gmail.com (Vitalii Aleksandrov) Date: Thu, 1 Nov 2018 13:46:17 +0200 Subject: [OpenSIPS-Users] async(wait_for_event()) statement in a branch_route[] Message-ID: Hi, I'm a bit new to opensips, while have some experience with kamailio. Trying to figure out whether it's possible to use async() statement from branch_route[]. From the documentation I understood that async() functionality is tightly connected to the TM module and creates some context attached to a transaction, suspends it and resumes a transaction on an event. If said above is correct I suppose calling async() from a branch_route[] to suspend only one branch and expect that other branches will continue their normal execution is not what it was designed for.. Is there any way to forward a branch while other branches wait for an async event or i/o? Would appreciate any ideas. From bogdan at opensips.org Thu Nov 1 12:54:23 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 1 Nov 2018 18:54:23 +0200 Subject: [OpenSIPS-Users] SUBSCRIBE In-Reply-To: <817377071.64924.1540833372746.JavaMail.zimbra@skillsearch.ca> References: <283498062.64714.1540821702719.JavaMail.zimbra@skillsearch.ca> <1322942341.64749.1540822238950.JavaMail.zimbra@skillsearch.ca> <8c19384f-1b71-1646-41cd-b2db4b5b098e@opensips.org> <817377071.64924.1540833372746.JavaMail.zimbra@skillsearch.ca> Message-ID: <20b99496-22d3-1c1a-1370-ba2d952c6f90@opensips.org> Hi Slava, I see you use 3.0, so I pushed a small commit to print more information - this will help to understand your failure. Update and be sure you have this commit https://github.com/OpenSIPS/opensips/commit/e4868acfb1bd92fd9ec32489c18b0aaebbf8fc5b and give it a try. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 10/29/2018 07:16 PM, Slava Bendersky wrote: > In that setup freeswitch boxes directly connected to opensips LAN side > and opensips listen for for two interfaces internal and external. UDP > and TCP. > > volga629 > > ------------------------------------------------------------------------ > *From: *"Bogdan-Andrei Iancu" > *To: *"OpenSIPS users mailling list" , > "Slava Bendersky" > *Sent: *Monday, October 29, 2018 2:13:06 PM > *Subject: *Re: [OpenSIPS-Users] SUBSCRIBE > > Hi Slava, > > Maybe the same destination may be reached (from the IP routing table > perspective) via multiple interfaces like via interface A and > interface B - and OS (via mhomed) selected interface A , but OpenSIPS > listen only on B. > > Such a setup may lead to the error you listed - OS picking up for > routing a local interface that is not used by OpenSIPS. > > Regards, > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Bootcamp 2018 > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > On 10/29/2018 04:10 PM, Slava Bendersky wrote: > > That error message in log > > Oct 29 09:00:16 vprx00 /usr/sbin/opensips[1506]: > ERROR:core:get_out_socket: no socket found > Oct 29 09:00:16 vprx00 /usr/sbin/opensips[1506]: ERROR:tm:t_uac: > no corresponding socket for af 2 > Oct 29 09:00:16 vprx00 /usr/sbin/opensips[1506]: > ERROR:pua:send_publish_int: failed to send PUBLISH > Oct 29 09:00:16 vprx00 /usr/sbin/opensips[1506]: > ERROR:pua_dialoginfo:dialog_publish: sending publish failed for > pres_uri [sip:106 at domain.com:5160] to server [] > Oct 29 09:00:16 vprx00 /usr/sbin/opensips[1506]: > ERROR:core:get_out_socket: no socket found > Oct 29 09:00:16 vprx00 /usr/sbin/opensips[1506]: ERROR:tm:t_uac: > no corresponding socket for af 2 > > LAN side is UDP and WAN side TCP. Multi home environment. > volga629 > ------------------------------------------------------------------------ > *From: *"Slava Bendersky" > *To: *"OpenSIPS users mailling list" > *Sent: *Monday, October 29, 2018 11:01:42 AM > *Subject: *[OpenSIPS-Users] SUBSCRIBE > > Hello Everyone, > What possible cause that opensips generate 500 error on SUBSCRIBE > > #### Presence > loadmodule "presence.so" > loadmodule "presence_mwi.so" > loadmodule "presence_xml.so" > loadmodule "presence_dialoginfo.so" > loadmodule "presence_callinfo.so" > loadmodule "pua.so" > loadmodule "pua_dialoginfo.so" > loadmodule "xcap.so" > modparam("presence|xcap|pua","db_url",") > #modparam("presence","server_address","sip:proxy@") multi home > modparam("presence", "clean_period", 30) > modparam("presence", "mix_dialog_presence", 1) > modparam("presence_xml", "force_active", 1) > #modparam("pua_dialoginfo", "presence_server", "sip:sbc@) multi home > > > 2018/10/29 08:50:30.061342 190.240.46.242:36868 -> 190.248.157.3:5084 > SUBSCRIBE sip:101 at domain.tld:5160 SIP/2.0 > Via: SIP/2.0/TCP > 192.168.1.65:19464;branch=z9hG4bK1855278236;rport;alias > From: ;tag=473433483 > To: > Call-ID: 733815006-19464-119 at BJC.BGI.B.GF > CSeq: 21200 SUBSCRIBE > Contact: > X-Grandstream-PBX: true > Max-Forwards: 70 > User-Agent: Grandstream GXP1760W 1.0.1.74 > Expires: 432000 > Supported: replaces, path, timer > Event: message-summary > Accept: application/simple-message-summary > Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, > REFER, UPDATE, MESSAGE > Content-Length: 0 > > > > 2018/10/29 08:50:30.062338 190.248.157.3:5084 -> 190.240.46.242:36868 > SIP/2.0 500 Server error occurred (1/SL) > Via: SIP/2.0/TCP > 192.168.1.65:19464;received=190.240.46.242;branch=z9hG4bK1855278236;rport=36868;alias > From: ;tag=473433483 > To: > ;tag=a3f6a7240e26af61ce52d7ac2261cc82.ee28 > Call-ID: 733815006-19464-119 at BJC.BGI.B.GF > CSeq: 21200 SUBSCRIBE > Server: OpenSIPS (3.0.0-dev (x86_64/linux)) > Content-Length: 0 > > > _______________________________________________ > 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 Thu Nov 1 13:29:40 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 1 Nov 2018 19:29:40 +0200 Subject: [OpenSIPS-Users] CPU 100% with TCP In-Reply-To: References: <55F1EF32-61C8-45E2-B77C-1A3D14C6FA32@genesys.com> <0bed4419-36ff-b0ad-425a-c5c8283e1943@opensips.org> <97134D2E-9BFE-4CDF-AC24-B4A211EA9BE7@genesys.com> <356959a3-97ba-38b3-7ca4-89403d0a3335@opensips.org> Message-ID: <8f6a5096-b719-bfae-dfac-b5dad161601f@opensips.org> Hi Ben, First be sure you have the DBG_LOCK option compiled in. Do the "opensips -V" and see the output flags. Next step will be to force an SIGSEGV to opensips (killall -11 opensips) when the deadlockoccurs - I need a core file to inspect (assuming that runtime inspection with gdb is not possible). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 10/31/2018 09:07 PM, Ben Newlin wrote: > > Bogdan, > > For the first test I have done as you suggested and disabled only > async operation for HEP, so it is still using TCP. I will send you the > trap info directly as it is too large. I also compiled with the > DBG_LOCK option, but am unsure whether that extra information will be > available in the trap output or do you need something else? > > I am now going to switch HEP to use UDP to mirror our production > environment and try to reproduce again. Wish me luck! ☺ > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > *Date: *Monday, October 29, 2018 at 2:19 PM > *To: *Ben Newlin , OpenSIPS users mailling > list > *Subject: *Re: [OpenSIPS-Users] CPU 100% with TCP > > Hi Ben, > > I checked the error trace and it should not leave any dangling lock > (due mishandled error). Before disabling HEP, try to disable the async > support for HEP. > > If you claim that the same 100% CPU happens with HEP + UDP, send me a > trap for that too, as in the previous case, the deadlock was > exclusively HEP + TCP related. > > Anyhow, as the original trap showed a deadlock, next step will be to > recompile with the DBG_LOCK option - this enables extra code to > debug/troubleshoot locking related issues - are you able to do it? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Bootcamp 2018 > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 10/26/2018 04:14 PM, Ben Newlin wrote: > > Bogdan, > > Actually, yes we do. Looking back I can see these errors just > before the issue occurs: > > Oct 24 19:00:36 [5700] ERROR:proto_hep:send_hep_message: Cannot > send hep message! > > Oct 24 19:00:36 [5700] ERROR:proto_hep:msg_send: send() to > 10.32.163.211:9061 for proto hep_tcp/9 failed > > Oct 24 19:00:36 [5700] ERROR:proto_hep:hep_tcp_send: failed to send > > Oct 24 19:00:36 [5700] ERROR:proto_hep:async_tsend_stream: Failed > first TCP async send : (32) Broken pipe > > I will try disabling HEP and see if we can reproduce. > > Just for information, I have been reproducing the issue in our > testing environment which uses TCP for HEP, however the issue is > occurring in our production environment as well which is still > using UDP for HEP. > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > > *Date: *Friday, October 26, 2018 at 3:06 AM > *To: *Ben Newlin > , OpenSIPS users mailling list > > *Subject: *Re: [OpenSIPS-Users] CPU 100% with TCP > > Hi Ben, > > Thank you for the info. > > It looks like the processes get stuck into a HEP related internal > lock - do you see any HEP related errors in your logs, prior to > the dead-lock ? > > Also, as PoC, could you disabled HEP tracing to see if the problem > goes away ? > > Thanks, > > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2018 > > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 10/24/2018 10:18 PM, Ben Newlin wrote: > > Bogdan, > > I have run the command but the output was too large for > pastebin so I have sent it to you directly. > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > > *Date: *Wednesday, October 24, 2018 at 5:17 AM > *To: *OpenSIPS users mailling list > , Ben Newlin > > *Subject: *Re: [OpenSIPS-Users] CPU 100% with TCP > > Hi Ben, > > Could you run "opensipsctl trap" ? > > Regards, > > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2018 > > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 10/24/2018 12:56 AM, Ben Newlin wrote: > > Hi, > > We have implemented TCP recently and are performing > TCP<->UDP translation on one of our proxy types. This > proxy only exists for that purpose; there are no DB > queries, REST calls, or anything like that. It is designed > to be very fast and high throughput. > > Recently we have found that when the remote endpoint of a > TCP connection is lost, i.e. the server goes down, while > under moderate load OpenSIPS quickly reaches 100% CPU and > becomes unresponsive. When this occurs, the “top” command > shows that between 30-90% CPU is in System (kernel) space, > and each OpenSIPS TCP process shows many times the normal > CPU. We are running OpenSIPS 2.4.2 on Amazon Linux. > > I obtained as much information as I could using ps, > strace, and gdb here: https://pastebin.com/JP3DnCqs > . We can reproduce the > failure consistently by removing a server during call traffic. > > A few things I noticed: > > * The number of running threads reported by OpenSIPS > doesn’t align with our configuration, copied here: > > ####### Global Parameters ######### > > children=32 > > #// Allow 503 to pass back to Control > > disable_503_translation=yes > > #// Even though we are not receiving HEP, > > #// this listener is required by OpenSIPS > > #// in order to use the proto_hep module. :/ > > listen=hep_tcp:10.32.40.245:9061 use_children 1 > > #// Configure the listeners > > listen=udp:10.32.40.245:5060 as XXX.XXX.XXX.XXX > > listen=tcp:10.32.40.245:5060 as XXX.XXX.XXX.XXX > > #// Transaction Module > > loadmodule "tm.so" > > modparam("tm", "restart_fr_on_each_reply", 0) > > modparam("tm", "timer_partitions", 8) > > modparam("tm", "onreply_avp_mode", 1) > > modparam("tm", "wt_timer", 10) > > According to the documentation if “tcp_children” is not > set then the value of “children” will be used [1], but we > have set “children” to 32 and only have the default 8 TCP > processes. Also we appear to only have 1 timer process, > although we have set the number of timer partitions to 8. > > * The server that is terminated was using TCP > connections exclusively, but all of the CPU seems to > be in the UDP threads. The one I looked at appeared to > be handling a CANCEL to one of the calls that was > active and was attempting to send it out via TCP. I’m > not sure why it would be trying to relay the CANCEL as > no 100 Trying had been received from the server. I > have noticed that in 2.x OpenSIPS will now send > CANCELs for transactions even when 100 Trying was not > received. Is that intentional? RFC 3261 states that no > CANCEL should be sent unless a provisional response > has been received. > > Any assistance with this would be appreciated. > > [1] - > http://www.opensips.org/Documentation/Script-CoreParameters-2-4#toc66 > > 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 Thu Nov 1 14:14:17 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 1 Nov 2018 20:14:17 +0200 Subject: [OpenSIPS-Users] mi_xmlrpc_ng for ul_dump In-Reply-To: References: Message-ID: Hi Schneur, Yes, the ul_dump has an optional param, so you can use it without any parameters. The error you get says there is no payload found in your HTTP request. Normally the HTTP request should carry the XML with the request description. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 10/31/2018 12:13 PM, Schneur Rosenberg wrote: > Hi, I was relying on the DB to get information regarding registered > users , but I'm having issues with the DB so I'm trying to get it > directly from opensips using mi_xmlrpc_ng, I'm using PHP with the > xmlrpc_encode_request function., but the ul_dump method does not seem > to have any params, when I remove the params I get a "401 Empty > request" reply, and if i place any other value in the params I get a > "500 Internal server error" reply, whats the proper way to get a > ul_dump using mi_xmlrpc_ng? > > S. Rosenberg > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From bogdan at opensips.org Thu Nov 1 14:16:30 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 1 Nov 2018 20:16:30 +0200 Subject: [OpenSIPS-Users] async(wait_for_event()) statement in a branch_route[] In-Reply-To: References: Message-ID: <740921f2-0a87-f226-5b10-8bdceed6f54d@opensips.org> Hi Vitalii, The async() statement can be used only in REQUEST route. Still, whatever you need to do in terms of async query from branch route can also be done from request route. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/01/2018 01:46 PM, Vitalii Aleksandrov wrote: > Hi, > > I'm a bit new to opensips, while have some experience with kamailio. > > Trying to figure out whether it's possible to use async() statement > from branch_route[]. > > From the documentation I understood that async() functionality is > tightly connected to the TM module and creates some context attached > to a transaction, suspends it and resumes a transaction on an event. > If said above is correct I suppose calling async() from a > branch_route[] to suspend only one branch and expect that other > branches will continue their normal execution is not what it was > designed for.. > > Is there any way to forward a branch while other branches wait for an > async event or i/o? > > Would appreciate any ideas. > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From adrian.fretwell at topgreen.co.uk Thu Nov 1 16:27:28 2018 From: adrian.fretwell at topgreen.co.uk (Adrian Fretwell) Date: Thu, 01 Nov 2018 20:27:28 +0000 Subject: [OpenSIPS-Users] Fraud Detection Module:check_fraud - problem with profile_id Message-ID: <1541104048.1085.33.camel@topgreen.co.uk> Hello, I have been using the Fraud detection module for two years now with no issues, but I only used one profile and call the check_fraud function as follows: check_fraud("$var(t_fd_user)", "$rU", "1"); Now I have the need for two profiles and wanted to call check_fraud using a PV as the profile_id parameter like so: $var(t_fd_p) = "1"; check_fraud("$var(t_fd_user)", "$rU", "$var(t_fd_p)"); But I always get a CORE PV error and the function fails. Is what I want to do possible? What am I doing wrong? Kind regards, Adrian Fretwell -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Fri Nov 2 05:02:03 2018 From: liviu at opensips.org (Liviu Chircu) Date: Fri, 2 Nov 2018 11:02:03 +0200 Subject: [OpenSIPS-Users] Fraud Detection Module:check_fraud - problem with profile_id In-Reply-To: <1541104048.1085.33.camel@topgreen.co.uk> References: <1541104048.1085.33.camel@topgreen.co.uk> Message-ID: <767c3fd4-191c-e2f9-260d-8fc24ef81673@opensips.org> Hi Adrian, Pretty annoying, indeed.  The idea is that the "profile ID" parameter must be integer or a pseudo-var holding an integer (only).  So you'd fix your code as follows: $var(t_fd_p) = 1; # notice I removed the quotes! check_fraud("$var(t_fd_user)", "$rU", "$var(t_fd_p)"); Best regards, Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 01.11.2018 22:27, Adrian Fretwell wrote: > Hello, > I have been using the Fraud detection module for two years now with no > issues, but I only used one profile and call the check_fraud function > as follows: > > check_fraud("$var(t_fd_user)", "$rU", "1"); > > Now I have the need for two profiles and wanted to call check_fraud > using a PV as the profile_id parameter like so: > > $var(t_fd_p) = "1"; > check_fraud("$var(t_fd_user)", "$rU", "$var(t_fd_p)"); > > But I always get a CORE PV error and the function fails. Is what I > want to do possible? What am I doing wrong? > > > Kind regards, > Adrian Fretwell > > _______________________________________________ > 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 adrian.fretwell at topgreen.co.uk Fri Nov 2 05:13:36 2018 From: adrian.fretwell at topgreen.co.uk (Adrian Fretwell) Date: Fri, 02 Nov 2018 09:13:36 +0000 Subject: [OpenSIPS-Users] Fraud Detection Module:check_fraud - problem with profile_id In-Reply-To: <767c3fd4-191c-e2f9-260d-8fc24ef81673@opensips.org> References: <1541104048.1085.33.camel@topgreen.co.uk> <767c3fd4-191c-e2f9-260d-8fc24ef81673@opensips.org> Message-ID: <1541150016.1085.37.camel@topgreen.co.uk> Hi Liviu,Thankyou for the information. I'm actually pulling the profile id from a char column in the Db so I just need to make sure I perform and integer conversion on it. Many thanks,Adrian.  On Fri, 2018-11-02 at 11:02 +0200, Liviu Chircu wrote: >     Hi Adrian, > >     Pretty annoying, indeed.  The idea is that the "profile ID" >         parameter must be integer or a pseudo-var holding an integer >         (only).  So you'd fix your code as follows: > >     $var(t_fd_p) = 1; # notice I removed the quotes! > >        > >      >      >     check_fraud("$var(t_fd_user)", "$rU", "$var(t_fd_p)"); > >      > >        > >     Best regards, > >        > >      >     On 01.11.2018 22:27, Adrian Fretwell >       wrote: > >      > >      > >        > >       Hello, > >       I have been using the Fraud detection module for two years > >         now with no issues, but I only used one profile and call > > the > >         check_fraud function as follows: > >        > > > >        > >       check_fraud("$var(t_fd_user)", "$rU", "1"); > >        > > > >        > >       Now I have the need for two profiles and wanted to call > >         check_fraud using a PV as the profile_id parameter like so: > >        > > > >        > >       $var(t_fd_p) = "1"; > >       check_fraud("$var(t_fd_user)", "$rU", "$var(t_fd_p)"); > >        > > > >        > >       But I always get a CORE PV error and the function fails. Is > >         what I want to do possible? What am I doing wrong? > >        > > > >        > >        > > > >        > >       Kind regards, > >       Adrian Fretwell > >        > > > >        > >       _______________________________________________ > > 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 vitalik.voip at gmail.com Fri Nov 2 06:04:44 2018 From: vitalik.voip at gmail.com (Vitalii Aleksandrov) Date: Fri, 2 Nov 2018 12:04:44 +0200 Subject: [OpenSIPS-Users] async(wait_for_event()) statement in a branch_route[] In-Reply-To: <740921f2-0a87-f226-5b10-8bdceed6f54d@opensips.org> References: <740921f2-0a87-f226-5b10-8bdceed6f54d@opensips.org> Message-ID: <02dc3d1c-72bc-bb7e-6c68-bb62992f1441@gmail.com> Hi Bogdan, Ok. Let's say I have 5 branches and before calling t_relay() want to iterate over branches and do some async() action. From my understanding if I do something via async() it will suspend the whole transaction with all branches and if I need the same action for every branch it will create a big PDD because all async() calls will be done sequentially. As a workaround I think I can send all branches to myself (spiral call) and receiving those INVITEs from loopback opensips will create a separate transaction for every destination so I'll be able to run async() tasks in parallel and forward a call to every destination immediately its async() task is finished. > Hi Vitalii, > > The async() statement can be used only in REQUEST route. Still, > whatever you need to do in terms of async query from branch route can > also be done from request route. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer >   http://www.opensips-solutions.com > OpenSIPS Bootcamp 2018 >   http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 11/01/2018 01:46 PM, Vitalii Aleksandrov wrote: >> Hi, >> >> I'm a bit new to opensips, while have some experience with kamailio. >> >> Trying to figure out whether it's possible to use async() statement >> from branch_route[]. >> >> From the documentation I understood that async() functionality is >> tightly connected to the TM module and creates some context attached >> to a transaction, suspends it and resumes a transaction on an event. >> If said above is correct I suppose calling async() from a >> branch_route[] to suspend only one branch and expect that other >> branches will continue their normal execution is not what it was >> designed for.. >> >> Is there any way to forward a branch while other branches wait for an >> async event or i/o? >> >> Would appreciate any ideas. >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > From razvan at opensips.org Fri Nov 2 08:11:40 2018 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Fri, 2 Nov 2018 14:11:40 +0200 Subject: [OpenSIPS-Users] [RELEASE] OpenSIPS 2.4.3 Release Planning Message-ID: Hi, Everyone! After a long series of bug fixes, especially the presence modules ones, we decided it's time to make a new OpenSIPS 2.4 release. Therefore I would like to announce that the new OpenSIPS 2.4.3 release is scheduled for next Thursday, 8th of November 2018. We are now pushing hard to solve all known issues - but if you do have any other problems with OpenSIPS 2.4.2, now is the time to let us know by opening a ticket[1]. Thank you all for using OpenSIPS! [1] https://github.com/OpenSIPS/opensips/issues Cheers, -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com From jeff at ugnd.org Fri Nov 2 19:26:00 2018 From: jeff at ugnd.org (Jeff Pyle) Date: Fri, 2 Nov 2018 19:26:00 -0400 Subject: [OpenSIPS-Users] combining topology_hiding and mid_registrar Message-ID: Hello, My v2.4 implementation is a type of SBC suitable for the edge of a customer network. I have topology_hiding working for INVITEs. I have mid_registrar working for REGISTERs. I do not have them working together. I'd like to completely hide the topology of the inside network when sending traffic to the outside main registrar / ITSP. topology_hiding does a nice job with that for INVITEs, but mid_registrar still leaks details through Via headers and such. Is it possible to combine them? - Jeff -------------- next part -------------- An HTML attachment was scrubbed... URL: From jehanzaib.kiani at gmail.com Sun Nov 4 16:55:05 2018 From: jehanzaib.kiani at gmail.com (J E H A N Z A I B) Date: Mon, 5 Nov 2018 10:55:05 +1300 Subject: [OpenSIPS-Users] =?utf-8?q?=28no_subject=29?= Message-ID: Hi team, I used redis cache for dialog storage. I have 2 different servers both are sharing the same redis. Is the profile size shared in this case? here is the dialog config. loadmodule "dialog.so" modparam("dialog", "enable_stats", 1) modparam("dialog", "cachedb_url", "redis:mysip://mysipx.xx.xx:xxxx/") This is how I check my profile size. create_dialog(); set_dlg_profile("myuniqprof","$avp(myprofile_id)"); get_profile_size("myuniqprof","$avp(myprofile_id )","$var(current_profile_size)"); Please note I am using version: opensips 1.11.3-notls -- Regards, Jehanzaib -------------- next part -------------- An HTML attachment was scrubbed... URL: From jehanzaib.kiani at gmail.com Mon Nov 5 05:27:27 2018 From: jehanzaib.kiani at gmail.com (J E H A N Z A I B) Date: Mon, 5 Nov 2018 23:27:27 +1300 Subject: [OpenSIPS-Users] redis cache sharing for dialog storage? Message-ID: Hi team, I used redis cache for dialog storage. I have 2 different servers both are sharing the same redis. Is the profile size shared in this case? here is the dialog config. loadmodule "dialog.so" modparam("dialog", "enable_stats", 1) modparam("dialog", "cachedb_url", "redis:mysip://mysipx.xx.xx:xxxx/") This is how I check my profile size. create_dialog(); set_dlg_profile("myuniqprof","$avp(myprofile_id)"); get_profile_size("myuniqprof","$avp(myprofile_id )","$var(current_profile_size)"); Please note I am using version: opensips 1.11.3-notls -- Regards, Jehanzaib -------------- next part -------------- An HTML attachment was scrubbed... URL: From john.quick at smartvox.co.uk Mon Nov 5 09:59:02 2018 From: john.quick at smartvox.co.uk (John Quick) Date: Mon, 5 Nov 2018 14:59:02 -0000 Subject: [OpenSIPS-Users] OpenSIPS Control Panel 8.2.4 was released Message-ID: <001801d47518$177f4a70$467ddf50$@smartvox.co.uk> Hello Bogdan, Thanks for releasing a new version of the Control Panel. There are some great new features in this version that I want to try. I have been testing OCP v8 on a CentOS 6 server. I'm finding quite a lot of errors - it is possible some of these are associated with the operating system or the version of PHP I am using. PHP is v 5.3.3 The most critical problem for me concerns administrator access: I am unable to change the password for the admin user. As a precaution, I created a new admin user with a different name and then tried to revoke the key privileges of the original admin user, but this made no difference until I edited the source code and changed the tool name passed to the get_priv() function as detailed below. Here are my full notes on the problems encountered so far and code adjustments I have made: In login.php: Warns me "It is not safe to rely on the systems timezone settings. You are required to use date.timezone or date_default_timezone_set() function". As a work-around, I added calls to date_default_timezone_set("Europe/London") in a couple of places to stop these warnings being written. Many of the tabs display a blank page. In the httpd log file; the httpd log files shows errors about "unexpected [". When I look up the file and line number shown in the httpd log message, it's always a line that calls the die() function to handle failure of an SQL request. I found it was due to the use of "errorInfo()[2]". If you remove the [2], it doesn't give an error and the page is rendered too. Used grep to find every case like this: # grep -rl "errorInfo()\[" . ./users/user_management/template/user_management.main.php ./users/acl_management/template/acl_management.main.php ./system/dialplan/template/dialplan.main.php ./system/dialplan/dialplan.php ./system/siptrace/template/tracer.main.php ./system/drouting/template/groups.main.php ./system/drouting/template/rules.main.php ./system/drouting/template/carriers.main.php ./system/drouting/template/gateways.main.php ./system/drouting/lib/carriers.test.inc.php ./system/drouting/lib/carriers.functions.inc.php ./system/tls_mgm/lib/data_loader.php ./system/loadbalancer/template/loadbalancer.main.php ./system/dispatcher/template/dispatcher.main.php ./system/dispatcher/template/dispatcher.form.php ./system/tviewer/lib/data_loader.php ./system/callcenter/lib/data_loader.php ./admin/list_admins/template/list_admins.main.php Admin Access Control I cannot find a way to change the admin password. It always accepts the password of "opensips" even after resetting it in the Admin User editing form and also deleting it from the field "password" in the ocp_admin_privileges table. On further testing, I then found that none of the changes I make to the admin user get saved to the DB - not just the password, but also the first name and last name fields. Access privileges were not working to restrict whether the logged-in user could change their own (and other people's) admin privileges. Found this was because the get_priv() function was being called with the argument set to "list_admins" and that the arrays used to store access privileges only have a key called "user_management" and do not have one for "list_admins". There are no lines of code to act as a "catch-all" in the get_priv() function so if the tool name passed to the function does not match any of the known tab names, then it fails to set the session variable read_only. The absence of any value is then treated as "grant full access" in the corresponding module so it is not very safe. As a work-around, I edited web/tools/admin/list_admins/index.php and changed the following line: << get_priv("list_admins"); >> get_priv("user_management"); John Quick Smartvox Limited From govoiper at gmail.com Mon Nov 5 10:29:11 2018 From: govoiper at gmail.com (SamyGo) Date: Mon, 5 Nov 2018 10:29:11 -0500 Subject: [OpenSIPS-Users] Raise event from MI/FIFO interface(externally) In-Reply-To: <9b3e0dbc-0190-4d15-96b5-c04edcaca90f@opensips.org> References: <9b3e0dbc-0190-4d15-96b5-c04edcaca90f@opensips.org> Message-ID: Hi Bogdan, Sure thing I'd open up a feature request. Yes thats what we decided to go with to use sipsak or something like that to trigger SIP packet and make use of it. Apologize for late reply; thanks a lot for your response. Regards, Sammy On Tue, Oct 16, 2018 at 4:29 AM Bogdan-Andrei Iancu wrote: > Hi Sammy, > > The Event Interface philosophy is to allow external apps to get access to > the events generated by OpenSIPS; basically a data flow from OpenSIPS to > outside world. So, there is no way to trigger an event from outside (via > MI). > Still, your example is valid, so please open a feature request on github > and we could add a "raise_event" MI function. > > As really (really) dirty hack, you can use (if possible) some SIP OPTIONS > to trigger the event. From the web send an OPTIONS via UDP, with a special > RURI -> you identify the RURI in script and do a raise_event() in script. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Bootcamp 2018 > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 10/11/2018 07:35 PM, SamyGo wrote: > > Hi, > > I'm trying to find out document pages regarding raising events from > outside the OpenSIPS via fifo/mi_* modules. All I have read so far is > opensips can send events and their data OUT to external "subscribing" > applications. There is even a fifo command to subscribe for an event from > FIFO layer. > > Kindly guide me as how do I tell OpenSIPS that a particular Event has > triggered. > > My usage scenario is (OpenSIPS 2.2.7) a caller is > waiting_on_event("OPENSIPS_BOOTCAMP"); > Now some external web monitoring bot just realized Bootcamp has started > and it wants to raise this event to the waiting caller ! > > opensipsctl fifo raise_event OPENSIPS_BOOTCAMP 22Oct18 Romania > > Best 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 z at startit.ru Mon Nov 5 10:30:25 2018 From: z at startit.ru (zzz) Date: Mon, 5 Nov 2018 15:30:25 +0000 Subject: [OpenSIPS-Users] OpenSIPS Control Panel 8.2.4 was released In-Reply-To: References: Message-ID: <8B36F227BD22B041AEA7015FD914CD9503817141FA@JET-EX02.jettel.ru> Hi Bogdan, Thanks for all you do. Ive tried the new release on debian8, it works great. Would you please change "domaini" to "domain" in the file acl_management.php https://github.com/OpenSIPS/opensips-cp/blob/8.2.4/web/tools/users/acl_management/acl_management.php lines 156 and 169. Thanks, Yuriy -----Original Message----- From: Users [mailto:users-bounces at lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Thursday, November 1, 2018 2:12 PM To: users at lists.opensips.org; developensips ; business at lists.opensips.org Subject: [OpenSIPS-Users] OpenSIPS Control Panel 8.2.4 was released Hi all, We are proud and happy to announce the release of the OpenSIPS Control Panel 8.2.4 . This OpenSIPS Control Panel is a provisioning (SIP users and OpenSIPS system) web interface design for and compatible with OpenSIPS 2.4 LTS - http://controlpanel.opensips.org . The OCP 8.2.4 has new awesome additions : * a new modern look by reworking the entire CSS and layout, see http://controlpanel.opensips.org/screenshots.php * a new tool for handling the RTPEngine instances via database * a new tool for provisioning TLS domains and certificates via database * a more powerful Tviewer engine to allow more complex displaying, correlation and validation of the data types and to create inter-tool web links. In the same time the OCP 8.2.4 brings great improvements under the hood: * compatibility with PHP 7, by migrating from MDB2 to PDO database support * better security of the DB operations by migrating to prepared statements, to avoid SQL injections * improved security for the WEB sessions and page access to avoid illegitimate direct access to sub-pages * extended tooltip and data validation support * code refactoring for easy further development This new OpenSIPS Control Panel release is a significant step forward for the project in terms of vision. All this happened thanks to the people and companies involved in the development of the project. Special thanks to the Voicenter team - https://www.voicenter.com - for its great contribution in re-styling and re-designing this web interface. Download and enjoy it as it's freshly baked for you, -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users From govoiper at gmail.com Mon Nov 5 10:46:41 2018 From: govoiper at gmail.com (SamyGo) Date: Mon, 5 Nov 2018 10:46:41 -0500 Subject: [OpenSIPS-Users] async(wait_for_event()) statement in a branch_route[] In-Reply-To: <02dc3d1c-72bc-bb7e-6c68-bb62992f1441@gmail.com> References: <740921f2-0a87-f226-5b10-8bdceed6f54d@opensips.org> <02dc3d1c-72bc-bb7e-6c68-bb62992f1441@gmail.com> Message-ID: Hi Bogdan, Intrigued by this thread, I recall in older threads I was told that I can do something like this, I'm going to use my example. event_route[ABCD]{ route(DO_SOMETHING); } and then in that DO_SOMETHING route I can use the redis/mysql functions or anything that is not acceptable to be used in the event_route. By the same analogy is this also the right way to write? branch_route[per_branch_ops]{ xlog("I'm a branch of the call , I cant do alot here"); route(WAIT_FOR_EVENT); } where; route[WAIT_FOR_EVENT]{ async(wait_for_event("ABCX","$avp(filter"),"40"),"resume_here"); } I think another way to ask Vitalii's question would be, how parallel is "parallel forking".? If there are 5 branches and each of them takes between A-Z amount of time to process script functions then all of them are going to be relayed out only when ALL branches are done processing? or each branch is relayed out independent of how much time other branch is taking. Hope to catch some insight on this. (in context of opensips version 2.4.2) Regards, Sammy On Fri, Nov 2, 2018 at 6:07 AM Vitalii Aleksandrov wrote: > Hi Bogdan, > > Ok. Let's say I have 5 branches and before calling t_relay() want to > iterate over branches and do some async() action. From my understanding > if I do something via async() it will suspend the whole transaction with > all branches and if I need the same action for every branch it will > create a big PDD because all async() calls will be done sequentially. > > As a workaround I think I can send all branches to myself (spiral call) > and receiving those INVITEs from loopback opensips will create a > separate transaction for every destination so I'll be able to run > async() tasks in parallel and forward a call to every destination > immediately its async() task is finished. > > > > Hi Vitalii, > > > > The async() statement can be used only in REQUEST route. Still, > > whatever you need to do in terms of async query from branch route can > > also be done from request route. > > > > Regards, > > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2018 > > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > > > On 11/01/2018 01:46 PM, Vitalii Aleksandrov wrote: > >> Hi, > >> > >> I'm a bit new to opensips, while have some experience with kamailio. > >> > >> Trying to figure out whether it's possible to use async() statement > >> from branch_route[]. > >> > >> From the documentation I understood that async() functionality is > >> tightly connected to the TM module and creates some context attached > >> to a transaction, suspends it and resumes a transaction on an event. > >> If said above is correct I suppose calling async() from a > >> branch_route[] to suspend only one branch and expect that other > >> branches will continue their normal execution is not what it was > >> designed for.. > >> > >> Is there any way to forward a branch while other branches wait for an > >> async event or i/o? > >> > >> Would appreciate any ideas. > >> > >> > >> _______________________________________________ > >> 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 Mon Nov 5 10:50:51 2018 From: govoiper at gmail.com (SamyGo) Date: Mon, 5 Nov 2018 10:50:51 -0500 Subject: [OpenSIPS-Users] redis cache sharing for dialog storage? In-Reply-To: References: Message-ID: I have a strong feeling that you're using an old version of opensips to expect it to share dialog states/profiles. I think you'll need to use newer opensips 2.4+ having dialog sharing capability using proto_bin and clusterer module: http://www.opensips.org/html/docs/modules/2.4.x/dialog.html#param_profile_replication_cluster On Mon, Nov 5, 2018 at 5:30 AM J E H A N Z A I B wrote: > Hi team, > > I used redis cache for dialog storage. I have 2 different servers both are > sharing the same redis. Is the profile size shared in this case? > > here is the dialog config. > > loadmodule "dialog.so" > modparam("dialog", "enable_stats", 1) > modparam("dialog", "cachedb_url", "redis:mysip://mysipx.xx.xx:xxxx/") > > This is how I check my profile size. > create_dialog(); > set_dlg_profile("myuniqprof","$avp(myprofile_id)"); > get_profile_size("myuniqprof","$avp(myprofile_id > )","$var(current_profile_size)"); > > Please note I am using version: opensips 1.11.3-notls > > > -- > Regards, > Jehanzaib > _______________________________________________ > 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 Nov 5 12:01:18 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 5 Nov 2018 19:01:18 +0200 Subject: [OpenSIPS-Users] OpenSIPS Control Panel 8.2.4 was released In-Reply-To: <8B36F227BD22B041AEA7015FD914CD9503817141FA@JET-EX02.jettel.ru> References: <8B36F227BD22B041AEA7015FD914CD9503817141FA@JET-EX02.jettel.ru> Message-ID: <29d0413d-5171-4323-b416-42359d90d6f4@opensips.org> Hi Yuriy, Thanks for spotting that, I just pushed a fix: https://github.com/OpenSIPS/opensips-cp/commit/9556483a80514509def862bd2c870f738adda29a Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/05/2018 05:30 PM, zzz wrote: > Hi Bogdan, > Thanks for all you do. > Ive tried the new release on debian8, it works great. > Would you please change "domaini" to "domain" in the file acl_management.php > https://github.com/OpenSIPS/opensips-cp/blob/8.2.4/web/tools/users/acl_management/acl_management.php > lines 156 and 169. > Thanks, > Yuriy > > > > -----Original Message----- > From: Users [mailto:users-bounces at lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu > Sent: Thursday, November 1, 2018 2:12 PM > To: users at lists.opensips.org; developensips ; business at lists.opensips.org > Subject: [OpenSIPS-Users] OpenSIPS Control Panel 8.2.4 was released > > Hi all, > > We are proud and happy to announce the release of the OpenSIPS Control Panel 8.2.4 . This OpenSIPS Control Panel is a provisioning (SIP users and OpenSIPS system) web interface design for and compatible with OpenSIPS 2.4 LTS - http://controlpanel.opensips.org . > > > The OCP 8.2.4 has new awesome additions : > > * a new modern look by reworking the entire CSS and layout, see http://controlpanel.opensips.org/screenshots.php > > * a new tool for handling the RTPEngine instances via database > > * a new tool for provisioning TLS domains and certificates via database > > * a more powerful Tviewer engine to allow more complex displaying, correlation and validation of the data types and to create inter-tool web links. > > > In the same time the OCP 8.2.4 brings great improvements under the hood: > > * compatibility with PHP 7, by migrating from MDB2 to PDO database support > > * better security of the DB operations by migrating to prepared statements, to avoid SQL injections > > * improved security for the WEB sessions and page access to avoid illegitimate direct access to sub-pages > > * extended tooltip and data validation support > > * code refactoring for easy further development > > > This new OpenSIPS Control Panel release is a significant step forward for the project in terms of vision. All this happened thanks to the people and companies involved in the development of the project. > > Special thanks to the Voicenter team - https://www.voicenter.com - for its great contribution in re-styling and re-designing this web interface. > > > Download and enjoy it as it's freshly baked for you, > From bogdan at opensips.org Mon Nov 5 12:07:47 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 5 Nov 2018 19:07:47 +0200 Subject: [OpenSIPS-Users] async(wait_for_event()) statement in a branch_route[] In-Reply-To: <02dc3d1c-72bc-bb7e-6c68-bb62992f1441@gmail.com> References: <740921f2-0a87-f226-5b10-8bdceed6f54d@opensips.org> <02dc3d1c-72bc-bb7e-6c68-bb62992f1441@gmail.com> Message-ID: <0bcfd5f7-0bbd-4ffd-5c1a-652d981c6764@opensips.org> Hi Vitalii, Yes, that is correct. The idea is you cannot do async() ops in the branch route as this type of route is strongly interfering with the SIP signaling - branch route is triggered when a new branch is set out (as a result of a t_relay) and no signaling operations are allowed (as you are already doing a signaling op). Still, this limitation is valid only for async calls () as async is based on the transaction status. But, you can do regular sync() queries from the branch route, with a risk of blocking the processes of opensips. The spiraling (as you described) is also a valid option, but you need to be a bit careful with the script logic to be sure you do not mix the loops :) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/02/2018 12:04 PM, Vitalii Aleksandrov wrote: > Hi Bogdan, > > Ok. Let's say I have 5 branches and before calling t_relay() want to > iterate over branches and do some async() action. From my > understanding if I do something via async() it will suspend the whole > transaction with all branches and if I need the same action for every > branch it will create a big PDD because all async() calls will be done > sequentially. > > As a workaround I think I can send all branches to myself (spiral > call) and receiving those INVITEs from loopback opensips will create a > separate transaction for every destination so I'll be able to run > async() tasks in parallel and forward a call to every destination > immediately its async() task is finished. > > >> Hi Vitalii, >> >> The async() statement can be used only in REQUEST route. Still, >> whatever you need to do in terms of async query from branch route can >> also be done from request route. >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> OpenSIPS Bootcamp 2018 >> http://opensips.org/training/OpenSIPS_Bootcamp_2018/ >> >> On 11/01/2018 01:46 PM, Vitalii Aleksandrov wrote: >>> Hi, >>> >>> I'm a bit new to opensips, while have some experience with kamailio. >>> >>> Trying to figure out whether it's possible to use async() statement >>> from branch_route[]. >>> >>> From the documentation I understood that async() functionality is >>> tightly connected to the TM module and creates some context attached >>> to a transaction, suspends it and resumes a transaction on an event. >>> If said above is correct I suppose calling async() from a >>> branch_route[] to suspend only one branch and expect that other >>> branches will continue their normal execution is not what it was >>> designed for.. >>> >>> Is there any way to forward a branch while other branches wait for >>> an async event or i/o? >>> >>> Would appreciate any ideas. >>> >>> >>> _______________________________________________ >>> 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 bogdan at opensips.org Mon Nov 5 12:10:09 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 5 Nov 2018 19:10:09 +0200 Subject: [OpenSIPS-Users] OpenSIPS Control Panel 8.2.4 was released In-Reply-To: <001801d47518$177f4a70$467ddf50$@smartvox.co.uk> References: <001801d47518$177f4a70$467ddf50$@smartvox.co.uk> Message-ID: Hi John, Thanks for all your testing - this is something that we so most appreciate ;). To help us even more, could you please split the report (you have in the email) as individual issues and create corresponding tickets on github: https://github.com/OpenSIPS/opensips-cp/issues Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/05/2018 04:59 PM, John Quick wrote: > Hello Bogdan, > > Thanks for releasing a new version of the Control Panel. There are some > great new features in this version that I want to try. > > I have been testing OCP v8 on a CentOS 6 server. I'm finding quite a lot of > errors - it is possible some of these are associated with the operating > system or the version of PHP I am using. PHP is v 5.3.3 > > The most critical problem for me concerns administrator access: I am unable > to change the password for the admin user. > As a precaution, I created a new admin user with a different name and then > tried to revoke the key privileges of the original admin user, but this made > no difference until I edited the source code and changed the tool name > passed to the get_priv() function as detailed below. > > Here are my full notes on the problems encountered so far and code > adjustments I have made: > In login.php: Warns me "It is not safe to rely on the systems timezone > settings. You are required to use date.timezone or > date_default_timezone_set() function". As a work-around, I added calls to > date_default_timezone_set("Europe/London") in a couple of places to stop > these warnings being written. > > Many of the tabs display a blank page. In the httpd log file; the httpd log > files shows errors about "unexpected [". When I look up the file and line > number shown in the httpd log message, it's always a line that calls the > die() function to handle failure of an SQL request. > I found it was due to the use of "errorInfo()[2]". If you remove the [2], > it doesn't give an error and the page is rendered too. > > Used grep to find every case like this: > # grep -rl "errorInfo()\[" . > ./users/user_management/template/user_management.main.php > ./users/acl_management/template/acl_management.main.php > ./system/dialplan/template/dialplan.main.php > ./system/dialplan/dialplan.php > ./system/siptrace/template/tracer.main.php > ./system/drouting/template/groups.main.php > ./system/drouting/template/rules.main.php > ./system/drouting/template/carriers.main.php > ./system/drouting/template/gateways.main.php > ./system/drouting/lib/carriers.test.inc.php > ./system/drouting/lib/carriers.functions.inc.php > ./system/tls_mgm/lib/data_loader.php > ./system/loadbalancer/template/loadbalancer.main.php > ./system/dispatcher/template/dispatcher.main.php > ./system/dispatcher/template/dispatcher.form.php > ./system/tviewer/lib/data_loader.php > ./system/callcenter/lib/data_loader.php > ./admin/list_admins/template/list_admins.main.php > > Admin Access Control > I cannot find a way to change the admin password. It always accepts the > password of "opensips" even after resetting it in the Admin User editing > form and also deleting it from the field "password" in the > ocp_admin_privileges table. On further testing, I then found that none of > the changes I make to the admin user get saved to the DB - not just the > password, but also the first name and last name fields. > > Access privileges were not working to restrict whether the logged-in user > could change their own (and other people's) admin privileges. > Found this was because the get_priv() function was being called with the > argument set to "list_admins" and that the arrays used to store access > privileges only have a key called "user_management" and do not have one for > "list_admins". There are no lines of code to act as a "catch-all" in the > get_priv() function so if the tool name passed to the function does not > match any of the known tab names, then it fails to set the session variable > read_only. The absence of any value is then treated as "grant full access" > in the corresponding module so it is not very safe. > As a work-around, I edited web/tools/admin/list_admins/index.php and changed > the following line: > << get_priv("list_admins"); >>> get_priv("user_management"); > John Quick > Smartvox Limited > > From jehanzaib.kiani at gmail.com Mon Nov 5 18:11:44 2018 From: jehanzaib.kiani at gmail.com (J E H A N Z A I B) Date: Tue, 6 Nov 2018 12:11:44 +1300 Subject: [OpenSIPS-Users] redis cache sharing for dialog storage? In-Reply-To: References: Message-ID: Hi there, I am not sure why the dialog stats will not be shared. If all records are going to redis then I have a redis cluster which synchs the cache. When I fetch the profile size it should be same (if it is being fetched from the redis) across all the opensips node. I am bit sceptical to upgrade without knowing what's happening. On Tue, Nov 6, 2018 at 4:53 AM SamyGo wrote: > I have a strong feeling that you're using an old version of opensips to > expect it to share dialog states/profiles. I think you'll need to use newer > opensips 2.4+ having dialog sharing capability using proto_bin and > clusterer module: > > http://www.opensips.org/html/docs/modules/2.4.x/dialog.html#param_profile_replication_cluster > > > On Mon, Nov 5, 2018 at 5:30 AM J E H A N Z A I B < > jehanzaib.kiani at gmail.com> wrote: > >> Hi team, >> >> I used redis cache for dialog storage. I have 2 different servers both >> are sharing the same redis. Is the profile size shared in this case? >> >> here is the dialog config. >> >> loadmodule "dialog.so" >> modparam("dialog", "enable_stats", 1) >> modparam("dialog", "cachedb_url", "redis:mysip://mysipx.xx.xx:xxxx/") >> >> This is how I check my profile size. >> create_dialog(); >> set_dlg_profile("myuniqprof","$avp(myprofile_id)"); >> get_profile_size("myuniqprof","$avp(myprofile_id >> )","$var(current_profile_size)"); >> >> Please note I am using version: opensips 1.11.3-notls >> >> >> -- >> Regards, >> Jehanzaib >> _______________________________________________ >> 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 > -- Regards, Jehanzaib -------------- next part -------------- An HTML attachment was scrubbed... URL: From art666 at hotmail.com Tue Nov 6 06:29:00 2018 From: art666 at hotmail.com (Arto Kuiri) Date: Tue, 6 Nov 2018 11:29:00 +0000 Subject: [OpenSIPS-Users] OpenSIPS Control Panel 8.2.4 was released In-Reply-To: References: <001801d47518$177f4a70$467ddf50$@smartvox.co.uk>, Message-ID: Hi, I think there is still something at control panel's user management. You cannot create another admin user from UI. Even if you select read-write permission to all modules for new user, it cannot access to Users (at top panel). If you change manually permissions field at database from "read-write, read-wri...." to "all" it will work. I think there should be own permission settings for "Control Panel Users" ? Best regards, Arto Kuiri Lähettäjä: Users käyttäjän Bogdan-Andrei Iancu puolesta Lähetetty: maanantai 5. marraskuuta 2018 19.10 Vastaanottaja: john.quick at smartvox.co.uk; users at lists.opensips.org Aihe: Re: [OpenSIPS-Users] OpenSIPS Control Panel 8.2.4 was released   Hi John, Thanks for all your testing - this is something that we so most appreciate ;). To help us even more, could you please split the report (you have in the email) as individual issues and create corresponding tickets on github:          https://github.com/OpenSIPS/opensips-cp/issues Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer    http://www.opensips-solutions.com OpenSIPS Bootcamp 2018    http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/05/2018 04:59 PM, John Quick wrote: > Hello Bogdan, > > Thanks for releasing a new version of the Control Panel. There are some > great new features in this version that I want to try. > > I have been testing OCP v8 on a CentOS 6 server. I'm finding quite a lot of > errors - it is possible some of these are associated with the operating > system or the version of PHP I am using. PHP is v 5.3.3 > > The most critical problem for me concerns administrator access: I am unable > to change the password for the admin user. > As a precaution, I created a new admin user with a different name and then > tried to revoke the key privileges of the original admin user, but this made > no difference until I edited the source code and changed the tool name > passed to the get_priv() function as detailed below. > > Here are my full notes on the problems encountered so far and code > adjustments I have made: > In login.php: Warns me "It is not safe to rely on the systems timezone > settings. You are required to use date.timezone or > date_default_timezone_set() function".   As a work-around, I added calls to > date_default_timezone_set("Europe/London") in a couple of places to stop > these warnings being written. > > Many of the tabs display a blank page. In the httpd log file; the httpd log > files shows errors about "unexpected [".  When I look up the file and line > number shown in the httpd log message, it's always a line that calls the > die() function to handle failure of an SQL request. > I found it was due to the use of "errorInfo()[2]".  If you remove the [2], > it doesn't give an error and the page is rendered too. > > Used grep to find every case like this: >        # grep -rl "errorInfo()\[" . >        ./users/user_management/template/user_management.main.php >        ./users/acl_management/template/acl_management.main.php >        ./system/dialplan/template/dialplan.main.php >        ./system/dialplan/dialplan.php >        ./system/siptrace/template/tracer.main.php >        ./system/drouting/template/groups.main.php >        ./system/drouting/template/rules.main.php >        ./system/drouting/template/carriers.main.php >        ./system/drouting/template/gateways.main.php >        ./system/drouting/lib/carriers.test.inc.php >        ./system/drouting/lib/carriers.functions.inc.php >        ./system/tls_mgm/lib/data_loader.php >        ./system/loadbalancer/template/loadbalancer.main.php >        ./system/dispatcher/template/dispatcher.main.php >        ./system/dispatcher/template/dispatcher.form.php >        ./system/tviewer/lib/data_loader.php >        ./system/callcenter/lib/data_loader.php >        ./admin/list_admins/template/list_admins.main.php > > Admin Access Control > I cannot find a way to change the admin password. It always accepts the > password of "opensips" even after resetting it in the Admin User editing > form and also deleting it from the field "password" in the > ocp_admin_privileges table. On further testing, I then found that none of > the changes I make to the admin user get saved to the DB - not just the > password, but also the first name and last name fields. > > Access privileges were not working to restrict whether the logged-in user > could change their own (and other people's) admin privileges. > Found this was because the get_priv() function was being called with the > argument set to "list_admins" and that the arrays used to store access > privileges only have a key called "user_management" and do not have one for > "list_admins".  There are no lines of code to act as a "catch-all" in the > get_priv() function so if the tool name passed to the function does not > match any of the known tab names, then it fails to set the session variable > read_only. The absence of any value is then treated as "grant full access" > in the corresponding module so it is not very safe. > As a work-around, I edited web/tools/admin/list_admins/index.php and changed > the following line: > << get_priv("list_admins"); >>> get_priv("user_management"); > John Quick > Smartvox Limited > > _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users From rosenberg11219 at gmail.com Tue Nov 6 07:41:55 2018 From: rosenberg11219 at gmail.com (Schneur Rosenberg) Date: Tue, 6 Nov 2018 14:41:55 +0200 Subject: [OpenSIPS-Users] ACK increased Cseq on reinvite Message-ID: Hi, I have this scenario, I'm sending to a carrier that requires authentication and I use uac_auth(), when I send a call to the carrier everything works fine, but when OpenSIPs sends a reinvite the carrier responds witha 200 OK, OpenSIPs replies with a ACK but it falsely increments the CSeq , causing the carrier to ignore it, and then the carrier will keep on resending 200 OK's and OpenSIPs still returns the wrong ACK and eventually the carrier hangs up. I'm running OpenSIPs 2.4.1 thanks Scot From john.quick at smartvox.co.uk Tue Nov 6 12:14:16 2018 From: john.quick at smartvox.co.uk (John Quick) Date: Tue, 6 Nov 2018 17:14:16 -0000 Subject: [OpenSIPS-Users] OpenSIPS Control Panel 8.2.4 was released In-Reply-To: References: <001801d47518$177f4a70$467ddf50$@smartvox.co.uk> Message-ID: <002601d475f4$25d0fbf0$7172f3d0$@smartvox.co.uk> Hi Bogdan, Several issues previously mentioned were raised as issues. Thanks for the fixes so far. I don't want to raise issues if it is just my testing at fault, so here are some new questions: 1. Should it be possible to edit a record in System>Domains? When I click Edit, the page doesn’t change. I have read-write permission 2. How do you activate the TLS Management tab? I cannot see it 3. Why has support for MI using FIFO been dropped? It seems you must use MI_JSON (which needs the HTTPD module) John Quick Smartvox Limited > -----Original Message----- > From: Bogdan-Andrei Iancu > Sent: 05 November 2018 17:10 > To: john.quick at smartvox.co.uk; users at lists.opensips.org > Subject: Re: [OpenSIPS-Users] OpenSIPS Control Panel 8.2.4 was released > > Hi John, > >Thanks for all your testing - this is something that we so most appreciate ;). > > To help us even more, could you please split the report (you have in the > email) as individual issues and create corresponding tickets on github: > https://github.com/OpenSIPS/opensips-cp/issues > > Best regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer From johan at democon.be Tue Nov 6 15:15:34 2018 From: johan at democon.be (Johan De Clercq) Date: Tue, 6 Nov 2018 20:15:34 +0000 Subject: [OpenSIPS-Users] OpenSIPS Control Panel 8.2.4 was released In-Reply-To: <002601d475f4$25d0fbf0$7172f3d0$@smartvox.co.uk> References: <001801d47518$177f4a70$467ddf50$@smartvox.co.uk> , <002601d475f4$25d0fbf0$7172f3d0$@smartvox.co.uk> Message-ID: 3. This I can confirm mi_json only. Outlook voor iOS downloaden ________________________________ Van: Users namens John Quick Verzonden: dinsdag, november 6, 2018 6:16 PM Aan: 'Bogdan-Andrei Iancu'; users at lists.opensips.org Onderwerp: Re: [OpenSIPS-Users] OpenSIPS Control Panel 8.2.4 was released Hi Bogdan, Several issues previously mentioned were raised as issues. Thanks for the fixes so far. I don't want to raise issues if it is just my testing at fault, so here are some new questions: 1. Should it be possible to edit a record in System>Domains? When I click Edit, the page doesn’t change. I have read-write permission 2. How do you activate the TLS Management tab? I cannot see it 3. Why has support for MI using FIFO been dropped? It seems you must use MI_JSON (which needs the HTTPD module) John Quick Smartvox Limited > -----Original Message----- > From: Bogdan-Andrei Iancu > Sent: 05 November 2018 17:10 > To: john.quick at smartvox.co.uk; users at lists.opensips.org > Subject: Re: [OpenSIPS-Users] OpenSIPS Control Panel 8.2.4 was released > > Hi John, > >Thanks for all your testing - this is something that we so most appreciate ;). > > To help us even more, could you please split the report (you have in the > email) as individual issues and create corresponding tickets on github: > https://github.com/OpenSIPS/opensips-cp/issues > > Best regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and 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 bogdan at opensips.org Tue Nov 6 16:11:41 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 6 Nov 2018 23:11:41 +0200 Subject: [OpenSIPS-Users] OpenSIPS Control Panel 8.2.4 was released In-Reply-To: <002601d475f4$25d0fbf0$7172f3d0$@smartvox.co.uk> References: <001801d47518$177f4a70$467ddf50$@smartvox.co.uk> <002601d475f4$25d0fbf0$7172f3d0$@smartvox.co.uk> Message-ID: HI John, Thanks for the reports :) 1) I tried to reproduce, but no luck - When Edit button is clicked, the value is loaded in the upper text area, so you change it. Do you use the mighty "admin" user for this operation ? or an access user with custom permissions ? 2) Indeed, there was missing entry for the tls_mgm module in the modules.inc.php file, I fixed it in the git repo. 3) starting with 7.2.3 (las year), the mi_json became the only supported MI backend in OCP - the reason is that mi_json is the only one to offer a structured / encoded data output that can be easily and reliably parsed by OCP. Best Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/06/2018 07:14 PM, John Quick wrote: > Hi Bogdan, > > Several issues previously mentioned were raised as issues. Thanks for the fixes so far. > > I don't want to raise issues if it is just my testing at fault, so here are some new questions: > 1. Should it be possible to edit a record in System>Domains? When I click Edit, the page doesn’t change. I have read-write permission > 2. How do you activate the TLS Management tab? I cannot see it > 3. Why has support for MI using FIFO been dropped? It seems you must use MI_JSON (which needs the HTTPD module) > > John Quick > Smartvox Limited > > >> -----Original Message----- >> From: Bogdan-Andrei Iancu >> Sent: 05 November 2018 17:10 >> To: john.quick at smartvox.co.uk; users at lists.opensips.org >> Subject: Re: [OpenSIPS-Users] OpenSIPS Control Panel 8.2.4 was released >> >> Hi John, >> >> Thanks for all your testing - this is something that we so most appreciate ;). >> >> To help us even more, could you please split the report (you have in the >> email) as individual issues and create corresponding tickets on github: >> https://github.com/OpenSIPS/opensips-cp/issues >> >> Best regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer > From bogdan at opensips.org Tue Nov 6 16:16:46 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 6 Nov 2018 23:16:46 +0200 Subject: [OpenSIPS-Users] OpenSIPS Control Panel 8.2.4 was released In-Reply-To: References: <001801d47518$177f4a70$467ddf50$@smartvox.co.uk> Message-ID: <892137fb-9156-5b30-e540-df782bcd693c@opensips.org> Hi Arto, Indeed, there is an issue: a newly created access user can got receive permissions to manage the access users. Basically only the initial "admin" has the ability to create new access users. We are working on a solution to this issue, thanks for the report. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/06/2018 01:29 PM, Arto Kuiri wrote: > Hi, > > I think there is still something at control panel's user management. You cannot create another admin user from UI. Even > if you select read-write permission to all modules for new user, it cannot access to Users (at top panel). If you change manually > permissions field at database from "read-write, read-wri...." to "all" it will work. > > I think there should be own permission settings for "Control Panel Users" ? > > Best regards, > Arto Kuiri > > > Lähettäjä: Users käyttäjän Bogdan-Andrei Iancu puolesta > Lähetetty: maanantai 5. marraskuuta 2018 19.10 > Vastaanottaja: john.quick at smartvox.co.uk; users at lists.opensips.org > Aihe: Re: [OpenSIPS-Users] OpenSIPS Control Panel 8.2.4 was released >   > Hi John, > > Thanks for all your testing - this is something that we so most > appreciate ;). > > To help us even more, could you please split the report (you have in the > email) as individual issues and create corresponding tickets on github: >          https://github.com/OpenSIPS/opensips-cp/issues > > Best regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer >    http://www.opensips-solutions.com > OpenSIPS Bootcamp 2018 >    http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 11/05/2018 04:59 PM, John Quick wrote: > > Hello Bogdan, > > > > Thanks for releasing a new version of the Control Panel. There are some > > great new features in this version that I want to try. > > > > I have been testing OCP v8 on a CentOS 6 server. I'm finding quite a lot of > > errors - it is possible some of these are associated with the operating > > system or the version of PHP I am using. PHP is v 5.3.3 > > > > The most critical problem for me concerns administrator access: I am unable > > to change the password for the admin user. > > As a precaution, I created a new admin user with a different name and then > > tried to revoke the key privileges of the original admin user, but this made > > no difference until I edited the source code and changed the tool name > > passed to the get_priv() function as detailed below. > > > > Here are my full notes on the problems encountered so far and code > > adjustments I have made: > > In login.php: Warns me "It is not safe to rely on the systems timezone > > settings. You are required to use date.timezone or > > date_default_timezone_set() function".   As a work-around, I added calls to > > date_default_timezone_set("Europe/London") in a couple of places to stop > > these warnings being written. > > > > Many of the tabs display a blank page. In the httpd log file; the httpd log > > files shows errors about "unexpected [".  When I look up the file and line > > number shown in the httpd log message, it's always a line that calls the > > die() function to handle failure of an SQL request. > > I found it was due to the use of "errorInfo()[2]".  If you remove the [2], > > it doesn't give an error and the page is rendered too. > > > > Used grep to find every case like this: > >        # grep -rl "errorInfo()\[" . > >        ./users/user_management/template/user_management.main.php > >        ./users/acl_management/template/acl_management.main.php > >        ./system/dialplan/template/dialplan.main.php > >        ./system/dialplan/dialplan.php > >        ./system/siptrace/template/tracer.main.php > >        ./system/drouting/template/groups.main.php > >        ./system/drouting/template/rules.main.php > >        ./system/drouting/template/carriers.main.php > >        ./system/drouting/template/gateways.main.php > >        ./system/drouting/lib/carriers.test.inc.php > >        ./system/drouting/lib/carriers.functions.inc.php > >        ./system/tls_mgm/lib/data_loader.php > >        ./system/loadbalancer/template/loadbalancer.main.php > >        ./system/dispatcher/template/dispatcher.main.php > >        ./system/dispatcher/template/dispatcher.form.php > >        ./system/tviewer/lib/data_loader.php > >        ./system/callcenter/lib/data_loader.php > >        ./admin/list_admins/template/list_admins.main.php > > > > Admin Access Control > > I cannot find a way to change the admin password. It always accepts the > > password of "opensips" even after resetting it in the Admin User editing > > form and also deleting it from the field "password" in the > > ocp_admin_privileges table. On further testing, I then found that none of > > the changes I make to the admin user get saved to the DB - not just the > > password, but also the first name and last name fields. > > > > Access privileges were not working to restrict whether the logged-in user > > could change their own (and other people's) admin privileges. > > Found this was because the get_priv() function was being called with the > > argument set to "list_admins" and that the arrays used to store access > > privileges only have a key called "user_management" and do not have one for > > "list_admins".  There are no lines of code to act as a "catch-all" in the > > get_priv() function so if the tool name passed to the function does not > > match any of the known tab names, then it fails to set the session variable > > read_only. The absence of any value is then treated as "grant full access" > > in the corresponding module so it is not very safe. > > As a work-around, I edited web/tools/admin/list_admins/index.php and changed > > the following line: > > << get_priv("list_admins"); > >>> get_priv("user_management"); > > John Quick > > Smartvox Limited > > > > > > > _______________________________________________ > 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 bogdan at opensips.org Tue Nov 6 16:21:40 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 6 Nov 2018 23:21:40 +0200 Subject: [OpenSIPS-Users] ACK increased Cseq on reinvite In-Reply-To: References: Message-ID: <7d7ad894-9cb6-6a2a-4809-3626625d656c@opensips.org> Hi Schneur, Could you please post a link to a pcap showing this exact problem - the trace should be from the opensips machine, covering the traffic from caller and callee side too for the whole duration of the call. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/06/2018 02:41 PM, Schneur Rosenberg wrote: > Hi, I have this scenario, I'm sending to a carrier that requires > authentication and I use uac_auth(), when I send a call to the carrier > everything works fine, but when OpenSIPs sends a reinvite the carrier > responds witha 200 OK, OpenSIPs replies with a ACK but it falsely > increments the CSeq , causing the carrier to ignore it, and then the > carrier will keep on resending 200 OK's and OpenSIPs still returns the > wrong ACK and eventually the carrier hangs up. > > I'm running OpenSIPs 2.4.1 > > thanks > Scot > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From bogdan at opensips.org Tue Nov 6 16:36:48 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 6 Nov 2018 23:36:48 +0200 Subject: [OpenSIPS-Users] Raise event from MI/FIFO interface(externally) In-Reply-To: References: <9b3e0dbc-0190-4d15-96b5-c04edcaca90f@opensips.org> Message-ID: <5bab39fb-8c23-1982-652a-ae9af0bf0d0f@opensips.org> Thanks Sammy for the follow up. For the sake of the completion of this discussion, just update here with the feature request link, so people can follow it later. Thanks and regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/05/2018 05:29 PM, SamyGo wrote: > Hi Bogdan, > Sure thing I'd open up a feature request. Yes thats what we decided to > go with to use sipsak or something like that to trigger SIP packet and > make use of it. > Apologize for late reply; thanks a lot for your response. > > Regards, > Sammy > > On Tue, Oct 16, 2018 at 4:29 AM Bogdan-Andrei Iancu > > wrote: > > Hi Sammy, > > The Event Interface philosophy is to allow external apps to get > access to the events generated by OpenSIPS; basically a data flow > from OpenSIPS to outside world. So, there is no way to trigger an > event from outside (via MI). > Still, your example is valid, so please open a feature request on > github and we could add a "raise_event" MI function. > > As really (really) dirty hack, you can use (if possible) some SIP > OPTIONS to trigger the event. From the web send an OPTIONS via > UDP, with a special RURI -> you identify the RURI in script and do > a raise_event() in script. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Bootcamp 2018 > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 10/11/2018 07:35 PM, SamyGo wrote: >> Hi, >> >> I'm trying to find out document pages regarding raising events >> from outside the OpenSIPS via fifo/mi_* modules. All I have read >> so far is opensips can send events and their data OUT to external >> "subscribing" applications. There is even a fifo command to >> subscribe for an event from FIFO layer. >> >> Kindly guide me as how do I tell OpenSIPS that a particular Event >> has triggered. >> >> My usage scenario is (OpenSIPS 2.2.7) a caller is >> waiting_on_event("OPENSIPS_BOOTCAMP"); >> Now some external web monitoring bot just realized Bootcamp has >> started and it wants to raise this event to the waiting caller ! >> >> opensipsctl fifo raise_event OPENSIPS_BOOTCAMP 22Oct18 Romania >> >> Best 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 rob.dyck at telus.net Tue Nov 6 21:30:16 2018 From: rob.dyck at telus.net (Robert Dyck) Date: Tue, 06 Nov 2018 18:30:16 -0800 Subject: [OpenSIPS-Users] Flush bad user data from from running opensips Message-ID: <11961787.XINqL1RPXL@blacky.mylan> I have updated my small test bed from 2.3.2 to 2.4.2. I didn't bother to back up the 'subscriber" table and it was wiped by the installation. No big deal, it was tiny. So I added the users but made an error. opensipsctl add abc xyz -- I didn't specify the domain. The UAC would not register. I corrected the user. opensipsctl rm abc, opensipsctl add abc at 192.168.1.2 xyz The UAC still cannot register. DBG:auth_db:get_ha1: no result for user 'abc@' Opensips is restarted and the UAC registers. Restaring a production machine is problematic. Is there a way to flush the bad data which I assume has been cached? Some error checking in opensipsctl or the DB interface would be helpful. Thanks for your time and the product. Rob -------------- next part -------------- An HTML attachment was scrubbed... URL: From john.quick at smartvox.co.uk Wed Nov 7 04:42:35 2018 From: john.quick at smartvox.co.uk (John Quick) Date: Wed, 7 Nov 2018 09:42:35 -0000 Subject: [OpenSIPS-Users] OpenSIPS Control Panel 8.2.4 was released In-Reply-To: References: <001801d47518$177f4a70$467ddf50$@smartvox.co.uk> <002601d475f4$25d0fbf0$7172f3d0$@smartvox.co.uk> Message-ID: <000b01d4767e$36dee080$a49ca180$@smartvox.co.uk> Hi Bogdan, On item 1: Sorry, my mistake. I was looking for the page to change and did not notice that the domain name is loaded in the existing text box at the top. John Quick Smartvox Limited -----Original Message----- From: Bogdan-Andrei Iancu Sent: 06 November 2018 21:12 To: john.quick at smartvox.co.uk; users at lists.opensips.org Subject: Re: [OpenSIPS-Users] OpenSIPS Control Panel 8.2.4 was released HI John, Thanks for the reports :) 1) I tried to reproduce, but no luck - When Edit button is clicked, the value is loaded in the upper text area, so you change it. Do you use the mighty "admin" user for this operation ? or an access user with custom permissions ? 2) Indeed, there was missing entry for the tls_mgm module in the modules.inc.php file, I fixed it in the git repo. 3) starting with 7.2.3 (las year), the mi_json became the only supported MI backend in OCP - the reason is that mi_json is the only one to offer a structured / encoded data output that can be easily and reliably parsed by OCP. Best Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com From rosenberg11219 at gmail.com Wed Nov 7 04:59:43 2018 From: rosenberg11219 at gmail.com (Schneur Rosenberg) Date: Wed, 7 Nov 2018 11:59:43 +0200 Subject: [OpenSIPS-Users] ACK increased Cseq on reinvite In-Reply-To: <7d7ad894-9cb6-6a2a-4809-3626625d656c@opensips.org> References: <7d7ad894-9cb6-6a2a-4809-3626625d656c@opensips.org> Message-ID: For security reasons I will email it directly to you. thanks On Tue, Nov 6, 2018 at 11:21 PM Bogdan-Andrei Iancu wrote: > > Hi Schneur, > > Could you please post a link to a pcap showing this exact problem - the > trace should be from the opensips machine, covering the traffic from > caller and callee side too for the whole duration of the call. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Bootcamp 2018 > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 11/06/2018 02:41 PM, Schneur Rosenberg wrote: > > Hi, I have this scenario, I'm sending to a carrier that requires > > authentication and I use uac_auth(), when I send a call to the carrier > > everything works fine, but when OpenSIPs sends a reinvite the carrier > > responds witha 200 OK, OpenSIPs replies with a ACK but it falsely > > increments the CSeq , causing the carrier to ignore it, and then the > > carrier will keep on resending 200 OK's and OpenSIPs still returns the > > wrong ACK and eventually the carrier hangs up. > > > > I'm running OpenSIPs 2.4.1 > > > > thanks > > Scot > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > From john.quick at smartvox.co.uk Wed Nov 7 10:08:43 2018 From: john.quick at smartvox.co.uk (John Quick) Date: Wed, 7 Nov 2018 15:08:43 -0000 Subject: [OpenSIPS-Users] OpenSIPS Control Panel 8.2.4 was released In-Reply-To: References: <001801d47518$177f4a70$467ddf50$@smartvox.co.uk> <002601d475f4$25d0fbf0$7172f3d0$@smartvox.co.uk> Message-ID: <001c01d476ab$c6560b80$53022280$@smartvox.co.uk> In the CDR Viewer tab, the Call-ID is shown as a link, but the link is not working. If I click this link, it displays a blank page. I think this used to take you to the SIP Trace page and display the data for the selected call. Data for this call is available if I go directly to the SIP Trace tab and click on the Call Info button for the same call. I've checked the settings in /config/tools/system/cdrviewer/local.inc.php and they look correct However, I have not set up the suggested cron job yet. I assume it not relevant here. John Quick Smartvox Limited From bpasquet at openip.fr Wed Nov 7 10:27:18 2018 From: bpasquet at openip.fr (Benjamin Pasquet) Date: Wed, 7 Nov 2018 15:27:18 +0000 Subject: [OpenSIPS-Users] Fraud Detection Module Message-ID: Hello, I have some questions about fraud detection module and more particulary about the sequential call statistics. I am actualy in opensips version 2.2.2 but I tried in 2.2.4 and 2.2.7, and I don't understand well behavior what I see on each version. 1) First, in relation to the behavior of the sequential call statistics, I will give you an exemple to try to explain what I expected and what I found. I have this following rules : ruleid profileid prefix ... 1 10000 0033 2 10000 0044 3 20000 0033 4 20000 0044 User 10000 call the 0033123456789, I do check_fraud(10000, 0033123456789, 10000), who match with the rule 1, the sequential call counter of the rule 1 goes from 0 to 1. User 10000 call the 0033123456789, I do check_fraud(10000, 0033123456789, 10000), who match with the rule 1, the sequential call counter of the rule 1 goes from 1 to 2. User 10000 call the 0044123456789, I do check_fraud(10000, 0044123456789, 10000), who match with the rule 2, the sequential call counter of the rule 2 goes from 0 to 1. User 10000 call the 0033123456789, I do check_fraud(10000, 0033123456789, 10000), who match with the rule 1, the sequential call counter of the rule 1 goes from 2 to 3 --> I was expecting that the counter to go back to 1 cause the last number called by this user is different. User 10000 call the 0033987654321, I do check_fraud(10000, 0033987654321, 10000), who match with the rule 1, the sequential call counter of the rule 1 goes from 3 to 4 --> I was expecting that the counter to go back to 1 for the same reasons than the previously case, and further, for this rule and prefix, le number called is different, that's why I was expecting even more that the counter to go back to 1 User 20000 call the 0033123456789, I do check_fraud(20000, 0033123456789, 20000), who match with the rule 1, the sequential call counter of the rule 3 goes from 0 to 1. User 20000 call the 0033123456789, I do check_fraud(20000, 0033123456789, 20000), who match with the rule 1, the sequential call counter of the rule 3 goes from 1 to 2. User 10000 call the 0033123456789, I do check_fraud(10000, 0033123456789, 10000), who match with the rule 1, the sequential call counter of the rule 1 goes from 4 to 5 --> For this user, this prefix, le called number is different than the previous one called, I was expected that the counter to go back to 1 even if another user have called this number just previously. For summarize, I was expected that the counter is reset per user for all its rules, from the time the number called by the user is different from the previous one. 2) Secondly, the FRAUD statistics are daily reset, but which parameter are concerned? Total calls Calls per minute Concurrent calls Number of sequential calls Call duration 3) Thirdly and the last point, is it possible to set a value for a parameter rule who permit to don't check this one? Like set the warning and critical parameter values of the sequential call to -1 for a rule for exemple (I have find this supposition into the mailing list). Thank you in advance for your answer, Best regards, Benjamin -------------- next part -------------- An HTML attachment was scrubbed... URL: From rob.dyck at telus.net Wed Nov 7 15:09:39 2018 From: rob.dyck at telus.net (Robert Dyck) Date: Wed, 07 Nov 2018 12:09:39 -0800 Subject: [OpenSIPS-Users] GRUU contact not found Message-ID: <1891753.yvUAHoSbzm@blacky.mylan> My understanding is that GRUU processing in opensips is automatic, provided it is not disabled. No further configuration or scripting is required. Is that correct. A GRUU capable UA rergisters and receives public and temporary GR identities. The UA establishes a dialog with another UA. The callee ends the call. The caller does not recive the BYE. Caller : Request-Line: INVITE sip:7 at 192.168.1.2 SIP/2.0 Contact URI: sip:4 at 192.168.1.2:5060;gr=urn:uuid:35dfa98a-2feb-482a-bde7-7568a86348b1 Callee: Status-Line: SIP/2.0 200 OK Caller: Request-Line: ACK sip:7 at 192.168.1.3:5062 SIP/2.0 Callee: Request-Line: BYE sip:4 at 192.168.1.2:5060;gr=urn:uuid:35dfa98a-2feb-482a- bde7-7568a86348b1 SIP/2.0 Proxy ( opensips @ 192.168.1.2 ) Status-Line: SIP/2.0 404 Not here Am I missing something? Should "opensipsctl ul show" show the GRUU? AOR:: 4 Contact:: sip:4 at 192.168.1.72:5062;transport=udp Q= -------------- next part -------------- An HTML attachment was scrubbed... URL: From rob.dyck at telus.net Wed Nov 7 15:22:51 2018 From: rob.dyck at telus.net (Robert Dyck) Date: Wed, 07 Nov 2018 12:22:51 -0800 Subject: [OpenSIPS-Users] test - please ignore Message-ID: <1887179.YrnlRj2BJX@blacky.mylan> From bogdan at opensips.org Wed Nov 7 15:56:39 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 7 Nov 2018 22:56:39 +0200 Subject: [OpenSIPS-Users] OpenSIPS Control Panel 8.2.4 was released In-Reply-To: <001c01d476ab$c6560b80$53022280$@smartvox.co.uk> References: <001801d47518$177f4a70$467ddf50$@smartvox.co.uk> <002601d475f4$25d0fbf0$7172f3d0$@smartvox.co.uk> <001c01d476ab$c6560b80$53022280$@smartvox.co.uk> Message-ID: <702a01d1-21a7-1950-a56f-2ca06cf8821e@opensips.org> Hi John, So you have the siptrace tool enabled, right ? but the link is broken ? Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/07/2018 05:08 PM, John Quick wrote: > In the CDR Viewer tab, the Call-ID is shown as a link, but the link is not working. > If I click this link, it displays a blank page. I think this used to take you to the SIP Trace page and display the data for the selected call. > Data for this call is available if I go directly to the SIP Trace tab and click on the Call Info button for the same call. > > I've checked the settings in /config/tools/system/cdrviewer/local.inc.php and they look correct > However, I have not set up the suggested cron job yet. I assume it not relevant here. > > John Quick > Smartvox Limited > From pasandev at ymail.com Wed Nov 7 21:39:50 2018 From: pasandev at ymail.com (Pasan Meemaduma) Date: Thu, 8 Nov 2018 02:39:50 +0000 (UTC) Subject: [OpenSIPS-Users] check for NULL values References: <1238611305.714858.1541644790120.ref@mail.yahoo.com> Message-ID: <1238611305.714858.1541644790120@mail.yahoo.com> Hi Guys, I have a check for NULL for $tu var in the script, But when the value is missing I'm getting the following error. ERROR:core:comp_scriptvar: cannot get left var value WARNING:core:do_action: error in expression at /etc/opensips/opensips.cfg:806 and line 806 contains following.     if ( $tu != NULL ) {         remove("location","$tu");     } any suggestion on how to test for NULL values without getting above error. I'm using opensips 2.3.5 -------------- next part -------------- An HTML attachment was scrubbed... URL: From john.quick at smartvox.co.uk Thu Nov 8 04:46:58 2018 From: john.quick at smartvox.co.uk (John Quick) Date: Thu, 8 Nov 2018 09:46:58 -0000 Subject: [OpenSIPS-Users] OpenSIPS Control Panel 8.2.4 was released In-Reply-To: <702a01d1-21a7-1950-a56f-2ca06cf8821e@opensips.org> References: <001801d47518$177f4a70$467ddf50$@smartvox.co.uk> <002601d475f4$25d0fbf0$7172f3d0$@smartvox.co.uk> <001c01d476ab$c6560b80$53022280$@smartvox.co.uk> <702a01d1-21a7-1950-a56f-2ca06cf8821e@opensips.org> Message-ID: <000201d47747$fe350d20$fa9f2760$@smartvox.co.uk> Yes, the SIP Trace tab is enabled and can be used directly to view the requests/responses for a call. The CDR Viewer tab is also enabled and it shows me the same call. However, if I click on the Sip Call ID for a given call, it acts as a link to a new blank page. It does not open the SIP Trace view as I expected it to do. This is the link URL. Perhaps "tracer=homer" is significant here - I'm not using Homer, just writing all data to MySQL tables: http://abc.xyz.com:8080/cp/tools/system/cdrviewer/trace.php?tracer=homer&callid=39e32b7a-5d45-1237-beb7-52540058cb88 John Quick Smartvox Limited Tel: 01727-221221 -----Original Message----- From: Bogdan-Andrei Iancu Sent: 07 November 2018 20:57 To: john.quick at smartvox.co.uk; users at lists.opensips.org Subject: Re: [OpenSIPS-Users] OpenSIPS Control Panel 8.2.4 was released Hi John, So you have the siptrace tool enabled, right ? but the link is broken ? Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/07/2018 05:08 PM, John Quick wrote: > In the CDR Viewer tab, the Call-ID is shown as a link, but the link is not working. > If I click this link, it displays a blank page. I think this used to take you to the SIP Trace page and display the data for the selected call. > Data for this call is available if I go directly to the SIP Trace tab and click on the Call Info button for the same call. > > I've checked the settings in > /config/tools/system/cdrviewer/local.inc.php and they look correct However, I have not set up the suggested cron job yet. I assume it not relevant here. > > John Quick > Smartvox Limited > From bogdan at opensips.org Thu Nov 8 10:47:36 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 8 Nov 2018 17:47:36 +0200 Subject: [OpenSIPS-Users] [RELEASE] OpenSIPS 2.4.3 minor releases Message-ID: <145befa0-234b-5318-694d-b73128e0b1b5@opensips.org> Hi all, As promised, the 2.4.3 minor OpenSIPS release is now available. Even if minor, this release is an important one on the 2.4 branch as it bring some crucial fixes in the presence engine - related to the DB operations - and in the dialog clustering - related to sharing tags and callback triggering. And, at least from the perspective of these two fixes, we strongly recommend to upgrade ! Of course, several other (smaller) fixes are included in this release, see the full OpenSIPS 2.4.3 ChangeLog here : http://opensips.org/pub/opensips/2.4.3/ChangeLog Many thanks to all the people who helped to get an even better 2.4 release, contributing with reports, with testing and fixes - this is a great community work! As 2.4.3 is a minor release (on 2.4 branch), there are no changes in the OpenSIPS behavior (coming from 2.4.x), but only fixes - so, you are safe in upgrading. Even more, again, we do encourage you to upgrade - it really improves the quality of your night sleep ;). Best regards, -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training From julian.santer at rolmail.net Thu Nov 8 11:52:13 2018 From: julian.santer at rolmail.net (Julian Santer) Date: Thu, 8 Nov 2018 17:52:13 +0100 Subject: [OpenSIPS-Users] Permission doesn't match Message-ID: Hi guys, I have some question to the permission module. We are using Opensips 2.2.6. The permissions are load from the address table located in a MySQL DB. My config looks like: ... else if (check_address("52", "$si", "$sp", "$proto", "$avp(ctx)", "$ua")) { xlog("L_INFO", "Entered here due permission 52 - LF_BASE"); } else if (check_address("54", "$si", "$sp", "$proto", "$avp(ctx)", "$ua")) { xlog("L_INFO", "Entered here due permission 54 - LF_BASE"); } ... address table: id grp ip mask port proto pattern context_info 41 52 192.168.1.0 24 0 any AVM*.06.* test 648 54 192.168.1.0 24 0 any AVM*.07.* test This line is matching: Nov 8 17:10:59 M=REGISTER RURI=sip:test.com F=sip:abc at test.com T=sip:abc at test.com SRC=192.168.1.46:5060 UAC=AVM FRITZ!Box Fon WLAN 7390 84.06.85 (Sep 10 2018) ID=9A0B1C90057A9126 at 192.168.1.46 B= Nov 8 17:10:59 Entered here due permission 52 - M=REGISTER RURI=sip:test.com F=sip:abc at test.com T=sip:abc at test.com SRC=192.168.1.46:5060 UAC=AVM FRITZ!Box Fon WLAN 7390 84.06.85 (Sep 10 2018) ID=9A0B1C90057A9126 at 192.168.146 B= But this line is not matching: Nov 8 17:35:19 M=REGISTER RURI=sip:test.com F=sip:def at test.com T=sip:def at test.com SRC=192.168.1.215:5060 UAC=AVM FRITZ!Box 7490 113.07.01 (Sep 11 2018) ID=5DC1E7DC326043BA at 192.168.1.215 B= I already did a opensipsctl address reload and several times restarted the whole opensips service. Have you maybe some hint for me? Kind regards, Julian Santer From rob.dyck at telus.net Thu Nov 8 15:05:16 2018 From: rob.dyck at telus.net (Robert Dyck) Date: Thu, 08 Nov 2018 12:05:16 -0800 Subject: [OpenSIPS-Users] GRUU contact not found Message-ID: <1643163.C2HuH20Fne@blacky.mylan> After some thought I realized that a lookup had to be invoked while in dialog. The BYE was directed at the proxy and the GRUU needed to be mapped to the device that was the intended target. Added the following to script for "in dialog" xlog("Check for GRUU, Method is $rm\n"); Nov 8 11:49:30 [24807] DBG:rr:after_loose: Topmost route URI: 'sip: 192.168.1.2;lr;ftag=SAjVc2sqm' is me Nov 8 11:49:30 [24807] DBG:core:parse_headers: flags=200 Nov 8 11:49:30 [24807] DBG:core:get_hdr_field: cseq : <252> Nov 8 11:49:30 [24807] DBG:core:get_hdr_field: content_length=0 Nov 8 11:49:30 [24807] DBG:core:get_hdr_field: found end of header Nov 8 11:49:30 [24807] DBG:rr:find_next_route: No next Route HF found Nov 8 11:49:30 [24807] DBG:rr:after_loose: No next URI found! Nov 8 11:49:30 [24807] DBG:core:parse_headers: flags=78 Nov 8 11:49:30 [24807] DBG:core:parse_to_param: tag=uqzwj Nov 8 11:49:30 [24807] DBG:core:_parse_to: end of header reached, state=29 Nov 8 11:49:30 [24807] DBG:core:_parse_to: display={}, ruri={sip:7 at 192.168.1.2} Nov 8 11:49:30 [24807] DBG:rr:check_route_param: params are <;lr;ftag=SAjVc2sqm> Nov 8 11:49:30 [24807] DBG:rr:check_route_param: params are <;lr;ftag=SAjVc2sqm> Nov 8 11:49:30 [24807] Check for GRUU, Method is BYE Nov 8 11:49:30 [24807] Found GRUU Nov 8 11:49:30 [24807] DBG:registrar:parse_lookup_flags: final flags: 1 Nov 8 11:49:30 [24807] DBG:registrar:extract_aor: has gruu Nov 8 11:49:30 [24807] DBG:registrar:extract_aor: public gruu Nov 8 11:49:30 [24807] DBG:registrar:select_contacts: ct: sip: 4 at 192.168.1.72:5062;transport=udp Nov 8 11:49:30 [24807] DBG:registrar:select_contacts: ruri has gruu Nov 8 11:49:30 [24807] DBG:registrar:select_contacts: matched sip instance Nov 8 11:49:30 [24807] DBG:registrar:push_branch: setting as ruri -------------- next part -------------- An HTML attachment was scrubbed... URL: From vitalik.voip at gmail.com Fri Nov 9 06:58:05 2018 From: vitalik.voip at gmail.com (Vitalii Aleksandrov) Date: Fri, 9 Nov 2018 13:58:05 +0200 Subject: [OpenSIPS-Users] Remove usrloc record after terminating a tcp connection. Message-ID: <5955ecc4-53ad-4e3b-2a26-5057ee65b2b2@gmail.com> Hi, Wonder if it's possible to delete a user location contact when opensips detects a connenction failure? Couldn't find either "unregister" function to remove a contact from a failure_route or an event_route called when opensips core detects connection failure. Is there any good way to achieve it without usrloc module hacking and trying to bind "ucontacts" to tcp connections? -------------- next part -------------- An HTML attachment was scrubbed... URL: From alexei.vasilyev at gmail.com Fri Nov 9 07:49:16 2018 From: alexei.vasilyev at gmail.com (vasilevalex) Date: Fri, 9 Nov 2018 05:49:16 -0700 (MST) Subject: [OpenSIPS-Users] Remove usrloc record after terminating a tcp connection. In-Reply-To: <5955ecc4-53ad-4e3b-2a26-5057ee65b2b2@gmail.com> References: <5955ecc4-53ad-4e3b-2a26-5057ee65b2b2@gmail.com> Message-ID: <1541767756519-0.post@n2.nabble.com> Hi Vitalii, You need just enable contact pinging and delete dead by nathelper module. Here is answer to the same my question: http://opensips-open-sip-server.1449251.n2.nabble.com/how-to-delete-contacts-on-dead-TCP-or-TLS-sessions-td7611306.html -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html From john.quick at smartvox.co.uk Fri Nov 9 08:42:15 2018 From: john.quick at smartvox.co.uk (John Quick) Date: Fri, 9 Nov 2018 13:42:15 -0000 Subject: [OpenSIPS-Users] WebRTC-SIP only working in one direction Message-ID: <000601d47832$06fe1600$14fa4200$@smartvox.co.uk> I have been trying to get WebRTC working with OpenSIPS v2.4 and rtpengine version "git-master-3102357" Testing has been tried using both Chrome and Firefox browsers running on Windows 10 PCs combined with two different web-host-servers: (1) A home-grown sip.js page and (2) the ctxSIP trial client available at collecttix.github.io/ctxSip Test calls from the WebRTC client to an Asterisk server, via OpenSIPS and rtpengine, are working okay. However, I cannot get calls to work the other way. The WebRTC client registers okay. It appears in the location table with typical values like this: Username=1002 Contact= sip:mch27rvr at bcdtsq8dhvt4.invalid;transport=ws Received= sip:123.45.67.89:15752;transport=wss Cflags= WS_DEVICE NATTED_CLIENT User-agent= SIP.js/0.11.3 (my sip.js) OR SIP.js/0.7.8 (ctx client) Here is what happens when I make a call from Asterisk to user 1002: Using Chrome or Firefox with my own sip.js web page: It initiates ringing on the browser-based client (the caller's name is shown correctly), but when I click the Answer button, the call fails. The SIP time sequence between OpenSIPS and the WebRTC client looks like this: -----> INVITE with SDP <---- 100 Trying <---- 180 Ringing Browser page shows incoming call and displays correct caller display name I click the Answer button <--- 480 Temporarily Unavailable On the web page it reports: Call cancelled: Cause: WebRTC Error Using Chrome with the ctxSip client: The browser-based client rejects the call as soon as the INVITE is received. -----> INVITE with SDP <---- 100 Trying <---- 488 Not Acceptable Here Using Firefox with the ctxSip client: The browser-based client shows an incoming call and the caller's display name is shown correctly. It offers a green and a red button to respond to the call. I press the green button and the call ends immediately. -----> INVITE with SDP <---- 100 Trying <---- 180 Ringing Browser page shows incoming call and displays correct caller display name I click the Answer button <--- 480 Temporarily Unavailable The browser window simply reports "Terminated". Any help or ideas would be very welcome. I've tried everything I can think of including changes to the codecs offered and various adjustments to the arguments passed (e.g. rtcp-mux, ICE) in rtpengine_offer(). My current arguments string is like this: "trust-address replace-origin replace-session-connection rtcp-mux-offer ICE=force codec-strip-G729 transcode-PCMU RTP/SAVPF" My JavaScript skills are minimal. I think there must be a way to get more debug info from sip.js, but I cannot figure it out. If anyone can post a sample of the code showing how to activate the debug logging that would be really useful. I'm not even sure where the log output would be written. Could it be relevant that rtpengine and OpenSIPS, although both running on one server, are bound to different IP addresses? John Quick Smartvox Limited From vitalik.voip at gmail.com Fri Nov 9 08:54:31 2018 From: vitalik.voip at gmail.com (Vitalii Aleksandrov) Date: Fri, 9 Nov 2018 15:54:31 +0200 Subject: [OpenSIPS-Users] Remove usrloc record after terminating a tcp connection. In-Reply-To: <1541767756519-0.post@n2.nabble.com> References: <5955ecc4-53ad-4e3b-2a26-5057ee65b2b2@gmail.com> <1541767756519-0.post@n2.nabble.com> Message-ID: <41c5d529-c028-ef68-3cb3-487cdee52330@gmail.com> Thanks. Had to search better before asking... This is close to what I want. I would be great to remove a contact immediately after opensips detects a closed socket. Anyhow definitely better than wait for a contact expiration. > Hi Vitalii, > You need just enable contact pinging and delete dead by nathelper module. > Here is answer to the same my question: > http://opensips-open-sip-server.1449251.n2.nabble.com/how-to-delete-contacts-on-dead-TCP-or-TLS-sessions-td7611306.html > > > > -- > Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From bogdan at opensips.org Fri Nov 9 08:54:51 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 9 Nov 2018 15:54:51 +0200 Subject: [OpenSIPS-Users] OpenSIPS Control Panel 8.2.4 was released In-Reply-To: <000201d47747$fe350d20$fa9f2760$@smartvox.co.uk> References: <001801d47518$177f4a70$467ddf50$@smartvox.co.uk> <002601d475f4$25d0fbf0$7172f3d0$@smartvox.co.uk> <001c01d476ab$c6560b80$53022280$@smartvox.co.uk> <702a01d1-21a7-1950-a56f-2ca06cf8821e@opensips.org> <000201d47747$fe350d20$fa9f2760$@smartvox.co.uk> Message-ID: <2e351279-7ede-6ae9-2d2d-db8f3b98b0d8@opensips.org> Hi John, I found a small typo that affected who the link was constructed . See https://github.com/OpenSIPS/opensips-cp/commit/d50503123477f99b00795703614077b685ca4579 In order to link siptrace to cdrviewer, you need to (a) be sure homer tool is disabled and (b) siptrace tool is enabled. Let me know if this fix does the trick for you. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/08/2018 11:46 AM, John Quick wrote: > Yes, the SIP Trace tab is enabled and can be used directly to view the requests/responses for a call. > The CDR Viewer tab is also enabled and it shows me the same call. However, if I click on the Sip Call ID for a given call, it acts as a link to a new blank page. > It does not open the SIP Trace view as I expected it to do. > > This is the link URL. Perhaps "tracer=homer" is significant here - I'm not using Homer, just writing all data to MySQL tables: > http://abc.xyz.com:8080/cp/tools/system/cdrviewer/trace.php?tracer=homer&callid=39e32b7a-5d45-1237-beb7-52540058cb88 > > John Quick > Smartvox Limited > Tel: 01727-221221 > > > -----Original Message----- > From: Bogdan-Andrei Iancu > Sent: 07 November 2018 20:57 > To: john.quick at smartvox.co.uk; users at lists.opensips.org > Subject: Re: [OpenSIPS-Users] OpenSIPS Control Panel 8.2.4 was released > > Hi John, > > So you have the siptrace tool enabled, right ? but the link is broken ? > > Best regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Bootcamp 2018 > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 11/07/2018 05:08 PM, John Quick wrote: >> In the CDR Viewer tab, the Call-ID is shown as a link, but the link is not working. >> If I click this link, it displays a blank page. I think this used to take you to the SIP Trace page and display the data for the selected call. >> Data for this call is available if I go directly to the SIP Trace tab and click on the Call Info button for the same call. >> >> I've checked the settings in >> /config/tools/system/cdrviewer/local.inc.php and they look correct However, I have not set up the suggested cron job yet. I assume it not relevant here. >> >> John Quick >> Smartvox Limited >> From bogdan at opensips.org Fri Nov 9 08:59:30 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 9 Nov 2018 15:59:30 +0200 Subject: [OpenSIPS-Users] WARNING:core:utimer_ticker: utimer task In-Reply-To: <005301d4774b$c9e0c330$5da24990$@web.de> References: <981bb92a-d8e3-26be-b9c8-d9672d297151@opensips.org> <005301d4774b$c9e0c330$5da24990$@web.de> Message-ID: <35c3aafe-2356-6b95-0c15-ac1119396164@opensips.org> Hi Xaled, (switching back to the list) Again, it seems to be related to your postgres server - the backtrace shows that all opensips procs are stuck while trying to connect to the postgres server. Why the connect phase takes so long? not sure, it is a postgres stuff. Try setting this param to 1 or 2: http://www.opensips.org/html/docs/modules/2.4.x/db_postgres.html#param_timeout And see how it works. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/08/2018 12:14 PM, xaled wrote: > > Hi Bogdan, > > here is the compiled version from git sources: > > root at fra-appsrv01:/usr/local/etc/opensips# /usr/local/sbin/opensips -V > > version: opensips 2.4.2 (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, sigio_rt, select. > > git revision: a9b9169 > > main.c compiled on 19:30:38 Nov 7 2018 with gcc 4.9.2 > > and here is the debian repo version: > > root at fra-appsrv01:/usr/local/etc/opensips# /usr/sbin/opensips -V > > version: opensips 2.4.2 (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, sigio_rt, select. > > main.c compiled on with gcc 4.9.2 > > Greetings, > Xaled > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* Thursday, November 08, 2018 10:21 AM > *To:* xaled > *Subject:* Re: [OpenSIPS-Users] WARNING:core:utimer_ticker: utimer > task > > Hi Xaled, > > What's the exact version / revision of OpenSIPS you are using ? > > Best regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Bootcamp 2018 > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 11/08/2018 01:44 AM, xaled wrote: > > Hi Bogdan, > > Is there something you can recommend to debug and fix it? > > The database is in the same VM as opensips and there are never > time out warnings during the actual database access. Warnings come > either at startup, or randomly if opensips gets through startup phase. > > I will take a closer look at the database connection performance. > It makes no sense to me though, that warning flood started after > migrating the opensips VM to a newer hardware without changing > anything else and then partially stopped after using compiled > version of opensips vs installed from Debian repo on the same VM. > > The older Debian VM with the same opensips/db setup on a > different physical server does not have the startup warning flood > at all. Only random warning during run time. > > Appreciate your time, > > Xaled > > On Nov 7, 2018 22:18, Bogdan-Andrei Iancu > wrote: > > Hi Xaled, > > As I suspected, all the worker processes in OpenSIPS are > blocked in the > startup sequence (so called child_init routine) trying to > connect to the > postgres database. This is holding up without rejecting or > accepting the > connection from OpenSIPS. > -------------- next part -------------- An HTML attachment was scrubbed... URL: From rosenberg11219 at gmail.com Fri Nov 9 09:12:01 2018 From: rosenberg11219 at gmail.com (Schneur Rosenberg) Date: Fri, 9 Nov 2018 16:12:01 +0200 Subject: [OpenSIPS-Users] ACK increased Cseq on reinvite In-Reply-To: References: <7d7ad894-9cb6-6a2a-4809-3626625d656c@opensips.org> Message-ID: Bogdan, any update? On Wed, Nov 7, 2018 at 11:59 AM Schneur Rosenberg wrote: > > For security reasons I will email it directly to you. > > thanks > On Tue, Nov 6, 2018 at 11:21 PM Bogdan-Andrei Iancu wrote: > > > > Hi Schneur, > > > > Could you please post a link to a pcap showing this exact problem - the > > trace should be from the opensips machine, covering the traffic from > > caller and callee side too for the whole duration of the call. > > > > Regards, > > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2018 > > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > > > On 11/06/2018 02:41 PM, Schneur Rosenberg wrote: > > > Hi, I have this scenario, I'm sending to a carrier that requires > > > authentication and I use uac_auth(), when I send a call to the carrier > > > everything works fine, but when OpenSIPs sends a reinvite the carrier > > > responds witha 200 OK, OpenSIPs replies with a ACK but it falsely > > > increments the CSeq , causing the carrier to ignore it, and then the > > > carrier will keep on resending 200 OK's and OpenSIPs still returns the > > > wrong ACK and eventually the carrier hangs up. > > > > > > I'm running OpenSIPs 2.4.1 > > > > > > thanks > > > Scot > > > > > > _______________________________________________ > > > Users mailing list > > > Users at lists.opensips.org > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > From atuxnull at gmail.com Fri Nov 9 09:31:07 2018 From: atuxnull at gmail.com (John Tuxies) Date: Fri, 9 Nov 2018 16:31:07 +0200 Subject: [OpenSIPS-Users] asterisk integration with control panel Message-ID: Hi. i am new to the area of opensips and i would like some help. I am doing my baby steps, but i have some experience with Asterisk. In a Debian 9 64bit system i did install opensips as described in here: https://www.powerpbx.org/content/opensips-v24-debian-v8-mariadb-apache-v1 I would like to integrate Asterisk in the same box. In the common sql of the users i will have users eg 4000-4099 and then asterisk will have the inbound/outbound trunks, codecs, conference, voicemail.. Opensips will manage traffic and security. Up to now all i have is a system with Opensips and its panel. I do not know how to integrate Asterisk in here. What changes to do in the database. Is there a guide on how to integrate it? I have seen the one in https://www.opensips.org/Documentation/Tutorials-OpenSIPSAsteriskIntegration but it is outdated and it broke my system when i followed it to Asterisk 11 and newer versions. Once having that setup running then i assume all users will be created in opensips panel, or in Asterisk? Sincerely yours, John -------------- next part -------------- An HTML attachment was scrubbed... URL: From xaled at web.de Fri Nov 9 10:08:51 2018 From: xaled at web.de (xaled) Date: Fri, 9 Nov 2018 16:08:51 +0100 Subject: [OpenSIPS-Users] WARNING:core:utimer_ticker: utimer task In-Reply-To: <35c3aafe-2356-6b95-0c15-ac1119396164@opensips.org> References: <981bb92a-d8e3-26be-b9c8-d9672d297151@opensips.org> <005301d4774b$c9e0c330$5da24990$@web.de> <35c3aafe-2356-6b95-0c15-ac1119396164@opensips.org> Message-ID: <00f001d4783e$2044c3f0$60ce4bd0$@web.de> Hi Bogdan, I found the source of the problem. It is the use of localhost to connect to local postgresql database. I changed it to 127.0.0.1 and opensips starts fast and without warnings. before: modparam("db_virtual", "db_urls", "postgres://user:pass at localhost/db") after: modparam("db_virtual", "db_urls", "postgres://user:pass at 127.0.0.1/db") There is this warning I don’t understand: Nov 8 09:52:40 fra-appsrv01 /usr/local/sbin/opensips[12635]: WARNING:avpops:avpops_db_bind: async() calls for DB URL [default] will work in normal mode due to driver limitations And once or twice a day I get timer warnings, but it is way better than it was before. Nov 8 09:59:45 fra-appsrv01 /usr/local/sbin/opensips[12640]: WARNING:core:handle_timer_job: timer job has a 70000 us delay in execution Nov 8 09:59:45 fra-appsrv01 /usr/local/sbin/opensips[12641]: WARNING:core:handle_timer_job: timer job has a 70000 us delay in execution Nov 8 18:10:10 fra-appsrv01 /usr/local/sbin/opensips[12640]: WARNING:core:handle_timer_job: timer job has a 60000 us delay in execution Nov 8 18:10:10 fra-appsrv01 /usr/local/sbin/opensips[12640]: WARNING:core:handle_timer_job: timer job has a 60000 us delay in execution Nov 8 18:10:10 fra-appsrv01 /usr/local/sbin/opensips[12640]: WARNING:core:handle_timer_job: timer job has a 60000 us delay in execution Thanks, Xaled From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Friday, November 09, 2018 3:00 PM To: xaled ; users at lists.opensips.org Subject: Re: [OpenSIPS-Users] WARNING:core:utimer_ticker: utimer task Hi Xaled, (switching back to the list) Again, it seems to be related to your postgres server - the backtrace shows that all opensips procs are stuck while trying to connect to the postgres server. Why the connect phase takes so long? not sure, it is a postgres stuff. Try setting this param to 1 or 2: http://www.opensips.org/html/docs/modules/2.4.x/db_postgres.html#param_timeout And see how it works. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/08/2018 12:14 PM, xaled wrote: Hi Bogdan, here is the compiled version from git sources: root at fra-appsrv01:/usr/local/etc/opensips# /usr/local/sbin/opensips -V version: opensips 2.4.2 (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, sigio_rt, select. git revision: a9b9169 main.c compiled on 19:30:38 Nov 7 2018 with gcc 4.9.2 and here is the debian repo version: root at fra-appsrv01:/usr/local/etc/opensips# /usr/sbin/opensips -V version: opensips 2.4.2 (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, sigio_rt, select. main.c compiled on with gcc 4.9.2 Greetings, Xaled From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Thursday, November 08, 2018 10:21 AM To: xaled Subject: Re: [OpenSIPS-Users] WARNING:core:utimer_ticker: utimer task Hi Xaled, What's the exact version / revision of OpenSIPS you are using ? Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/08/2018 01:44 AM, xaled wrote: Hi Bogdan, Is there something you can recommend to debug and fix it? The database is in the same VM as opensips and there are never time out warnings during the actual database access. Warnings come either at startup, or randomly if opensips gets through startup phase. I will take a closer look at the database connection performance. It makes no sense to me though, that warning flood started after migrating the opensips VM to a newer hardware without changing anything else and then partially stopped after using compiled version of opensips vs installed from Debian repo on the same VM. The older Debian VM with the same opensips/db setup on a different physical server does not have the startup warning flood at all. Only random warning during run time. Appreciate your time, Xaled On Nov 7, 2018 22:18, Bogdan-Andrei Iancu wrote: Hi Xaled, As I suspected, all the worker processes in OpenSIPS are blocked in the startup sequence (so called child_init routine) trying to connect to the postgres database. This is holding up without rejecting or accepting the connection from OpenSIPS. -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Nov 12 09:31:53 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 12 Nov 2018 16:31:53 +0200 Subject: [OpenSIPS-Users] mi_xmlrpc_ng for ul_dump In-Reply-To: References: Message-ID: <15e683bd-573e-c16d-8563-e72697fb1bac@opensips.org> Hi Schneur, Do you see any errors in the OpenSIPS logs when you get these 500 MI replies ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/12/2018 04:29 PM, Schneur Rosenberg wrote: > I upgraded OpenSIPs to 2.4.3 and the problem still remains, right > after I restart OpenSIPs it works fine but after a approx 2 minutes I > get a 500 Internal server error. > > I also tried changing ports and no success, the http server is > running, as I can see the server listening to the port, and it is > returning a 500 and not a failed to open stream: message that I would > get when the server is not listsening. > On Sat, Nov 10, 2018 at 9:15 PM Schneur Rosenberg > wrote: >> Bogdan, my script works when I start up OpenSIPs, but after running it >> a few times it returns a "500 Internal server error" and only a >> restart on OpenSIPs clears it up, >> >> Im runining Ver 2.4.1 >> On Thu, Nov 1, 2018 at 8:14 PM Bogdan-Andrei Iancu wrote: >>> Hi Schneur, >>> >>> Yes, the ul_dump has an optional param, so you can use it without any >>> parameters. The error you get says there is no payload found in your >>> HTTP request. Normally the HTTP request should carry the XML with the >>> request description. >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> http://www.opensips-solutions.com >>> OpenSIPS Bootcamp 2018 >>> http://opensips.org/training/OpenSIPS_Bootcamp_2018/ >>> >>> On 10/31/2018 12:13 PM, Schneur Rosenberg wrote: >>>> Hi, I was relying on the DB to get information regarding registered >>>> users , but I'm having issues with the DB so I'm trying to get it >>>> directly from opensips using mi_xmlrpc_ng, I'm using PHP with the >>>> xmlrpc_encode_request function., but the ul_dump method does not seem >>>> to have any params, when I remove the params I get a "401 Empty >>>> request" reply, and if i place any other value in the params I get a >>>> "500 Internal server error" reply, whats the proper way to get a >>>> ul_dump using mi_xmlrpc_ng? >>>> >>>> S. Rosenberg >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users From bogdan at opensips.org Mon Nov 12 11:42:24 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 12 Nov 2018 18:42:24 +0200 Subject: [OpenSIPS-Users] ACK increased Cseq on reinvite In-Reply-To: References: <7d7ad894-9cb6-6a2a-4809-3626625d656c@opensips.org> Message-ID: <7b09cd7a-24b1-3ab0-d521-a4857f969f0b@opensips.org> Thanks Schneur, Indeed, I see that during the re-INVITe, the cseq of the ACK bumps to 105 instead of 104 (from 103 on caller leg). Please open a bug report on the github tracker, to have this issue in our attention. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/07/2018 12:01 PM, Schneur Rosenberg wrote: > Attached please find the trace, Im sending you the trace in private > for security and privacy reasons, thank you > > > On Wed, Nov 7, 2018 at 11:59 AM Schneur Rosenberg > wrote: >> For security reasons I will email it directly to you. >> >> thanks >> On Tue, Nov 6, 2018 at 11:21 PM Bogdan-Andrei Iancu wrote: >>> Hi Schneur, >>> >>> Could you please post a link to a pcap showing this exact problem - the >>> trace should be from the opensips machine, covering the traffic from >>> caller and callee side too for the whole duration of the call. >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> http://www.opensips-solutions.com >>> OpenSIPS Bootcamp 2018 >>> http://opensips.org/training/OpenSIPS_Bootcamp_2018/ >>> >>> On 11/06/2018 02:41 PM, Schneur Rosenberg wrote: >>>> Hi, I have this scenario, I'm sending to a carrier that requires >>>> authentication and I use uac_auth(), when I send a call to the carrier >>>> everything works fine, but when OpenSIPs sends a reinvite the carrier >>>> responds witha 200 OK, OpenSIPs replies with a ACK but it falsely >>>> increments the CSeq , causing the carrier to ignore it, and then the >>>> carrier will keep on resending 200 OK's and OpenSIPs still returns the >>>> wrong ACK and eventually the carrier hangs up. >>>> >>>> I'm running OpenSIPs 2.4.1 >>>> >>>> thanks >>>> Scot >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users From pasandev at ymail.com Tue Nov 13 03:56:42 2018 From: pasandev at ymail.com (Pasan Meemaduma) Date: Tue, 13 Nov 2018 08:56:42 +0000 (UTC) Subject: [OpenSIPS-Users] check for NULL values In-Reply-To: <1238611305.714858.1541644790120@mail.yahoo.com> References: <1238611305.714858.1541644790120.ref@mail.yahoo.com> <1238611305.714858.1541644790120@mail.yahoo.com> Message-ID: <1506752889.1453845.1542099402376@mail.yahoo.com> Hey, Anyone have a suggestion for this? On Thursday, 8 November 2018, 8:09:50 AM GMT+5:30, Pasan Meemaduma wrote: Hi Guys, I have a check for NULL for $tu var in the script, But when the value is missing I'm getting the following error. ERROR:core:comp_scriptvar: cannot get left var value WARNING:core:do_action: error in expression at /etc/opensips/opensips.cfg:806 and line 806 contains following.     if ( $tu != NULL ) {         remove("location","$tu");     } any suggestion on how to test for NULL values without getting above error. I'm using opensips 2.3.5 -------------- next part -------------- An HTML attachment was scrubbed... URL: From rob.dyck at telus.net Tue Nov 13 10:59:19 2018 From: rob.dyck at telus.net (Robert Dyck) Date: Tue, 13 Nov 2018 07:59:19 -0800 Subject: [OpenSIPS-Users] check for NULL values References: <1238611305.714858.1541644790120.ref@mail.yahoo.com> <1238611305.714858.1541644790120@mail.yahoo.com> Message-ID: <1986278.jP8lLNN5P7@blacky.mylan> Just a guess. Try if $tu { remove("location","$tu"); } Not tested. A nonzero value may evaluate as TRUE. On Tuesday, November 13, 2018 12:56:42 AM PST Pasan Meemaduma via Users wrote: Hey, Anyone have a suggestion for this? On Thursday, 8 November 2018, 8:09:50 AM GMT+5:30, Pasan Meemaduma wrote: ERROR:core:comp_scriptvar: cannot get left var value WARNING:core:do_action: error in expression at /etc/opensips/opensips.cfg:806 and line 806 contains following. if ( $tu != NULL ) { remove("location","$tu"); } any suggestion on how to test for NULL values without getting above error. I'm using opensips 2.3.5 -------------- next part -------------- An HTML attachment was scrubbed... URL: From atuxnull at gmail.com Tue Nov 13 11:06:59 2018 From: atuxnull at gmail.com (John Tuxies) Date: Tue, 13 Nov 2018 18:06:59 +0200 Subject: [OpenSIPS-Users] asterisk integration with control panel In-Reply-To: References: Message-ID: Someone plese? Opensips 2.4.3 and control panel 8.2.4 has been installed Cannot find how to integrate Asterisk On Friday, November 9, 2018, John Tuxies wrote: > Hi. i am new to the area of opensips and i would like some help. I am > doing my baby steps, but i have some experience with Asterisk. > In a Debian 9 64bit system i did install opensips as described in here: > https://www.powerpbx.org/content/opensips-v24-debian-v8-mariadb-apache-v1 > I would like to integrate Asterisk in the same box. > In the common sql of the users i will have users eg 4000-4099 and then > asterisk will have the inbound/outbound trunks, codecs, conference, > voicemail.. Opensips will manage traffic and security. > Up to now all i have is a system with Opensips and its panel. I do not > know how to integrate Asterisk in here. What changes to do in the database. > Is there a guide on how to integrate it? I have seen the one in > https://www.opensips.org/Documentation/Tutorials-OpenSIPSAst > eriskIntegration but it is outdated and it broke my system when i > followed it to Asterisk 11 and newer versions. > > Once having that setup running then i assume all users will be created in > opensips panel, or in Asterisk? > > Sincerely yours, > John > -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Tue Nov 6 15:14:07 2018 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Tue, 6 Nov 2018 20:14:07 +0000 Subject: [OpenSIPS-Users] CPU 100% with TCP In-Reply-To: <8f6a5096-b719-bfae-dfac-b5dad161601f@opensips.org> References: <55F1EF32-61C8-45E2-B77C-1A3D14C6FA32@genesys.com> <0bed4419-36ff-b0ad-425a-c5c8283e1943@opensips.org> <97134D2E-9BFE-4CDF-AC24-B4A211EA9BE7@genesys.com> <356959a3-97ba-38b3-7ca4-89403d0a3335@opensips.org> <8f6a5096-b719-bfae-dfac-b5dad161601f@opensips.org> Message-ID: <23D5BF0D-9B02-4EED-BA2F-3E18D1327887@genesys.com> Bogdan, I am trying to obtain this information for you but I am having trouble getting the core files. Is it really necessary to kill every opensips process? This generates almost 40 core files and each is quite large (~1GB). I simply don’t have that disk space currently. I can make a change to get more but it is slowing the process. Would it be sufficient to get just one core file? Also, runtime inspection with gdb is possible in this case if you can provide me with the commands you would want to see. I would need very specific commands as I am not very familiar with gdb. Ben Newlin From: Bogdan-Andrei Iancu Date: Thursday, November 1, 2018 at 1:29 PM To: Ben Newlin , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] CPU 100% with TCP Hi Ben, First be sure you have the DBG_LOCK option compiled in. Do the "opensips -V" and see the output flags. Next step will be to force an SIGSEGV to opensips (killall -11 opensips) when the deadlock occurs - I need a core file to inspect (assuming that runtime inspection with gdb is not possible). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 10/31/2018 09:07 PM, Ben Newlin wrote: Bogdan, For the first test I have done as you suggested and disabled only async operation for HEP, so it is still using TCP. I will send you the trap info directly as it is too large. I also compiled with the DBG_LOCK option, but am unsure whether that extra information will be available in the trap output or do you need something else? I am now going to switch HEP to use UDP to mirror our production environment and try to reproduce again. Wish me luck! ☺ Ben Newlin From: Bogdan-Andrei Iancu Date: Monday, October 29, 2018 at 2:19 PM To: Ben Newlin , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] CPU 100% with TCP Hi Ben, I checked the error trace and it should not leave any dangling lock (due mishandled error). Before disabling HEP, try to disable the async support for HEP. If you claim that the same 100% CPU happens with HEP + UDP, send me a trap for that too, as in the previous case, the deadlock was exclusively HEP + TCP related. Anyhow, as the original trap showed a deadlock, next step will be to recompile with the DBG_LOCK option - this enables extra code to debug/troubleshoot locking related issues - are you able to do it? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 10/26/2018 04:14 PM, Ben Newlin wrote: Bogdan, Actually, yes we do. Looking back I can see these errors just before the issue occurs: Oct 24 19:00:36 [5700] ERROR:proto_hep:send_hep_message: Cannot send hep message! Oct 24 19:00:36 [5700] ERROR:proto_hep:msg_send: send() to 10.32.163.211:9061 for proto hep_tcp/9 failed Oct 24 19:00:36 [5700] ERROR:proto_hep:hep_tcp_send: failed to send Oct 24 19:00:36 [5700] ERROR:proto_hep:async_tsend_stream: Failed first TCP async send : (32) Broken pipe I will try disabling HEP and see if we can reproduce. Just for information, I have been reproducing the issue in our testing environment which uses TCP for HEP, however the issue is occurring in our production environment as well which is still using UDP for HEP. Ben Newlin From: Bogdan-Andrei Iancu Date: Friday, October 26, 2018 at 3:06 AM To: Ben Newlin , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] CPU 100% with TCP Hi Ben, Thank you for the info. It looks like the processes get stuck into a HEP related internal lock - do you see any HEP related errors in your logs, prior to the dead-lock ? Also, as PoC, could you disabled HEP tracing to see if the problem goes away ? Thanks, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 10/24/2018 10:18 PM, Ben Newlin wrote: Bogdan, I have run the command but the output was too large for pastebin so I have sent it to you directly. Ben Newlin From: Bogdan-Andrei Iancu Date: Wednesday, October 24, 2018 at 5:17 AM To: OpenSIPS users mailling list , Ben Newlin Subject: Re: [OpenSIPS-Users] CPU 100% with TCP Hi Ben, Could you run "opensipsctl trap" ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 10/24/2018 12:56 AM, Ben Newlin wrote: Hi, We have implemented TCP recently and are performing TCP<->UDP translation on one of our proxy types. This proxy only exists for that purpose; there are no DB queries, REST calls, or anything like that. It is designed to be very fast and high throughput. Recently we have found that when the remote endpoint of a TCP connection is lost, i.e. the server goes down, while under moderate load OpenSIPS quickly reaches 100% CPU and becomes unresponsive. When this occurs, the “top” command shows that between 30-90% CPU is in System (kernel) space, and each OpenSIPS TCP process shows many times the normal CPU. We are running OpenSIPS 2.4.2 on Amazon Linux. I obtained as much information as I could using ps, strace, and gdb here: https://pastebin.com/JP3DnCqs. We can reproduce the failure consistently by removing a server during call traffic. A few things I noticed: * The number of running threads reported by OpenSIPS doesn’t align with our configuration, copied here: ####### Global Parameters ######### children=32 #// Allow 503 to pass back to Control disable_503_translation=yes #// Even though we are not receiving HEP, #// this listener is required by OpenSIPS #// in order to use the proto_hep module. :/ listen=hep_tcp:10.32.40.245:9061 use_children 1 #// Configure the listeners listen=udp:10.32.40.245:5060 as XXX.XXX.XXX.XXX listen=tcp:10.32.40.245:5060 as XXX.XXX.XXX.XXX #// Transaction Module loadmodule "tm.so" modparam("tm", "restart_fr_on_each_reply", 0) modparam("tm", "timer_partitions", 8) modparam("tm", "onreply_avp_mode", 1) modparam("tm", "wt_timer", 10) According to the documentation if “tcp_children” is not set then the value of “children” will be used [1], but we have set “children” to 32 and only have the default 8 TCP processes. Also we appear to only have 1 timer process, although we have set the number of timer partitions to 8. * The server that is terminated was using TCP connections exclusively, but all of the CPU seems to be in the UDP threads. The one I looked at appeared to be handling a CANCEL to one of the calls that was active and was attempting to send it out via TCP. I’m not sure why it would be trying to relay the CANCEL as no 100 Trying had been received from the server. I have noticed that in 2.x OpenSIPS will now send CANCELs for transactions even when 100 Trying was not received. Is that intentional? RFC 3261 states that no CANCEL should be sent unless a provisional response has been received. Any assistance with this would be appreciated. [1] - http://www.opensips.org/Documentation/Script-CoreParameters-2-4#toc66 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 trance_maniak at o2.pl Thu Nov 8 03:03:50 2018 From: trance_maniak at o2.pl (maatohewetbi) Date: Thu, 8 Nov 2018 01:03:50 -0700 (MST) Subject: [OpenSIPS-Users] Opensips 2.1.2 - can't pass var in rewritehostport() Message-ID: <1541664230763-0.post@n2.nabble.com> I have Opensips 2.1.2 and wanna to pass IP variable using sip header and parse it in Opensips to use with rewritehostport() function. I've checked that when I use it like rewritehostport("var(IP):5060") I can't pass this variable, it's empty. I've read it's impossible to do it. How can I do it right? -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html From bogdan at opensips.org Tue Nov 13 11:11:16 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 13 Nov 2018 18:11:16 +0200 Subject: [OpenSIPS-Users] CPU 100% with TCP In-Reply-To: <23D5BF0D-9B02-4EED-BA2F-3E18D1327887@genesys.com> References: <55F1EF32-61C8-45E2-B77C-1A3D14C6FA32@genesys.com> <0bed4419-36ff-b0ad-425a-c5c8283e1943@opensips.org> <97134D2E-9BFE-4CDF-AC24-B4A211EA9BE7@genesys.com> <356959a3-97ba-38b3-7ca4-89403d0a3335@opensips.org> <8f6a5096-b719-bfae-dfac-b5dad161601f@opensips.org> <23D5BF0D-9B02-4EED-BA2F-3E18D1327887@genesys.com> Message-ID: Hi Ben, Sorry for not being able to answer you before sending the new set of BTs. Indeed, getting the corefile of only one process will do it as the locks (and debug info) are in the shared memory. So, the deadlock happens again, do the "opensipsctl trap" and get the corefile of one process (ideally an UDP worker - get its pid via "opensipsctl fifo ps"). Keep the core as we will have to dig into it together :). Many thanks, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/06/2018 10:14 PM, Ben Newlin wrote: > > Bogdan, > > I am trying to obtain this information for you but I am having trouble > getting the core files. Is it really necessary to kill every opensips > process? This generates almost 40 core files and each is quite large > (~1GB). I simply don’t have that disk space currently. I can make a > change to get more but it is slowing the process. Would it be > sufficient to get just one core file? > > Also, runtime inspection with gdb is possible in this case if you can > provide me with the commands you would want to see. I would need very > specific commands as I am not very familiar with gdb. > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > *Date: *Thursday, November 1, 2018 at 1:29 PM > *To: *Ben Newlin , OpenSIPS users mailling > list > *Subject: *Re: [OpenSIPS-Users] CPU 100% with TCP > > Hi Ben, > > First be sure you have the DBG_LOCK option compiled in. Do the > "opensips -V" and see the output flags. > > Next step will be to force an SIGSEGV to opensips (killall -11 > opensips) when the deadlock occurs - I need a core file to inspect > (assuming that runtime inspection with gdb is not possible). > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Bootcamp 2018 > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 10/31/2018 09:07 PM, Ben Newlin wrote: > > Bogdan, > > For the first test I have done as you suggested and disabled only > async operation for HEP, so it is still using TCP. I will send you > the trap info directly as it is too large. I also compiled with > the DBG_LOCK option, but am unsure whether that extra information > will be available in the trap output or do you need something else? > > I am now going to switch HEP to use UDP to mirror our production > environment and try to reproduce again. Wish me luck! ☺ > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > > *Date: *Monday, October 29, 2018 at 2:19 PM > *To: *Ben Newlin > , OpenSIPS users mailling list > > *Subject: *Re: [OpenSIPS-Users] CPU 100% with TCP > > Hi Ben, > > I checked the error trace and it should not leave any dangling > lock (due mishandled error). Before disabling HEP, try to disable > the async support for HEP. > > If you claim that the same 100% CPU happens with HEP + UDP, send > me a trap for that too, as in the previous case, the deadlock was > exclusively HEP + TCP related. > > Anyhow, as the original trap showed a deadlock, next step will be > to recompile with the DBG_LOCK option - this enables extra code to > debug/troubleshoot locking related issues - are you able to do it? > > Regards, > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2018 > > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 10/26/2018 04:14 PM, Ben Newlin wrote: > > Bogdan, > > Actually, yes we do. Looking back I can see these errors just > before the issue occurs: > > Oct 24 19:00:36 [5700] ERROR:proto_hep:send_hep_message: > Cannot send hep message! > > Oct 24 19:00:36 [5700] ERROR:proto_hep:msg_send: send() to > 10.32.163.211:9061 for proto hep_tcp/9 failed > > Oct 24 19:00:36 [5700] ERROR:proto_hep:hep_tcp_send: failed to > send > > Oct 24 19:00:36 [5700] ERROR:proto_hep:async_tsend_stream: > Failed first TCP async send : (32) Broken pipe > > I will try disabling HEP and see if we can reproduce. > > Just for information, I have been reproducing the issue in our > testing environment which uses TCP for HEP, however the issue > is occurring in our production environment as well which is > still using UDP for HEP. > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > > *Date: *Friday, October 26, 2018 at 3:06 AM > *To: *Ben Newlin > , OpenSIPS users mailling list > > *Subject: *Re: [OpenSIPS-Users] CPU 100% with TCP > > Hi Ben, > > Thank you for the info. > > It looks like the processes get stuck into a HEP related > internal lock - do you see any HEP related errors in your > logs, prior to the dead-lock ? > > Also, as PoC, could you disabled HEP tracing to see if the > problem goes away ? > > Thanks, > > > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2018 > > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 10/24/2018 10:18 PM, Ben Newlin wrote: > > Bogdan, > > I have run the command but the output was too large for > pastebin so I have sent it to you directly. > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > > *Date: *Wednesday, October 24, 2018 at 5:17 AM > *To: *OpenSIPS users mailling list > > , Ben Newlin > > *Subject: *Re: [OpenSIPS-Users] CPU 100% with TCP > > Hi Ben, > > Could you run "opensipsctl trap" ? > > Regards, > > > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2018 > > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 10/24/2018 12:56 AM, Ben Newlin wrote: > > Hi, > > We have implemented TCP recently and are performing > TCP<->UDP translation on one of our proxy types. This > proxy only exists for that purpose; there are no DB > queries, REST calls, or anything like that. It is > designed to be very fast and high throughput. > > Recently we have found that when the remote endpoint > of a TCP connection is lost, i.e. the server goes > down, while under moderate load OpenSIPS quickly > reaches 100% CPU and becomes unresponsive. When this > occurs, the “top” command shows that between 30-90% > CPU is in System (kernel) space, and each OpenSIPS TCP > process shows many times the normal CPU. We are > running OpenSIPS 2.4.2 on Amazon Linux. > > I obtained as much information as I could using ps, > strace, and gdb here: https://pastebin.com/JP3DnCqs > . We can reproduce the > failure consistently by removing a server during call > traffic. > > A few things I noticed: > > * The number of running threads reported by OpenSIPS > doesn’t align with our configuration, copied here: > > ####### Global Parameters ######### > > children=32 > > #// Allow 503 to pass back to Control > > disable_503_translation=yes > > #// Even though we are not receiving HEP, > > #// this listener is required by OpenSIPS > > #// in order to use the proto_hep module. :/ > > listen=hep_tcp:10.32.40.245:9061 use_children 1 > > #// Configure the listeners > > listen=udp:10.32.40.245:5060 as XXX.XXX.XXX.XXX > > listen=tcp:10.32.40.245:5060 as XXX.XXX.XXX.XXX > > #// Transaction Module > > loadmodule "tm.so" > > modparam("tm", "restart_fr_on_each_reply", 0) > > modparam("tm", "timer_partitions", 8) > > modparam("tm", "onreply_avp_mode", 1) > > modparam("tm", "wt_timer", 10) > > According to the documentation if “tcp_children” is > not set then the value of “children” will be used [1], > but we have set “children” to 32 and only have the > default 8 TCP processes. Also we appear to only have 1 > timer process, although we have set the number of > timer partitions to 8. > > * The server that is terminated was using TCP > connections exclusively, but all of the CPU seems > to be in the UDP threads. The one I looked at > appeared to be handling a CANCEL to one of the > calls that was active and was attempting to send > it out via TCP. I’m not sure why it would be > trying to relay the CANCEL as no 100 Trying had > been received from the server. I have noticed that > in 2.x OpenSIPS will now send CANCELs for > transactions even when 100 Trying was not > received. Is that intentional? RFC 3261 states > that no CANCEL should be sent unless a provisional > response has been received. > > Any assistance with this would be appreciated. > > [1] - > http://www.opensips.org/Documentation/Script-CoreParameters-2-4#toc66 > > 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 Nov 13 12:11:07 2018 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Tue, 13 Nov 2018 19:11:07 +0200 Subject: [OpenSIPS-Users] Opensips 2.1.2 - can't pass var in rewritehostport() In-Reply-To: <1541664230763-0.post@n2.nabble.com> References: <1541664230763-0.post@n2.nabble.com> Message-ID: You can achieve this by using: $rd = $hdr(X-IP-header); # X-IP-header is the header's name that holds the IP Best regards, Răzvan On 11/8/18 10:03 AM, maatohewetbi wrote: > I have Opensips 2.1.2 and wanna to pass IP variable using sip header and > parse it in Opensips to use with rewritehostport() function. I've checked > that when I use it like rewritehostport("var(IP):5060") I can't pass this > variable, it's empty. I've read it's impossible to do it. How can I do it > right? > > > > -- > Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com From Ben.Newlin at genesys.com Tue Nov 13 12:36:08 2018 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Tue, 13 Nov 2018 17:36:08 +0000 Subject: [OpenSIPS-Users] CPU 100% with TCP In-Reply-To: References: <55F1EF32-61C8-45E2-B77C-1A3D14C6FA32@genesys.com> <0bed4419-36ff-b0ad-425a-c5c8283e1943@opensips.org> <97134D2E-9BFE-4CDF-AC24-B4A211EA9BE7@genesys.com> <356959a3-97ba-38b3-7ca4-89403d0a3335@opensips.org> <8f6a5096-b719-bfae-dfac-b5dad161601f@opensips.org> <23D5BF0D-9B02-4EED-BA2F-3E18D1327887@genesys.com> Message-ID: Bogdan, Can you clarify if you’re saying you need more information beyond the dumps I’ve just provided to you off-list? Ben Newlin From: Bogdan-Andrei Iancu Date: Tuesday, November 13, 2018 at 11:11 AM To: Ben Newlin , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] CPU 100% with TCP Hi Ben, Sorry for not being able to answer you before sending the new set of BTs. Indeed, getting the corefile of only one process will do it as the locks (and debug info) are in the shared memory. So, the deadlock happens again, do the "opensipsctl trap" and get the corefile of one process (ideally an UDP worker - get its pid via "opensipsctl fifo ps"). Keep the core as we will have to dig into it together :). Many thanks, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/06/2018 10:14 PM, Ben Newlin wrote: Bogdan, I am trying to obtain this information for you but I am having trouble getting the core files. Is it really necessary to kill every opensips process? This generates almost 40 core files and each is quite large (~1GB). I simply don’t have that disk space currently. I can make a change to get more but it is slowing the process. Would it be sufficient to get just one core file? Also, runtime inspection with gdb is possible in this case if you can provide me with the commands you would want to see. I would need very specific commands as I am not very familiar with gdb. Ben Newlin From: Bogdan-Andrei Iancu Date: Thursday, November 1, 2018 at 1:29 PM To: Ben Newlin , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] CPU 100% with TCP Hi Ben, First be sure you have the DBG_LOCK option compiled in. Do the "opensips -V" and see the output flags. Next step will be to force an SIGSEGV to opensips (killall -11 opensips) when the deadlock occurs - I need a core file to inspect (assuming that runtime inspection with gdb is not possible). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 10/31/2018 09:07 PM, Ben Newlin wrote: Bogdan, For the first test I have done as you suggested and disabled only async operation for HEP, so it is still using TCP. I will send you the trap info directly as it is too large. I also compiled with the DBG_LOCK option, but am unsure whether that extra information will be available in the trap output or do you need something else? I am now going to switch HEP to use UDP to mirror our production environment and try to reproduce again. Wish me luck! ☺ Ben Newlin From: Bogdan-Andrei Iancu Date: Monday, October 29, 2018 at 2:19 PM To: Ben Newlin , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] CPU 100% with TCP Hi Ben, I checked the error trace and it should not leave any dangling lock (due mishandled error). Before disabling HEP, try to disable the async support for HEP. If you claim that the same 100% CPU happens with HEP + UDP, send me a trap for that too, as in the previous case, the deadlock was exclusively HEP + TCP related. Anyhow, as the original trap showed a deadlock, next step will be to recompile with the DBG_LOCK option - this enables extra code to debug/troubleshoot locking related issues - are you able to do it? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 10/26/2018 04:14 PM, Ben Newlin wrote: Bogdan, Actually, yes we do. Looking back I can see these errors just before the issue occurs: Oct 24 19:00:36 [5700] ERROR:proto_hep:send_hep_message: Cannot send hep message! Oct 24 19:00:36 [5700] ERROR:proto_hep:msg_send: send() to 10.32.163.211:9061 for proto hep_tcp/9 failed Oct 24 19:00:36 [5700] ERROR:proto_hep:hep_tcp_send: failed to send Oct 24 19:00:36 [5700] ERROR:proto_hep:async_tsend_stream: Failed first TCP async send : (32) Broken pipe I will try disabling HEP and see if we can reproduce. Just for information, I have been reproducing the issue in our testing environment which uses TCP for HEP, however the issue is occurring in our production environment as well which is still using UDP for HEP. Ben Newlin From: Bogdan-Andrei Iancu Date: Friday, October 26, 2018 at 3:06 AM To: Ben Newlin , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] CPU 100% with TCP Hi Ben, Thank you for the info. It looks like the processes get stuck into a HEP related internal lock - do you see any HEP related errors in your logs, prior to the dead-lock ? Also, as PoC, could you disabled HEP tracing to see if the problem goes away ? Thanks, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 10/24/2018 10:18 PM, Ben Newlin wrote: Bogdan, I have run the command but the output was too large for pastebin so I have sent it to you directly. Ben Newlin From: Bogdan-Andrei Iancu Date: Wednesday, October 24, 2018 at 5:17 AM To: OpenSIPS users mailling list , Ben Newlin Subject: Re: [OpenSIPS-Users] CPU 100% with TCP Hi Ben, Could you run "opensipsctl trap" ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 10/24/2018 12:56 AM, Ben Newlin wrote: Hi, We have implemented TCP recently and are performing TCP<->UDP translation on one of our proxy types. This proxy only exists for that purpose; there are no DB queries, REST calls, or anything like that. It is designed to be very fast and high throughput. Recently we have found that when the remote endpoint of a TCP connection is lost, i.e. the server goes down, while under moderate load OpenSIPS quickly reaches 100% CPU and becomes unresponsive. When this occurs, the “top” command shows that between 30-90% CPU is in System (kernel) space, and each OpenSIPS TCP process shows many times the normal CPU. We are running OpenSIPS 2.4.2 on Amazon Linux. I obtained as much information as I could using ps, strace, and gdb here: https://pastebin.com/JP3DnCqs. We can reproduce the failure consistently by removing a server during call traffic. A few things I noticed: * The number of running threads reported by OpenSIPS doesn’t align with our configuration, copied here: ####### Global Parameters ######### children=32 #// Allow 503 to pass back to Control disable_503_translation=yes #// Even though we are not receiving HEP, #// this listener is required by OpenSIPS #// in order to use the proto_hep module. :/ listen=hep_tcp:10.32.40.245:9061 use_children 1 #// Configure the listeners listen=udp:10.32.40.245:5060 as XXX.XXX.XXX.XXX listen=tcp:10.32.40.245:5060 as XXX.XXX.XXX.XXX #// Transaction Module loadmodule "tm.so" modparam("tm", "restart_fr_on_each_reply", 0) modparam("tm", "timer_partitions", 8) modparam("tm", "onreply_avp_mode", 1) modparam("tm", "wt_timer", 10) According to the documentation if “tcp_children” is not set then the value of “children” will be used [1], but we have set “children” to 32 and only have the default 8 TCP processes. Also we appear to only have 1 timer process, although we have set the number of timer partitions to 8. * The server that is terminated was using TCP connections exclusively, but all of the CPU seems to be in the UDP threads. The one I looked at appeared to be handling a CANCEL to one of the calls that was active and was attempting to send it out via TCP. I’m not sure why it would be trying to relay the CANCEL as no 100 Trying had been received from the server. I have noticed that in 2.x OpenSIPS will now send CANCELs for transactions even when 100 Trying was not received. Is that intentional? RFC 3261 states that no CANCEL should be sent unless a provisional response has been received. Any assistance with this would be appreciated. [1] - http://www.opensips.org/Documentation/Script-CoreParameters-2-4#toc66 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 Nov 13 15:29:13 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 13 Nov 2018 22:29:13 +0200 Subject: [OpenSIPS-Users] CPU 100% with TCP In-Reply-To: References: <55F1EF32-61C8-45E2-B77C-1A3D14C6FA32@genesys.com> <0bed4419-36ff-b0ad-425a-c5c8283e1943@opensips.org> <97134D2E-9BFE-4CDF-AC24-B4A211EA9BE7@genesys.com> <356959a3-97ba-38b3-7ca4-89403d0a3335@opensips.org> <8f6a5096-b719-bfae-dfac-b5dad161601f@opensips.org> <23D5BF0D-9B02-4EED-BA2F-3E18D1327887@genesys.com> Message-ID: <47329ac3-a572-ce08-9956-b5f80bd0d5dd@opensips.org> Ben, The dump you send is actually the backtrace from all the procs. What is the Holly Grail is the corefile to be inspected via gdb(off list, of course). Thanks and regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/13/2018 07:36 PM, Ben Newlin wrote: > > Bogdan, > > Can you clarify if you’re saying you need more information beyond the > dumps I’ve just provided to you off-list? > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > *Date: *Tuesday, November 13, 2018 at 11:11 AM > *To: *Ben Newlin , OpenSIPS users mailling > list > *Subject: *Re: [OpenSIPS-Users] CPU 100% with TCP > > Hi Ben, > > Sorry for not being able to answer you before sending the new set of > BTs. Indeed, getting the corefile of only one process will do it as > the locks (and debug info) are in the shared memory. So, the deadlock > happens again, do the "opensipsctl trap" and get the corefile of one > process (ideally an UDP worker - get its pid via "opensipsctl fifo ps"). > Keep the core as we will have to dig into it together :). > > Many thanks, > > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Bootcamp 2018 > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 11/06/2018 10:14 PM, Ben Newlin wrote: > > Bogdan, > > I am trying to obtain this information for you but I am having > trouble getting the core files. Is it really necessary to kill > every opensips process? This generates almost 40 core files and > each is quite large (~1GB). I simply don’t have that disk space > currently. I can make a change to get more but it is slowing the > process. Would it be sufficient to get just one core file? > > Also, runtime inspection with gdb is possible in this case if you > can provide me with the commands you would want to see. I would > need very specific commands as I am not very familiar with gdb. > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > > *Date: *Thursday, November 1, 2018 at 1:29 PM > *To: *Ben Newlin > , OpenSIPS users mailling list > > *Subject: *Re: [OpenSIPS-Users] CPU 100% with TCP > > Hi Ben, > > First be sure you have the DBG_LOCK option compiled in. Do the > "opensips -V" and see the output flags. > > Next step will be to force an SIGSEGV to opensips (killall -11 > opensips) when the deadlock occurs - I need a core file to inspect > (assuming that runtime inspection with gdb is not possible). > > Regards, > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2018 > > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 10/31/2018 09:07 PM, Ben Newlin wrote: > > Bogdan, > > For the first test I have done as you suggested and disabled > only async operation for HEP, so it is still using TCP. I will > send you the trap info directly as it is too large. I also > compiled with the DBG_LOCK option, but am unsure whether that > extra information will be available in the trap output or do > you need something else? > > I am now going to switch HEP to use UDP to mirror our > production environment and try to reproduce again. Wish me luck! ☺ > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > > *Date: *Monday, October 29, 2018 at 2:19 PM > *To: *Ben Newlin > , OpenSIPS users mailling list > > *Subject: *Re: [OpenSIPS-Users] CPU 100% with TCP > > Hi Ben, > > I checked the error trace and it should not leave any dangling > lock (due mishandled error). Before disabling HEP, try to > disable the async support for HEP. > > If you claim that the same 100% CPU happens with HEP + UDP, > send me a trap for that too, as in the previous case, the > deadlock was exclusively HEP + TCP related. > > Anyhow, as the original trap showed a deadlock, next step will > be to recompile with the DBG_LOCK option - this enables extra > code to debug/troubleshoot locking related issues - are you > able to do it? > > Regards, > > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2018 > > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 10/26/2018 04:14 PM, Ben Newlin wrote: > > Bogdan, > > Actually, yes we do. Looking back I can see these errors > just before the issue occurs: > > Oct 24 19:00:36 [5700] ERROR:proto_hep:send_hep_message: > Cannot send hep message! > > Oct 24 19:00:36 [5700] ERROR:proto_hep:msg_send: send() to > 10.32.163.211:9061 for proto hep_tcp/9 failed > > Oct 24 19:00:36 [5700] ERROR:proto_hep:hep_tcp_send: > failed to send > > Oct 24 19:00:36 [5700] ERROR:proto_hep:async_tsend_stream: > Failed first TCP async send : (32) Broken pipe > > I will try disabling HEP and see if we can reproduce. > > Just for information, I have been reproducing the issue in > our testing environment which uses TCP for HEP, however > the issue is occurring in our production environment as > well which is still using UDP for HEP. > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > > *Date: *Friday, October 26, 2018 at 3:06 AM > *To: *Ben Newlin > , OpenSIPS users mailling > list > > *Subject: *Re: [OpenSIPS-Users] CPU 100% with TCP > > Hi Ben, > > Thank you for the info. > > It looks like the processes get stuck into a HEP related > internal lock - do you see any HEP related errors in your > logs, prior to the dead-lock ? > > Also, as PoC, could you disabled HEP tracing to see if the > problem goes away ? > > Thanks, > > > > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2018 > > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 10/24/2018 10:18 PM, Ben Newlin wrote: > > Bogdan, > > I have run the command but the output was too large > for pastebin so I have sent it to you directly. > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > > *Date: *Wednesday, October 24, 2018 at 5:17 AM > *To: *OpenSIPS users mailling list > > , Ben Newlin > > *Subject: *Re: [OpenSIPS-Users] CPU 100% with TCP > > Hi Ben, > > Could you run "opensipsctl trap" ? > > Regards, > > > > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2018 > > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 10/24/2018 12:56 AM, Ben Newlin wrote: > > Hi, > > We have implemented TCP recently and are > performing TCP<->UDP translation on one of our > proxy types. This proxy only exists for that > purpose; there are no DB queries, REST calls, or > anything like that. It is designed to be very fast > and high throughput. > > Recently we have found that when the remote > endpoint of a TCP connection is lost, i.e. the > server goes down, while under moderate load > OpenSIPS quickly reaches 100% CPU and becomes > unresponsive. When this occurs, the “top” command > shows that between 30-90% CPU is in System > (kernel) space, and each OpenSIPS TCP process > shows many times the normal CPU. We are running > OpenSIPS 2.4.2 on Amazon Linux. > > I obtained as much information as I could using > ps, strace, and gdb here: > https://pastebin.com/JP3DnCqs > . We can reproduce > the failure consistently by removing a server > during call traffic. > > A few things I noticed: > > * The number of running threads reported by > OpenSIPS doesn’t align with our configuration, > copied here: > > ####### Global Parameters ######### > > children=32 > > #// Allow 503 to pass back to Control > > disable_503_translation=yes > > #// Even though we are not receiving HEP, > > #// this listener is required by OpenSIPS > > #// in order to use the proto_hep module. :/ > > listen=hep_tcp:10.32.40.245:9061 use_children 1 > > #// Configure the listeners > > listen=udp:10.32.40.245:5060 as XXX.XXX.XXX.XXX > > listen=tcp:10.32.40.245:5060 as XXX.XXX.XXX.XXX > > #// Transaction Module > > loadmodule "tm.so" > > modparam("tm", "restart_fr_on_each_reply", 0) > > modparam("tm", "timer_partitions", 8) > > modparam("tm", "onreply_avp_mode", 1) > > modparam("tm", "wt_timer", 10) > > According to the documentation if “tcp_children” > is not set then the value of “children” will be > used [1], but we have set “children” to 32 and > only have the default 8 TCP processes. Also we > appear to only have 1 timer process, although we > have set the number of timer partitions to 8. > > * The server that is terminated was using TCP > connections exclusively, but all of the CPU > seems to be in the UDP threads. The one I > looked at appeared to be handling a CANCEL to > one of the calls that was active and was > attempting to send it out via TCP. I’m not > sure why it would be trying to relay the > CANCEL as no 100 Trying had been received from > the server. I have noticed that in 2.x > OpenSIPS will now send CANCELs for > transactions even when 100 Trying was not > received. Is that intentional? RFC 3261 states > that no CANCEL should be sent unless a > provisional response has been received. > > Any assistance with this would be appreciated. > > [1] - > http://www.opensips.org/Documentation/Script-CoreParameters-2-4#toc66 > > 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 genesys.com Tue Nov 13 15:34:42 2018 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Tue, 13 Nov 2018 20:34:42 +0000 Subject: [OpenSIPS-Users] CPU 100% with TCP In-Reply-To: <47329ac3-a572-ce08-9956-b5f80bd0d5dd@opensips.org> References: <55F1EF32-61C8-45E2-B77C-1A3D14C6FA32@genesys.com> <0bed4419-36ff-b0ad-425a-c5c8283e1943@opensips.org> <97134D2E-9BFE-4CDF-AC24-B4A211EA9BE7@genesys.com> <356959a3-97ba-38b3-7ca4-89403d0a3335@opensips.org> <8f6a5096-b719-bfae-dfac-b5dad161601f@opensips.org> <23D5BF0D-9B02-4EED-BA2F-3E18D1327887@genesys.com> <47329ac3-a572-ce08-9956-b5f80bd0d5dd@opensips.org> Message-ID: <8CA919C5-EDFE-45F3-87E8-9B7A313DFF10@genesys.com> Bogdan, I do still have the corefiles, but isn’t that useless to you without my exact system (binaries, modules,etc)? Either way, I’m happy to send it. Ben Newlin From: Bogdan-Andrei Iancu Date: Tuesday, November 13, 2018 at 3:29 PM To: Ben Newlin , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] CPU 100% with TCP Ben, The dump you send is actually the backtrace from all the procs. What is the Holly Grail is the corefile to be inspected via gdb (off list, of course). Thanks and regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/13/2018 07:36 PM, Ben Newlin wrote: Bogdan, Can you clarify if you’re saying you need more information beyond the dumps I’ve just provided to you off-list? Ben Newlin From: Bogdan-Andrei Iancu Date: Tuesday, November 13, 2018 at 11:11 AM To: Ben Newlin , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] CPU 100% with TCP Hi Ben, Sorry for not being able to answer you before sending the new set of BTs. Indeed, getting the corefile of only one process will do it as the locks (and debug info) are in the shared memory. So, the deadlock happens again, do the "opensipsctl trap" and get the corefile of one process (ideally an UDP worker - get its pid via "opensipsctl fifo ps"). Keep the core as we will have to dig into it together :). Many thanks, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/06/2018 10:14 PM, Ben Newlin wrote: Bogdan, I am trying to obtain this information for you but I am having trouble getting the core files. Is it really necessary to kill every opensips process? This generates almost 40 core files and each is quite large (~1GB). I simply don’t have that disk space currently. I can make a change to get more but it is slowing the process. Would it be sufficient to get just one core file? Also, runtime inspection with gdb is possible in this case if you can provide me with the commands you would want to see. I would need very specific commands as I am not very familiar with gdb. Ben Newlin From: Bogdan-Andrei Iancu Date: Thursday, November 1, 2018 at 1:29 PM To: Ben Newlin , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] CPU 100% with TCP Hi Ben, First be sure you have the DBG_LOCK option compiled in. Do the "opensips -V" and see the output flags. Next step will be to force an SIGSEGV to opensips (killall -11 opensips) when the deadlock occurs - I need a core file to inspect (assuming that runtime inspection with gdb is not possible). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 10/31/2018 09:07 PM, Ben Newlin wrote: Bogdan, For the first test I have done as you suggested and disabled only async operation for HEP, so it is still using TCP. I will send you the trap info directly as it is too large. I also compiled with the DBG_LOCK option, but am unsure whether that extra information will be available in the trap output or do you need something else? I am now going to switch HEP to use UDP to mirror our production environment and try to reproduce again. Wish me luck! ☺ Ben Newlin From: Bogdan-Andrei Iancu Date: Monday, October 29, 2018 at 2:19 PM To: Ben Newlin , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] CPU 100% with TCP Hi Ben, I checked the error trace and it should not leave any dangling lock (due mishandled error). Before disabling HEP, try to disable the async support for HEP. If you claim that the same 100% CPU happens with HEP + UDP, send me a trap for that too, as in the previous case, the deadlock was exclusively HEP + TCP related. Anyhow, as the original trap showed a deadlock, next step will be to recompile with the DBG_LOCK option - this enables extra code to debug/troubleshoot locking related issues - are you able to do it? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 10/26/2018 04:14 PM, Ben Newlin wrote: Bogdan, Actually, yes we do. Looking back I can see these errors just before the issue occurs: Oct 24 19:00:36 [5700] ERROR:proto_hep:send_hep_message: Cannot send hep message! Oct 24 19:00:36 [5700] ERROR:proto_hep:msg_send: send() to 10.32.163.211:9061 for proto hep_tcp/9 failed Oct 24 19:00:36 [5700] ERROR:proto_hep:hep_tcp_send: failed to send Oct 24 19:00:36 [5700] ERROR:proto_hep:async_tsend_stream: Failed first TCP async send : (32) Broken pipe I will try disabling HEP and see if we can reproduce. Just for information, I have been reproducing the issue in our testing environment which uses TCP for HEP, however the issue is occurring in our production environment as well which is still using UDP for HEP. Ben Newlin From: Bogdan-Andrei Iancu Date: Friday, October 26, 2018 at 3:06 AM To: Ben Newlin , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] CPU 100% with TCP Hi Ben, Thank you for the info. It looks like the processes get stuck into a HEP related internal lock - do you see any HEP related errors in your logs, prior to the dead-lock ? Also, as PoC, could you disabled HEP tracing to see if the problem goes away ? Thanks, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 10/24/2018 10:18 PM, Ben Newlin wrote: Bogdan, I have run the command but the output was too large for pastebin so I have sent it to you directly. Ben Newlin From: Bogdan-Andrei Iancu Date: Wednesday, October 24, 2018 at 5:17 AM To: OpenSIPS users mailling list , Ben Newlin Subject: Re: [OpenSIPS-Users] CPU 100% with TCP Hi Ben, Could you run "opensipsctl trap" ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 10/24/2018 12:56 AM, Ben Newlin wrote: Hi, We have implemented TCP recently and are performing TCP<->UDP translation on one of our proxy types. This proxy only exists for that purpose; there are no DB queries, REST calls, or anything like that. It is designed to be very fast and high throughput. Recently we have found that when the remote endpoint of a TCP connection is lost, i.e. the server goes down, while under moderate load OpenSIPS quickly reaches 100% CPU and becomes unresponsive. When this occurs, the “top” command shows that between 30-90% CPU is in System (kernel) space, and each OpenSIPS TCP process shows many times the normal CPU. We are running OpenSIPS 2.4.2 on Amazon Linux. I obtained as much information as I could using ps, strace, and gdb here: https://pastebin.com/JP3DnCqs. We can reproduce the failure consistently by removing a server during call traffic. A few things I noticed: * The number of running threads reported by OpenSIPS doesn’t align with our configuration, copied here: ####### Global Parameters ######### children=32 #// Allow 503 to pass back to Control disable_503_translation=yes #// Even though we are not receiving HEP, #// this listener is required by OpenSIPS #// in order to use the proto_hep module. :/ listen=hep_tcp:10.32.40.245:9061 use_children 1 #// Configure the listeners listen=udp:10.32.40.245:5060 as XXX.XXX.XXX.XXX listen=tcp:10.32.40.245:5060 as XXX.XXX.XXX.XXX #// Transaction Module loadmodule "tm.so" modparam("tm", "restart_fr_on_each_reply", 0) modparam("tm", "timer_partitions", 8) modparam("tm", "onreply_avp_mode", 1) modparam("tm", "wt_timer", 10) According to the documentation if “tcp_children” is not set then the value of “children” will be used [1], but we have set “children” to 32 and only have the default 8 TCP processes. Also we appear to only have 1 timer process, although we have set the number of timer partitions to 8. * The server that is terminated was using TCP connections exclusively, but all of the CPU seems to be in the UDP threads. The one I looked at appeared to be handling a CANCEL to one of the calls that was active and was attempting to send it out via TCP. I’m not sure why it would be trying to relay the CANCEL as no 100 Trying had been received from the server. I have noticed that in 2.x OpenSIPS will now send CANCELs for transactions even when 100 Trying was not received. Is that intentional? RFC 3261 states that no CANCEL should be sent unless a provisional response has been received. Any assistance with this would be appreciated. [1] - http://www.opensips.org/Documentation/Script-CoreParameters-2-4#toc66 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 kurgan-rus at inbox.ru Wed Nov 14 05:06:40 2018 From: kurgan-rus at inbox.ru (=?UTF-8?B?QWxleGV5IEthemFudHNldg==?=) Date: Wed, 14 Nov 2018 13:06:40 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?uac=5Fregistrant_clustering?= Message-ID: <1542190000.342851450@f436.i.mail.ru> Hi all, if there's a way to share outboung registrations for high availability? There's nothing about cluster support in uac_registrant documentation. Assuming that I'd like to register 2 OpenSIPS boxes against VoIP providers' in the Internet. ----------------------------------------------- BR, Alexey -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Nov 14 07:09:35 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 14 Nov 2018 14:09:35 +0200 Subject: [OpenSIPS-Users] asterisk integration with control panel In-Reply-To: References: Message-ID: <10b5c1ab-22e5-e385-4db4-0d2a7d602ff8@opensips.org> Hi John, There is also a newer (but not newest) version of the tutorial: https://www.opensips.org/Documentation/Tutorials-OpenSIPSAsteriskIntegration-1-8 As per the setup description (in the tutorial), Asterisk is driven via the OpenSIPS database, so you do not have to do any user provisioning in Asterisk, but only in OpenSIPS. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/13/2018 06:06 PM, John Tuxies wrote: > Someone plese? > > Opensips 2.4.3 and control panel 8.2.4 > has been installed > Cannot find how to integrate Asterisk > > > > > > On Friday, November 9, 2018, John Tuxies > wrote: > > Hi. i am new to the area of opensips and i would like some help. I > am doing my baby steps, but i have some experience with Asterisk. > In a Debian 9 64bit system i did install opensips as described in > here: > https://www.powerpbx.org/content/opensips-v24-debian-v8-mariadb-apache-v1 > > I would like to integrate Asterisk in the same box. > In the common sql of the users i will have users eg 4000-4099 and > then asterisk will have the inbound/outbound trunks, codecs, > conference, voicemail.. Opensips will manage traffic and security. > Up to now all i have is a system with Opensips and its panel. I do > not know how to integrate Asterisk in here. What changes to do in > the database. Is there a guide on how to integrate it? I have seen > the one in > https://www.opensips.org/Documentation/Tutorials-OpenSIPSAsteriskIntegration > > but it is outdated and it broke my system when i followed it to > Asterisk 11 and newer versions. > > Once having that setup running then i assume all users will be > created in opensips panel, or in Asterisk? > > Sincerely yours, > John > > > > _______________________________________________ > 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 vitalik.voip at gmail.com Wed Nov 14 08:17:38 2018 From: vitalik.voip at gmail.com (Vitalii Aleksandrov) Date: Wed, 14 Nov 2018 15:17:38 +0200 Subject: [OpenSIPS-Users] EBR and wait_for_event() timeout Message-ID: Hi, event_routing module provides the great async function wait_for_event().  If script subscribes for a event and received it it calls some "resume_route". What I can't understand is what happens with a transaction if wait_for_event() never catches an event and reaches its timeout. Is the any way to continue script execution from the place where "wait_for_event() was called or to execute some "timeout_route" to handle transaction properly? From bogdan at opensips.org Wed Nov 14 08:49:35 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 14 Nov 2018 15:49:35 +0200 Subject: [OpenSIPS-Users] check for NULL values In-Reply-To: <1506752889.1453845.1542099402376@mail.yahoo.com> References: <1238611305.714858.1541644790120.ref@mail.yahoo.com> <1238611305.714858.1541644790120@mail.yahoo.com> <1506752889.1453845.1542099402376@mail.yahoo.com> Message-ID: <38f2caf6-04a3-27f3-9a33-08c1cbe0d2b2@opensips.org> Hi Pasan, It should be fine if $tu would translate to NULL. The script handles this case. But I'm afraid you have something else there, like another deeper error that prevents the $tu variable to be evaluated. Do you see any other errors before the mentioned ones ? what is the type of route where you do the test ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/13/2018 10:56 AM, Pasan Meemaduma via Users wrote: > Hey, > > Anyone have a suggestion for this? > > On Thursday, 8 November 2018, 8:09:50 AM GMT+5:30, Pasan Meemaduma > wrote: > > > Hi Guys, > > I have a check for NULL for $tu var in the script, But when the value > is missing I'm getting the following error. > > ERROR:core:comp_scriptvar: cannot get left var value > > WARNING:core:do_action: error in expression at > /etc/opensips/opensips.cfg:806 > > and line 806 contains following. > > if ( $tu != NULL ) { > remove("location","$tu"); > } > > any suggestion on how to test for NULL values without getting above > error. I'm using opensips 2.3.5 > > > > > > _______________________________________________ > 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 Nov 14 08:51:42 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 14 Nov 2018 15:51:42 +0200 Subject: [OpenSIPS-Users] (no subject) In-Reply-To: References: Message-ID: <5c616812-2d91-7cb7-3cfa-0c93c49cb220@opensips.org> Hi Jehanzaib, For a profile to be shared via a noSQL db it must have the '/s' marker at the end (when you define it and when you use it). Regards, Bogdan Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/04/2018 11:55 PM, J E H A N Z A I B wrote: > Hi team, > > I used redis cache for dialog storage. I have 2 different servers both > are sharing the same redis. Is the profile size shared in this case? > > here is the dialog config. > > loadmodule "dialog.so" > modparam("dialog", "enable_stats", 1) > modparam("dialog", "cachedb_url", "redis:mysip://mysipx.xx.xx:xxxx/") > > This is how I check my profile size. > create_dialog(); > set_dlg_profile("myuniqprof","$avp(myprofile_id)"); > > get_profile_size("myuniqprof","$avp(myprofile_id)","$var(current_profile_size)"); > > Please note I am using version: opensips 1.11.3-notls > > > > > -- > Regards, > Jehanzaib > > > _______________________________________________ > 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 Nov 14 09:04:17 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 14 Nov 2018 16:04:17 +0200 Subject: [OpenSIPS-Users] redis cache sharing for dialog storage? In-Reply-To: References: Message-ID: <2022c9e0-ce7a-e06a-419b-76d69854b015@opensips.org> Hi Jehanzaib, Again, use the /s marker - see http://www.opensips.org/html/docs/modules/2.4.x/dialog.html#param_profiles_with_value . Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/06/2018 01:11 AM, J E H A N Z A I B wrote: > Hi there, > > I am not sure why the dialog stats will not be shared. If all records > are going to redis then I have a redis cluster which synchs the cache. > When I fetch the profile size it should be same (if it is being > fetched from the redis) across all the opensips node. I am bit > sceptical to upgrade without knowing what's happening. > > > On Tue, Nov 6, 2018 at 4:53 AM SamyGo > wrote: > > I have a strong feeling that you're using an old version of > opensips to expect it to share dialog states/profiles. I think > you'll need to use newer opensips 2.4+ having dialog sharing > capability using proto_bin and clusterer module: > http://www.opensips.org/html/docs/modules/2.4.x/dialog.html#param_profile_replication_cluster > > > On Mon, Nov 5, 2018 at 5:30 AM J E H A N Z A I B > > wrote: > > Hi team, > > I used redis cache for dialog storage. I have 2 different > servers both are sharing the same redis. Is the profile size > shared in this case? > > here is the dialog config. > > loadmodule "dialog.so" > modparam("dialog", "enable_stats", 1) > modparam("dialog", "cachedb_url", > "redis:mysip://mysipx.xx.xx:xxxx/") > > This is how I check my profile size. > create_dialog(); > set_dlg_profile("myuniqprof","$avp(myprofile_id)"); > get_profile_size("myuniqprof","$avp(myprofile_id)","$var(current_profile_size)"); > > Please note I am using version: opensips 1.11.3-notls > > > -- > Regards, > Jehanzaib > _______________________________________________ > 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 > > > > -- > Regards, > Jehanzaib > > > _______________________________________________ > 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 Nov 14 09:11:52 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 14 Nov 2018 16:11:52 +0200 Subject: [OpenSIPS-Users] Flush bad user data from from running opensips In-Reply-To: <11961787.XINqL1RPXL@blacky.mylan> References: <11961787.XINqL1RPXL@blacky.mylan> Message-ID: Hi Robert, Do you have the "use_domain" parameter enabled in the auth_db module ? http://www.opensips.org/html/docs/modules/2.4.x/auth_db.html#param_use_domain Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/07/2018 04:30 AM, Robert Dyck wrote: > > I have updated my small test bed from 2.3.2 to 2.4.2. I didn't bother > to back up the 'subscriber" table and it was wiped by the > installation. No big deal, it was tiny. > > So I added the users but made an error. > > opensipsctl add abc xyz -- I didn't specify the domain. The UAC would > not register. > > I corrected the user. > > opensipsctl rm abc, opensipsctl add abc at 192.168.1.2 xyz > > The UAC still cannot register. > > DBG:auth_db:get_ha1: no result for user 'abc@' > > Opensips is restarted and the UAC registers. > > Restaring a production machine is problematic. Is there a way to flush > the bad data which I assume has been cached? > > Some error checking in opensipsctl or the DB interface would be helpful. > > Thanks for your time and the product. > > Rob > > > > _______________________________________________ > 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 Nov 14 09:14:20 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 14 Nov 2018 16:14:20 +0200 Subject: [OpenSIPS-Users] GRUU contact not found In-Reply-To: <1891753.yvUAHoSbzm@blacky.mylan> References: <1891753.yvUAHoSbzm@blacky.mylan> Message-ID: <51a0382b-655f-7705-9756-df0eaae3b8b7@opensips.org> Hi Robert, According to docs, the gruu is by default off - see http://www.opensips.org/html/docs/modules/2.3.x/registrar.html#idp5567984 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/07/2018 10:09 PM, Robert Dyck wrote: > > My understanding is that GRUU processing in opensips is automatic, > provided it is not disabled. No further configuration or scripting is > required. Is that correct. > > A GRUU capable UA rergisters and receives public and temporary GR > identities. The UA establishes a dialog with another UA. The callee > ends the call. The caller does not recive the BYE. > > Caller : > > Request-Line: INVITE sip:7 at 192.168.1.2 SIP/2.0 > > Contact URI: > sip:4 at 192.168.1.2:5060;gr=urn:uuid:35dfa98a-2feb-482a-bde7-7568a86348b1 > > Callee: > > Status-Line: SIP/2.0 200 OK > > Caller: > > Request-Line: ACK sip:7 at 192.168.1.3:5062 SIP/2.0 > > Callee: > > Request-Line: BYE > sip:4 at 192.168.1.2:5060;gr=urn:uuid:35dfa98a-2feb-482a-bde7-7568a86348b1 > SIP/2.0 > > Proxy ( opensips @ 192.168.1.2 ) > > Status-Line: SIP/2.0 404 Not here > > Am I missing something? > > Should "opensipsctl ul show" show the GRUU? > > AOR:: 4 Contact:: sip:4 at 192.168.1.72:5062;transport=udp Q= > ContactID:: 3518589640418194 Expires:: > 3586 Callid:: OL1gvsViBJ Cseq:: 21 > User-agent:: LinphoneAndroid/4.0.1 (belle-sip/1.6.3) > State:: CS_NEW Flags:: 0 > Cflags:: Socket:: udp:192.168.1.2:5060 > Methods:: 4294967295 SIP_instance:: > > > > > _______________________________________________ > 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 Nov 14 09:20:52 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 14 Nov 2018 16:20:52 +0200 Subject: [OpenSIPS-Users] Permission doesn't match In-Reply-To: References: Message-ID: <7a999860-c3b2-246a-5d6a-738e05db5678@opensips.org> Hi Julian, If you do a "subnet_dump" (see http://www.opensips.org/html/docs/modules/2.4.x/permissions.html#mi_subnet_dump), do you see both records ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/08/2018 06:52 PM, Julian Santer wrote: > Hi guys, > > I have some question to the permission module. We are using Opensips > 2.2.6. > The permissions are load from the address table located in a MySQL DB. > > My config looks like: > > ... > else if (check_address("52", "$si", "$sp", "$proto", "$avp(ctx)", "$ua")) > { > xlog("L_INFO", "Entered here due permission 52 - LF_BASE"); > } > else if (check_address("54", "$si", "$sp", "$proto", "$avp(ctx)", "$ua")) > { > xlog("L_INFO", "Entered here due permission 54 - LF_BASE"); > } > ... > > address table: > id grp ip mask port proto pattern context_info > 41 52 192.168.1.0 24 0 any AVM*.06.* test > 648 54 192.168.1.0 24 0 any AVM*.07.* test > > This line is matching: > Nov 8 17:10:59 M=REGISTER RURI=sip:test.com F=sip:abc at test.com > T=sip:abc at test.com SRC=192.168.1.46:5060 UAC=AVM FRITZ!Box Fon WLAN > 7390 84.06.85 (Sep 10 2018) ID=9A0B1C90057A9126 at 192.168.1.46 B= > Nov 8 17:10:59 Entered here due permission 52 - M=REGISTER > RURI=sip:test.com F=sip:abc at test.com T=sip:abc at test.com > SRC=192.168.1.46:5060 UAC=AVM FRITZ!Box Fon WLAN 7390 84.06.85 (Sep 10 > 2018) ID=9A0B1C90057A9126 at 192.168.146 B= > > But this line is not matching: > Nov 8 17:35:19 M=REGISTER RURI=sip:test.com F=sip:def at test.com > T=sip:def at test.com SRC=192.168.1.215:5060 UAC=AVM FRITZ!Box 7490 > 113.07.01 (Sep 11 2018) ID=5DC1E7DC326043BA at 192.168.1.215 B= > > I already did a opensipsctl address reload and several times restarted > the whole opensips service. > Have you maybe some hint for me? > > Kind regards, > Julian Santer > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From bogdan at opensips.org Wed Nov 14 09:26:20 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 14 Nov 2018 16:26:20 +0200 Subject: [OpenSIPS-Users] uac_registrant clustering In-Reply-To: <1542190000.342851450@f436.i.mail.ru> References: <1542190000.342851450@f436.i.mail.ru> Message-ID: Hi Alexey, I fail to see the need for such a sharing - correct me if I'm wrong, but if "backup" OpenSIPS kicks in, it should simply register again against the provider Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/14/2018 12:06 PM, Alexey Kazantsev via Users wrote: > Hi all, > > if there's a way to share outboung registrations for high availability? > > There's nothing about cluster support in uac_registrant documentation. > > Assuming that I'd like to register 2 OpenSIPS boxes against VoIP > providers' > in the Internet. > > > ----------------------------------------------- > BR, Alexey > > > > _______________________________________________ > 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 Nov 14 09:28:48 2018 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 14 Nov 2018 16:28:48 +0200 Subject: [OpenSIPS-Users] Fraud Detection Module In-Reply-To: References: Message-ID: <18893c8d-38fe-1653-6975-ed5ccaa385a1@opensips.org> Hi Benjamin, As I've told you before [1], I haven't lost track of this one, rather I haven't found any time for it. [1]: http://lists.opensips.org/pipermail/users/2018-October/040093.html Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 07.11.2018 17:27, Benjamin Pasquet wrote: > Hello, > > I have some questions about fraud detection module and more > particulary about the sequential call statistics. > I am actualy in opensips version 2.2.2 but I tried in 2.2.4 and 2.2.7, > and I don't understand well behavior what I see on each version. > > 1) First, in relation to the behavior of the sequential call > statistics, I will give you an exemple to try to explain what I > expected and what I found. > > I have this following rules : > > ruleid profileid prefix ... > 1 10000 0033 > 2 10000 0044 > 3 20000 0033 > 4 20000 0044 > > User 10000 call the 0033123456789, I do check_fraud(10000, > 0033123456789, 10000), who match with the rule 1, the sequential call > counter of the rule 1 goes from 0 to 1. > User 10000 call the 0033123456789, I do check_fraud(10000, > 0033123456789, 10000), who match with the rule 1, the sequential call > counter of the rule 1 goes from 1 to 2. > User 10000 call the 0044123456789, I do check_fraud(10000, > 0044123456789, 10000), who match with the rule 2, the sequential call > counter of the rule 2 goes from 0 to 1. > User 10000 call the 0033123456789, I do check_fraud(10000, > 0033123456789, 10000), who match with the rule 1, the sequential call > counter of the rule 1 goes from 2 to 3 --> I was expecting that the > counter to go back to 1 cause the last number called by this user is > different. > User 10000 call the 0033987654321, I do check_fraud(10000, > 0033987654321, 10000), who match with the rule 1, the sequential call > counter of the rule 1 goes from 3 to 4 --> I was expecting that the > counter to go back to 1 for the same reasons than the previously case, > and further, for this rule and prefix, le number called is different, > that's why I was expecting even more that the counter to go back to 1 > > User 20000 call the 0033123456789, I do check_fraud(20000, > 0033123456789, 20000), who match with the rule 1, the sequential call > counter of the rule 3 goes from 0 to 1. > User 20000 call the 0033123456789, I do check_fraud(20000, > 0033123456789, 20000), who match with the rule 1, the sequential call > counter of the rule 3 goes from 1 to 2. > > User 10000 call the 0033123456789, I do check_fraud(10000, > 0033123456789, 10000), who match with the rule 1, the sequential call > counter of the rule 1 goes from 4 to 5 --> For this user, this prefix, > le called number is different than the previous one called, I was > expected that the counter to go back to 1 even if another user have > called this number just previously. > > For summarize, I was expected that the counter is reset per user for > all its rules, from the time the number called by the user is > different from the previous one. > > 2) Secondly, the FRAUD statistics are daily reset, but which parameter > are concerned? > Total calls > Calls per minute > Concurrent calls > Number of sequential calls > Call duration > > 3) Thirdly and the last point, is it possible to set a value for a > parameter rule who permit to don't check this one? Like set the > warning and critical parameter values of the sequential call to -1 for > a rule for exemple (I have find this supposition into the mailing list). > > Thank you in advance for your answer, > Best regards, Benjamin > > > _______________________________________________ > 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 genesys.com Wed Nov 14 10:04:39 2018 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Wed, 14 Nov 2018 15:04:39 +0000 Subject: [OpenSIPS-Users] OpenSIPS 2.4.3 Rest Client module Message-ID: Hello, After upgrading to 2.4.3 I have found that my tests using the REST client module are failing. It seems that some parameters to the rest_post client may have been inadvertently switched. Per the documentation [1], I am passing the body in the second parameter and the content type in the third. async(rest_post("$var(uri)", "$json(resourceRequest)", "$var(send_ctype)", "$var(body)", "$var(recv_ctype)", "$var(rcode)"), resume); But the request being sent out has the body and content type reversed: { "method" : "POST", "path" : "/manager/v1/resources", "headers" : { "Host" : [ "203.0.113.6:1080" ], "Accept" : [ "*/*" ], "Content-Type" : [ "{ \"resources\": 1, \"capabilities\": [ \"sip-service\"" ], "Content-Length" : [ "17" ] }, "keepAlive" : true, "secure" : false, "body" : "application\\/json" } I’ve verified that in 2.4.2 the request is being created properly. [1] https://opensips.org/html/docs/modules/2.4.x/rest_client.html#afunc_rest_post Ben Newlin -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Nov 14 10:16:59 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 14 Nov 2018 17:16:59 +0200 Subject: [OpenSIPS-Users] EBR and wait_for_event() timeout In-Reply-To: References: Message-ID: Hi Vitalii, For the wait_for_event(), the timeout seems to have no effect, the waiting being for ever :-| . The transaction has no timeout as you didn;t sent out any branch yet (the transaction timeout is for waiting on replies). Could you please open bug report on the opensips tracker on github ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/14/2018 03:17 PM, Vitalii Aleksandrov wrote: > Hi, > > event_routing module provides the great async function > wait_for_event(). If script subscribes for a event and received it it > calls some "resume_route". > What I can't understand is what happens with a transaction if > wait_for_event() never catches an event and reaches its timeout. > Is the any way to continue script execution from the place where > "wait_for_event() was called or to execute some "timeout_route" to > handle transaction properly? > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From bogdan at opensips.org Wed Nov 14 10:41:19 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 14 Nov 2018 17:41:19 +0200 Subject: [OpenSIPS-Users] Payload location In-Reply-To: <03F087CF-A2F5-47E2-824C-7AE6E863D569@free.fr> References: <03F087CF-A2F5-47E2-824C-7AE6E863D569@free.fr> Message-ID: Hi Alain, There is fix in maintained versions (2.4 and 2.2) to address this issue. 1.11 is not maintained anymore, but feel free to backport the patch : https://github.com/OpenSIPS/opensips/commit/42934c42400842a2e72e6766e040037f366643f4 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 06/07/2018 12:11 PM, Alain Bieuzent wrote: > > Hi, > > I’m using opensips 1.11.11 and since severals day i have a lot of > error like: > > Jun 7 11:06:08 ded-lb-in-master opensips[29450]: > ERROR:core:set_sdp_payload_attr: Invalid payload location > > Jun 7 11:06:08 ded-lb-in-master opensips[29452]: > ERROR:core:set_sdp_payload_attr: Invalid payload location > > Jun 7 11:06:11 ded-lb-in-master opensips[29449]: > ERROR:core:set_sdp_payload_attr: Invalid payload location > > Jun 7 11:06:12 ded-lb-in-master opensips[29449]: > ERROR:core:set_sdp_payload_attr: Invalid payload location > > I changed the log level to 4 but there is no more trace. > > Where can find more information about this error? > > Thanks > > Alain > > > > _______________________________________________ > 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 Nov 14 10:53:46 2018 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 14 Nov 2018 17:53:46 +0200 Subject: [OpenSIPS-Users] OpenSIPS 2.4.3 Rest Client module In-Reply-To: References: Message-ID: <0b937e05-97c1-4ff1-e8d2-52b670d5dd1d@opensips.org> Hi Ben, Looking into this asap.  I did lots of tests for the updated return codes / granular error handling, however I _did not_ assume I could have broken anything in that area. Thanks, Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 14.11.2018 17:04, Ben Newlin wrote: > > Hello, > > After upgrading to 2.4.3 I have found that my tests using the REST > client module are failing. It seems that some parameters to the > rest_post client may have been inadvertently switched. Per the > documentation [1], I am passing the body in the second parameter and > the content type in the third. > > async(rest_post("$var(uri)", "$json(resourceRequest)", > "$var(send_ctype)", "$var(body)", "$var(recv_ctype)", "$var(rcode)"), > resume); > > But the request being sent out has the body and content type reversed: > > { > >   "method" : "POST", > >   "path" : "/manager/v1/resources", > >   "headers" : { > >     "Host" : [ "203.0.113.6:1080" ], > >     "Accept" : [ "*/*" ], > >     "Content-Type" : [ "{ \"resources\": 1, \"capabilities\": [ > \"sip-service\"" ], > >     "Content-Length" : [ "17" ] > >   }, > >   "keepAlive" : true, > >   "secure" : false, > >   "body" : "application\\/json" > > } > > I’ve verified that in 2.4.2 the request is being created properly. > > [1] > https://opensips.org/html/docs/modules/2.4.x/rest_client.html#afunc_rest_post > > 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 liviu at opensips.org Wed Nov 14 11:10:21 2018 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 14 Nov 2018 18:10:21 +0200 Subject: [OpenSIPS-Users] OpenSIPS 2.4.3 Rest Client module In-Reply-To: <0b937e05-97c1-4ff1-e8d2-52b670d5dd1d@opensips.org> References: <0b937e05-97c1-4ff1-e8d2-52b670d5dd1d@opensips.org> Message-ID: <94e57c16-8796-15af-e4c1-bfcbfadce1a7@opensips.org> Many thanks for catching this one, Ben, I just fixed it [1]. This bug breaks all async(rest_post()) and async(rest_put()) calls of the 2.4.3 original release.  Maybe we should do another one... Cheers, [1]: https://github.com/OpenSIPS/opensips/commit/fb2aaf65ed993f429b2f12b547dc872aa8632992 Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 14.11.2018 17:53, Liviu Chircu wrote: > > Hi Ben, > > Looking into this asap.  I did lots of tests for the updated return > codes / granular error handling, however I _did not_ assume I could > have broken anything in that area. > > Thanks, > > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > On 14.11.2018 17:04, Ben Newlin wrote: >> >> Hello, >> >> After upgrading to 2.4.3 I have found that my tests using the REST >> client module are failing. It seems that some parameters to the >> rest_post client may have been inadvertently switched. Per the >> documentation [1], I am passing the body in the second parameter and >> the content type in the third. >> >> async(rest_post("$var(uri)", "$json(resourceRequest)", >> "$var(send_ctype)", "$var(body)", "$var(recv_ctype)", "$var(rcode)"), >> resume); >> >> But the request being sent out has the body and content type reversed: >> >> { >> >>   "method" : "POST", >> >>   "path" : "/manager/v1/resources", >> >>   "headers" : { >> >>     "Host" : [ "203.0.113.6:1080" ], >> >>     "Accept" : [ "*/*" ], >> >>     "Content-Type" : [ "{ \"resources\": 1, \"capabilities\": [ >> \"sip-service\"" ], >> >>     "Content-Length" : [ "17" ] >> >>   }, >> >>   "keepAlive" : true, >> >>   "secure" : false, >> >>   "body" : "application\\/json" >> >> } >> >> I’ve verified that in 2.4.2 the request is being created properly. >> >> [1] >> https://opensips.org/html/docs/modules/2.4.x/rest_client.html#afunc_rest_post >> >> 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 rob.dyck at telus.net Wed Nov 14 11:52:01 2018 From: rob.dyck at telus.net (Robert Dyck) Date: Wed, 14 Nov 2018 08:52:01 -0800 Subject: [OpenSIPS-Users] Flush bad user data from from running opensips References: <11961787.XINqL1RPXL@blacky.mylan> Message-ID: <1668240.Sym1kIPFuu@blacky.mylan> I do not have that parameter set and I do not use multiple domains. The problem was that after I corrected the error ( missing domain ), opensips continued to look for abc@ rather than abc. I was looking for a graceful way to correct the internal representation of the user name. Restarting opensips is no problem on a small installation but it is less than ideal. On Wednesday, November 14, 2018 6:11:52 AM PST Bogdan-Andrei Iancu wrote: Hi Robert, Do you have the "use_domain" parameter enabled in the auth_db module ? http://www.opensips.org/html/docs/modules/2.4.x/auth_db.html#param_use_domain[1] Bogdan-Andrei IancuOpenSIPS Founder and Developer http://www.opensips-solutions.com[2]OpenSIPS Bootcamp 2018 http://opensips.org/training/ OpenSIPS_Bootcamp_2018/[3] On 11/07/2018 04:30 AM, Robert Dyck wrote: I have updated my small test bed from 2.3.2 to 2.4.2. I didn't bother to back up the 'subscriber" table and it was wiped by the installation. No big deal, it was tiny. So I added the users but made an error. opensipsctl add abc xyz -- I didn't specify the domain. The UAC would not register. I corrected the user. opensipsctl rm abc, opensipsctl add abc at 192.168.1.2[4] xyz The UAC still cannot register. DBG:auth_db:get_ha1: no result for user 'abc@' Opensips is restarted and the UAC registers. Restaring a production machine is problematic. Is there a way to flush the bad data which I assume has been cached? Some error checking in opensipsctl or the DB interface would be helpful. Thanks for your time and the product. Rob _______________________________________________Users mailing listUsers at lists.opensips.org[5]http://lists.opensips.org/cgi-bin/mailman/listinfo/users[6] -------- [1] http://www.opensips.org/html/docs/modules/2.4.x/auth_db.html#param_use_domain [2] http://www.opensips-solutions.com [3] http://opensips.org/training/OpenSIPS_Bootcamp_2018/ [4] mailto:abc at 192.168.1.2 [5] mailto:Users at lists.opensips.org [6] http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Nov 14 11:59:10 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 14 Nov 2018 18:59:10 +0200 Subject: [OpenSIPS-Users] Flush bad user data from from running opensips In-Reply-To: <1668240.Sym1kIPFuu@blacky.mylan> References: <11961787.XINqL1RPXL@blacky.mylan> <1668240.Sym1kIPFuu@blacky.mylan> Message-ID: <1bf83534-d124-ad49-d4e3-5572e8881183@opensips.org> That's the whole idea - if the "use_domain" is on 0, OpenSIPS will reference the users only by username. So try "opensipsctl add abc xyz" and post what record you get into the subscriber table. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/14/2018 06:52 PM, Robert Dyck wrote: > > I do not have that parameter set and I do not use multiple domains. > > The problem was that after I corrected the error ( missing domain ), > opensips continued to look for abc@ rather than abc. I was looking for > a graceful way to correct the internal representation of the user > name. Restarting opensips is no problem on a small installation but it > is less than ideal. > > On Wednesday, November 14, 2018 6:11:52 AM PST Bogdan-Andrei Iancu wrote: > > Hi Robert, Do you have the "use_domain" parameter enabled in the > auth_db module ? > http://www.opensips.org/html/docs/modules/2.4.x/auth_db.html#param_use_domainRegards, > > > Bogdan-Andrei IancuOpenSIPS Founder and Developer > http://www.opensips-solutions.comOpenSIPS Bootcamp 2018 > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 11/07/2018 04:30 AM, Robert Dyck wrote: > > I have updated my small test bed from 2.3.2 to 2.4.2. I didn't bother > to back up the 'subscriber" table and it was wiped by the > installation. No big deal, it was tiny. > > So I added the users but made an error. > > opensipsctl add abc xyz -- I didn't specify the domain. The UAC would > not register. > > I corrected the user. > > opensipsctl rm abc, opensipsctl add abc at 192.168.1.2 > xyz > > The UAC still cannot register. > > DBG:auth_db:get_ha1: no result for user 'abc@' > > Opensips is restarted and the UAC registers. > > Restaring a production machine is problematic. Is there a way to flush > the bad data which I assume has been cached? > > Some error checking in opensipsctl or the DB interface would be helpful. > > Thanks for your time and the product. > > Rob > > _______________________________________________Users mailing > listUsers at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From rob.dyck at telus.net Wed Nov 14 12:09:03 2018 From: rob.dyck at telus.net (Robert Dyck) Date: Wed, 14 Nov 2018 09:09:03 -0800 Subject: [OpenSIPS-Users] GRUU contact not found References: <1891753.yvUAHoSbzm@blacky.mylan> Message-ID: <1563251.d96nG9f5RN@blacky.mylan> I started with the sample residential script from some time back. GRUU was enabled in the sample. When I started working with a UA that registers a GRUU I noticed it was not receiving BYE when the other end released the call. I have since experimented with checking for GRUU while in dialog. That seems to work. The documentation doesn't mention anything about modifying the script other than enabling or disabling GRUU. Do you have any tips regarding GRUU in the script? Are there corner cases I should be aware of? Rob On Wednesday, November 14, 2018 6:14:20 AM PST Bogdan-Andrei Iancu wrote: Hi Robert, According to docs, the gruu is by default off - seehttp://www.opensips.org/ html/docs/modules/2.3.x/registrar.html#idp5567984[1] Bogdan-Andrei IancuOpenSIPS Founder and Developer http://www.opensips-solutions.com[2]OpenSIPS Bootcamp 2018 http://opensips.org/training/ OpenSIPS_Bootcamp_2018/[3] On 11/07/2018 10:09 PM, Robert Dyck wrote: My understanding is that GRUU processing in opensips is automatic, provided it is not disabled. No further configuration or scripting is required. Is that correct. A GRUU capable UA rergisters and receives public and temporary GR identities. The UA establishes a dialog with another UA. The callee ends the call. The caller does not recive the BYE. Caller : Request-Line: INVITE sip:7 at 192.168.1.2[4] SIP/2.0 Contact URI: sip:4 at 192.168.1.2:5060;gr=urn:uuid:35dfa98a-2feb-482a-bde7-7568a86348b1[5] Callee: Status-Line: SIP/2.0 200 OK Caller: Request-Line: ACK sip:7 at 192.168.1.3:5062[6] SIP/2.0 Callee: Request-Line: BYE sip:4 at 192.168.1.2:5060;gr=urn:uuid:35dfa98a-2feb-482a- bde7-7568a86348b1[5] SIP/2.0 Proxy ( opensips @ 192.168.1.2 ) Status-Line: SIP/2.0 404 Not here Am I missing something? Should "opensipsctl ul show" show the GRUU? AOR:: 4 Contact:: sip:4 at 192.168.1.72:5062;transport=udp[7] Q= ContactID:: 3518589640418194 Expires:: 3586 Callid:: OL1gvsViBJ Cseq:: 21 User-agent:: LinphoneAndroid/4.0.1 (belle-sip/1.6.3) State:: CS_NEW Flags:: 0 Cflags:: Socket:: udp:192.168.1.2:5060 Methods:: 4294967295 SIP_instance:: _______________________________________________Users mailing listUsers at lists.opensips.org[8]http://lists.opensips.org/cgi-bin/mailman/listinfo/users[9] -------- [1] http://www.opensips.org/html/docs/modules/2.3.x/registrar.html#idp5567984 [2] http://www.opensips-solutions.com [3] http://opensips.org/training/OpenSIPS_Bootcamp_2018/ [4] sip:7 at 192.168.1.2 [5] sip:4 at 192.168.1.2:5060;gr=urn:uuid:35dfa98a-2feb-482a-bde7-7568a86348b1 [6] sip:7 at 192.168.1.3:5062 [7] sip:4 at 192.168.1.72:5062;transport=udp [8] mailto:Users at lists.opensips.org [9] http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From govoiper at gmail.com Wed Nov 14 12:20:32 2018 From: govoiper at gmail.com (SamyGo) Date: Wed, 14 Nov 2018 12:20:32 -0500 Subject: [OpenSIPS-Users] ACC module with JSON events In-Reply-To: References: <1519631511.3267.4.camel@gmail.com> Message-ID: Howdy again, Thanks team for creating the event_jsonrpc module in 2.4 - I'm back to this topic and trying to give this module a try however opensips fail to start with following error: Nov 14 17:13:03 opsips /usr/local/sbin/opensips[18997]: INFO:uac:mod_init: initializing... Nov 14 17:13:03 opsips /usr/local/sbin/opensips[18997]: NOTICE:event_jsonrpc:mod_init: initializing module ... Nov 14 17:13:03 opsips /usr/local/sbin/opensips[18997]: ERROR:core:register_event_mod: duplicate flag 4000000 Nov 14 17:13:03 opsips /usr/local/sbin/opensips[18997]: ERROR:event_jsonrpc:mod_init: cannot register transport functions for jsonrpc Nov 14 17:13:03 opsips /usr/local/sbin/opensips[18997]: ERROR:core:init_mod: failed to initialize module event_jsonrpc Nov 14 17:13:03 opsips /usr/local/sbin/opensips[18997]: ERROR:core:main: error while initializing modules I do have these two module loaded as well, (which I can't remove). loadmodule "event_routing.so" loadmodule "event_route.so" Any insight as to why these error show up? Big thanks, Sammy. On Mon, Feb 26, 2018 at 12:07 PM SamyGo wrote: > Hi DanB, > I can certainly try CGRates, but...surely enough its an overkill for > simple CDRs. Anyways, my request is for an event_JSONRPC which maybe very > helpful in events outside of ACC. > > Best Regards, > Sammy > > > > > On Mon, Feb 26, 2018 at 6:13 AM, Khalil Khamlichi < > khamlichi.khalil at gmail.com> wrote: > >> Hi Dan, >> >> Can you show us how to do that with some step by step tutorial, we had >> a lot of trouble trying to figure out how to use CGRates, with this >> functionality of json export, we will at least use some of CGRates >> functionalities and hopefully slowly get familiar with it. >> >> Thanks in advance. >> >> On Mon, Feb 26, 2018 at 7:51 AM, DanB wrote: >> > Sammy, >> > >> > Another option on short term until the new feature will be implemented >> > in OpenSIPS would be to use CGRateS as CDR format converter: receive >> > CDRs from "cgrates" module in OpenSIPS and use online export of CGRateS >> > to further export the CDR in the JSON over http (customizable fields). >> > You don't need to configure much on CGRateS side in this case since no >> > billing needs to be involved. >> > >> > DanB >> > >> > >> > >> > >> > >> > _______________________________________________ >> > 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 rob.dyck at telus.net Wed Nov 14 12:36:34 2018 From: rob.dyck at telus.net (Robert Dyck) Date: Wed, 14 Nov 2018 09:36:34 -0800 Subject: [OpenSIPS-Users] Flush bad user data from from running opensips References: <11961787.XINqL1RPXL@blacky.mylan> <1668240.Sym1kIPFuu@blacky.mylan> Message-ID: <1561693.mbn41HfsKT@blacky.mylan> [root at slim opensips]# opensipsctl add abc xyz *new user 'abc' added* 10:abc:localhost:xyz:: 6c7faf173d3b8e26d95e7f26dd0388d6:e091cc8c08b19e1d50ee3891d3f37153: [root at slim opensips]# opensipsctl rm abc [root at slim opensips]# opensipsctl add abc at 192.168.1.2 xyz *new user 'abc at 192.168.1.2' added* 10:abc:192.168.1.2:xyz:: 9ce761c3a9f328510ea011bd5c9bd2c5:cc312796ec331326cd537f3a3ffad7b6: The difference being localhost vs 192.168.1.2 abc@ not found. On Wednesday, November 14, 2018 8:59:10 AM PST Bogdan-Andrei Iancu wrote: That's the whole idea - if the "use_domain" is on 0, OpenSIPS will reference the users only by username. So try "opensipsctl add abc xyz" and post what record you get into the subscriber table. Regards, Bogdan-Andrei IancuOpenSIPS Founder and Developer http://www.opensips-solutions.com[1]OpenSIPS Bootcamp 2018 http://opensips.org/training/ OpenSIPS_Bootcamp_2018/[2] On 11/14/2018 06:52 PM, Robert Dyck wrote: I do not have that parameter set and I do not use multiple domains. The problem was that after I corrected the error ( missing domain ), opensips continued to look for abc@ rather than abc. I was looking for a graceful way to correct the internal representation of the user name. Restarting opensips is no problem on a small installation but it is less than ideal. On Wednesday, November 14, 2018 6:11:52 AM PST Bogdan-Andrei Iancu wrote: Hi Robert, Do you have the "use_domain" parameter enabled in the auth_db module ? http://www.opensips.org/html/docs/modules/2.4.x/auth_db.html#param_use_domain[3] Regards, Bogdan-Andrei IancuOpenSIPS Founder and Developer http://www.opensips-solutions.com[1]OpenSIPS Bootcamp 2018 http://opensips.org/training/ OpenSIPS_Bootcamp_2018/[2] On 11/07/2018 04:30 AM, Robert Dyck wrote: I have updated my small test bed from 2.3.2 to 2.4.2. I didn't bother to back up the 'subscriber" table and it was wiped by the installation. No big deal, it was tiny. So I added the users but made an error. opensipsctl add abc xyz -- I didn't specify the domain. The UAC would not register. I corrected the user. opensipsctl rm abc, opensipsctl add abc at 192.168.1.2[4] xyz The UAC still cannot register. DBG:auth_db:get_ha1: no result for user 'abc@' Opensips is restarted and the UAC registers. Restaring a production machine is problematic. Is there a way to flush the bad data which I assume has been cached? Some error checking in opensipsctl or the DB interface would be helpful. Thanks for your time and the product. Rob _______________________________________________Users mailing listUsers at lists.opensips.org[5]http://lists.opensips.org/cgi-bin/mailman/listinfo/users[6] -------- [1] http://www.opensips-solutions.com [2] http://opensips.org/training/OpenSIPS_Bootcamp_2018/ [3] http://www.opensips.org/html/docs/modules/2.4.x/auth_db.html#param_use_domain [4] mailto:abc at 192.168.1.2 [5] mailto:Users at lists.opensips.org [6] http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From aronp at guaranteedplus.com Wed Nov 14 12:38:10 2018 From: aronp at guaranteedplus.com (Podrigal, Aron) Date: Wed, 14 Nov 2018 11:38:10 -0600 Subject: [OpenSIPS-Users] Forking sip MESSAGE to multiple endpoints Message-ID: Hi. I want to fork a MESSAGE request to multiple endpoints and handle the delivery status for each on reply. currently I append_branch() for each additional destination. But the problem is, that if any branch received a 200 reply, any other branch which did not relay the request yet (ie socket connection wasn't established) or TM timer is triggered, then those branches are canceled without triggering onreply route or failure route. How could I implement it, so that I can handle the timeout or failure for each endpoint? Maybe t_replicate? Or is there an event raised for a canceled branch which I can subscribe to? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Wed Nov 14 12:58:45 2018 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Wed, 14 Nov 2018 17:58:45 +0000 Subject: [OpenSIPS-Users] OpenSIPS 2.4.3 Rest Client module In-Reply-To: <94e57c16-8796-15af-e4c1-bfcbfadce1a7@opensips.org> References: <0b937e05-97c1-4ff1-e8d2-52b670d5dd1d@opensips.org> <94e57c16-8796-15af-e4c1-bfcbfadce1a7@opensips.org> Message-ID: <2D96AC08-6DD8-4EEB-83E3-764E3B06399A@genesys.com> Liviu, Thanks for the quick turnaround on this one! I’ve confirmed the fix in that commit. Ben Newlin From: Users on behalf of Liviu Chircu Reply-To: OpenSIPS users mailling list Date: Wednesday, November 14, 2018 at 11:10 AM To: "users at lists.opensips.org" Subject: Re: [OpenSIPS-Users] OpenSIPS 2.4.3 Rest Client module Many thanks for catching this one, Ben, I just fixed it [1]. This bug breaks all async(rest_post()) and async(rest_put()) calls of the 2.4.3 original release. Maybe we should do another one... Cheers, [1]: https://github.com/OpenSIPS/opensips/commit/fb2aaf65ed993f429b2f12b547dc872aa8632992 Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 14.11.2018 17:53, Liviu Chircu wrote: Hi Ben, Looking into this asap. I did lots of tests for the updated return codes / granular error handling, however I _did not_ assume I could have broken anything in that area. Thanks, Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 14.11.2018 17:04, Ben Newlin wrote: Hello, After upgrading to 2.4.3 I have found that my tests using the REST client module are failing. It seems that some parameters to the rest_post client may have been inadvertently switched. Per the documentation [1], I am passing the body in the second parameter and the content type in the third. async(rest_post("$var(uri)", "$json(resourceRequest)", "$var(send_ctype)", "$var(body)", "$var(recv_ctype)", "$var(rcode)"), resume); But the request being sent out has the body and content type reversed: { "method" : "POST", "path" : "/manager/v1/resources", "headers" : { "Host" : [ "203.0.113.6:1080" ], "Accept" : [ "*/*" ], "Content-Type" : [ "{ \"resources\": 1, \"capabilities\": [ \"sip-service\"" ], "Content-Length" : [ "17" ] }, "keepAlive" : true, "secure" : false, "body" : "application\\/json" } I’ve verified that in 2.4.2 the request is being created properly. [1] https://opensips.org/html/docs/modules/2.4.x/rest_client.html#afunc_rest_post 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 rob.dyck at telus.net Wed Nov 14 13:03:06 2018 From: rob.dyck at telus.net (Robert Dyck) Date: Wed, 14 Nov 2018 10:03:06 -0800 Subject: [OpenSIPS-Users] Flush bad user data from from running opensips References: <11961787.XINqL1RPXL@blacky.mylan> <1668240.Sym1kIPFuu@blacky.mylan> Message-ID: <3727916.AF73741jUM@blacky.mylan> I added "modparam("auth_db", "use_domain", 1)" but it doesn't make a difference to the subscriber table. On Wednesday, November 14, 2018 9:36:34 AM PST Robert Dyck wrote: [root at slim opensips]# opensipsctl add abc xyz *new user 'abc' added* 10:abc:localhost:xyz:: 6c7faf173d3b8e26d95e7f26dd0388d6:e091cc8c08b19e1d50ee3891d3f37153: [root at slim opensips]# opensipsctl rm abc [root at slim opensips]# opensipsctl add abc at 192.168.1.2 xyz *new user 'abc at 192.168.1.2' added* 10:abc:192.168.1.2:xyz:: 9ce761c3a9f328510ea011bd5c9bd2c5:cc312796ec331326cd537f3a3ffad7b6: The difference being localhost vs 192.168.1.2 abc@ not found. On Wednesday, November 14, 2018 8:59:10 AM PST Bogdan-Andrei Iancu wrote: That's the whole idea - if the "use_domain" is on 0, OpenSIPS will reference the users only by username. So try "opensipsctl add abc xyz" and post what record you get into the subscriber table. Regards, Bogdan-Andrei IancuOpenSIPS Founder and Developer http://www.opensips-solutions.com[1]OpenSIPS Bootcamp 2018 http://opensips.org/training/ OpenSIPS_Bootcamp_2018/[2] On 11/14/2018 06:52 PM, Robert Dyck wrote: I do not have that parameter set and I do not use multiple domains. The problem was that after I corrected the error ( missing domain ), opensips continued to look for abc@ rather than abc. I was looking for a graceful way to correct the internal representation of the user name. Restarting opensips is no problem on a small installation but it is less than ideal. On Wednesday, November 14, 2018 6:11:52 AM PST Bogdan-Andrei Iancu wrote: Hi Robert, Do you have the "use_domain" parameter enabled in the auth_db module ? http://www.opensips.org/html/docs/modules/2.4.x/auth_db.html#param_use_domain[3] Regards, Bogdan-Andrei IancuOpenSIPS Founder and Developer http://www.opensips-solutions.com[1]OpenSIPS Bootcamp 2018 http://opensips.org/training/ OpenSIPS_Bootcamp_2018/[2] On 11/07/2018 04:30 AM, Robert Dyck wrote: I have updated my small test bed from 2.3.2 to 2.4.2. I didn't bother to back up the 'subscriber" table and it was wiped by the installation. No big deal, it was tiny. So I added the users but made an error. opensipsctl add abc xyz -- I didn't specify the domain. The UAC would not register. I corrected the user. opensipsctl rm abc, opensipsctl add abc at 192.168.1.2[4] xyz The UAC still cannot register. DBG:auth_db:get_ha1: no result for user 'abc@' Opensips is restarted and the UAC registers. Restaring a production machine is problematic. Is there a way to flush the bad data which I assume has been cached? Some error checking in opensipsctl or the DB interface would be helpful. Thanks for your time and the product. Rob _______________________________________________Users mailing listUsers at lists.opensips.org[5]http://lists.opensips.org/cgi-bin/mailman/listinfo/users[6] -------- [1] http://www.opensips-solutions.com [2] http://opensips.org/training/OpenSIPS_Bootcamp_2018/ [3] http://www.opensips.org/html/docs/modules/2.4.x/auth_db.html#param_use_domain [4] mailto:abc at 192.168.1.2 [5] mailto:Users at lists.opensips.org [6] http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From govoiper at gmail.com Wed Nov 14 13:08:02 2018 From: govoiper at gmail.com (SamyGo) Date: Wed, 14 Nov 2018 13:08:02 -0500 Subject: [OpenSIPS-Users] ACC module with JSON events In-Reply-To: References: <1519631511.3267.4.camel@gmail.com> Message-ID: Possibly found the reason: The event_jsonrpc.c has this flag: JSONRPC_FLAG with same value "1 << 26" same as event_route.c modules/event_route/event_route.h:#define SCRIPTROUTE_FLAG (1 << 26) modules/event_jsonrpc/event_jsonrpc.h:#define JSONRPC_FLAG (1 << 26) While other event routes have unique value. Is this intentional? can I change this flag value to something unused i.e 1 << 31 ? On Wed, Nov 14, 2018 at 12:20 PM SamyGo wrote: > Howdy again, > > Thanks team for creating the event_jsonrpc module in 2.4 - I'm back to > this topic and trying to give this module a try however opensips fail to > start with following error: > > Nov 14 17:13:03 opsips /usr/local/sbin/opensips[18997]: INFO:uac:mod_init: > initializing... > Nov 14 17:13:03 opsips /usr/local/sbin/opensips[18997]: > NOTICE:event_jsonrpc:mod_init: initializing module ... > Nov 14 17:13:03 opsips /usr/local/sbin/opensips[18997]: > ERROR:core:register_event_mod: duplicate flag 4000000 > Nov 14 17:13:03 opsips /usr/local/sbin/opensips[18997]: > ERROR:event_jsonrpc:mod_init: cannot register transport functions for > jsonrpc > Nov 14 17:13:03 opsips /usr/local/sbin/opensips[18997]: > ERROR:core:init_mod: failed to initialize module event_jsonrpc > Nov 14 17:13:03 opsips /usr/local/sbin/opensips[18997]: > ERROR:core:main: error while initializing modules > > I do have these two module loaded as well, (which I can't remove). > loadmodule "event_routing.so" > loadmodule "event_route.so" > > Any insight as to why these error show up? > > Big thanks, > Sammy. > > > > On Mon, Feb 26, 2018 at 12:07 PM SamyGo wrote: > >> Hi DanB, >> I can certainly try CGRates, but...surely enough its an overkill for >> simple CDRs. Anyways, my request is for an event_JSONRPC which maybe very >> helpful in events outside of ACC. >> >> Best Regards, >> Sammy >> >> >> >> >> On Mon, Feb 26, 2018 at 6:13 AM, Khalil Khamlichi < >> khamlichi.khalil at gmail.com> wrote: >> >>> Hi Dan, >>> >>> Can you show us how to do that with some step by step tutorial, we had >>> a lot of trouble trying to figure out how to use CGRates, with this >>> functionality of json export, we will at least use some of CGRates >>> functionalities and hopefully slowly get familiar with it. >>> >>> Thanks in advance. >>> >>> On Mon, Feb 26, 2018 at 7:51 AM, DanB wrote: >>> > Sammy, >>> > >>> > Another option on short term until the new feature will be implemented >>> > in OpenSIPS would be to use CGRateS as CDR format converter: receive >>> > CDRs from "cgrates" module in OpenSIPS and use online export of CGRateS >>> > to further export the CDR in the JSON over http (customizable fields). >>> > You don't need to configure much on CGRateS side in this case since no >>> > billing needs to be involved. >>> > >>> > DanB >>> > >>> > >>> > >>> > >>> > >>> > _______________________________________________ >>> > 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 rosenberg11219 at gmail.com Wed Nov 14 13:08:16 2018 From: rosenberg11219 at gmail.com (Schneur Rosenberg) Date: Wed, 14 Nov 2018 20:08:16 +0200 Subject: [OpenSIPS-Users] Flush bad user data from from running opensips In-Reply-To: <1bf83534-d124-ad49-d4e3-5572e8881183@opensips.org> References: <11961787.XINqL1RPXL@blacky.mylan> <1668240.Sym1kIPFuu@blacky.mylan> <1bf83534-d124-ad49-d4e3-5572e8881183@opensips.org> Message-ID: Bogdan what ever happened to the reload feature, it was discussed in the past, it can greatly improve our lives :-) On Wed, Nov 14, 2018, 7:02 PM Bogdan-Andrei Iancu That's the whole idea - if the "use_domain" is on 0, OpenSIPS will > reference the users only by username. So try "opensipsctl add abc xyz" and > post what record you get into the subscriber table. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Bootcamp 2018 > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 11/14/2018 06:52 PM, Robert Dyck wrote: > > I do not have that parameter set and I do not use multiple domains. > > > > The problem was that after I corrected the error ( missing domain ), > opensips continued to look for abc@ rather than abc. I was looking for a > graceful way to correct the internal representation of the user name. > Restarting opensips is no problem on a small installation but it is less > than ideal. > > > > On Wednesday, November 14, 2018 6:11:52 AM PST Bogdan-Andrei Iancu wrote: > > Hi Robert, Do you have the "use_domain" parameter enabled in the auth_db > module ? > http://www.opensips.org/html/docs/modules/2.4.x/auth_db.html#param_use_domain > Regards, > > Bogdan-Andrei IancuOpenSIPS Founder and Developer > http://www.opensips-solutions.comOpenSIPS Bootcamp 2018 > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 11/07/2018 04:30 AM, Robert Dyck wrote: > > I have updated my small test bed from 2.3.2 to 2.4.2. I didn't bother to > back up the 'subscriber" table and it was wiped by the installation. No big > deal, it was tiny. > > So I added the users but made an error. > > opensipsctl add abc xyz -- I didn't specify the domain. The UAC would not > register. > > I corrected the user. > > opensipsctl rm abc, opensipsctl add abc at 192.168.1.2 xyz > > The UAC still cannot register. > > DBG:auth_db:get_ha1: no result for user 'abc@' > > Opensips is restarted and the UAC registers. > > Restaring a production machine is problematic. Is there a way to flush the > bad data which I assume has been cached? > > Some error checking in opensipsctl or the DB interface would be helpful. > > Thanks for your time and the product. > > Rob > > _______________________________________________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 pasandev at ymail.com Thu Nov 15 03:04:45 2018 From: pasandev at ymail.com (Pasan Meemaduma) Date: Thu, 15 Nov 2018 08:04:45 +0000 (UTC) Subject: [OpenSIPS-Users] check for NULL values In-Reply-To: <38f2caf6-04a3-27f3-9a33-08c1cbe0d2b2@opensips.org> References: <1238611305.714858.1541644790120.ref@mail.yahoo.com> <1238611305.714858.1541644790120@mail.yahoo.com> <1506752889.1453845.1542099402376@mail.yahoo.com> <38f2caf6-04a3-27f3-9a33-08c1cbe0d2b2@opensips.org> Message-ID: <1820042280.937452.1542269085544@mail.yahoo.com> Hi Bogdan, I'm not seeing any other related error msg prior to this. I'm calling this test in request route for a REGISTER request. Before the error pops up I could see the received msg successfully parse by opensips and can see value of To uri Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_msg: SIP Request: Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_msg:  method:  Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_msg:  uri:     Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_msg:  version: Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_headers: flags=2 Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_via_param: found param type 232, = ; state=16 Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_via: end of header reached, state=5 Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_headers: via found, flags=2 Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_headers: this is the first via Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:receive_msg: After parse_msg... Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:receive_msg: preparing to run routing scripts... Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_headers: flags=100 Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:_parse_to: end of header reached, state=10 Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:_parse_to: display={XXXXXXXXXX}, ruri={sip:XXXXXXXXXX at xx.xx.xx.x} Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:get_hdr_field: [48]; uri=[sip:XXXXXXXXXX at x.x.x.x] Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:get_hdr_field: to body [XXXXXXXXXX #015#012] Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:get_hdr_field: cseq : <79474> Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:maxfwd:is_maxfwd_present: value = 70 Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_to_param: tag=46474fbe5728f700o0 Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:_parse_to: end of header reached, state=29 Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:_parse_to: display={XXXXXXXXXX}, ruri={sip:XXXXXXXXXX at x.x.x.x} Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_headers: flags=200 Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:get_hdr_field: content_length=0 Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:get_hdr_field: found end of header and following are the line that I get before the error If I set log level to 6 Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_headers: flags=ffffffffffffffff Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_headers: flags=ffffffffffffffff Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_headers: flags=ffffffffffffffff Nov 15 18:19:19  /usr/sbin/opensips[21608]: ERROR:core:comp_scriptvar: cannot get left var value Nov 15 18:19:19  /usr/sbin/opensips[21608]: WARNING:core:do_action: error in expression at /etc/opensips/opensips.cfg:807 On Wednesday, 14 November 2018, 7:19:44 PM GMT+5:30, Bogdan-Andrei Iancu wrote: Hi Pasan, It should be fine if $tu would translate to NULL. The script handles this case. But I'm afraid you have something else there, like another deeper error that prevents the $tu variable to be evaluated. Do you see any other errors before the mentioned ones ? what is the type of route where you do the test ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/13/2018 10:56 AM, Pasan Meemaduma via Users wrote: Hey, Anyone have a suggestion for this? On Thursday, 8 November 2018, 8:09:50 AM GMT+5:30, Pasan Meemaduma wrote: Hi Guys, I have a check for NULL for $tu var in the script, But when the value is missing I'm getting the following error. ERROR:core:comp_scriptvar: cannot get left var value WARNING:core:do_action: error in expression at /etc/opensips/opensips.cfg:806 and line 806 contains following.     if ( $tu != NULL ) {         remove("location","$tu");     } any suggestion on how to test for NULL values without getting above error. I'm using opensips 2.3.5 _______________________________________________ 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 alexei.vasilyev at gmail.com Thu Nov 15 04:05:56 2018 From: alexei.vasilyev at gmail.com (vasilevalex) Date: Thu, 15 Nov 2018 02:05:56 -0700 (MST) Subject: [OpenSIPS-Users] uac_registrant clustering In-Reply-To: References: <1542190000.342851450@f436.i.mail.ru> Message-ID: <1542272756241-0.post@n2.nabble.com> Hi, all. In cluster environment it makes sense to have some way to enable/disable outbound registrations. To keep backup server silent. -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html From razvan at opensips.org Thu Nov 15 04:53:02 2018 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Thu, 15 Nov 2018 11:53:02 +0200 Subject: [OpenSIPS-Users] ACC module with JSON events In-Reply-To: References: <1519631511.3267.4.camel@gmail.com> Message-ID: <02183342-ab9f-9aad-924b-b0fbb35475f6@opensips.org> You're write, the issue is that the values are not unique! I have committed a fix now, but this should definitely be changed, as its prone to errors. I will try to rework the flags mechanism to something more flexible. Thanks for reporting this! Best regards, Răzvan On 11/14/18 8:08 PM, SamyGo wrote: > Possibly found the reason: > > The event_jsonrpc.c has this flag: JSONRPC_FLAG with same value "1 << > 26" same as event_route.c > > modules/event_route/event_route.h:#define SCRIPTROUTE_FLAG > (1 << 26) > modules/event_jsonrpc/event_jsonrpc.h:#define JSONRPC_FLAG > (1 << 26) > > While other event routes have unique value. Is this intentional? can I > change this flag value to something unused i.e 1 << 31  ? > > > On Wed, Nov 14, 2018 at 12:20 PM SamyGo > wrote: > > Howdy again, > > Thanks team for creating the event_jsonrpc module in 2.4 - I'm back > to this topic and trying to give this module a try however opensips > fail to start with following error: > > Nov 14 17:13:03 opsips /usr/local/sbin/opensips[18997]: > INFO:uac:mod_init: initializing... > Nov 14 17:13:03 opsips  /usr/local/sbin/opensips[18997]: > NOTICE:event_jsonrpc:mod_init: initializing module ... > Nov 14 17:13:03 opsips  /usr/local/sbin/opensips[18997]: > ERROR:core:register_event_mod: duplicate flag 4000000 > Nov 14 17:13:03 opsips  /usr/local/sbin/opensips[18997]: > ERROR:event_jsonrpc:mod_init: cannot register transport functions > for jsonrpc > Nov 14 17:13:03 opsips  /usr/local/sbin/opensips[18997]: > ERROR:core:init_mod: failed to initialize module event_jsonrpc > Nov 14 17:13:03 opsips  /usr/local/sbin/opensips[18997]: > ERROR:core:main: error while initializing modules > > I do have these two module loaded as well, (which I can't remove). > loadmodule "event_routing.so" > loadmodule "event_route.so" > > Any insight as to why these error show up? > > Big thanks, > Sammy. > > > > On Mon, Feb 26, 2018 at 12:07 PM SamyGo > wrote: > > Hi DanB, > I can certainly try CGRates, but...surely enough its an overkill > for simple CDRs. Anyways, my request is for an event_JSONRPC > which maybe very helpful in events outside of ACC. > > Best Regards, > Sammy > > > > > On Mon, Feb 26, 2018 at 6:13 AM, Khalil Khamlichi > > > wrote: > > Hi Dan, > > Can you show us how to do that with some step by step > tutorial, we had > a lot of trouble trying to figure out how to use CGRates, > with this > functionality of json export, we will at least use some of > CGRates > functionalities and hopefully slowly get familiar with it. > > Thanks in advance. > > On Mon, Feb 26, 2018 at 7:51 AM, DanB > wrote: > > Sammy, > > > > Another option on short term until the new feature will > be implemented > > in OpenSIPS would be to use CGRateS as CDR format > converter: receive > > CDRs from "cgrates" module in OpenSIPS and use online > export of CGRateS > > to further export the CDR in the JSON over http > (customizable fields). > > You don't need to configure much on CGRateS side in this > case since no > > billing needs to be involved. > > > > DanB > > > > > > > > > > > > _______________________________________________ > > 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 > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com From razvan at opensips.org Thu Nov 15 05:13:20 2018 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Thu, 15 Nov 2018 12:13:20 +0200 Subject: [OpenSIPS-Users] ACC module with JSON events In-Reply-To: <02183342-ab9f-9aad-924b-b0fbb35475f6@opensips.org> References: <1519631511.3267.4.camel@gmail.com> <02183342-ab9f-9aad-924b-b0fbb35475f6@opensips.org> Message-ID: <6bcba03c-1bba-40f0-6531-80ac67e4cb8b@opensips.org> Apologies, I forgot to give you credit in the commit, but I've done it in the issue I've opened: https://github.com/OpenSIPS/opensips/issues/1533 Best regards, Răzvan On 11/15/18 11:53 AM, Răzvan Crainea wrote: > You're write, the issue is that the values are not unique! I have > committed a fix now, but this should definitely be changed, as its prone > to errors. I will try to rework the flags mechanism to something more > flexible. > > Thanks for reporting this! > Best regards, > Răzvan > > On 11/14/18 8:08 PM, SamyGo wrote: >> Possibly found the reason: >> >> The event_jsonrpc.c has this flag: JSONRPC_FLAG with same value "1 << >> 26" same as event_route.c >> >> modules/event_route/event_route.h:#define SCRIPTROUTE_FLAG (1 << 26) >> modules/event_jsonrpc/event_jsonrpc.h:#define JSONRPC_FLAG (1 << 26) >> >> While other event routes have unique value. Is this intentional? can I >> change this flag value to something unused i.e 1 << 31  ? >> >> >> On Wed, Nov 14, 2018 at 12:20 PM SamyGo > > wrote: >> >>     Howdy again, >> >>     Thanks team for creating the event_jsonrpc module in 2.4 - I'm back >>     to this topic and trying to give this module a try however opensips >>     fail to start with following error: >> >>     Nov 14 17:13:03 opsips /usr/local/sbin/opensips[18997]: >>     INFO:uac:mod_init: initializing... >>     Nov 14 17:13:03 opsips  /usr/local/sbin/opensips[18997]: >>     NOTICE:event_jsonrpc:mod_init: initializing module ... >>     Nov 14 17:13:03 opsips  /usr/local/sbin/opensips[18997]: >>     ERROR:core:register_event_mod: duplicate flag 4000000 >>     Nov 14 17:13:03 opsips  /usr/local/sbin/opensips[18997]: >>     ERROR:event_jsonrpc:mod_init: cannot register transport functions >>     for jsonrpc >>     Nov 14 17:13:03 opsips  /usr/local/sbin/opensips[18997]: >>     ERROR:core:init_mod: failed to initialize module event_jsonrpc >>     Nov 14 17:13:03 opsips  /usr/local/sbin/opensips[18997]: >>     ERROR:core:main: error while initializing modules >> >>     I do have these two module loaded as well, (which I can't remove). >>     loadmodule "event_routing.so" >>     loadmodule "event_route.so" >> >>     Any insight as to why these error show up? >> >>     Big thanks, >>     Sammy. >> >> >> >>     On Mon, Feb 26, 2018 at 12:07 PM SamyGo >     > wrote: >> >>         Hi DanB, >>         I can certainly try CGRates, but...surely enough its an overkill >>         for simple CDRs. Anyways, my request is for an event_JSONRPC >>         which maybe very helpful in events outside of ACC. >> >>         Best Regards, >>         Sammy >> >> >> >> >>         On Mon, Feb 26, 2018 at 6:13 AM, Khalil Khamlichi >>         > >>         wrote: >> >>             Hi Dan, >> >>             Can you show us how to do that with some step by step >>             tutorial, we had >>             a lot of trouble trying to figure out how to use CGRates, >>             with this >>             functionality of json export, we will at least use some of >>             CGRates >>             functionalities and hopefully slowly get familiar with it. >> >>             Thanks in advance. >> >>             On Mon, Feb 26, 2018 at 7:51 AM, DanB >             > wrote: >>              > Sammy, >>              > >>              > Another option on short term until the new feature will >>             be implemented >>              > in OpenSIPS would be to use CGRateS as CDR format >>             converter: receive >>              > CDRs from "cgrates" module in OpenSIPS and use online >>             export of CGRateS >>              > to further export the CDR in the JSON over http >>             (customizable fields). >>              > You don't need to configure much on CGRateS side in this >>             case since no >>              > billing needs to be involved. >>              > >>              > DanB >>              > >>              > >>              > >>              > >>              > >>              > _______________________________________________ >>              > 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 >> > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com From bogdan at opensips.org Thu Nov 15 06:55:44 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 15 Nov 2018 13:55:44 +0200 Subject: [OpenSIPS-Users] check for NULL values In-Reply-To: <1820042280.937452.1542269085544@mail.yahoo.com> References: <1238611305.714858.1541644790120.ref@mail.yahoo.com> <1238611305.714858.1541644790120@mail.yahoo.com> <1506752889.1453845.1542099402376@mail.yahoo.com> <38f2caf6-04a3-27f3-9a33-08c1cbe0d2b2@opensips.org> <1820042280.937452.1542269085544@mail.yahoo.com> Message-ID: <28ec9806-c8b6-eb33-0795-1af3308ec1cd@opensips.org> Hi Pasan, Indeed, the logs show that the TO hdr is successfully parsed - are you 100% sure about the reported line ? maybe it is not the $tu related ? try to activate the script_trace() [http://www.opensips.org/Documentation/Script-CoreFunctions-2-4#toc42] function to see when exactly the error is generated. Maybe you can actually print $tu before, to see what you get. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/15/2018 10:04 AM, Pasan Meemaduma wrote: > Hi Bogdan, > > I'm not seeing any other related error msg prior to this. I'm calling > this test in request route for a REGISTER request. Before the error > pops up I could see the received msg successfully parse by opensips > and can see value of To uri > > Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:parse_msg: SIP > Request: > Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:parse_msg: > method: > Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:parse_msg: > uri: > Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:parse_msg: > version: > Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:parse_headers: > flags=2 > Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:parse_via_param: > found param type 232, = ; state=16 > Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:parse_via: end of > header reached, state=5 > Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:parse_headers: > via found, flags=2 > Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:parse_headers: > this is the first via > Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:receive_msg: > After parse_msg... > Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:receive_msg: > preparing to run routing scripts... > Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:parse_headers: > flags=100 > Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:_parse_to: end of > header reached, state=10 > Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:_parse_to: > display={XXXXXXXXXX}, ruri={sip:XXXXXXXXXX at xx.xx.xx.x} > Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:get_hdr_field: > [48]; uri=[sip:XXXXXXXXXX at x.x.x.x] > Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:get_hdr_field: to > body [XXXXXXXXXX #015#012] > Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:get_hdr_field: > cseq : <79474> > Nov 15 18:19:19 /usr/sbin/opensips[21608]: > DBG:maxfwd:is_maxfwd_present: value = 70 > Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:parse_to_param: > tag=46474fbe5728f700o0 > Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:_parse_to: end of > header reached, state=29 > Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:_parse_to: > display={XXXXXXXXXX}, ruri={sip:XXXXXXXXXX at x.x.x.x} > Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:parse_headers: > flags=200 > Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:get_hdr_field: > content_length=0 > Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:get_hdr_field: > found end of header > > and following are the line that I get before the error If I set log > level to 6 > > Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:parse_headers: > flags=ffffffffffffffff > Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:parse_headers: > flags=ffffffffffffffff > Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:parse_headers: > flags=ffffffffffffffff > Nov 15 18:19:19 /usr/sbin/opensips[21608]: ERROR:core:comp_scriptvar: > cannot get left var value > Nov 15 18:19:19 /usr/sbin/opensips[21608]: WARNING:core:do_action: > error in expression at /etc/opensips/opensips.cfg:807 > > > > > On Wednesday, 14 November 2018, 7:19:44 PM GMT+5:30, Bogdan-Andrei > Iancu wrote: > > > Hi Pasan, > > It should be fine if $tu would translate to NULL. The script handles > this case. But I'm afraid you have something else there, like another > deeper error that prevents the $tu variable to be evaluated. > > Do you see any other errors before the mentioned ones ? what is the > type of route where you do the test ? > > Regards, > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Bootcamp 2018 > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > On 11/13/2018 10:56 AM, Pasan Meemaduma via Users wrote: >> Hey, >> >> Anyone have a suggestion for this? >> >> On Thursday, 8 November 2018, 8:09:50 AM GMT+5:30, Pasan Meemaduma >> wrote: >> >> >> Hi Guys, >> >> I have a check for NULL for $tu var in the script, But when the value >> is missing I'm getting the following error. >> >> ERROR:core:comp_scriptvar: cannot get left var value >> >> WARNING:core:do_action: error in expression at >> /etc/opensips/opensips.cfg:806 >> >> and line 806 contains following. >> >> if ( $tu != NULL ) { >> remove("location","$tu"); >> } >> >> any suggestion on how to test for NULL values without getting above >> error. I'm using opensips 2.3.5 >> >> >> >> >> >> _______________________________________________ >> 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 Nov 15 07:24:47 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 15 Nov 2018 14:24:47 +0200 Subject: [OpenSIPS-Users] GRUU contact not found In-Reply-To: <1563251.d96nG9f5RN@blacky.mylan> References: <1891753.yvUAHoSbzm@blacky.mylan> <1563251.d96nG9f5RN@blacky.mylan> Message-ID: <35db8e62-07ad-88ba-18e5-038d475a474b@opensips.org> Hi Robert, So you experience the issues during the sequential requests in a call, and not during the registration. Outside the registration context, the gruu does not require special support. Could you email (off-list) a pacap showing your problem ? The pcap should be from OpenSIPS machine, showing in and out traffic, from the very beginning of the call. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/14/2018 07:09 PM, Robert Dyck wrote: > > I started with the sample residential script from some time back. GRUU > was enabled in the sample. When I started working with a UA that > registers a GRUU I noticed it was not receiving BYE when the other end > released the call. I have since experimented with checking for GRUU > while in dialog. That seems to work. The documentation doesn't mention > anything about modifying the script other than enabling or disabling > GRUU. Do you have any tips regarding GRUU in the script? Are there > corner cases I should be aware of? > > Rob > > On Wednesday, November 14, 2018 6:14:20 AM PST Bogdan-Andrei Iancu wrote: > > Hi Robert, According to docs, the gruu is by default off - > seehttp://www.opensips.org/html/docs/modules/2.3.x/registrar.html#idp5567984Regards, > > > Bogdan-Andrei IancuOpenSIPS Founder and Developer > http://www.opensips-solutions.comOpenSIPS Bootcamp 2018 > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 11/07/2018 10:09 PM, Robert Dyck wrote: > > My understanding is that GRUU processing in opensips is automatic, > provided it is not disabled. No further configuration or scripting is > required. Is that correct. > > A GRUU capable UA rergisters and receives public and temporary GR > identities. The UA establishes a dialog with another UA. The callee > ends the call. The caller does not recive the BYE. > > Caller : > > Request-Line: INVITE sip:7 at 192.168.1.2 SIP/2.0 > > Contact URI: > sip:4 at 192.168.1.2:5060;gr=urn:uuid:35dfa98a-2feb-482a-bde7-7568a86348b1 > > Callee: > > Status-Line: SIP/2.0 200 OK > > Caller: > > Request-Line: ACK sip:7 at 192.168.1.3:5062 SIP/2.0 > > Callee: > > Request-Line: BYE > sip:4 at 192.168.1.2:5060;gr=urn:uuid:35dfa98a-2feb-482a-bde7-7568a86348b1 > SIP/2.0 > > Proxy ( opensips @ 192.168.1.2 ) > > Status-Line: SIP/2.0 404 Not here > > Am I missing something? > > Should "opensipsctl ul show" show the GRUU? > > AOR:: 4 Contact:: sip:4 at 192.168.1.72:5062;transport=udpQ= > ContactID:: 3518589640418194 Expires:: > 3586 Callid:: OL1gvsViBJ Cseq:: 21 > User-agent:: LinphoneAndroid/4.0.1 (belle-sip/1.6.3) > State:: CS_NEW Flags:: 0 > Cflags:: Socket:: udp:192.168.1.2:5060 > Methods:: 4294967295 SIP_instance:: > > > _______________________________________________Users mailing > listUsers 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 Nov 15 08:50:08 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 15 Nov 2018 15:50:08 +0200 Subject: [OpenSIPS-Users] Flush bad user data from from running opensips In-Reply-To: <3727916.AF73741jUM@blacky.mylan> References: <11961787.XINqL1RPXL@blacky.mylan> <1668240.Sym1kIPFuu@blacky.mylan> <3727916.AF73741jUM@blacky.mylan> Message-ID: Hi Robert, So, the data in DB is ok, but the auth still fails - please paste the exact setting/modparams you have for the auth_db module along with the return code provided by the www_auth function - see http://www.opensips.org/html/docs/modules/2.4.x/auth_db.html#func_www_authorize Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/14/2018 08:03 PM, Robert Dyck wrote: > > I added "modparam("auth_db", "use_domain", 1)" but it doesn't make a > difference to the subscriber table. > > On Wednesday, November 14, 2018 9:36:34 AM PST Robert Dyck wrote: > > [root at slim opensips]# opensipsctl add abc xyz Updated subscriber, rows > affected: 1 new user 'abc' added > > 10:abc:localhost:xyz::6c7faf173d3b8e26d95e7f26dd0388d6:e091cc8c08b19e1d50ee3891d3f37153: > > [root at slim opensips]# opensipsctl rm abc Updated dbaliases, rows > affected: 0 [root at slim opensips]# opensipsctl add abc at 192.168.1.2 xyz > Updated subscriber, rows affected: 1 new user 'abc at 192.168.1.2' added > > 10:abc:192.168.1.2:xyz::9ce761c3a9f328510ea011bd5c9bd2c5:cc312796ec331326cd537f3a3ffad7b6: > > The difference being localhost vs 192.168.1.2 > > abc@ not found. > > On Wednesday, November 14, 2018 8:59:10 AM PST Bogdan-Andrei Iancu wrote: > > That's the whole idea - if the "use_domain" is on 0, OpenSIPS will > reference the users only by username. So try "opensipsctl add abc xyz" > and post what record you get into the subscriber table. Regards, > > Bogdan-Andrei IancuOpenSIPS Founder and Developer > http://www.opensips-solutions.comOpenSIPS Bootcamp 2018 > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 11/14/2018 06:52 PM, Robert Dyck wrote: > > I do not have that parameter set and I do not use multiple domains. > > The problem was that after I corrected the error ( missing domain ), > opensips continued to look for abc@ rather than abc. I was looking for > a graceful way to correct the internal representation of the user > name. Restarting opensips is no problem on a small installation but it > is less than ideal. > > On Wednesday, November 14, 2018 6:11:52 AM PST Bogdan-Andrei Iancu wrote: > > Hi Robert, Do you have the "use_domain" parameter enabled in the > auth_db module ? > http://www.opensips.org/html/docs/modules/2.4.x/auth_db.html#param_use_domainRegards, > > > Bogdan-Andrei IancuOpenSIPS Founder and Developer > http://www.opensips-solutions.comOpenSIPS Bootcamp 2018 > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 11/07/2018 04:30 AM, Robert Dyck wrote: > > I have updated my small test bed from 2.3.2 to 2.4.2. I didn't bother > to back up the 'subscriber" table and it was wiped by the > installation. No big deal, it was tiny. > > So I added the users but made an error. > > opensipsctl add abc xyz -- I didn't specify the domain. The UAC would > not register. > > I corrected the user. > > opensipsctl rm abc, opensipsctl add abc at 192.168.1.2 > xyz > > The UAC still cannot register. > > DBG:auth_db:get_ha1: no result for user 'abc@' > > Opensips is restarted and the UAC registers. > > Restaring a production machine is problematic. Is there a way to flush > the bad data which I assume has been cached? > > Some error checking in opensipsctl or the DB interface would be helpful. > > Thanks for your time and the product. > > Rob > > _______________________________________________Users mailing > listUsers 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 Nov 15 08:53:32 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 15 Nov 2018 15:53:32 +0200 Subject: [OpenSIPS-Users] uac_registrant clustering In-Reply-To: <1542272756241-0.post@n2.nabble.com> References: <1542190000.342851450@f436.i.mail.ru> <1542272756241-0.post@n2.nabble.com> Message-ID: <77c94162-6e18-d808-42b9-5eaedaa2f700@opensips.org> Hi Alexey, So basically you want to take advantage of the clustering layer in order to decide which server (active/backup) to register further ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/15/2018 11:05 AM, vasilevalex wrote: > Hi, all. > In cluster environment it makes sense to have some way to enable/disable > outbound registrations. To keep backup server silent. > > > > -- > Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From alexei.vasilyev at gmail.com Thu Nov 15 09:43:18 2018 From: alexei.vasilyev at gmail.com (vasilevalex) Date: Thu, 15 Nov 2018 07:43:18 -0700 (MST) Subject: [OpenSIPS-Users] uac_registrant clustering In-Reply-To: <77c94162-6e18-d808-42b9-5eaedaa2f700@opensips.org> References: <1542190000.342851450@f436.i.mail.ru> <1542272756241-0.post@n2.nabble.com> <77c94162-6e18-d808-42b9-5eaedaa2f700@opensips.org> Message-ID: <1542292998279-0.post@n2.nabble.com> Hi Bogdan, I hope, I didn't confuse you as original question was by another Alexey. And I also think that sharing outbound registrations between two active servers is something that has not much practical appliance. But when we have Active/Passive cluster with virtual IP (both servers running), it's better to have some way to disable outbound registrations on backup server and to enable them when switching Backup->Active. Perfect if it will be the same way, as Razvan wrote here: https://github.com/OpenSIPS/opensips/issues/1532 -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html From rob.dyck at telus.net Thu Nov 15 13:50:39 2018 From: rob.dyck at telus.net (Robert Dyck) Date: Thu, 15 Nov 2018 10:50:39 -0800 Subject: [OpenSIPS-Users] Flush bad user data from from running opensips References: <11961787.XINqL1RPXL@blacky.mylan> <3727916.AF73741jUM@blacky.mylan> Message-ID: <3127500.jTXMe0R1NV@blacky.mylan> So now I am confused. I am unable to reproduce the problem regardless of the "use_domain" setting. I even tried eliminating that line from the configuration ( that is how it was originally ) and the client can register whether I had created the user as abc xyz or abc at 192.168.1.2 xyz. A few days ago I reproduced it several times. the line "DBG:auth_db:get_ha1: no result for user 'abc@'" no longer appears in the debug. For your curiosity here are my auth settings. #### AUTHentication modules On Thursday, November 15, 2018 5:50:08 AM PST Bogdan-Andrei Iancu wrote: Hi Robert, So, the data in DB is ok, but the auth still fails - please paste the exact setting/modparams you have for the auth_db module along with the return code provided by the www_auth function - seehttp://www.opensips.org/html/docs/modules/2.4.x/ auth_db.html#func_www_authorize[1] Bogdan-Andrei IancuOpenSIPS Founder and Developer http://www.opensips-solutions.com[2]OpenSIPS Bootcamp 2018 http://opensips.org/training/ OpenSIPS_Bootcamp_2018/[3] On 11/14/2018 08:03 PM, Robert Dyck wrote: I added "modparam("auth_db", "use_domain", 1)" but it doesn't make a difference to the subscriber table. On Wednesday, November 14, 2018 9:36:34 AM PST Robert Dyck wrote: [root at slim opensips]# opensipsctl add abc xyz Updated subscriber, rows affected: 1 *new user 'abc' added* 10:abc:localhost:xyz:: 6c7faf173d3b8e26d95e7f26dd0388d6:e091cc8c08b19e1d50ee3891d3f37153: [root at slim opensips]# opensipsctl rm abc Updated dbaliases, rows affected: 0[root at slim opensips]# opensipsctl add abc at 192.168.1.2[4] xyz Updated subscriber, rows affected: 1 *new user '_abc at 192.168.1.2_' added* 10:abc:192.168.1.2:xyz:: 9ce761c3a9f328510ea011bd5c9bd2c5:cc312796ec331326cd537f3a3ffad7b6: The difference being localhost vs 192.168.1.2 abc@ not found. On Wednesday, November 14, 2018 8:59:10 AM PST Bogdan-Andrei Iancu wrote: That's the whole idea - if the "use_domain" is on 0, OpenSIPS will reference the users only by username. So try "opensipsctl add abc xyz" and post what record you get into the subscriber table. Regards, Bogdan-Andrei IancuOpenSIPS Founder and Developer http://www.opensips-solutions.com[2]OpenSIPS Bootcamp 2018 http://opensips.org/training/ OpenSIPS_Bootcamp_2018/[3] On 11/14/2018 06:52 PM, Robert Dyck wrote: I do not have that parameter set and I do not use multiple domains. The problem was that after I corrected the error ( missing domain ), opensips continued to look for abc@ rather than abc. I was looking for a graceful way to correct the internal representation of the user name. Restarting opensips is no problem on a small installation but it is less than ideal. On Wednesday, November 14, 2018 6:11:52 AM PST Bogdan-Andrei Iancu wrote: Hi Robert, Do you have the "use_domain" parameter enabled in the auth_db module ? http://www.opensips.org/html/docs/modules/2.4.x/auth_db.html#param_use_domain[5] Regards, Bogdan-Andrei IancuOpenSIPS Founder and Developer http://www.opensips-solutions.com[2]OpenSIPS Bootcamp 2018 http://opensips.org/training/ OpenSIPS_Bootcamp_2018/[3] On 11/07/2018 04:30 AM, Robert Dyck wrote: I have updated my small test bed from 2.3.2 to 2.4.2. I didn't bother to back up the 'subscriber" table and it was wiped by the installation. No big deal, it was tiny. So I added the users but made an error. opensipsctl add abc xyz -- I didn't specify the domain. The UAC would not register. I corrected the user. opensipsctl rm abc, opensipsctl add abc at 192.168.1.2[4] xyz The UAC still cannot register. DBG:auth_db:get_ha1: no result for user 'abc@' Opensips is restarted and the UAC registers. Restaring a production machine is problematic. Is there a way to flush the bad data which I assume has been cached? Some error checking in opensipsctl or the DB interface would be helpful. Thanks for your time and the product. Rob _______________________________________________Users mailing listUsers at lists.opensips.org[6]http://lists.opensips.org/cgi-bin/mailman/listinfo/users[7] -------- [1] http://www.opensips.org/html/docs/modules/2.4.x/auth_db.html#func_www_authorize [2] http://www.opensips-solutions.com [3] http://opensips.org/training/OpenSIPS_Bootcamp_2018/ [4] mailto:abc at 192.168.1.2 [5] http://www.opensips.org/html/docs/modules/2.4.x/auth_db.html#param_use_domain [6] mailto:Users at lists.opensips.org [7] http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From pasandev at ymail.com Fri Nov 16 00:20:28 2018 From: pasandev at ymail.com (Pasan Meemaduma) Date: Fri, 16 Nov 2018 05:20:28 +0000 (UTC) Subject: [OpenSIPS-Users] check for NULL values In-Reply-To: <28ec9806-c8b6-eb33-0795-1af3308ec1cd@opensips.org> References: <1238611305.714858.1541644790120.ref@mail.yahoo.com> <1238611305.714858.1541644790120@mail.yahoo.com> <1506752889.1453845.1542099402376@mail.yahoo.com> <38f2caf6-04a3-27f3-9a33-08c1cbe0d2b2@opensips.org> <1820042280.937452.1542269085544@mail.yahoo.com> <28ec9806-c8b6-eb33-0795-1af3308ec1cd@opensips.org> Message-ID: <2053587817.1497481.1542345628843@mail.yahoo.com> Hi Bogdan, I tried scrip_trace function and it also indicate same behavior, I can see the value of $tu but it still gives the error. Interesting thing is, it evaluates to true and invokes the remove() function as well. Nov 16 16:06:46  /usr/sbin/opensips[16582]: [Script Trace][/etc/opensips/opensips.cfg:814][reg-debug][core if] -> (REGISTER from Y.Y.Y.Y F=sip:XXX at x.x.x.x, T=sip:XXX at x.x.x.x ID=cd7921be-54514c80 at v.v.v.v#012) Nov 16 16:06:46  /usr/sbin/opensips[16582]: ERROR:core:comp_scriptvar: cannot get left var value Nov 16 16:06:46  /usr/sbin/opensips[16582]: WARNING:core:do_action: error in expression at /etc/opensips/opensips.cfg:814 Nov 16 16:06:52  /usr/sbin/opensips[16582]: [Script Trace][/etc/opensips/opensips.cfg:815][reg-debug][module remove] -> (REGISTER from Y.Y.Y.Y F=sip:XXX at x.x.x.x, T=sip:XXX at x.x.x.x ID=4afba829-fef90105 at v.v.v.v#012) /etc/opensips/opensips.cfg:814     if ( $tu != NULL ) {         remove("location","$tu");     } On Thursday, 15 November 2018, 5:25:54 PM GMT+5:30, Bogdan-Andrei Iancu wrote: Hi Pasan, Indeed, the logs show that the TO hdr is successfully parsed - are you 100% sure about the reported line ? maybe it is not the $tu related ? try to activate the script_trace() [http://www.opensips.org/Documentation/Script-CoreFunctions-2-4#toc42] function to see when exactly the error is generated. Maybe you can actually print $tu before, to see what you get. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/15/2018 10:04 AM, Pasan Meemaduma wrote: Hi Bogdan, I'm not seeing any other related error msg prior to this. I'm calling this test in request route for a REGISTER request. Before the error pops up I could see the received msg successfully parse by opensips and can see value of To uri Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_msg: SIP Request: Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_msg:  method:  Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_msg:  uri:     Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_msg:  version: Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_headers: flags=2 Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_via_param: found param type 232, = ; state=16 Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_via: end of header reached, state=5 Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_headers: via found, flags=2 Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_headers: this is the first via Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:receive_msg: After parse_msg... Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:receive_msg: preparing to run routing scripts... Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_headers: flags=100 Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:_parse_to: end of header reached, state=10 Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:_parse_to: display={XXXXXXXXXX}, ruri={sip:XXXXXXXXXX at xx.xx.xx.x} Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:get_hdr_field: [48]; uri=[sip:XXXXXXXXXX at x.x.x.x] Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:get_hdr_field: to body [XXXXXXXXXX #015#012] Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:get_hdr_field: cseq : <79474> Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:maxfwd:is_maxfwd_present: value = 70 Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_to_param: tag=46474fbe5728f700o0 Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:_parse_to: end of header reached, state=29 Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:_parse_to: display={XXXXXXXXXX}, ruri={sip:XXXXXXXXXX at x.x.x.x} Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_headers: flags=200 Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:get_hdr_field: content_length=0 Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:get_hdr_field: found end of header and following are the line that I get before the error If I set log level to 6 Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_headers: flags=ffffffffffffffff Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_headers: flags=ffffffffffffffff Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_headers: flags=ffffffffffffffff Nov 15 18:19:19  /usr/sbin/opensips[21608]: ERROR:core:comp_scriptvar: cannot get left var value Nov 15 18:19:19  /usr/sbin/opensips[21608]: WARNING:core:do_action: error in expression at /etc/opensips/opensips.cfg:807 On Wednesday, 14 November 2018, 7:19:44 PM GMT+5:30, Bogdan-Andrei Iancu wrote: Hi Pasan, It should be fine if $tu would translate to NULL. The script handles this case. But I'm afraid you have something else there, like another deeper error that prevents the $tu variable to be evaluated. Do you see any other errors before the mentioned ones ? what is the type of route where you do the test ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/13/2018 10:56 AM, Pasan Meemaduma via Users wrote: Hey, Anyone have a suggestion for this? On Thursday, 8 November 2018, 8:09:50 AM GMT+5:30, Pasan Meemaduma wrote: Hi Guys, I have a check for NULL for $tu var in the script, But when the value is missing I'm getting the following error. ERROR:core:comp_scriptvar: cannot get left var value WARNING:core:do_action: error in expression at /etc/opensips/opensips.cfg:806 and line 806 contains following.     if ( $tu != NULL ) {         remove("location","$tu");     } any suggestion on how to test for NULL values without getting above error. I'm using opensips 2.3.5 _______________________________________________ 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 julian.santer at rolmail.net Fri Nov 16 08:06:16 2018 From: julian.santer at rolmail.net (Julian Santer) Date: Fri, 16 Nov 2018 14:06:16 +0100 Subject: [OpenSIPS-Users] Permission doesn't match In-Reply-To: <7a999860-c3b2-246a-5d6a-738e05db5678@opensips.org> References: <7a999860-c3b2-246a-5d6a-738e05db5678@opensips.org> Message-ID: Hi Bogdan, if I exec "subnet_dump", I receive the records in grp "52", but the records in grp "54" are missing. Kind regards, Julian Santer Am 14.11.18 um 15:20 schrieb Bogdan-Andrei Iancu: > Hi Julian, > > If you do a "subnet_dump" (see http://www.opensips.org/html/docs/modules/2.4.x/permissions.html#mi_subnet_dump), do you see both records ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer >   http://www.opensips-solutions.com > OpenSIPS Bootcamp 2018 >   http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 11/08/2018 06:52 PM, Julian Santer wrote: >> Hi guys, >> >> I have some question to the permission module. We are using Opensips 2.2.6. >> The permissions are load from the address table located in a MySQL DB. >> >> My config looks like: >> >> ... >> else if (check_address("52", "$si", "$sp", "$proto", "$avp(ctx)", "$ua")) >> { >>     xlog("L_INFO", "Entered here due permission 52 - LF_BASE"); >> } >> else if (check_address("54", "$si", "$sp", "$proto", "$avp(ctx)", "$ua")) >> { >>     xlog("L_INFO", "Entered here due permission 54 - LF_BASE"); >> } >> ... >> >> address table: >> id    grp    ip             mask    port    proto    pattern context_info >> 41    52     192.168.1.0    24      0       any      AVM*.06.* test >> 648   54     192.168.1.0    24      0       any      AVM*.07.* test >> >> This line is matching: >> Nov  8 17:10:59 M=REGISTER RURI=sip:test.com F=sip:abc at test.com T=sip:abc at test.com SRC=192.168.1.46:5060 UAC=AVM FRITZ!Box Fon WLAN 7390 84.06.85 >> (Sep 10 2018) ID=9A0B1C90057A9126 at 192.168.1.46 B= >> Nov  8 17:10:59 Entered here due permission 52 - M=REGISTER RURI=sip:test.com F=sip:abc at test.com T=sip:abc at test.com SRC=192.168.1.46:5060 UAC=AVM >> FRITZ!Box Fon WLAN 7390 84.06.85 (Sep 10 2018) ID=9A0B1C90057A9126 at 192.168.146 B= >> >> But this line is not matching: >> Nov  8 17:35:19 M=REGISTER RURI=sip:test.com F=sip:def at test.com T=sip:def at test.com SRC=192.168.1.215:5060 UAC=AVM FRITZ!Box 7490 113.07.01 (Sep 11 >> 2018) ID=5DC1E7DC326043BA at 192.168.1.215 B= >> >> I already did a opensipsctl address reload and several times restarted the whole opensips service. >> Have you maybe some hint for me? >> >> Kind regards, >> Julian Santer >> >> >> _______________________________________________ >> 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 bogdan at opensips.org Fri Nov 16 11:12:12 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 16 Nov 2018 18:12:12 +0200 Subject: [OpenSIPS-Users] Permission doesn't match In-Reply-To: References: <7a999860-c3b2-246a-5d6a-738e05db5678@opensips.org> Message-ID: <6adb2c03-b2aa-f558-9395-15723d0ec039@opensips.org> Hi Julian, When you perform a "address_reload", do you see any errors or warnings in the opensips logs ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/16/2018 03:06 PM, Julian Santer wrote: > Hi Bogdan, > > if I exec "subnet_dump", I receive the records in grp "52", but the > records in grp "54" are missing. > > Kind regards, > Julian Santer > > Am 14.11.18 um 15:20 schrieb Bogdan-Andrei Iancu: >> Hi Julian, >> >> If you do a "subnet_dump" (see >> http://www.opensips.org/html/docs/modules/2.4.x/permissions.html#mi_subnet_dump), >> do you see both records ? >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> OpenSIPS Bootcamp 2018 >> http://opensips.org/training/OpenSIPS_Bootcamp_2018/ >> >> On 11/08/2018 06:52 PM, Julian Santer wrote: >>> Hi guys, >>> >>> I have some question to the permission module. We are using Opensips >>> 2.2.6. >>> The permissions are load from the address table located in a MySQL DB. >>> >>> My config looks like: >>> >>> ... >>> else if (check_address("52", "$si", "$sp", "$proto", "$avp(ctx)", >>> "$ua")) >>> { >>> xlog("L_INFO", "Entered here due permission 52 - LF_BASE"); >>> } >>> else if (check_address("54", "$si", "$sp", "$proto", "$avp(ctx)", >>> "$ua")) >>> { >>> xlog("L_INFO", "Entered here due permission 54 - LF_BASE"); >>> } >>> ... >>> >>> address table: >>> id grp ip mask port proto pattern >>> context_info >>> 41 52 192.168.1.0 24 0 any AVM*.06.* test >>> 648 54 192.168.1.0 24 0 any AVM*.07.* test >>> >>> This line is matching: >>> Nov 8 17:10:59 M=REGISTER RURI=sip:test.com F=sip:abc at test.com >>> T=sip:abc at test.com SRC=192.168.1.46:5060 UAC=AVM FRITZ!Box Fon WLAN >>> 7390 84.06.85 (Sep 10 2018) ID=9A0B1C90057A9126 at 192.168.1.46 B= >>> Nov 8 17:10:59 Entered here due permission 52 - M=REGISTER >>> RURI=sip:test.com F=sip:abc at test.com T=sip:abc at test.com >>> SRC=192.168.1.46:5060 UAC=AVM FRITZ!Box Fon WLAN 7390 84.06.85 (Sep >>> 10 2018) ID=9A0B1C90057A9126 at 192.168.146 B= >>> >>> But this line is not matching: >>> Nov 8 17:35:19 M=REGISTER RURI=sip:test.com F=sip:def at test.com >>> T=sip:def at test.com SRC=192.168.1.215:5060 UAC=AVM FRITZ!Box 7490 >>> 113.07.01 (Sep 11 2018) ID=5DC1E7DC326043BA at 192.168.1.215 B= >>> >>> I already did a opensipsctl address reload and several times >>> restarted the whole opensips service. >>> Have you maybe some hint for me? >>> >>> Kind regards, >>> Julian Santer >>> >>> >>> _______________________________________________ >>> 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 From trance_maniak at o2.pl Wed Nov 14 05:39:28 2018 From: trance_maniak at o2.pl (maatohewetbi) Date: Wed, 14 Nov 2018 03:39:28 -0700 (MST) Subject: [OpenSIPS-Users] Opensips 2.1.2 - can't pass var in rewritehostport() In-Reply-To: References: <1541664230763-0.post@n2.nabble.com> Message-ID: <1542191968913-0.post@n2.nabble.com> How can I use it, like that? $rd=$hdr(X-IP-Header) rewritehostport($rd:5060) -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html From julian.santer at rolmail.net Fri Nov 16 11:30:14 2018 From: julian.santer at rolmail.net (Julian Santer) Date: Fri, 16 Nov 2018 17:30:14 +0100 Subject: [OpenSIPS-Users] Permission doesn't match In-Reply-To: <6adb2c03-b2aa-f558-9395-15723d0ec039@opensips.org> References: <7a999860-c3b2-246a-5d6a-738e05db5678@opensips.org> <6adb2c03-b2aa-f558-9395-15723d0ec039@opensips.org> Message-ID: <7bb3ea33-20bf-57c9-0a20-3f4de0e2116e@rolmail.net> Hi Bogdan, yes we got the following critical errors: CRITICAL:permissions:subnet_table_insert: subnet table is full How many records could be stored and is there a way to increase the limit? Kind regards, Julian Santer Am 16.11.18 um 17:12 schrieb Bogdan-Andrei Iancu: > Hi Julian, > > When you perform a "address_reload", do you see any errors or warnings in the opensips logs ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer >   http://www.opensips-solutions.com > OpenSIPS Bootcamp 2018 >   http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 11/16/2018 03:06 PM, Julian Santer wrote: >> Hi Bogdan, >> >> if I exec "subnet_dump", I receive the records in grp "52", but the records in grp "54" are missing. >> >> Kind regards, >> Julian Santer >> >> Am 14.11.18 um 15:20 schrieb Bogdan-Andrei Iancu: >>> Hi Julian, >>> >>> If you do a "subnet_dump" (see http://www.opensips.org/html/docs/modules/2.4.x/permissions.html#mi_subnet_dump), do you see both records ? >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>>    http://www.opensips-solutions.com >>> OpenSIPS Bootcamp 2018 >>>    http://opensips.org/training/OpenSIPS_Bootcamp_2018/ >>> >>> On 11/08/2018 06:52 PM, Julian Santer wrote: >>>> Hi guys, >>>> >>>> I have some question to the permission module. We are using Opensips 2.2.6. >>>> The permissions are load from the address table located in a MySQL DB. >>>> >>>> My config looks like: >>>> >>>> ... >>>> else if (check_address("52", "$si", "$sp", "$proto", "$avp(ctx)", "$ua")) >>>> { >>>>     xlog("L_INFO", "Entered here due permission 52 - LF_BASE"); >>>> } >>>> else if (check_address("54", "$si", "$sp", "$proto", "$avp(ctx)", "$ua")) >>>> { >>>>     xlog("L_INFO", "Entered here due permission 54 - LF_BASE"); >>>> } >>>> ... >>>> >>>> address table: >>>> id    grp    ip             mask    port    proto    pattern context_info >>>> 41    52     192.168.1.0    24      0       any AVM*.06.* test >>>> 648   54     192.168.1.0    24      0       any AVM*.07.* test >>>> >>>> This line is matching: >>>> Nov  8 17:10:59 M=REGISTER RURI=sip:test.com F=sip:abc at test.com T=sip:abc at test.com SRC=192.168.1.46:5060 UAC=AVM FRITZ!Box Fon WLAN 7390 84.06.85 >>>> (Sep 10 2018) ID=9A0B1C90057A9126 at 192.168.1.46 B= >>>> Nov  8 17:10:59 Entered here due permission 52 - M=REGISTER RURI=sip:test.com F=sip:abc at test.com T=sip:abc at test.com SRC=192.168.1.46:5060 UAC=AVM >>>> FRITZ!Box Fon WLAN 7390 84.06.85 (Sep 10 2018) ID=9A0B1C90057A9126 at 192.168.146 B= >>>> >>>> But this line is not matching: >>>> Nov  8 17:35:19 M=REGISTER RURI=sip:test.com F=sip:def at test.com T=sip:def at test.com SRC=192.168.1.215:5060 UAC=AVM FRITZ!Box 7490 113.07.01 (Sep >>>> 11 2018) ID=5DC1E7DC326043BA at 192.168.1.215 B= >>>> >>>> I already did a opensipsctl address reload and several times restarted the whole opensips service. >>>> Have you maybe some hint for me? >>>> >>>> Kind regards, >>>> Julian Santer >>>> >>>> >>>> _______________________________________________ >>>> 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 > From vitalik.voip at gmail.com Fri Nov 16 12:07:01 2018 From: vitalik.voip at gmail.com (Vitalii Aleksandrov) Date: Fri, 16 Nov 2018 19:07:01 +0200 Subject: [OpenSIPS-Users] EBR and wait_for_event() timeout In-Reply-To: References: Message-ID: Thanks for the information. Have one more  related question. What If I call somewhere, opensips calls wait_for_event() and before the event happens or async timeout (will create a bug report) fired I CANCEL the call. Since async() keeps some context in transaction structure and this transaction is already canceled should I expect that async() task is also canceled and will never call a callback route? Or should I always check  t_was_cancelled() in the beginning of a callback route? > Hi Vitalii, > > For the wait_for_event(), the timeout seems to have no effect, the > waiting being for ever :-| . The transaction has no timeout as you > didn;t sent out any branch yet (the transaction timeout is for waiting > on replies). > > Could you please open bug report on the opensips tracker on github ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer >   http://www.opensips-solutions.com > OpenSIPS Bootcamp 2018 >   http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 11/14/2018 03:17 PM, Vitalii Aleksandrov wrote: >> Hi, >> >> event_routing module provides the great async function >> wait_for_event().  If script subscribes for a event and received it >> it calls some "resume_route". >> What I can't understand is what happens with a transaction if >> wait_for_event() never catches an event and reaches its timeout. >> Is the any way to continue script execution from the place where >> "wait_for_event() was called or to execute some "timeout_route" to >> handle transaction properly? >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > From bogdan at opensips.org Fri Nov 16 12:14:18 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 16 Nov 2018 19:14:18 +0200 Subject: [OpenSIPS-Users] Permission doesn't match In-Reply-To: <7bb3ea33-20bf-57c9-0a20-3f4de0e2116e@rolmail.net> References: <7a999860-c3b2-246a-5d6a-738e05db5678@opensips.org> <6adb2c03-b2aa-f558-9395-15723d0ec039@opensips.org> <7bb3ea33-20bf-57c9-0a20-3f4de0e2116e@rolmail.net> Message-ID: <2eae64f3-5cff-eb9a-43bd-20a94b7749fb@opensips.org> Hi Julian, And you have *only* those 2 subnet records in the address table ?? You should get that error only if you use more than 128 subnet records. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/16/2018 06:30 PM, Julian Santer wrote: > Hi Bogdan, > > yes we got the following critical errors: > CRITICAL:permissions:subnet_table_insert: subnet table is full > > How many records could be stored and is there a way to increase the > limit? > > Kind regards, > Julian Santer > > Am 16.11.18 um 17:12 schrieb Bogdan-Andrei Iancu: >> Hi Julian, >> >> When you perform a "address_reload", do you see any errors or >> warnings in the opensips logs ? >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> OpenSIPS Bootcamp 2018 >> http://opensips.org/training/OpenSIPS_Bootcamp_2018/ >> >> On 11/16/2018 03:06 PM, Julian Santer wrote: >>> Hi Bogdan, >>> >>> if I exec "subnet_dump", I receive the records in grp "52", but the >>> records in grp "54" are missing. >>> >>> Kind regards, >>> Julian Santer >>> >>> Am 14.11.18 um 15:20 schrieb Bogdan-Andrei Iancu: >>>> Hi Julian, >>>> >>>> If you do a "subnet_dump" (see >>>> http://www.opensips.org/html/docs/modules/2.4.x/permissions.html#mi_subnet_dump), >>>> do you see both records ? >>>> >>>> Regards, >>>> >>>> Bogdan-Andrei Iancu >>>> >>>> OpenSIPS Founder and Developer >>>> http://www.opensips-solutions.com >>>> OpenSIPS Bootcamp 2018 >>>> http://opensips.org/training/OpenSIPS_Bootcamp_2018/ >>>> >>>> On 11/08/2018 06:52 PM, Julian Santer wrote: >>>>> Hi guys, >>>>> >>>>> I have some question to the permission module. We are using >>>>> Opensips 2.2.6. >>>>> The permissions are load from the address table located in a MySQL >>>>> DB. >>>>> >>>>> My config looks like: >>>>> >>>>> ... >>>>> else if (check_address("52", "$si", "$sp", "$proto", "$avp(ctx)", >>>>> "$ua")) >>>>> { >>>>> xlog("L_INFO", "Entered here due permission 52 - LF_BASE"); >>>>> } >>>>> else if (check_address("54", "$si", "$sp", "$proto", "$avp(ctx)", >>>>> "$ua")) >>>>> { >>>>> xlog("L_INFO", "Entered here due permission 54 - LF_BASE"); >>>>> } >>>>> ... >>>>> >>>>> address table: >>>>> id grp ip mask port proto pattern >>>>> context_info >>>>> 41 52 192.168.1.0 24 0 any AVM*.06.* test >>>>> 648 54 192.168.1.0 24 0 any AVM*.07.* test >>>>> >>>>> This line is matching: >>>>> Nov 8 17:10:59 M=REGISTER RURI=sip:test.com F=sip:abc at test.com >>>>> T=sip:abc at test.com SRC=192.168.1.46:5060 UAC=AVM FRITZ!Box Fon >>>>> WLAN 7390 84.06.85 (Sep 10 2018) ID=9A0B1C90057A9126 at 192.168.1.46 >>>>> B= >>>>> Nov 8 17:10:59 Entered here due permission 52 - M=REGISTER >>>>> RURI=sip:test.com F=sip:abc at test.com T=sip:abc at test.com >>>>> SRC=192.168.1.46:5060 UAC=AVM FRITZ!Box Fon WLAN 7390 84.06.85 >>>>> (Sep 10 2018) ID=9A0B1C90057A9126 at 192.168.146 B= >>>>> >>>>> But this line is not matching: >>>>> Nov 8 17:35:19 M=REGISTER RURI=sip:test.com F=sip:def at test.com >>>>> T=sip:def at test.com SRC=192.168.1.215:5060 UAC=AVM FRITZ!Box 7490 >>>>> 113.07.01 (Sep 11 2018) ID=5DC1E7DC326043BA at 192.168.1.215 B= >>>>> >>>>> I already did a opensipsctl address reload and several times >>>>> restarted the whole opensips service. >>>>> Have you maybe some hint for me? From bogdan at opensips.org Fri Nov 16 12:25:36 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 16 Nov 2018 19:25:36 +0200 Subject: [OpenSIPS-Users] Forking sip MESSAGE to multiple endpoints In-Reply-To: References: Message-ID: <8ef10dd0-409b-d839-c471-26df583bb431@opensips.org> Hi Aron, The standard SIP forking (as per RFC3261) says only one destination should get the reply. Nevertheless, there is the RFC3841 defining "Content-Disposition: no-cancel" to prevent the proxy to send Cancel upon first 200 OK. We added support for it in OpenSIPS, starting 2.4 - see flag 0x10 - http://www.opensips.org/html/docs/modules/2.4.x/tm.html#func_t_relay but it works only for INVITEs :D Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/14/2018 07:38 PM, Podrigal, Aron wrote: > Hi. > > I want to fork a MESSAGE request to multiple endpoints and handle the > delivery status for each on reply. > > currently I append_branch() for each additional destination. But the > problem is, that if any branch received a 200 reply, any other branch > which did not relay the request yet (ie socket connection wasn't > established) or TM timer is triggered, then those branches are > canceled without triggering onreply route or failure route. > > How could I implement it, so that I can handle the timeout or failure > for each endpoint? > > Maybe t_replicate? Or is there an event raised for a canceled branch > which I can subscribe to? > > > Thanks > > > _______________________________________________ > 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 aronp at guaranteedplus.com Fri Nov 16 12:48:13 2018 From: aronp at guaranteedplus.com (Podrigal, Aron) Date: Fri, 16 Nov 2018 12:48:13 -0500 Subject: [OpenSIPS-Users] Forking sip MESSAGE to multiple endpoints In-Reply-To: <8ef10dd0-409b-d839-c471-26df583bb431@opensips.org> References: <8ef10dd0-409b-d839-c471-26df583bb431@opensips.org> Message-ID: Thank you Bogdan. The question is if we can add some flag, to always trigger onreply_route for each branch, so that one can do some cleanup / update database based on the reply for each branch. On Fri, Nov 16, 2018, 12:25 PM Bogdan-Andrei Iancu Hi Aron, > > The standard SIP forking (as per RFC3261) says only one destination should > get the reply. Nevertheless, there is the RFC3841 defining > "Content-Disposition: no-cancel" to prevent the proxy to send Cancel upon > first 200 OK. > We added support for it in OpenSIPS, starting 2.4 - see flag 0x10 - > http://www.opensips.org/html/docs/modules/2.4.x/tm.html#func_t_relay > but it works only for INVITEs :D > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Bootcamp 2018 > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 11/14/2018 07:38 PM, Podrigal, Aron wrote: > > Hi. > > I want to fork a MESSAGE request to multiple endpoints and handle the > delivery status for each on reply. > > currently I append_branch() for each additional destination. But the > problem is, that if any branch received a 200 reply, any other branch which > did not relay the request yet (ie socket connection wasn't established) or > TM timer is triggered, then those branches are canceled without triggering > onreply route or failure route. > > How could I implement it, so that I can handle the timeout or failure for > each endpoint? > > Maybe t_replicate? Or is there an event raised for a canceled branch which > I can subscribe to? > > > Thanks > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From jehanzaib.kiani at gmail.com Sun Nov 18 00:50:32 2018 From: jehanzaib.kiani at gmail.com (J E H A N Z A I B) Date: Sun, 18 Nov 2018 18:50:32 +1300 Subject: [OpenSIPS-Users] redis cache sharing for dialog storage? In-Reply-To: <2022c9e0-ce7a-e06a-419b-76d69854b015@opensips.org> References: <2022c9e0-ce7a-e06a-419b-76d69854b015@opensips.org> Message-ID: Thank you Bogdan. Do I have to use /s marker for setting as well ? set_dlg_profile("test/s" ? what about get_profile_size("test/s On Thu, Nov 15, 2018 at 3:04 AM Bogdan-Andrei Iancu wrote: > Hi Jehanzaib, > > Again, use the /s marker - see > http://www.opensips.org/html/docs/modules/2.4.x/dialog.html#param_profiles_with_value > . > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Bootcamp 2018 > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 11/06/2018 01:11 AM, J E H A N Z A I B wrote: > > Hi there, > > I am not sure why the dialog stats will not be shared. If all records are > going to redis then I have a redis cluster which synchs the cache. > When I fetch the profile size it should be same (if it is being fetched > from the redis) across all the opensips node. I am bit sceptical to upgrade > without knowing what's happening. > > > On Tue, Nov 6, 2018 at 4:53 AM SamyGo wrote: > >> I have a strong feeling that you're using an old version of opensips to >> expect it to share dialog states/profiles. I think you'll need to use newer >> opensips 2.4+ having dialog sharing capability using proto_bin and >> clusterer module: >> >> http://www.opensips.org/html/docs/modules/2.4.x/dialog.html#param_profile_replication_cluster >> >> >> On Mon, Nov 5, 2018 at 5:30 AM J E H A N Z A I B < >> jehanzaib.kiani at gmail.com> wrote: >> >>> Hi team, >>> >>> I used redis cache for dialog storage. I have 2 different servers both >>> are sharing the same redis. Is the profile size shared in this case? >>> >>> here is the dialog config. >>> >>> loadmodule "dialog.so" >>> modparam("dialog", "enable_stats", 1) >>> modparam("dialog", "cachedb_url", "redis:mysip://mysipx.xx.xx:xxxx/") >>> >>> This is how I check my profile size. >>> create_dialog(); >>> set_dlg_profile("myuniqprof","$avp(myprofile_id)"); >>> get_profile_size("myuniqprof","$avp(myprofile_id >>> )","$var(current_profile_size)"); >>> >>> Please note I am using version: opensips 1.11.3-notls >>> >>> >>> -- >>> Regards, >>> Jehanzaib >>> _______________________________________________ >>> 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 >> > > > -- > Regards, > Jehanzaib > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -- Regards, Jehanzaib -------------- next part -------------- An HTML attachment was scrubbed... URL: From dhruv.shah at ecosmob.com Mon Nov 19 01:28:48 2018 From: dhruv.shah at ecosmob.com (Dhruv Shah) Date: Mon, 19 Nov 2018 11:58:48 +0530 Subject: [OpenSIPS-Users] TLS->UDP Re-Invite Issue Message-ID: Hello, I am using opensips-2.2, and using TLS for calling. From opensips using dispatcher call is forwarded on freeswitch which is on UDP, then freeswitch forwards it to external gateway. It works perfectly, but when freeswitch generates reinvite, it is send to some another port i.e port is different from which the ACK is received, due to this call gets hangup due to request timeout. Please suggest solution to solve this issue. -- *Thanks & Regards* *Dhruv Shah* *Jr. Software Developer* [image: Ecosmob Technologies Pvt. Ltd.] *Ecosmob Technologies Pvt. Ltd. * https://www.ecosmob.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From goup2010 at gmail.com Mon Nov 19 03:55:04 2018 From: goup2010 at gmail.com (Dragomir Haralambiev) Date: Mon, 19 Nov 2018 10:55:04 +0200 Subject: [OpenSIPS-Users] system metrics Message-ID: Hello, For system metrics i need to get number active dialogs. How to do that easy. Best regards, Dragomir -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Nov 19 06:49:45 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 19 Nov 2018 13:49:45 +0200 Subject: [OpenSIPS-Users] Forking sip MESSAGE to multiple endpoints In-Reply-To: References: <8ef10dd0-409b-d839-c471-26df583bb431@opensips.org> Message-ID: <12bed5c4-0ea4-8c7f-7778-4bfc2e1b0754@opensips.org> Hi Aron, Right now, the onreply_route is triggered all the time for all replies (1xx, 2xx, or higher) received on any of the branches. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/16/2018 07:48 PM, Podrigal, Aron wrote: > Thank you Bogdan. > The question is if we can add some flag, to always trigger > onreply_route for each branch, so that one can do some cleanup / > update database based on the reply for each branch. > > On Fri, Nov 16, 2018, 12:25 PM Bogdan-Andrei Iancu > wrote: > > Hi Aron, > > The standard SIP forking (as per RFC3261) says only one > destination should get the reply. Nevertheless, there is the > RFC3841 defining "Content-Disposition: no-cancel" to prevent the > proxy to send Cancel upon first 200 OK. > We added support for it in OpenSIPS, starting 2.4 - see flag 0x10 > - http://www.opensips.org/html/docs/modules/2.4.x/tm.html#func_t_relay > but it works only for INVITEs :D > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Bootcamp 2018 > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 11/14/2018 07:38 PM, Podrigal, Aron wrote: >> Hi. >> >> I want to fork a MESSAGE request to multiple endpoints and handle >> the delivery status for each on reply. >> >> currently I append_branch() for each additional destination. But >> the problem is, that if any branch received a 200 reply, any >> other branch which did not relay the request yet (ie socket >> connection wasn't established) or TM timer is triggered, then >> those branches are canceled without triggering onreply route or >> failure route. >> >> How could I implement it, so that I can handle the timeout or >> failure for each endpoint? >> >> Maybe t_replicate? Or is there an event raised for a canceled >> branch which I can subscribe to? >> >> >> Thanks >> >> >> _______________________________________________ >> 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 Nov 19 06:51:48 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 19 Nov 2018 13:51:48 +0200 Subject: [OpenSIPS-Users] redis cache sharing for dialog storage? In-Reply-To: References: <2022c9e0-ce7a-e06a-419b-76d69854b015@opensips.org> Message-ID: <7da66ce6-51cb-cfb3-1f0b-b0972ff68aee@opensips.org> Yes, that is correct - you need to use the /s marker in all the places where you mention the profile. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/18/2018 07:50 AM, J E H A N Z A I B wrote: > Thank you Bogdan. > Do I have to use /s marker for setting as well ? > > set_dlg_profile("test/s" ? > > what about get_profile_size("test/s > > > On Thu, Nov 15, 2018 at 3:04 AM Bogdan-Andrei Iancu > > wrote: > > Hi Jehanzaib, > > Again, use the /s marker - see > http://www.opensips.org/html/docs/modules/2.4.x/dialog.html#param_profiles_with_value > . > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Bootcamp 2018 > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 11/06/2018 01:11 AM, J E H A N Z A I B wrote: >> Hi there, >> >> I am not sure why the dialog stats will not be shared. If all >> records are going to redis then I have a redis cluster which >> synchs the cache. >> When I fetch the profile size it should be same (if it is being >> fetched from the redis) across all the opensips node. I am bit >> sceptical to upgrade without knowing what's happening. >> >> >> On Tue, Nov 6, 2018 at 4:53 AM SamyGo > > wrote: >> >> I have a strong feeling that you're using an old version of >> opensips to expect it to share dialog states/profiles. I >> think you'll need to use newer opensips 2.4+ having dialog >> sharing capability using proto_bin and clusterer module: >> http://www.opensips.org/html/docs/modules/2.4.x/dialog.html#param_profile_replication_cluster >> >> >> On Mon, Nov 5, 2018 at 5:30 AM J E H A N Z A I B >> > > wrote: >> >> Hi team, >> >> I used redis cache for dialog storage. I have 2 different >> servers both are sharing the same redis. Is the profile >> size shared in this case? >> >> here is the dialog config. >> >> loadmodule "dialog.so" >> modparam("dialog", "enable_stats", 1) >> modparam("dialog", "cachedb_url", >> "redis:mysip://mysipx.xx.xx:xxxx/") >> >> This is how I check my profile size. >> create_dialog(); >> set_dlg_profile("myuniqprof","$avp(myprofile_id)"); >> get_profile_size("myuniqprof","$avp(myprofile_id)","$var(current_profile_size)"); >> >> Please note I am using version: opensips 1.11.3-notls >> >> >> -- >> Regards, >> Jehanzaib >> _______________________________________________ >> 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 >> >> >> >> -- >> Regards, >> Jehanzaib >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > Regards, > Jehanzaib -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Nov 19 07:12:27 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 19 Nov 2018 14:12:27 +0200 Subject: [OpenSIPS-Users] system metrics In-Reply-To: References: Message-ID: <56ba4366-f74e-a93d-5bf3-bec2f6589613@opensips.org> Hi Dragomir, See the statistics provided by the dialog module: http://www.opensips.org/html/docs/modules/2.4.x/dialog.html#exported_statistics Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/19/2018 10:55 AM, Dragomir Haralambiev wrote: > Hello, > > For system metrics i need to get number active dialogs. > How to do that easy. > > 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 john.quick at smartvox.co.uk Mon Nov 19 08:59:36 2018 From: john.quick at smartvox.co.uk (John Quick) Date: Mon, 19 Nov 2018 13:59:36 -0000 Subject: [OpenSIPS-Users] OpenSIPS Control Panel 8.2.4 was released Message-ID: <001001d48010$1bb732d0$53259870$@smartvox.co.uk> Hello Bogdan, The link works okay now, but (I think) only if the selected call is on the first page of results in the siptrace tab. John Quick Smartvox Limited > Hi John, > > I found a small typo that affected who the link was constructed . See > https://github.com/OpenSIPS/opensips-cp/commit/d50503123477f99b0079570361407 7b685ca4579 > > In order to link siptrace to cdrviewer, you need to (a) be sure homer > tool is disabled and (b) siptrace tool is enabled. > > Let me know if this fix does the trick for you. > > Best regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com From bogdan at opensips.org Mon Nov 19 09:11:40 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 19 Nov 2018 16:11:40 +0200 Subject: [OpenSIPS-Users] check for NULL values In-Reply-To: <2053587817.1497481.1542345628843@mail.yahoo.com> References: <1238611305.714858.1541644790120.ref@mail.yahoo.com> <1238611305.714858.1541644790120@mail.yahoo.com> <1506752889.1453845.1542099402376@mail.yahoo.com> <38f2caf6-04a3-27f3-9a33-08c1cbe0d2b2@opensips.org> <1820042280.937452.1542269085544@mail.yahoo.com> <28ec9806-c8b6-eb33-0795-1af3308ec1cd@opensips.org> <2053587817.1497481.1542345628843@mail.yahoo.com> Message-ID: Hi Pasan, That is really weird I would say. Could you "pack" the minimal script to get this error (plus the pcap of the REGISTER) ? you can send them off list to me. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/16/2018 07:20 AM, Pasan Meemaduma wrote: > Hi Bogdan, > > I tried scrip_trace function and it also indicate same behavior, I can > see the value of $tu but it still gives the error. Interesting thing > is, it evaluates to true and invokes the remove() function as well. > > Nov 16 16:06:46 /usr/sbin/opensips[16582]: [Script > Trace][/etc/opensips/opensips.cfg:814][reg-debug][core if] -> > (REGISTER from Y.Y.Y.Y F=sip:XXX at x.x.x.x, T=sip:XXX at x.x.x.x > ID=cd7921be-54514c80 at v.v.v.v#012) > Nov 16 16:06:46 /usr/sbin/opensips[16582]: ERROR:core:comp_scriptvar: > cannot get left var value > Nov 16 16:06:46 /usr/sbin/opensips[16582]: WARNING:core:do_action: > error in expression at /etc/opensips/opensips.cfg:814 > Nov 16 16:06:52 /usr/sbin/opensips[16582]: [Script > Trace][/etc/opensips/opensips.cfg:815][reg-debug][module remove] -> > (REGISTER from Y.Y.Y.Y F=sip:XXX at x.x.x.x, T=sip:XXX at x.x.x.x > ID=4afba829-fef90105 at v.v.v.v#012) > > /etc/opensips/opensips.cfg:814 > if ( $tu != NULL ) { > remove("location","$tu"); > } > > > > > On Thursday, 15 November 2018, 5:25:54 PM GMT+5:30, Bogdan-Andrei > Iancu wrote: > > > Hi Pasan, > > Indeed, the logs show that the TO hdr is successfully parsed - are you > 100% sure about the reported line ? maybe it is not the $tu related ? > try to activate the script_trace() > [http://www.opensips.org/Documentation/Script-CoreFunctions-2-4#toc42] > function to see when exactly the error is generated. Maybe you can > actually print $tu before, to see what you get. > > Regards, > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Bootcamp 2018 > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > On 11/15/2018 10:04 AM, Pasan Meemaduma wrote: >> Hi Bogdan, >> >> I'm not seeing any other related error msg prior to this. I'm calling >> this test in request route for a REGISTER request. Before the error >> pops up I could see the received msg successfully parse by opensips >> and can see value of To uri >> >> Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:parse_msg: SIP >> Request: >> Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:parse_msg: >> method: >> Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:parse_msg: uri: >> >> Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:parse_msg: >> version: >> Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:parse_headers: >> flags=2 >> Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:parse_via_param: >> found param type 232, = ; state=16 >> Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:parse_via: end >> of header reached, state=5 >> Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:parse_headers: >> via found, flags=2 >> Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:parse_headers: >> this is the first via >> Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:receive_msg: >> After parse_msg... >> Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:receive_msg: >> preparing to run routing scripts... >> Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:parse_headers: >> flags=100 >> Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:_parse_to: end >> of header reached, state=10 >> Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:_parse_to: >> display={XXXXXXXXXX}, ruri={sip:XXXXXXXXXX at xx.xx.xx.x} >> Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:get_hdr_field: >> [48]; uri=[sip:XXXXXXXXXX at x.x.x.x] >> Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:get_hdr_field: >> to body [XXXXXXXXXX #015#012] >> Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:get_hdr_field: >> cseq : <79474> >> Nov 15 18:19:19 /usr/sbin/opensips[21608]: >> DBG:maxfwd:is_maxfwd_present: value = 70 >> Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:parse_to_param: >> tag=46474fbe5728f700o0 >> Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:_parse_to: end >> of header reached, state=29 >> Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:_parse_to: >> display={XXXXXXXXXX}, ruri={sip:XXXXXXXXXX at x.x.x.x} >> Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:parse_headers: >> flags=200 >> Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:get_hdr_field: >> content_length=0 >> Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:get_hdr_field: >> found end of header >> >> and following are the line that I get before the error If I set log >> level to 6 >> >> Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:parse_headers: >> flags=ffffffffffffffff >> Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:parse_headers: >> flags=ffffffffffffffff >> Nov 15 18:19:19 /usr/sbin/opensips[21608]: DBG:core:parse_headers: >> flags=ffffffffffffffff >> Nov 15 18:19:19 /usr/sbin/opensips[21608]: >> ERROR:core:comp_scriptvar: cannot get left var value >> Nov 15 18:19:19 /usr/sbin/opensips[21608]: WARNING:core:do_action: >> error in expression at /etc/opensips/opensips.cfg:807 >> >> >> >> >> On Wednesday, 14 November 2018, 7:19:44 PM GMT+5:30, Bogdan-Andrei >> Iancu wrote: >> >> >> Hi Pasan, >> >> It should be fine if $tu would translate to NULL. The script handles >> this case. But I'm afraid you have something else there, like another >> deeper error that prevents the $tu variable to be evaluated. >> >> Do you see any other errors before the mentioned ones ? what is the >> type of route where you do the test ? >> >> Regards, >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> OpenSIPS Bootcamp 2018 >> http://opensips.org/training/OpenSIPS_Bootcamp_2018/ >> On 11/13/2018 10:56 AM, Pasan Meemaduma via Users wrote: >>> Hey, >>> >>> Anyone have a suggestion for this? >>> >>> On Thursday, 8 November 2018, 8:09:50 AM GMT+5:30, Pasan Meemaduma >>> wrote: >>> >>> >>> Hi Guys, >>> >>> I have a check for NULL for $tu var in the script, But when the >>> value is missing I'm getting the following error. >>> >>> ERROR:core:comp_scriptvar: cannot get left var value >>> >>> WARNING:core:do_action: error in expression at >>> /etc/opensips/opensips.cfg:806 >>> >>> and line 806 contains following. >>> >>> if ( $tu != NULL ) { >>> remove("location","$tu"); >>> } >>> >>> any suggestion on how to test for NULL values without getting above >>> error. I'm using opensips 2.3.5 >>> >>> >>> >>> >>> >>> _______________________________________________ >>> 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 mickael at winlux.fr Mon Nov 19 11:01:18 2018 From: mickael at winlux.fr (Mickael Hubert) Date: Mon, 19 Nov 2018 17:01:18 +0100 Subject: [OpenSIPS-Users] rtpengine (setid_avp) issue with BYE and CANCEL Message-ID: Hi all, I have a rtpengine with opensips, when I use rtpengine without setid_avp all works like a charm modparam("rtpengine", "rtpengine_sock", "udp:10.13.0.129:12221") *When I want use setid_avp:* modparam("rtpengine", "setid_avp", "$avp(setid)") modparam("rtpengine", "rtpengine_sock", "1 == udp:10.13.0.129:12221") *I call manage function in initial INVITE:* $avp(setid) = 1; create_dialog(); xlog("L_INFO","setid: $avp(setid)\n"); if(rtpengine_manage("ICE=remove")) { xlog("L_INFO","SDP Offer: $ci for INVITE\n"); } for an reply $avp(setid) is populate, and rtpengine works. *but for BYE and CANCEL I have an issue:* you can see the setid variable isn't populate for this CANCEL, but it is populate for 487 reply ! I have this error log: ERROR:rtpengine:select_rtpe_node: script error -no valid set selected / ERROR:rtpengine:rtpe_function_call: no available proxies When I force the setid to 1 for BYE and CANCEL, there isn't issue. However, the module documentation explains : *IMPORTANT: if you use multiple sets, take care and use the same set for both rtpengine_offer()/rtpengine_answer() and rtpengine_delete()!! * *If the set was selected using setid_avp, the avp needs to be set only once before rtpengine_offer() or rtpengine_manage() call. * Can you help me please ? thanks in advance PS: same issue with BYE method Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22531]: ################################################################################ Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22531]: CANCEL END OF CALL : call-id 1def6e7f-66b6-1237-7b9d-0050569229dc Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22531]: ################################################################################ Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22531]: setid: Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22531]: *ERROR:rtpengine:select_rtpe_node: script error -no valid set selected* Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22531]: *ERROR:rtpengine:rtpe_function_call: no available proxies* Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22521]: 1def6e7f-66b6-1237-7b9d-0050569229dc In ONREPLY ROUTE 2 - fu : sip:+******@am-isbc1-******** , ru : , si : 10.13.0.80, status : 487 Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22521]: setid: 1 if (is_method("CANCEL")) { xlog("L_INFO","################################################################################ \n"); xlog("L_INFO","CANCEL END OF CALL : call-id $ci \n"); xlog("L_INFO","################################################################################ \n"); xlog("L_INFO","setid: $avp(setid)\n"); if(rtpengine_manage("ICE=remove")) { xlog("L_INFO","SDP Offer: $ci for CANCEL\n"); } ............. -------------- next part -------------- An HTML attachment was scrubbed... URL: From pasandev at ymail.com Mon Nov 19 20:45:06 2018 From: pasandev at ymail.com (Pasan Meemaduma) Date: Tue, 20 Nov 2018 01:45:06 +0000 (UTC) Subject: [OpenSIPS-Users] check for NULL values In-Reply-To: References: <1238611305.714858.1541644790120.ref@mail.yahoo.com> <1238611305.714858.1541644790120@mail.yahoo.com> <1506752889.1453845.1542099402376@mail.yahoo.com> <38f2caf6-04a3-27f3-9a33-08c1cbe0d2b2@opensips.org> <1820042280.937452.1542269085544@mail.yahoo.com> <28ec9806-c8b6-eb33-0795-1af3308ec1cd@opensips.org> <2053587817.1497481.1542345628843@mail.yahoo.com> Message-ID: <1247814082.3263724.1542678306636@mail.yahoo.com> Hi Bogdan, I'll try to get that done. I'm seeing this on a production server. I'll try to get the info you requested. On Monday, 19 November 2018, 7:41:54 PM GMT+5:30, Bogdan-Andrei Iancu wrote: Hi Pasan, That is really weird I would say. Could you "pack" the minimal script to get this error (plus the pcap of the REGISTER) ? you can send them off list to me. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/16/2018 07:20 AM, Pasan Meemaduma wrote: Hi Bogdan, I tried scrip_trace function and it also indicate same behavior, I can see the value of $tu but it still gives the error. Interesting thing is, it evaluates to true and invokes the remove() function as well. Nov 16 16:06:46  /usr/sbin/opensips[16582]: [Script Trace][/etc/opensips/opensips.cfg:814][reg-debug][core if] -> (REGISTER from Y.Y.Y.Y F=sip:XXX at x.x.x.x, T=sip:XXX at x.x.x.x ID=cd7921be-54514c80 at v.v.v.v#012) Nov 16 16:06:46  /usr/sbin/opensips[16582]: ERROR:core:comp_scriptvar: cannot get left var value Nov 16 16:06:46  /usr/sbin/opensips[16582]: WARNING:core:do_action: error in expression at /etc/opensips/opensips.cfg:814 Nov 16 16:06:52  /usr/sbin/opensips[16582]: [Script Trace][/etc/opensips/opensips.cfg:815][reg-debug][module remove] -> (REGISTER from Y.Y.Y.Y F=sip:XXX at x.x.x.x, T=sip:XXX at x.x.x.x ID=4afba829-fef90105 at v.v.v.v#012) /etc/opensips/opensips.cfg:814     if ( $tu != NULL ) {         remove("location","$tu");     } On Thursday, 15 November 2018, 5:25:54 PM GMT+5:30, Bogdan-Andrei Iancu wrote: Hi Pasan, Indeed, the logs show that the TO hdr is successfully parsed - are you 100% sure about the reported line ? maybe it is not the $tu related ? try to activate the script_trace() [http://www.opensips.org/Documentation/Script-CoreFunctions-2-4#toc42] function to see when exactly the error is generated. Maybe you can actually print $tu before, to see what you get. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/15/2018 10:04 AM, Pasan Meemaduma wrote: Hi Bogdan, I'm not seeing any other related error msg prior to this. I'm calling this test in request route for a REGISTER request. Before the error pops up I could see the received msg successfully parse by opensips and can see value of To uri Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_msg: SIP Request: Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_msg:  method:  Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_msg:  uri:     Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_msg:  version: Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_headers: flags=2 Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_via_param: found param type 232, = ; state=16 Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_via: end of header reached, state=5 Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_headers: via found, flags=2 Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_headers: this is the first via Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:receive_msg: After parse_msg... Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:receive_msg: preparing to run routing scripts... Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_headers: flags=100 Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:_parse_to: end of header reached, state=10 Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:_parse_to: display={XXXXXXXXXX}, ruri={sip:XXXXXXXXXX at xx.xx.xx.x} Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:get_hdr_field: [48]; uri=[sip:XXXXXXXXXX at x.x.x.x] Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:get_hdr_field: to body [XXXXXXXXXX #015#012] Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:get_hdr_field: cseq : <79474> Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:maxfwd:is_maxfwd_present: value = 70 Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_to_param: tag=46474fbe5728f700o0 Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:_parse_to: end of header reached, state=29 Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:_parse_to: display={XXXXXXXXXX}, ruri={sip:XXXXXXXXXX at x.x.x.x} Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_headers: flags=200 Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:get_hdr_field: content_length=0 Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:get_hdr_field: found end of header and following are the line that I get before the error If I set log level to 6 Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_headers: flags=ffffffffffffffff Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_headers: flags=ffffffffffffffff Nov 15 18:19:19  /usr/sbin/opensips[21608]: DBG:core:parse_headers: flags=ffffffffffffffff Nov 15 18:19:19  /usr/sbin/opensips[21608]: ERROR:core:comp_scriptvar: cannot get left var value Nov 15 18:19:19  /usr/sbin/opensips[21608]: WARNING:core:do_action: error in expression at /etc/opensips/opensips.cfg:807 On Wednesday, 14 November 2018, 7:19:44 PM GMT+5:30, Bogdan-Andrei Iancu wrote: Hi Pasan, It should be fine if $tu would translate to NULL. The script handles this case. But I'm afraid you have something else there, like another deeper error that prevents the $tu variable to be evaluated. Do you see any other errors before the mentioned ones ? what is the type of route where you do the test ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/13/2018 10:56 AM, Pasan Meemaduma via Users wrote: Hey, Anyone have a suggestion for this? On Thursday, 8 November 2018, 8:09:50 AM GMT+5:30, Pasan Meemaduma wrote: Hi Guys, I have a check for NULL for $tu var in the script, But when the value is missing I'm getting the following error. ERROR:core:comp_scriptvar: cannot get left var value WARNING:core:do_action: error in expression at/etc/opensips/opensips.cfg:806 and line 806 contains following.     if ( $tu != NULL ) {         remove("location","$tu");     } any suggestion on how to test for NULL values without getting above error. I'm using opensips 2.3.5 _______________________________________________ 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 Nov 20 03:38:12 2018 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Tue, 20 Nov 2018 10:38:12 +0200 Subject: [OpenSIPS-Users] TLS->UDP Re-Invite Issue In-Reply-To: References: Message-ID: <2027acf9-c76a-3bd9-385f-f0196e78da91@opensips.org> Hello! Can you send us a sip trace, or pcap with tthe call? It will help us understand what's going on. Best regards, Razvan On 11/19/18 8:28 AM, Dhruv Shah wrote: > Hello, > > I am using opensips-2.2, and using TLS for calling. From opensips using > dispatcher call is forwarded on freeswitch which is on UDP, then > freeswitch forwards it to external gateway. It works perfectly, but when > freeswitch generates reinvite, it is send to some another port i.e port > is different from which the ACK is received, due to this call gets > hangup due to request timeout. > > > Please suggest solution to solve this issue. > > -- > *Thanks & Regards* > *Dhruv Shah* > *Jr. Software Developer* > Ecosmob Technologies Pvt. Ltd. > *Ecosmob Technologies Pvt. Ltd. * > https://www.ecosmob.com > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com From razvan at opensips.org Tue Nov 20 03:42:31 2018 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Tue, 20 Nov 2018 10:42:31 +0200 Subject: [OpenSIPS-Users] rtpengine (setid_avp) issue with BYE and CANCEL In-Reply-To: References: Message-ID: Hi, Mickael! AVPs are only available during transaction - initial request and its replies, but it is not avaialble to sequential requests (such as CANCEL or BYE). If you want to store a value throughout the entire dialog, you should save the variable in a dialog variable[1], something like this: # initial INVITE $avp(setid) = 1; create_dialog(); $dlg_val(setid) = $avp(setid); # CANCEL or BYE, *after* loose_route() is called # so that the dialog is matched/found loose_route(); $avp(setid) = $dlg_val(setid); Hope this helps. [1] https://opensips.org/html/docs/modules/2.4.x/dialog#pv_dlg_val Best regards, Răzvan On 11/19/18 6:01 PM, Mickael Hubert wrote: > Hi all, > I have a rtpengine with opensips, when I use rtpengine without setid_avp > all works like a charm > > modparam("rtpengine", "rtpengine_sock", "udp:10.13.0.129:12221 > ") > > _When I want use setid_avp:_ > > modparam("rtpengine", "setid_avp", "$avp(setid)") > modparam("rtpengine", "rtpengine_sock", "1 == udp:10.13.0.129:12221 > ") > > _I call manage function in initial INVITE:_ > > $avp(setid) =  1; > create_dialog(); > xlog("L_INFO","setid: $avp(setid)\n"); > if(rtpengine_manage("ICE=remove")) > { >   xlog("L_INFO","SDP Offer: $ci for INVITE\n"); > } > > for an reply $avp(setid) is populate, and rtpengine works. > > _but for BYE and CANCEL I have an issue:_ > _ > _ > you can see the setid variable isn't populate for this CANCEL, but it is > populate for 487 reply ! > I have this error log: ERROR:rtpengine:select_rtpe_node: script error > -no valid set selected / ERROR:rtpengine:rtpe_function_call: no > available proxies > > When I force the setid to 1 for BYE and CANCEL, there isn't issue. > > However, the module documentation explains : > /IMPORTANT: if you use multiple sets, take care and use the same set for > both rtpengine_offer()/rtpengine_answer() and rtpengine_delete()!! > / > /If the set was selected using setid_avp, the avp needs to be set only > once before rtpengine_offer() or rtpengine_manage() call. / > > Can you help me please ? > > thanks in advance > > PS: same issue with BYE method > __ > > __ > > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22531]: > ################################################################################ > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22531]: CANCEL END > OF CALL : call-id 1def6e7f-66b6-1237-7b9d-0050569229dc > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22531]: > ################################################################################ > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22531]: setid: > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22531]: > *ERROR:rtpengine:select_rtpe_node: script error -no valid set selected* > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22531]: > *ERROR:rtpengine:rtpe_function_call: no available proxies* > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22521]: > 1def6e7f-66b6-1237-7b9d-0050569229dc In ONREPLY ROUTE 2 - fu : > sip:+******@am-isbc1-******** , ru : , si : 10.13.0.80, status : 487 > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22521]: setid: 1 > > if (is_method("CANCEL")) >   { > > xlog("L_INFO","################################################################################ > \n"); >     xlog("L_INFO","CANCEL END OF CALL : call-id $ci \n"); > > xlog("L_INFO","################################################################################ > \n"); >     xlog("L_INFO","setid: $avp(setid)\n"); >     if(rtpengine_manage("ICE=remove")) >     { >       xlog("L_INFO","SDP Offer: $ci for CANCEL\n"); >     } > ............. > > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com From razvan at opensips.org Tue Nov 20 03:44:15 2018 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Tue, 20 Nov 2018 10:44:15 +0200 Subject: [OpenSIPS-Users] Opensips 2.1.2 - can't pass var in rewritehostport() In-Reply-To: <1542191968913-0.post@n2.nabble.com> References: <1541664230763-0.post@n2.nabble.com> <1542191968913-0.post@n2.nabble.com> Message-ID: All you have to do is to assingn the $rd pseudo-variable, which does the same thing as rewritehostport() function. On 11/14/18 12:39 PM, maatohewetbi wrote: > How can I use it, like that? > $rd=$hdr(X-IP-Header) > rewritehostport($rd:5060) > > > > > -- > Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com From mickael at winlux.fr Tue Nov 20 05:15:44 2018 From: mickael at winlux.fr (Mickael Hubert) Date: Tue, 20 Nov 2018 11:15:44 +0100 Subject: [OpenSIPS-Users] rtpengine (setid_avp) issue with BYE and CANCEL In-Reply-To: References: Message-ID: Hi Răzvan, thanks a lot for your answer ! Ok I will try this workaround But I don't understand the rtpengine module documentation ( http://www.opensips.org/html/docs/modules/2.4.x/rtpengine.html) If I have -> modparam("rtpengine", "setid_avp", "$avp(setid)") and I set setid_avp to 1 (for my example) just once, before rtpengine_manage, this variable should survives until the end no ? ++ Le mar. 20 nov. 2018 à 09:43, Răzvan Crainea a écrit : > Hi, Mickael! > > AVPs are only available during transaction - initial request and its > replies, but it is not avaialble to sequential requests (such as CANCEL > or BYE). If you want to store a value throughout the entire dialog, you > should save the variable in a dialog variable[1], something like this: > > # initial INVITE > $avp(setid) = 1; > create_dialog(); > $dlg_val(setid) = $avp(setid); > > # CANCEL or BYE, *after* loose_route() is called > # so that the dialog is matched/found > loose_route(); > $avp(setid) = $dlg_val(setid); > > Hope this helps. > > [1] https://opensips.org/html/docs/modules/2.4.x/dialog#pv_dlg_val > > Best regards, > Răzvan > > On 11/19/18 6:01 PM, Mickael Hubert wrote: > > Hi all, > > I have a rtpengine with opensips, when I use rtpengine without setid_avp > > all works like a charm > > > > modparam("rtpengine", "rtpengine_sock", "udp:10.13.0.129:12221 > > ") > > > > _When I want use setid_avp:_ > > > > modparam("rtpengine", "setid_avp", "$avp(setid)") > > modparam("rtpengine", "rtpengine_sock", "1 == udp:10.13.0.129:12221 > > ") > > > > _I call manage function in initial INVITE:_ > > > > $avp(setid) = 1; > > create_dialog(); > > xlog("L_INFO","setid: $avp(setid)\n"); > > if(rtpengine_manage("ICE=remove")) > > { > > xlog("L_INFO","SDP Offer: $ci for INVITE\n"); > > } > > > > for an reply $avp(setid) is populate, and rtpengine works. > > > > _but for BYE and CANCEL I have an issue:_ > > _ > > _ > > you can see the setid variable isn't populate for this CANCEL, but it is > > populate for 487 reply ! > > I have this error log: ERROR:rtpengine:select_rtpe_node: script error > > -no valid set selected / ERROR:rtpengine:rtpe_function_call: no > > available proxies > > > > When I force the setid to 1 for BYE and CANCEL, there isn't issue. > > > > However, the module documentation explains : > > /IMPORTANT: if you use multiple sets, take care and use the same set for > > both rtpengine_offer()/rtpengine_answer() and rtpengine_delete()!! > > / > > /If the set was selected using setid_avp, the avp needs to be set only > > once before rtpengine_offer() or rtpengine_manage() call. / > > > > Can you help me please ? > > > > thanks in advance > > > > PS: same issue with BYE method > > __ > > > > __ > > > > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22531]: > > > ################################################################################ > > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22531]: CANCEL END > > OF CALL : call-id 1def6e7f-66b6-1237-7b9d-0050569229dc > > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22531]: > > > ################################################################################ > > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22531]: setid: > > > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22531]: > > *ERROR:rtpengine:select_rtpe_node: script error -no valid set selected* > > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22531]: > > *ERROR:rtpengine:rtpe_function_call: no available proxies* > > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22521]: > > 1def6e7f-66b6-1237-7b9d-0050569229dc In ONREPLY ROUTE 2 - fu : > > sip:+******@am-isbc1-******** , ru : , si : 10.13.0.80, status : > 487 > > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22521]: setid: 1 > > > > if (is_method("CANCEL")) > > { > > > > > xlog("L_INFO","################################################################################ > > > \n"); > > xlog("L_INFO","CANCEL END OF CALL : call-id $ci \n"); > > > > > xlog("L_INFO","################################################################################ > > > \n"); > > xlog("L_INFO","setid: $avp(setid)\n"); > > if(rtpengine_manage("ICE=remove")) > > { > > xlog("L_INFO","SDP Offer: $ci for CANCEL\n"); > > } > > ............. > > > > > > > > > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.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 julian.santer at rolmail.net Mon Nov 19 12:41:15 2018 From: julian.santer at rolmail.net (Julian Santer) Date: Mon, 19 Nov 2018 18:41:15 +0100 Subject: [OpenSIPS-Users] Nathelper keepalive issue with received column in usrloc Message-ID: <39a84907-a8a7-4423-0952-53b2802960ae@rolmail.net> Hi guys, we need to switch from nat_traversal to nathelper. The reason is the keepalive mechanism. The nat_traversal module sends OPTIONS with the following to header: sip:UAC_IP:UAC_PORT Most of the UAC's answers with a 404 Not found. On AVM Fritzbox with firmware >= 6.04, this OPTIONS may activate a security feature. So after a certain time, the Fritzbox blocks all packages send from our proxy. As we have ca. 80% AVM Fritzbox as UAC, we got a big problem. So we deactivated the nat_keepalive vor this UAC's and we have to enable the keepalive Feature on the Fritzbox. The better solution would be, if we could send OPTIONS with a to header like: sip:username at UAC_IP:UAC_PORT. As I understood the nathelper module could send OPTIONS like this. Because it is looking into the userloc table. Right? The nathelper module is on our edge server, the registrar on our core server. For the "normal" UAC's (no received entry in usrloc) the keepalive's are now sent as expected. But for the "nated" UAC's (received entry in usrloc) the keepalive's are like before: sip:UAC_IP:UAC_PORT (values in the received column from usrloc). The REGISTER send to the core got the path header looking like: Path: Is there a possibility to add the $fU on the received part of the path header (the user in the path module adds a string to the path part, but not to the received part)? Or is there a possiblity on the registrar to store the $fU in the received column? On the nathelper keepalive mechanism I don't see any possibility to add the $fU. We are using the version 2.2.6 from the official debian source list. The config on the edge server's looks like: #### nat helper module loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 0) modparam("nathelper", "ping_nated_only", 0) modparam("nathelper", "natping_partitions", 1) modparam("nathelper", "natping_tcp", 0) ### REGISTER $var(nat) = null; if (nat_uac_test("127")) {     $var(nat) = TRUE; } else {     $var(nat) = FALSE; } consume_credentials(); if ($var(nat) == TRUE) {     if (! add_path_received())     {         xlog("L_ERR", "Adding PATH (with received) failed - LF_BASE");         send_reply("500", "Internal path error, registration not stored");         exit;     } } else {     if (! add_path())     {         send_reply("500", "Internal path error, registration not stored");         xlog("L_ERR", "Adding PATH (with received) failed - LF_BASE");         exit;     } } route("R_RELAY_TO_REGISTRAR"); exit; ### OPTIONS if (method=="OPTIONS") {     if ($si == "CORE")     {         topology_hiding("U");         if (! t_relay("0x05"))         {             send_reply("500", "Internal server error - failed to relay");             xlog("L_ERR", "Unable to relay OPTIONS - LF_BASE");         }     } } The config on the core server looks like: loadmodule "usrloc.so" modparam("usrloc", "user_column",           "username") modparam("usrloc", "domain_column",         "domain") modparam("usrloc", "contact_column",        "contact") modparam("usrloc", "expires_column",        "expires") modparam("usrloc", "q_column",              "q") modparam("usrloc", "callid_column",         "callid") modparam("usrloc", "cseq_column",           "cseq") modparam("usrloc", "methods_column",        "methods") modparam("usrloc", "flags_column",          "flags") modparam("usrloc", "user_agent_column",     "user_agent") modparam("usrloc", "received_column",       "received") modparam("usrloc", "path_column",           "path") modparam("usrloc", "socket_column",         "socket") modparam("usrloc", "use_domain",            0) modparam("usrloc", "desc_time_order",       0) modparam("usrloc", "timer_interval",        60) modparam("usrloc", "db_url",                "DBURL") modparam("usrloc", "db_mode",               2) modparam("usrloc", "matching_mode",         0) modparam("usrloc", "cseq_delay",            20) modparam("usrloc", "nat_bflag",             6) #### nat helper module loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 56) modparam("nathelper", "ping_nated_only", 0) modparam("nathelper", "natping_partitions", 1) modparam("nathelper", "sipping_bflag", 8) modparam("nathelper", "sipping_from", "sip:keepalive at DEFAULT_REALM") modparam("nathelper", "sipping_method", "OPTIONS") # We want to send a keepalive on each registered UAC if (proto == UDP) {     setbflag(8);     xlog("L_INFO", "Nat keepalive sip_ping_flag - LF_BASE"); } if (! save("location", "vp1")) {     xlog("L_ERR", "Saving contact from EDGE failed - LF_BASE");     exit; } Thank you for any hint. Kind regards, Julian Santer Raiffeisen OnLine ps: @Bogdan: this is why we have ca. 550 entry's in the address table (permission module). If we solve the keepalives, only ca. 50 entry's are remaining. From liviu at opensips.org Tue Nov 20 06:23:43 2018 From: liviu at opensips.org (Liviu Chircu) Date: Tue, 20 Nov 2018 13:23:43 +0200 Subject: [OpenSIPS-Users] Opensips 2.1.2 - can't pass var in rewritehostport() In-Reply-To: References: <1541664230763-0.post@n2.nabble.com> <1542191968913-0.post@n2.nabble.com> Message-ID: <8fd06ec8-8bd2-6f0c-6698-06a42d4e0503@opensips.org> Minor nit: $rd is not exactly the equivalent of rewritehostport(), rather $rd used together with $rp [1]. [1]: https://www.opensips.org/Documentation/Script-CoreVar-3-0#toc70 Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 20.11.2018 10:44, Răzvan Crainea wrote: > All you have to do is to assingn the $rd pseudo-variable, which does > the same thing as rewritehostport() function. > > On 11/14/18 12:39 PM, maatohewetbi wrote: >> How can I use it, like that? >> $rd=$hdr(X-IP-Header) >> rewritehostport($rd:5060) >> >> >> >> >> -- >> Sent from: >> http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > From razvan at opensips.org Tue Nov 20 07:44:34 2018 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Tue, 20 Nov 2018 14:44:34 +0200 Subject: [OpenSIPS-Users] rtpengine (setid_avp) issue with BYE and CANCEL In-Reply-To: References: Message-ID: <283af46f-7f59-d8f3-4674-c1eac65fd895@opensips.org> No, it only survives until the end of transaction :). I agree it would be nice to have it persistent throughout the dialog, but that's not in there yet. But you can always open a feature request for this :): https://github.com/OpenSIPS/opensips/issues Best regards, Răzvan On 11/20/18 12:15 PM, Mickael Hubert wrote: > Hi Răzvan, > thanks a lot for your answer ! > > Ok I will try this workaround > But I don't understand the rtpengine module documentation > (http://www.opensips.org/html/docs/modules/2.4.x/rtpengine.html) > > If I have -> modparam("rtpengine", "setid_avp", "$avp(setid)") and I set > setid_avp to 1 (for my example) just once, before rtpengine_manage, this > variable should survives until the end no ? > > > ++ > > > > Le mar. 20 nov. 2018 à 09:43, Răzvan Crainea > a écrit : > > Hi, Mickael! > > AVPs are only available during transaction - initial request and its > replies, but it is not avaialble to sequential requests (such as CANCEL > or BYE). If you want to store a value throughout the entire dialog, you > should save the variable in a dialog variable[1], something like this: > > # initial INVITE > $avp(setid) = 1; > create_dialog(); > $dlg_val(setid) = $avp(setid); > > # CANCEL or BYE, *after* loose_route() is called > # so that the dialog is matched/found > loose_route(); > $avp(setid) = $dlg_val(setid); > > Hope this helps. > > [1] https://opensips.org/html/docs/modules/2.4.x/dialog#pv_dlg_val > > Best regards, > Răzvan > > On 11/19/18 6:01 PM, Mickael Hubert wrote: > > Hi all, > > I have a rtpengine with opensips, when I use rtpengine without > setid_avp > > all works like a charm > > > > modparam("rtpengine", "rtpengine_sock", "udp:10.13.0.129:12221 > > > ") > > > > _When I want use setid_avp:_ > > > > modparam("rtpengine", "setid_avp", "$avp(setid)") > > modparam("rtpengine", "rtpengine_sock", "1 == > udp:10.13.0.129:12221 > > ") > > > > _I call manage function in initial INVITE:_ > > > > $avp(setid) =  1; > > create_dialog(); > > xlog("L_INFO","setid: $avp(setid)\n"); > > if(rtpengine_manage("ICE=remove")) > > { > >    xlog("L_INFO","SDP Offer: $ci for INVITE\n"); > > } > > > > for an reply $avp(setid) is populate, and rtpengine works. > > > > _but for BYE and CANCEL I have an issue:_ > > _ > > _ > > you can see the setid variable isn't populate for this CANCEL, > but it is > > populate for 487 reply ! > > I have this error log: ERROR:rtpengine:select_rtpe_node: script > error > > -no valid set selected / ERROR:rtpengine:rtpe_function_call: no > > available proxies > > > > When I force the setid to 1 for BYE and CANCEL, there isn't issue. > > > > However, the module documentation explains : > > /IMPORTANT: if you use multiple sets, take care and use the same > set for > > both rtpengine_offer()/rtpengine_answer() and rtpengine_delete()!! > > / > > /If the set was selected using setid_avp, the avp needs to be set > only > > once before rtpengine_offer() or rtpengine_manage() call. / > > > > Can you help me please ? > > > > thanks in advance > > > > PS: same issue with BYE method > > __ > > > > __ > > > > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22531]: > > > ################################################################################ > > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22531]: > CANCEL END > > OF CALL : call-id 1def6e7f-66b6-1237-7b9d-0050569229dc > > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22531]: > > > ################################################################################ > > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22531]: > setid: > > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22531]: > > *ERROR:rtpengine:select_rtpe_node: script error -no valid set > selected* > > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22531]: > > *ERROR:rtpengine:rtpe_function_call: no available proxies* > > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22521]: > > 1def6e7f-66b6-1237-7b9d-0050569229dc In ONREPLY ROUTE 2 - fu : > > sip:+******@am-isbc1-******** , ru : , si : 10.13.0.80, > status : 487 > > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22521]: > setid: 1 > > > > if (is_method("CANCEL")) > >    { > > > > > xlog("L_INFO","################################################################################ > > > \n"); > >      xlog("L_INFO","CANCEL END OF CALL : call-id $ci \n"); > > > > > xlog("L_INFO","################################################################################ > > > \n"); > >      xlog("L_INFO","setid: $avp(setid)\n"); > >      if(rtpengine_manage("ICE=remove")) > >      { > >        xlog("L_INFO","SDP Offer: $ci for CANCEL\n"); > >      } > > ............. > > > > > > > > > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.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 > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com From mickael at winlux.fr Tue Nov 20 07:58:04 2018 From: mickael at winlux.fr (Mickael Hubert) Date: Tue, 20 Nov 2018 13:58:04 +0100 Subject: [OpenSIPS-Users] rtpengine (setid_avp) issue with BYE and CANCEL In-Reply-To: <283af46f-7f59-d8f3-4674-c1eac65fd895@opensips.org> References: <283af46f-7f59-d8f3-4674-c1eac65fd895@opensips.org> Message-ID: OK thanks, I will write something about that ;) I tested your solution, it works for BYE but not force CANCEL. Indeed, for BYE from my SBC enters in loose_route statement, but not for a CANCEL. is there a simplest way to choose the good rtpengine ? I want use a group of rtpengine (like id:1) for Europe and other group (like id:2) for USA. ++ Le mar. 20 nov. 2018 à 13:45, Răzvan Crainea a écrit : > No, it only survives until the end of transaction :). > I agree it would be nice to have it persistent throughout the dialog, > but that's not in there yet. But you can always open a feature request > for this :): > > https://github.com/OpenSIPS/opensips/issues > > Best regards, > Răzvan > > On 11/20/18 12:15 PM, Mickael Hubert wrote: > > Hi Răzvan, > > thanks a lot for your answer ! > > > > Ok I will try this workaround > > But I don't understand the rtpengine module documentation > > (http://www.opensips.org/html/docs/modules/2.4.x/rtpengine.html) > > > > If I have -> modparam("rtpengine", "setid_avp", "$avp(setid)") and I set > > setid_avp to 1 (for my example) just once, before rtpengine_manage, this > > variable should survives until the end no ? > > > > > > ++ > > > > > > > > Le mar. 20 nov. 2018 à 09:43, Răzvan Crainea > > a écrit : > > > > Hi, Mickael! > > > > AVPs are only available during transaction - initial request and its > > replies, but it is not avaialble to sequential requests (such as > CANCEL > > or BYE). If you want to store a value throughout the entire dialog, > you > > should save the variable in a dialog variable[1], something like > this: > > > > # initial INVITE > > $avp(setid) = 1; > > create_dialog(); > > $dlg_val(setid) = $avp(setid); > > > > # CANCEL or BYE, *after* loose_route() is called > > # so that the dialog is matched/found > > loose_route(); > > $avp(setid) = $dlg_val(setid); > > > > Hope this helps. > > > > [1] https://opensips.org/html/docs/modules/2.4.x/dialog#pv_dlg_val > > > > Best regards, > > Răzvan > > > > On 11/19/18 6:01 PM, Mickael Hubert wrote: > > > Hi all, > > > I have a rtpengine with opensips, when I use rtpengine without > > setid_avp > > > all works like a charm > > > > > > modparam("rtpengine", "rtpengine_sock", "udp:10.13.0.129:12221 > > > > > ") > > > > > > _When I want use setid_avp:_ > > > > > > modparam("rtpengine", "setid_avp", "$avp(setid)") > > > modparam("rtpengine", "rtpengine_sock", "1 == > > udp:10.13.0.129:12221 > > > ") > > > > > > _I call manage function in initial INVITE:_ > > > > > > $avp(setid) = 1; > > > create_dialog(); > > > xlog("L_INFO","setid: $avp(setid)\n"); > > > if(rtpengine_manage("ICE=remove")) > > > { > > > xlog("L_INFO","SDP Offer: $ci for INVITE\n"); > > > } > > > > > > for an reply $avp(setid) is populate, and rtpengine works. > > > > > > _but for BYE and CANCEL I have an issue:_ > > > _ > > > _ > > > you can see the setid variable isn't populate for this CANCEL, > > but it is > > > populate for 487 reply ! > > > I have this error log: ERROR:rtpengine:select_rtpe_node: script > > error > > > -no valid set selected / ERROR:rtpengine:rtpe_function_call: no > > > available proxies > > > > > > When I force the setid to 1 for BYE and CANCEL, there isn't issue. > > > > > > However, the module documentation explains : > > > /IMPORTANT: if you use multiple sets, take care and use the same > > set for > > > both rtpengine_offer()/rtpengine_answer() and rtpengine_delete()!! > > > / > > > /If the set was selected using setid_avp, the avp needs to be set > > only > > > once before rtpengine_offer() or rtpengine_manage() call. / > > > > > > Can you help me please ? > > > > > > thanks in advance > > > > > > PS: same issue with BYE method > > > __ > > > > > > __ > > > > > > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22531]: > > > > > > ################################################################################ > > > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22531]: > > CANCEL END > > > OF CALL : call-id 1def6e7f-66b6-1237-7b9d-0050569229dc > > > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22531]: > > > > > > ################################################################################ > > > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22531]: > > setid: > > > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22531]: > > > *ERROR:rtpengine:select_rtpe_node: script error -no valid set > > selected* > > > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22531]: > > > *ERROR:rtpengine:rtpe_function_call: no available proxies* > > > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22521]: > > > 1def6e7f-66b6-1237-7b9d-0050569229dc In ONREPLY ROUTE 2 - fu : > > > sip:+******@am-isbc1-******** , ru : , si : 10.13.0.80, > > status : 487 > > > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22521]: > > setid: 1 > > > > > > if (is_method("CANCEL")) > > > { > > > > > > > > > xlog("L_INFO","################################################################################ > > > > > \n"); > > > xlog("L_INFO","CANCEL END OF CALL : call-id $ci \n"); > > > > > > > > > xlog("L_INFO","################################################################################ > > > > > \n"); > > > xlog("L_INFO","setid: $avp(setid)\n"); > > > if(rtpengine_manage("ICE=remove")) > > > { > > > xlog("L_INFO","SDP Offer: $ci for CANCEL\n"); > > > } > > > ............. > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > Users mailing list > > > Users at lists.opensips.org > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > -- > > Răzvan Crainea > > OpenSIPS Core Developer > > http://www.opensips-solutions.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 > > > > -- > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.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 mickael at winlux.fr Tue Nov 20 09:23:37 2018 From: mickael at winlux.fr (Mickael Hubert) Date: Tue, 20 Nov 2018 15:23:37 +0100 Subject: [OpenSIPS-Users] rtpengine (setid_avp) issue with BYE and CANCEL In-Reply-To: References: <283af46f-7f59-d8f3-4674-c1eac65fd895@opensips.org> Message-ID: I have to use local cache to workaround. *In INVITE statement:* $avp(setid) = 1; # group of rtp engine servers / Ex: 1 for FR / 2 for USA if(cache_store("local","$ci","$avp(setid)",1200)) { xlog("L_INFO","$ci -- Cache Store Name: $ci with port: $avp(setid)\n"); *in CANCEL statement:* if(cache_fetch("local","$ci",$avp(setid))) { xlog("L_INFO","$ci -- Cache fetch Name: $ci with port: $avp(setid)\n"); cache_remove("local","$ci"); } $avp(setid) = $(avp(setid){s.int}); And it's not clean but works like a charm. @Răzvan if you have an idea about the CANCEL issue with $dlg_val is not populate, I'm interesting ;) thanks Le mar. 20 nov. 2018 à 13:58, Mickael Hubert a écrit : > OK thanks, I will write something about that ;) > > I tested your solution, it works for BYE but not force CANCEL. > Indeed, for BYE from my SBC enters in loose_route statement, but not for a > CANCEL. > > is there a simplest way to choose the good rtpengine ? I want use a group > of rtpengine (like id:1) for Europe and other group (like id:2) for USA. > > ++ > > Le mar. 20 nov. 2018 à 13:45, Răzvan Crainea a > écrit : > >> No, it only survives until the end of transaction :). >> I agree it would be nice to have it persistent throughout the dialog, >> but that's not in there yet. But you can always open a feature request >> for this :): >> >> https://github.com/OpenSIPS/opensips/issues >> >> Best regards, >> Răzvan >> >> On 11/20/18 12:15 PM, Mickael Hubert wrote: >> > Hi Răzvan, >> > thanks a lot for your answer ! >> > >> > Ok I will try this workaround >> > But I don't understand the rtpengine module documentation >> > (http://www.opensips.org/html/docs/modules/2.4.x/rtpengine.html) >> > >> > If I have -> modparam("rtpengine", "setid_avp", "$avp(setid)") and I >> set >> > setid_avp to 1 (for my example) just once, before rtpengine_manage, >> this >> > variable should survives until the end no ? >> > >> > >> > ++ >> > >> > >> > >> > Le mar. 20 nov. 2018 à 09:43, Răzvan Crainea > > > a écrit : >> > >> > Hi, Mickael! >> > >> > AVPs are only available during transaction - initial request and its >> > replies, but it is not avaialble to sequential requests (such as >> CANCEL >> > or BYE). If you want to store a value throughout the entire dialog, >> you >> > should save the variable in a dialog variable[1], something like >> this: >> > >> > # initial INVITE >> > $avp(setid) = 1; >> > create_dialog(); >> > $dlg_val(setid) = $avp(setid); >> > >> > # CANCEL or BYE, *after* loose_route() is called >> > # so that the dialog is matched/found >> > loose_route(); >> > $avp(setid) = $dlg_val(setid); >> > >> > Hope this helps. >> > >> > [1] https://opensips.org/html/docs/modules/2.4.x/dialog#pv_dlg_val >> > >> > Best regards, >> > Răzvan >> > >> > On 11/19/18 6:01 PM, Mickael Hubert wrote: >> > > Hi all, >> > > I have a rtpengine with opensips, when I use rtpengine without >> > setid_avp >> > > all works like a charm >> > > >> > > modparam("rtpengine", "rtpengine_sock", "udp:10.13.0.129:12221 >> > >> > > ") >> > > >> > > _When I want use setid_avp:_ >> > > >> > > modparam("rtpengine", "setid_avp", "$avp(setid)") >> > > modparam("rtpengine", "rtpengine_sock", "1 == >> > udp:10.13.0.129:12221 >> > > ") >> > > >> > > _I call manage function in initial INVITE:_ >> > > >> > > $avp(setid) = 1; >> > > create_dialog(); >> > > xlog("L_INFO","setid: $avp(setid)\n"); >> > > if(rtpengine_manage("ICE=remove")) >> > > { >> > > xlog("L_INFO","SDP Offer: $ci for INVITE\n"); >> > > } >> > > >> > > for an reply $avp(setid) is populate, and rtpengine works. >> > > >> > > _but for BYE and CANCEL I have an issue:_ >> > > _ >> > > _ >> > > you can see the setid variable isn't populate for this CANCEL, >> > but it is >> > > populate for 487 reply ! >> > > I have this error log: ERROR:rtpengine:select_rtpe_node: script >> > error >> > > -no valid set selected / ERROR:rtpengine:rtpe_function_call: no >> > > available proxies >> > > >> > > When I force the setid to 1 for BYE and CANCEL, there isn't >> issue. >> > > >> > > However, the module documentation explains : >> > > /IMPORTANT: if you use multiple sets, take care and use the same >> > set for >> > > both rtpengine_offer()/rtpengine_answer() and >> rtpengine_delete()!! >> > > / >> > > /If the set was selected using setid_avp, the avp needs to be set >> > only >> > > once before rtpengine_offer() or rtpengine_manage() call. / >> > > >> > > Can you help me please ? >> > > >> > > thanks in advance >> > > >> > > PS: same issue with BYE method >> > > __ >> > > >> > > __ >> > > >> > > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22531]: >> > > >> > >> ################################################################################ >> > > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22531]: >> > CANCEL END >> > > OF CALL : call-id 1def6e7f-66b6-1237-7b9d-0050569229dc >> > > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22531]: >> > > >> > >> ################################################################################ >> > > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22531]: >> > setid: >> > > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22531]: >> > > *ERROR:rtpengine:select_rtpe_node: script error -no valid set >> > selected* >> > > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22531]: >> > > *ERROR:rtpengine:rtpe_function_call: no available proxies* >> > > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22521]: >> > > 1def6e7f-66b6-1237-7b9d-0050569229dc In ONREPLY ROUTE 2 - fu : >> > > sip:+******@am-isbc1-******** , ru : , si : 10.13.0.80, >> > status : 487 >> > > Nov 19 16:53:41 am-scr1-test /usr/local/sbin/opensips[22521]: >> > setid: 1 >> > > >> > > if (is_method("CANCEL")) >> > > { >> > > >> > > >> > >> xlog("L_INFO","################################################################################ >> > >> > > \n"); >> > > xlog("L_INFO","CANCEL END OF CALL : call-id $ci \n"); >> > > >> > > >> > >> xlog("L_INFO","################################################################################ >> > >> > > \n"); >> > > xlog("L_INFO","setid: $avp(setid)\n"); >> > > if(rtpengine_manage("ICE=remove")) >> > > { >> > > xlog("L_INFO","SDP Offer: $ci for CANCEL\n"); >> > > } >> > > ............. >> > > >> > > >> > > >> > > >> > > >> > > >> > > _______________________________________________ >> > > Users mailing list >> > > Users at lists.opensips.org >> > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > >> > >> > -- >> > Răzvan Crainea >> > OpenSIPS Core Developer >> > http://www.opensips-solutions.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 >> > >> >> -- >> Răzvan Crainea >> OpenSIPS Core Developer >> http://www.opensips-solutions.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 julian.santer at rolmail.net Tue Nov 20 09:51:44 2018 From: julian.santer at rolmail.net (Julian Santer) Date: Tue, 20 Nov 2018 15:51:44 +0100 Subject: [OpenSIPS-Users] Nathelper keepalive issue with received column in usrloc In-Reply-To: <39a84907-a8a7-4423-0952-53b2802960ae@rolmail.net> References: <39a84907-a8a7-4423-0952-53b2802960ae@rolmail.net> Message-ID: <597fe3da-452d-115e-ae1a-fc1e1b3d9f28@rolmail.net> Hi guys, if I don't use the received column on the edge server, but I call fix_nated_contact instead, it seems to work. if (nat_uac_test("127")) {     fix_nated_contact(); } consume_credentials(); if (! add_path()) {     send_reply("500", "Internal path error, registration not stored");     xlog("L_ERR", "Adding PATH (with received) failed - LF_BASE");     exit; } Is this the right way or could I break something else with this change? Kind regards, Julian Santer Am 19.11.18 um 18:41 schrieb Julian Santer: > Hi guys, > > we need to switch from nat_traversal to nathelper. > The reason is the keepalive mechanism. > > The nat_traversal module sends OPTIONS with the following to header: sip:UAC_IP:UAC_PORT > Most of the UAC's answers with a 404 Not found. > On AVM Fritzbox with firmware >= 6.04, this OPTIONS may activate a security feature. > So after a certain time, the Fritzbox blocks all packages send from our proxy. > As we have ca. 80% AVM Fritzbox as UAC, we got a big problem. > So we deactivated the nat_keepalive vor this UAC's and we have to enable the keepalive Feature on the Fritzbox. > > The better solution would be, if we could send OPTIONS with a to header like: sip:username at UAC_IP:UAC_PORT. > As I understood the nathelper module could send OPTIONS like this. Because it is looking into the userloc table. Right? > > The nathelper module is on our edge server, the registrar on our core server. > For the "normal" UAC's (no received entry in usrloc) the keepalive's are now sent as expected. > But for the "nated" UAC's (received entry in usrloc) the keepalive's are like before: sip:UAC_IP:UAC_PORT (values in the received column from usrloc). > The REGISTER send to the core got the path header looking like: > Path: > > Is there a possibility to add the $fU on the received part of the path header (the user in the path module adds a string to the path part, but not > to the received part)? > Or is there a possiblity on the registrar to store the $fU in the received column? > On the nathelper keepalive mechanism I don't see any possibility to add the $fU. > > We are using the version 2.2.6 from the official debian source list. > > The config on the edge server's looks like: > #### nat helper module > loadmodule "nathelper.so" > modparam("nathelper", "natping_interval", 0) > modparam("nathelper", "ping_nated_only", 0) > modparam("nathelper", "natping_partitions", 1) > modparam("nathelper", "natping_tcp", 0) > > ### REGISTER > > $var(nat) = null; > > if (nat_uac_test("127")) > { >     $var(nat) = TRUE; > } > else > { >     $var(nat) = FALSE; > } > > consume_credentials(); > > if ($var(nat) == TRUE) > { >     if (! add_path_received()) >     { >         xlog("L_ERR", "Adding PATH (with received) failed - LF_BASE"); >         send_reply("500", "Internal path error, registration not stored"); >         exit; >     } > } > else > { >     if (! add_path()) >     { >         send_reply("500", "Internal path error, registration not stored"); >         xlog("L_ERR", "Adding PATH (with received) failed - LF_BASE"); >         exit; >     } > } > > route("R_RELAY_TO_REGISTRAR"); > exit; > > ### OPTIONS > > if (method=="OPTIONS") > { >     if ($si == "CORE") >     { >         topology_hiding("U"); >         if (! t_relay("0x05")) >         { >             send_reply("500", "Internal server error - failed to relay"); >             xlog("L_ERR", "Unable to relay OPTIONS - LF_BASE"); >         } >     } > } > > > The config on the core server looks like: > loadmodule "usrloc.so" > modparam("usrloc", "user_column",           "username") > modparam("usrloc", "domain_column",         "domain") > modparam("usrloc", "contact_column",        "contact") > modparam("usrloc", "expires_column",        "expires") > modparam("usrloc", "q_column",              "q") > modparam("usrloc", "callid_column",         "callid") > modparam("usrloc", "cseq_column",           "cseq") > modparam("usrloc", "methods_column",        "methods") > modparam("usrloc", "flags_column",          "flags") > modparam("usrloc", "user_agent_column",     "user_agent") > modparam("usrloc", "received_column",       "received") > modparam("usrloc", "path_column",           "path") > modparam("usrloc", "socket_column",         "socket") > modparam("usrloc", "use_domain",            0) > modparam("usrloc", "desc_time_order",       0) > modparam("usrloc", "timer_interval",        60) > modparam("usrloc", "db_url",                "DBURL") > modparam("usrloc", "db_mode",               2) > modparam("usrloc", "matching_mode",         0) > modparam("usrloc", "cseq_delay",            20) > modparam("usrloc", "nat_bflag",             6) > > #### nat helper module > loadmodule "nathelper.so" > modparam("nathelper", "natping_interval", 56) > modparam("nathelper", "ping_nated_only", 0) > modparam("nathelper", "natping_partitions", 1) > modparam("nathelper", "sipping_bflag", 8) > modparam("nathelper", "sipping_from", "sip:keepalive at DEFAULT_REALM") > modparam("nathelper", "sipping_method", "OPTIONS") > > # We want to send a keepalive on each registered UAC > if (proto == UDP) > { >     setbflag(8); >     xlog("L_INFO", "Nat keepalive sip_ping_flag - LF_BASE"); > } > > if (! save("location", "vp1")) > { >     xlog("L_ERR", "Saving contact from EDGE failed - LF_BASE"); >     exit; > } > > Thank you for any hint. > > Kind regards, > Julian Santer > Raiffeisen OnLine > > ps: @Bogdan: this is why we have ca. 550 entry's in the address table (permission module). If we solve the keepalives, only ca. 50 entry's are > remaining. From julian.santer at rolmail.net Tue Nov 20 10:01:19 2018 From: julian.santer at rolmail.net (Julian Santer) Date: Tue, 20 Nov 2018 16:01:19 +0100 Subject: [OpenSIPS-Users] Nathelper keepalive issue with received column in usrloc In-Reply-To: <597fe3da-452d-115e-ae1a-fc1e1b3d9f28@rolmail.net> References: <39a84907-a8a7-4423-0952-53b2802960ae@rolmail.net> <597fe3da-452d-115e-ae1a-fc1e1b3d9f28@rolmail.net> Message-ID: Hi guys, as I see in the tcpdump: - with the received header the R-RURI in the INVITE looks like: Request-Line: INVITE sip:dev-lab1 at 192.168.44.101:40885;line=fjafxbr9 SIP/2.0 - without the received header the R-URI in the INVITE looks like: Request-Line: INVITE sip:dev-lab1 at 212.46.162.97:40885;line=7wsv12yg SIP/2.0 Could the INVITE been rejeced/dropped by the UAC, if the R-RURI contains the public instead the private IP or another port etc.? From UAC perspective, the R-RURI (puclic IP) by incoming packets doesn't match the R-RURI (private IP) by outgoing packets. Kind regards, Julian Santer Am 20.11.18 um 15:51 schrieb Julian Santer: > Hi guys, > > if I don't use the received column on the edge server, but I call fix_nated_contact instead, it seems to work. > > if (nat_uac_test("127")) > { >     fix_nated_contact(); > } > > consume_credentials(); > > if (! add_path()) > { >     send_reply("500", "Internal path error, registration not stored"); >     xlog("L_ERR", "Adding PATH (with received) failed - LF_BASE"); >     exit; > } > > Is this the right way or could I break something else with this change? > > > Kind regards, > > Julian Santer > > Am 19.11.18 um 18:41 schrieb Julian Santer: >> Hi guys, >> >> we need to switch from nat_traversal to nathelper. >> The reason is the keepalive mechanism. >> >> The nat_traversal module sends OPTIONS with the following to header: sip:UAC_IP:UAC_PORT >> Most of the UAC's answers with a 404 Not found. >> On AVM Fritzbox with firmware >= 6.04, this OPTIONS may activate a security feature. >> So after a certain time, the Fritzbox blocks all packages send from our proxy. >> As we have ca. 80% AVM Fritzbox as UAC, we got a big problem. >> So we deactivated the nat_keepalive vor this UAC's and we have to enable the keepalive Feature on the Fritzbox. >> >> The better solution would be, if we could send OPTIONS with a to header like: sip:username at UAC_IP:UAC_PORT. >> As I understood the nathelper module could send OPTIONS like this. Because it is looking into the userloc table. Right? >> >> The nathelper module is on our edge server, the registrar on our core server. >> For the "normal" UAC's (no received entry in usrloc) the keepalive's are now sent as expected. >> But for the "nated" UAC's (received entry in usrloc) the keepalive's are like before: sip:UAC_IP:UAC_PORT (values in the received column from usrloc). >> The REGISTER send to the core got the path header looking like: >> Path: >> >> Is there a possibility to add the $fU on the received part of the path header (the user in the path module adds a string to the path part, but not >> to the received part)? >> Or is there a possiblity on the registrar to store the $fU in the received column? >> On the nathelper keepalive mechanism I don't see any possibility to add the $fU. >> >> We are using the version 2.2.6 from the official debian source list. >> >> The config on the edge server's looks like: >> #### nat helper module >> loadmodule "nathelper.so" >> modparam("nathelper", "natping_interval", 0) >> modparam("nathelper", "ping_nated_only", 0) >> modparam("nathelper", "natping_partitions", 1) >> modparam("nathelper", "natping_tcp", 0) >> >> ### REGISTER >> >> $var(nat) = null; >> >> if (nat_uac_test("127")) >> { >>     $var(nat) = TRUE; >> } >> else >> { >>     $var(nat) = FALSE; >> } >> >> consume_credentials(); >> >> if ($var(nat) == TRUE) >> { >>     if (! add_path_received()) >>     { >>         xlog("L_ERR", "Adding PATH (with received) failed - LF_BASE"); >>         send_reply("500", "Internal path error, registration not stored"); >>         exit; >>     } >> } >> else >> { >>     if (! add_path()) >>     { >>         send_reply("500", "Internal path error, registration not stored"); >>         xlog("L_ERR", "Adding PATH (with received) failed - LF_BASE"); >>         exit; >>     } >> } >> >> route("R_RELAY_TO_REGISTRAR"); >> exit; >> >> ### OPTIONS >> >> if (method=="OPTIONS") >> { >>     if ($si == "CORE") >>     { >>         topology_hiding("U"); >>         if (! t_relay("0x05")) >>         { >>             send_reply("500", "Internal server error - failed to relay"); >>             xlog("L_ERR", "Unable to relay OPTIONS - LF_BASE"); >>         } >>     } >> } >> >> >> The config on the core server looks like: >> loadmodule "usrloc.so" >> modparam("usrloc", "user_column",           "username") >> modparam("usrloc", "domain_column",         "domain") >> modparam("usrloc", "contact_column",        "contact") >> modparam("usrloc", "expires_column",        "expires") >> modparam("usrloc", "q_column",              "q") >> modparam("usrloc", "callid_column",         "callid") >> modparam("usrloc", "cseq_column",           "cseq") >> modparam("usrloc", "methods_column",        "methods") >> modparam("usrloc", "flags_column",          "flags") >> modparam("usrloc", "user_agent_column",     "user_agent") >> modparam("usrloc", "received_column",       "received") >> modparam("usrloc", "path_column",           "path") >> modparam("usrloc", "socket_column",         "socket") >> modparam("usrloc", "use_domain",            0) >> modparam("usrloc", "desc_time_order",       0) >> modparam("usrloc", "timer_interval",        60) >> modparam("usrloc", "db_url",                "DBURL") >> modparam("usrloc", "db_mode",               2) >> modparam("usrloc", "matching_mode",         0) >> modparam("usrloc", "cseq_delay",            20) >> modparam("usrloc", "nat_bflag",             6) >> >> #### nat helper module >> loadmodule "nathelper.so" >> modparam("nathelper", "natping_interval", 56) >> modparam("nathelper", "ping_nated_only", 0) >> modparam("nathelper", "natping_partitions", 1) >> modparam("nathelper", "sipping_bflag", 8) >> modparam("nathelper", "sipping_from", "sip:keepalive at DEFAULT_REALM") >> modparam("nathelper", "sipping_method", "OPTIONS") >> >> # We want to send a keepalive on each registered UAC >> if (proto == UDP) >> { >>     setbflag(8); >>     xlog("L_INFO", "Nat keepalive sip_ping_flag - LF_BASE"); >> } >> >> if (! save("location", "vp1")) >> { >>     xlog("L_ERR", "Saving contact from EDGE failed - LF_BASE"); >>     exit; >> } >> >> Thank you for any hint. >> >> Kind regards, >> Julian Santer >> Raiffeisen OnLine >> >> ps: @Bogdan: this is why we have ca. 550 entry's in the address table (permission module). If we solve the keepalives, only ca. 50 entry's are >> remaining. From johnkiniston at gmail.com Tue Nov 20 18:34:59 2018 From: johnkiniston at gmail.com (John Kiniston) Date: Tue, 20 Nov 2018 16:34:59 -0700 Subject: [OpenSIPS-Users] Crash with nathelper Message-ID: I've just started playing with TLS and turned on nathelper and I'm seeing crashes. What information do I need to collect for a bug report and to hopefully diagnose this issue? Nov 20 16:32:35 sip2 /usr/sbin/opensips[867]: ERROR:nathelper:nh_timer: sip msg_send failed Nov 20 16:32:35 sip2 /usr/sbin/opensips[860]: CRITICAL:core:sig_usr: segfault in process pid: 860, id: 15 Nov 20 16:32:35 sip2 kernel: opensips[860]: segfault at 7feaffffffff ip 00007fead73a0324 sp 00007ffd39239110 error 4 in nathelper.so[7fead739b000+12000] Nov 20 16:32:35 sip2 abrt-hook-ccpp: Can't open 'core.860' at '/': Permission denied Nov 20 16:32:35 sip2 abrt-hook-ccpp: Process 860 (opensips) of user 996 killed by SIGSEGV - dumping core Nov 20 16:32:35 sip2 /usr/sbin/opensips[868]: CRITICAL:core:handle_tcp_worker: dead tcp worker 1 (EOF received), pid 860 Nov 20 16:32:35 sip2 /usr/sbin/opensips[868]: CRITICAL:core:handle_worker: dead child 15 (EOF received), pid 860 Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: INFO:core:handle_sigs: child process 860 exited by a signal 11 Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: INFO:core:handle_sigs: core was generated Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: INFO:core:handle_sigs: terminating due to SIGCHLD Nov 20 16:32:35 sip2 /usr/sbin/opensips[846]: INFO:core:sig_usr: signal 15 received Nov 20 16:32:35 sip2 /usr/sbin/opensips[847]: INFO:core:sig_usr: signal 15 received Nov 20 16:32:35 sip2 /usr/sbin/opensips[848]: INFO:core:sig_usr: signal 15 received Nov 20 16:32:35 sip2 /usr/sbin/opensips[849]: INFO:core:sig_usr: signal 15 received Nov 20 16:32:35 sip2 /usr/sbin/opensips[850]: INFO:core:sig_usr: signal 15 received Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: INFO:core:shutdown_opensips: process 1(846) [HTTPD 192.168.84.176:8888] terminated, still waiting for 21 more Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: INFO:core:shutdown_opensips: process 2(847) [event-route handler] terminated, still waiting for 20 more Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: INFO:core:shutdown_opensips: process 3(848) [MI FIFO] terminated, still waiting for 19 more Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: INFO:core:shutdown_opensips: process 4(849) [time_keeper] terminated, still waiting for 18 more Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: INFO:core:shutdown_opensips: process 5(850) [timer] terminated, still waiting for 17 more Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: INFO:core:shutdown_opensips: process 6(851) [SIP receiver udp:67.212.192.99:5060 ] terminated, still waiting for 16 more Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: INFO:core:shutdown_opensips: process 7(852) [SIP receiver udp:67.212.192.99:5060 ] terminated, still waiting for 15 more Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: INFO:core:shutdown_opensips: process 8(853) [SIP receiver udp:67.212.192.99:5060 ] terminated, still waiting for 14 more Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: INFO:core:shutdown_opensips: process 9(854) [SIP receiver udp:67.212.192.99:5060 ] terminated, still waiting for 13 more Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: INFO:core:shutdown_opensips: process 10(855) [SIP receiver udp:67.212.192.99:3478 ] terminated, still waiting for 12 more Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: INFO:core:shutdown_opensips: process 11(856) [SIP receiver udp:67.212.192.99:3478 ] terminated, still waiting for 11 more Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: INFO:core:shutdown_opensips: process 12(857) [SIP receiver udp:67.212.192.99:3478 ] terminated, still waiting for 10 more Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: INFO:core:shutdown_opensips: process 13(858) [SIP receiver udp:67.212.192.99:3478 ] terminated, still waiting for 9 more Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: INFO:core:shutdown_opensips: process 14(859) [TCP receiver] terminated, still waiting for 8 more Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: INFO:core:shutdown_opensips: process 16(861) [TCP receiver] terminated, still waiting for 7 more Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: INFO:core:shutdown_opensips: process 17(862) [TCP receiver] terminated, still waiting for 6 more Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: INFO:core:shutdown_opensips: process 18(863) [TCP receiver] terminated, still waiting for 5 more Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: INFO:core:shutdown_opensips: process 19(864) [TCP receiver] terminated, still waiting for 4 more Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: INFO:core:shutdown_opensips: process 20(865) [TCP receiver] terminated, still waiting for 3 more Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: INFO:core:shutdown_opensips: process 21(866) [TCP receiver] terminated, still waiting for 2 more Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: INFO:core:shutdown_opensips: process 22(867) [Timer handler] terminated, still waiting for 1 more Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: INFO:core:shutdown_opensips: process 23(868) [TCP main] terminated, still waiting for 0 more Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: INFO:core:cleanup: cleanup Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: NOTICE:event_route:destroy: destroy module ... Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: INFO:pike:pike_exit: destroying... Nov 20 16:32:36 sip2 systemd: opensips.service: main process exited, code=exited, status=139/n/a Nov 20 16:32:36 sip2 systemd: opensips.service: control process exited, code=exited status=1 Nov 20 16:32:36 sip2 pkill: pkill: pidfile not valid Nov 20 16:32:36 sip2 pkill: Try `pkill --help' for more information. Nov 20 16:32:36 sip2 systemd: Unit opensips.service entered failed state. Nov 20 16:32:36 sip2 systemd: opensips.service failed. Nov 20 16:32:36 sip2 abrt-server: Duplicate: core backtrace Nov 20 16:32:36 sip2 abrt-server: DUP_OF_DIR: /var/spool/abrt/ccpp-2018-11-20-14:29:38-26330 Nov 20 16:32:36 sip2 abrt-server: Deleting problem directory ccpp-2018-11-20-16:32:35-860 (dup of ccpp-2018-11-20-14:29:38-26330) Nov 20 16:32:36 sip2 systemd: opensips.service holdoff time over, scheduling restart. Nov 20 16:32:36 sip2 dbus[678]: [system] Activating service name='org.freedesktop.problems' (using servicehelper) Nov 20 16:32:36 sip2 systemd: Starting OpenSIPS is a very fast and flexible SIP (RFC3261) server... Nov 20 16:32:36 sip2 dbus[678]: [system] Activated service 'org.freedesktop.problems' failed: Failed to execute program /lib64/dbus-1/dbus-daemon-launch-helper: Success Nov 20 16:32:36 sip2 dbus[678]: [system] Activating service name='org.freedesktop.problems' (using servicehelper) Nov 20 16:32:36 sip2 dbus[678]: [system] Activated service 'org.freedesktop.problems' failed: Failed to execute program /lib64/dbus-1/dbus-daemon-launch-hel -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Wed Nov 21 04:15:59 2018 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Wed, 21 Nov 2018 11:15:59 +0200 Subject: [OpenSIPS-Users] Crash with nathelper In-Reply-To: References: Message-ID: <206a6859-a54a-0434-6880-f6eae1b94bbb@opensips.org> Hi, John! Please send us the output of `opensips -V`. Best regards, Răzvan On 11/21/18 1:34 AM, John Kiniston wrote: > I've just started playing with TLS and turned on nathelper and I'm > seeing crashes. > > What information do I need to collect for a bug report and to hopefully > diagnose this issue? > > Nov 20 16:32:35 sip2 /usr/sbin/opensips[867]: ERROR:nathelper:nh_timer: > sip msg_send failed > Nov 20 16:32:35 sip2 /usr/sbin/opensips[860]: CRITICAL:core:sig_usr: > segfault in process pid: 860, id: 15 > Nov 20 16:32:35 sip2 kernel: opensips[860]: segfault at 7feaffffffff ip > 00007fead73a0324 sp 00007ffd39239110 error 4 in > nathelper.so[7fead739b000+12000] > Nov 20 16:32:35 sip2 abrt-hook-ccpp: Can't open 'core.860' at '/': > Permission denied > Nov 20 16:32:35 sip2 abrt-hook-ccpp: Process 860 (opensips) of user 996 > killed by SIGSEGV - dumping core > Nov 20 16:32:35 sip2 /usr/sbin/opensips[868]: > CRITICAL:core:handle_tcp_worker: dead tcp worker 1 (EOF received), pid 860 > Nov 20 16:32:35 sip2 /usr/sbin/opensips[868]: > CRITICAL:core:handle_worker: dead child 15 (EOF received), pid 860 > Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: INFO:core:handle_sigs: > child process 860 exited by a signal 11 > Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: INFO:core:handle_sigs: > core was generated > Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: INFO:core:handle_sigs: > terminating due to SIGCHLD > Nov 20 16:32:35 sip2 /usr/sbin/opensips[846]: INFO:core:sig_usr: signal > 15 received > Nov 20 16:32:35 sip2 /usr/sbin/opensips[847]: INFO:core:sig_usr: signal > 15 received > Nov 20 16:32:35 sip2 /usr/sbin/opensips[848]: INFO:core:sig_usr: signal > 15 received > Nov 20 16:32:35 sip2 /usr/sbin/opensips[849]: INFO:core:sig_usr: signal > 15 received > Nov 20 16:32:35 sip2 /usr/sbin/opensips[850]: INFO:core:sig_usr: signal > 15 received > Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: > INFO:core:shutdown_opensips: process 1(846) [HTTPD 192.168.84.176:8888 > ] terminated, still waiting for 21 more > Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: > INFO:core:shutdown_opensips: process 2(847) [event-route handler] > terminated, still waiting for 20 more > Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: > INFO:core:shutdown_opensips: process 3(848) [MI FIFO] terminated, still > waiting for 19 more > Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: > INFO:core:shutdown_opensips: process 4(849) [time_keeper] terminated, > still waiting for 18 more > Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: > INFO:core:shutdown_opensips: process 5(850) [timer] terminated, still > waiting for 17 more > Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: > INFO:core:shutdown_opensips: process 6(851) [SIP receiver > udp:67.212.192.99:5060 ] terminated, still > waiting for 16 more > Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: > INFO:core:shutdown_opensips: process 7(852) [SIP receiver > udp:67.212.192.99:5060 ] terminated, still > waiting for 15 more > Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: > INFO:core:shutdown_opensips: process 8(853) [SIP receiver > udp:67.212.192.99:5060 ] terminated, still > waiting for 14 more > Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: > INFO:core:shutdown_opensips: process 9(854) [SIP receiver > udp:67.212.192.99:5060 ] terminated, still > waiting for 13 more > Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: > INFO:core:shutdown_opensips: process 10(855) [SIP receiver > udp:67.212.192.99:3478 ] terminated, still > waiting for 12 more > Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: > INFO:core:shutdown_opensips: process 11(856) [SIP receiver > udp:67.212.192.99:3478 ] terminated, still > waiting for 11 more > Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: > INFO:core:shutdown_opensips: process 12(857) [SIP receiver > udp:67.212.192.99:3478 ] terminated, still > waiting for 10 more > Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: > INFO:core:shutdown_opensips: process 13(858) [SIP receiver > udp:67.212.192.99:3478 ] terminated, still > waiting for 9 more > Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: > INFO:core:shutdown_opensips: process 14(859) [TCP receiver] terminated, > still waiting for 8 more > Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: > INFO:core:shutdown_opensips: process 16(861) [TCP receiver] terminated, > still waiting for 7 more > Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: > INFO:core:shutdown_opensips: process 17(862) [TCP receiver] terminated, > still waiting for 6 more > Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: > INFO:core:shutdown_opensips: process 18(863) [TCP receiver] terminated, > still waiting for 5 more > Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: > INFO:core:shutdown_opensips: process 19(864) [TCP receiver] terminated, > still waiting for 4 more > Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: > INFO:core:shutdown_opensips: process 20(865) [TCP receiver] terminated, > still waiting for 3 more > Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: > INFO:core:shutdown_opensips: process 21(866) [TCP receiver] terminated, > still waiting for 2 more > Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: > INFO:core:shutdown_opensips: process 22(867) [Timer handler] terminated, > still waiting for 1 more > Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: > INFO:core:shutdown_opensips: process 23(868) [TCP main] terminated, > still waiting for 0 more > Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: INFO:core:cleanup: cleanup > Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: > NOTICE:event_route:destroy: destroy module ... > Nov 20 16:32:35 sip2 /usr/sbin/opensips[840]: INFO:pike:pike_exit: > destroying... > Nov 20 16:32:36 sip2 systemd: opensips.service: main process exited, > code=exited, status=139/n/a > Nov 20 16:32:36 sip2 systemd: opensips.service: control process exited, > code=exited status=1 > Nov 20 16:32:36 sip2 pkill: pkill: pidfile not valid > Nov 20 16:32:36 sip2 pkill: Try `pkill --help' for more information. > Nov 20 16:32:36 sip2 systemd: Unit opensips.service entered failed state. > Nov 20 16:32:36 sip2 systemd: opensips.service failed. > Nov 20 16:32:36 sip2 abrt-server: Duplicate: core backtrace > Nov 20 16:32:36 sip2 abrt-server: DUP_OF_DIR: > /var/spool/abrt/ccpp-2018-11-20-14:29:38-26330 > Nov 20 16:32:36 sip2 abrt-server: Deleting problem directory > ccpp-2018-11-20-16:32:35-860 (dup of ccpp-2018-11-20-14:29:38-26330) > Nov 20 16:32:36 sip2 systemd: opensips.service holdoff time over, > scheduling restart. > Nov 20 16:32:36 sip2 dbus[678]: [system] Activating service > name='org.freedesktop.problems' (using servicehelper) > Nov 20 16:32:36 sip2 systemd: Starting OpenSIPS is a very fast and > flexible SIP (RFC3261) server... > Nov 20 16:32:36 sip2 dbus[678]: [system] Activated service > 'org.freedesktop.problems' failed: Failed to execute program > /lib64/dbus-1/dbus-daemon-launch-helper: Success > Nov 20 16:32:36 sip2 dbus[678]: [system] Activating service > name='org.freedesktop.problems' (using servicehelper) > Nov 20 16:32:36 sip2 dbus[678]: [system] Activated service > 'org.freedesktop.problems' failed: Failed to execute program > /lib64/dbus-1/dbus-daemon-launch-hel > > -- > A human being should be able to change a diaper, plan an invasion, > butcher a hog, conn a ship, design a building, write a sonnet, balance > accounts, build a wall, set a bone, comfort the dying, take orders, give > orders, cooperate, act alone, solve equations, analyze a new problem, > pitch manure, program a computer, cook a tasty meal, fight efficiently, > die gallantly. Specialization is for insects. > ---Heinlein > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com From kurgan-rus at inbox.ru Thu Nov 22 03:01:15 2018 From: kurgan-rus at inbox.ru (=?UTF-8?B?QWxleGV5IEthemFudHNldg==?=) Date: Thu, 22 Nov 2018 11:01:15 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?uac=5Fregistrant_clustering?= In-Reply-To: <1542292998279-0.post@n2.nabble.com> References: <1542190000.342851450@f436.i.mail.ru> <77c94162-6e18-d808-42b9-5eaedaa2f700@opensips.org> <1542292998279-0.post@n2.nabble.com> Message-ID: <1542873675.306728524@f407.i.mail.ru> assuming that un-registration is just a REGISTER request with Expiry: 0, maybe this will not be a difficult task for our respected developers to implement this :) >vasilevalex : > >... >But when we have Active/Passive cluster with virtual IP (both servers >running), it's better to have some way to disable outbound registrations on >backup server and to enable them when switching Backup->Active. > ----------------------------------------------- BR, Alexey http://alexeyka.zantsev.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From johan at democon.be Fri Nov 23 01:41:31 2018 From: johan at democon.be (Johan De Clercq) Date: Fri, 23 Nov 2018 06:41:31 +0000 Subject: [OpenSIPS-Users] Im with message. Message-ID: Hi, I need to be able to handle SIP MESSAGE for IM. Does this mean that I need to setup openxcap and use xcap client module in my script? Best regards, Outlook voor iOS downloaden -------------- next part -------------- An HTML attachment was scrubbed... URL: From vitalik.voip at gmail.com Fri Nov 23 05:34:41 2018 From: vitalik.voip at gmail.com (Vitalii Aleksandrov) Date: Fri, 23 Nov 2018 12:34:41 +0200 Subject: [OpenSIPS-Users] t_on_reply() behavior Message-ID: <313adf78-404b-d9ae-9fe5-b9065d15b39c@gmail.com> Hi, TM documentation mentions that it's possible to call t_on_reply() from a branch_route and it will set a branch specific reply route. I understood that if t_on_reply() is set for a branch it overwrites the global reply_route and the global one won't be called for that branch anymore. Test shows that branch on_reply is added into a list and after its invocation the global one is also called. Is it expected behavior? Maybe we can update documentation to make it clear regarding this detail. From bogdan at opensips.org Fri Nov 23 08:45:42 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 23 Nov 2018 15:45:42 +0200 Subject: [OpenSIPS-Users] Im with message. In-Reply-To: References: Message-ID: No :) Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/23/2018 08:41 AM, Johan De Clercq wrote: > Hi, I need to be able to handle SIP MESSAGE for IM. Does this mean > that I need to setup openxcap and use xcap client module in my script? > > Best regards, > > Outlook voor iOS downloaden > > > _______________________________________________ > 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 ag at ag-projects.com Fri Nov 23 09:20:01 2018 From: ag at ag-projects.com (Adrian Georgescu) Date: Fri, 23 Nov 2018 11:20:01 -0300 Subject: [OpenSIPS-Users] Im with message. In-Reply-To: References: Message-ID: SIP MESSAGE uses only the SIP proxy, you don’t need other components. Adrian > On 23 Nov 2018, at 03:41, Johan De Clercq wrote: > > Hi, I need to be able to handle SIP MESSAGE for IM. Does this mean that I need to setup openxcap and use xcap client module in my script? > > Best regards, > > Outlook voor iOS downloaden > _______________________________________________ > 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: signature.asc Type: application/pgp-signature Size: 235 bytes Desc: Message signed with OpenPGP URL: From Johan at democon.be Fri Nov 23 09:35:36 2018 From: Johan at democon.be (Johan De Clercq) Date: Fri, 23 Nov 2018 15:35:36 +0100 Subject: [OpenSIPS-Users] Im with message. In-Reply-To: References: Message-ID: Okay, so I just can handle it trhough my normal script as I would do with an initial INVITE ? Secondly, when exactly is XCAP needed ? And where does this fit with SIP SIMPLE ? I am really sorry but I am not really used to presence and IM. Op vr 23 nov. 2018 om 14:45 schreef Bogdan-Andrei Iancu : > No :) > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Bootcamp 2018 > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 11/23/2018 08:41 AM, Johan De Clercq wrote: > > Hi, I need to be able to handle SIP MESSAGE for IM. Does this mean that I > need to setup openxcap and use xcap client module in my script? > > Best regards, > > Outlook voor iOS downloaden > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From ag at ag-projects.com Fri Nov 23 15:28:39 2018 From: ag at ag-projects.com (Adrian Georgescu) Date: Fri, 23 Nov 2018 17:28:39 -0300 Subject: [OpenSIPS-Users] Im with message. In-Reply-To: References: Message-ID: <5E982A3E-843C-4D7D-AE35-F487E4672C5B@ag-projects.com> SIP MESSAGE has nothing to do with SIP SIMPLE suite of standards. SIP SIMPLE is a suite of standards for session based IM and file transfers using MSRP media plane and Presence. Presence requires contacts management for which XCAP server is used. XCAP is built on top of an HTTP server by serving XML documents defined for contacts and policies used by Presence. These documents can be changed by a SIP client with XCAP support. More information about the SIP SIMPLE standards are found here: https://datatracker.ietf.org/wg/simple/documents/ Adrian > On 23 Nov 2018, at 11:35, Johan De Clercq wrote: > > Okay, so I just can handle it trhough my normal script as I would do with an initial INVITE ? > Secondly, when exactly is XCAP needed ? > And where does this fit with SIP SIMPLE ? > > I am really sorry but I am not really used to presence and IM. > > Op vr 23 nov. 2018 om 14:45 schreef Bogdan-Andrei Iancu >: > No :) > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Bootcamp 2018 > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > On 11/23/2018 08:41 AM, Johan De Clercq wrote: >> Hi, I need to be able to handle SIP MESSAGE for IM. Does this mean that I need to setup openxcap and use xcap client module in my script? >> >> Best regards, >> >> Outlook voor iOS downloaden >> >> >> _______________________________________________ >> 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: -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 235 bytes Desc: Message signed with OpenPGP URL: From z at startit.ru Sat Nov 24 05:18:16 2018 From: z at startit.ru (zzz) Date: Sat, 24 Nov 2018 10:18:16 +0000 Subject: [OpenSIPS-Users] snmp stat Message-ID: <8B36F227BD22B041AEA7015FD914CD95038176482B@JET-EX02.jettel.ru> Hello, I'm able to collect total Opensips load with SNMP OID .1.3.6.1.4.1.27483.3.1.3.1.3.2.1.0 Is it possible to get snmp load stats per gateway (carrier) which I have in the dr_gateways (dr_carriers) table ? Thsnks, Yuriy -------------- next part -------------- An HTML attachment was scrubbed... URL: From johnkiniston at gmail.com Mon Nov 26 10:19:21 2018 From: johnkiniston at gmail.com (John Kiniston) Date: Mon, 26 Nov 2018 08:19:21 -0700 Subject: [OpenSIPS-Users] Crash with nathelper In-Reply-To: <206a6859-a54a-0434-6880-f6eae1b94bbb@opensips.org> References: <206a6859-a54a-0434-6880-f6eae1b94bbb@opensips.org> Message-ID: version: opensips 2.4.3 (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, sigio_rt, select. main.c compiled on 01:38:42 Nov 9 2018 with gcc 4.8.5 On Wed, Nov 21, 2018 at 2:19 AM Răzvan Crainea wrote: > Hi, John! > > Please send us the output of `opensips -V`. > > Best regards, > Răzvan > > On 11/21/18 1:34 AM, John Kiniston wrote: > > I've just started playing with TLS and turned on nathelper and I'm > > seeing crashes. > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From vishalmpai at gmail.com Mon Nov 26 11:42:38 2018 From: vishalmpai at gmail.com (Vishal Pai) Date: Mon, 26 Nov 2018 22:12:38 +0530 Subject: [OpenSIPS-Users] Replace $tU value Message-ID: Hello Everyone Is it possible to update the $tU value to new one. Let say $tU = 1234567890 need to overwrite to $tU = 9876543210 how i can achieve that. -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Mon Nov 26 12:22:25 2018 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 26 Nov 2018 17:22:25 +0000 Subject: [OpenSIPS-Users] Replace $tU value In-Reply-To: References: Message-ID: Just do: $tU = 9876543210 Simple :) On Mon, 26 Nov 2018 at 16:46, Vishal Pai wrote: > Hello Everyone > > Is it possible to update the $tU value to new one. Let say $tU = > 1234567890 need to overwrite to $tU = 9876543210 how i can achieve that. > _______________________________________________ > 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 Nov 26 12:29:24 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 26 Nov 2018 19:29:24 +0200 Subject: [OpenSIPS-Users] Replace $tU value In-Reply-To: References: Message-ID: <1e181116-0088-83c6-0e46-17ce022e5914@opensips.org> Hi Vishal, use uac_replace_to() function: http://www.opensips.org/html/docs/modules/2.4.x/uac.html#func_uac_replace_from Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/26/2018 06:42 PM, Vishal Pai wrote: > Hello Everyone > > Is it possible to update the $tU value to new one. Let say $tU = > 1234567890 need to overwrite to $tU = 9876543210 how i can achieve that. > > > _______________________________________________ > 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 Nov 26 12:29:59 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 26 Nov 2018 19:29:59 +0200 Subject: [OpenSIPS-Users] Replace $tU value In-Reply-To: References: Message-ID: David, That does not work as the $tU is a read-only variable. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/26/2018 07:22 PM, David Villasmil wrote: > Just do: > > $tU = 9876543210 > > Simple :) > On Mon, 26 Nov 2018 at 16:46, Vishal Pai > wrote: > > Hello Everyone > > Is it possible to update the $tU value to new one. Let say $tU = > 1234567890 need to overwrite to $tU = 9876543210 how i can achieve > that. > _______________________________________________ > 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 vishalmpai at gmail.com Mon Nov 26 14:01:11 2018 From: vishalmpai at gmail.com (Vishal Pai) Date: Tue, 27 Nov 2018 00:31:11 +0530 Subject: [OpenSIPS-Users] Replace $tU value In-Reply-To: <1e181116-0088-83c6-0e46-17ce022e5914@opensips.org> References: <1e181116-0088-83c6-0e46-17ce022e5914@opensips.org> Message-ID: Hi Bogdan, Thank you for your quick response. I have tried both the way still unable to overwrite the value. uac_replace_to("$tU","9876543210"); uac_replace_to("9876543210","$tU"); Can you please let me know the right way to achieve this. Thanks On Mon, Nov 26, 2018 at 10:59 PM Bogdan-Andrei Iancu wrote: > Hi Vishal, > > use uac_replace_to() function: > > http://www.opensips.org/html/docs/modules/2.4.x/uac.html#func_uac_replace_from > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Bootcamp 2018 > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 11/26/2018 06:42 PM, Vishal Pai wrote: > > Hello Everyone > > Is it possible to update the $tU value to new one. Let say $tU = > 1234567890 need to overwrite to $tU = 9876543210 how i can achieve that. > > > _______________________________________________ > 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 Nov 26 15:50:18 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 26 Nov 2018 22:50:18 +0200 Subject: [OpenSIPS-Users] Replace $tU value In-Reply-To: References: <1e181116-0088-83c6-0e46-17ce022e5914@opensips.org> Message-ID: <036240cf-e17c-3bc1-6c7e-24b11ebf50be@opensips.org> Hi, uac_replace_to("sip:9876543210@$td"); should do it - you need to call it only for the initial INVITE. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/26/2018 09:01 PM, Vishal Pai wrote: > Hi Bogdan, > > Thank you for your quick response. > > I have tried both the way still unable to overwrite the value. > > uac_replace_to("$tU","9876543210"); > uac_replace_to("9876543210","$tU"); > > Can you please let me know the right way to achieve this. > > Thanks > > > On Mon, Nov 26, 2018 at 10:59 PM Bogdan-Andrei Iancu > > wrote: > > Hi Vishal, > > use uac_replace_to() function: > http://www.opensips.org/html/docs/modules/2.4.x/uac.html#func_uac_replace_from > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Bootcamp 2018 > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 11/26/2018 06:42 PM, Vishal Pai wrote: >> Hello Everyone >> >> Is it possible to update the $tU value to new one. Let say $tU = >> 1234567890 need to overwrite to $tU = 9876543210 how i can >> achieve that. >> >> >> _______________________________________________ >> 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 Nov 26 16:00:41 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 26 Nov 2018 23:00:41 +0200 Subject: [OpenSIPS-Users] t_on_reply() behavior In-Reply-To: <313adf78-404b-d9ae-9fe5-b9065d15b39c@gmail.com> References: <313adf78-404b-d9ae-9fe5-b9065d15b39c@gmail.com> Message-ID: Hi Vitalii, I can confirm (by checking the code) that both types of reply routes are executed - first the per-branch one and then the per-transaction one. I checked the docs (http://www.opensips.org/Documentation/Script-Routes-2-4#toc4) for errors, but I haven;t found the "overwriting" part - could you point me to place where you read that ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/23/2018 12:34 PM, Vitalii Aleksandrov wrote: > Hi, > > TM documentation mentions that it's possible to call t_on_reply() from > a branch_route and it will set a branch specific reply route. > I understood that if t_on_reply() is set for a branch it overwrites > the global reply_route and the global one won't be called for that > branch anymore. > Test shows that branch on_reply is added into a list and after its > invocation the global one is also called. > > Is it expected behavior? Maybe we can update documentation to make it > clear regarding this detail. > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From bogdan at opensips.org Mon Nov 26 16:03:08 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 26 Nov 2018 23:03:08 +0200 Subject: [OpenSIPS-Users] uac_registrant clustering In-Reply-To: <1542292998279-0.post@n2.nabble.com> References: <1542190000.342851450@f436.i.mail.ru> <1542272756241-0.post@n2.nabble.com> <77c94162-6e18-d808-42b9-5eaedaa2f700@opensips.org> <1542292998279-0.post@n2.nabble.com> Message-ID: Hi Alexei, Thanks for pointing it out and sorry for the mixing :( I agree on what you said, I already sync'ed with Razvan on the matter, such an option will be available for 3.0 Thanks and regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/15/2018 04:43 PM, vasilevalex wrote: > Hi Bogdan, > > I hope, I didn't confuse you as original question was by another Alexey. > > And I also think that sharing outbound registrations between two active > servers is something that has not much practical appliance. > But when we have Active/Passive cluster with virtual IP (both servers > running), it's better to have some way to disable outbound registrations on > backup server and to enable them when switching Backup->Active. > Perfect if it will be the same way, as Razvan wrote here: > https://github.com/OpenSIPS/opensips/issues/1532 > > > > -- > Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From bogdan at opensips.org Mon Nov 26 16:04:43 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 26 Nov 2018 23:04:43 +0200 Subject: [OpenSIPS-Users] uac_registrant clustering In-Reply-To: <1542873675.306728524@f407.i.mail.ru> References: <1542190000.342851450@f436.i.mail.ru> <77c94162-6e18-d808-42b9-5eaedaa2f700@opensips.org> <1542292998279-0.post@n2.nabble.com> <1542873675.306728524@f407.i.mail.ru> Message-ID: Hi Alexey, What you are suggesting is to have the module to "un-register" itself when going into backup mode according to the sharing tag ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/22/2018 10:01 AM, Alexey Kazantsev via Users wrote: > assuming that un-registration is just a REGISTER request with Expiry: 0, > maybe this will not be a difficult task for our respected developers to > implement this :) > > > vasilevalex : > > ... > But when we have Active/Passive cluster with virtual IP (both servers > running), it's better to have some way to disable outbound > registrations on > backup server and to enable them when switching Backup->Active. > > > > ----------------------------------------------- > BR, Alexey > http://alexeyka.zantsev.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 Nov 26 16:14:43 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 26 Nov 2018 23:14:43 +0200 Subject: [OpenSIPS-Users] snmp stat In-Reply-To: <8B36F227BD22B041AEA7015FD914CD95038176482B@JET-EX02.jettel.ru> References: <8B36F227BD22B041AEA7015FD914CD95038176482B@JET-EX02.jettel.ru> Message-ID: <51e73b7c-c7fe-94f4-e755-9f10d93a8268@opensips.org> Hi Yuriy, By default no, but you can generate such stats by using the "statistics" module - this module allows you to create and manage (inc/dec) stats from the script. Second step is use AgentX to grab the stats from OpenSIPS - via the MI interface - and publish them as mibs. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/24/2018 12:18 PM, zzz wrote: > > Hello, > > I’m able to collect total Opensips load with SNMP OID > .1.3.6.1.4.1.27483.3.1.3.1.3.2.1.0 > > Is it possible to get snmp load stats per gateway (carrier) which I > have in the dr_gateways (dr_carriers) table ? > > Thsnks, > > Yuriy > > > > _______________________________________________ > 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 z at startit.ru Mon Nov 26 16:53:20 2018 From: z at startit.ru (zzz) Date: Mon, 26 Nov 2018 21:53:20 +0000 Subject: [OpenSIPS-Users] snmp stat In-Reply-To: <51e73b7c-c7fe-94f4-e755-9f10d93a8268@opensips.org> References: <8B36F227BD22B041AEA7015FD914CD95038176482B@JET-EX02.jettel.ru> <51e73b7c-c7fe-94f4-e755-9f10d93a8268@opensips.org> Message-ID: <8B36F227BD22B041AEA7015FD914CD950381769A51@JET-EX02.jettel.ru> Hi Bogdan, As I understand I’ll need to update each variable individually for each GW. Shell I create as many variables for all gateways I have or the one variable would be enough ? Thanks, Yuriy From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Monday, November 26, 2018 10:15 PM To: OpenSIPS users mailling list ; zzz Subject: Re: [OpenSIPS-Users] snmp stat Hi Yuriy, By default no, but you can generate such stats by using the "statistics" module - this module allows you to create and manage (inc/dec) stats from the script. Second step is use AgentX to grab the stats from OpenSIPS - via the MI interface - and publish them as mibs. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/24/2018 12:18 PM, zzz wrote: Hello, I’m able to collect total Opensips load with SNMP OID .1.3.6.1.4.1.27483.3.1.3.1.3.2.1.0 Is it possible to get snmp load stats per gateway (carrier) which I have in the dr_gateways (dr_carriers) table ? Thsnks, Yuriy _______________________________________________ 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 Nov 27 04:55:25 2018 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Tue, 27 Nov 2018 11:55:25 +0200 Subject: [OpenSIPS-Users] Crash with nathelper In-Reply-To: References: <206a6859-a54a-0434-6880-f6eae1b94bbb@opensips.org> Message-ID: <0698980c-867b-12b4-a27e-0521f2565bc4@opensips.org> Hi, John! I see that you don't have permissions to write a corefile. Can you set the working directory (-w parameter) to a path that is writeable, to get a proper core file that we can examine further. Best regards, Răzvan On 11/26/18 5:19 PM, John Kiniston wrote: > version: opensips 2.4.3 (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, sigio_rt, select. > main.c compiled on 01:38:42 Nov  9 2018 with gcc 4.8.5 > > > On Wed, Nov 21, 2018 at 2:19 AM Răzvan Crainea > wrote: > > Hi, John! > > Please send us the output of `opensips -V`. > > Best regards, > Răzvan > > On 11/21/18 1:34 AM, John Kiniston wrote: > > I've just started playing with TLS and turned on nathelper and I'm > > seeing crashes. > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com From bogdan at opensips.org Tue Nov 27 06:09:09 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 27 Nov 2018 13:09:09 +0200 Subject: [OpenSIPS-Users] EBR and wait_for_event() timeout In-Reply-To: References: Message-ID: <2f50553c-368b-d48e-2376-768b6e12d49c@opensips.org> Hi Vitalii, The waiting for the event is not strong correlated with the transaction. The transaction may disappear while waiting. The waiting context is only keeping a reference to the transaction, so the t_inject_branch() can look it up (if still exists). So, if the transaction is canceled, the event waiting is not affected, it will still be triggered, but the t_inject will fail (either with transaction not found, either with not able to create more branches on a canceled transaction). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/16/2018 07:07 PM, Vitalii Aleksandrov wrote: > Thanks for the information. Have one more related question. > > What If I call somewhere, opensips calls wait_for_event() and before > the event happens or async timeout (will create a bug report) fired I > CANCEL the call. > Since async() keeps some context in transaction structure and this > transaction is already canceled should I expect that async() task is > also canceled and will never call a callback route? Or should I always > check t_was_cancelled() in the beginning of a callback route? > > >> Hi Vitalii, >> >> For the wait_for_event(), the timeout seems to have no effect, the >> waiting being for ever :-| . The transaction has no timeout as you >> didn;t sent out any branch yet (the transaction timeout is for >> waiting on replies). >> >> Could you please open bug report on the opensips tracker on github ? >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> OpenSIPS Bootcamp 2018 >> http://opensips.org/training/OpenSIPS_Bootcamp_2018/ >> >> On 11/14/2018 03:17 PM, Vitalii Aleksandrov wrote: >>> Hi, >>> >>> event_routing module provides the great async function >>> wait_for_event(). If script subscribes for a event and received it >>> it calls some "resume_route". >>> What I can't understand is what happens with a transaction if >>> wait_for_event() never catches an event and reaches its timeout. >>> Is the any way to continue script execution from the place where >>> "wait_for_event() was called or to execute some "timeout_route" to >>> handle transaction properly? >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> From vishalmpai at gmail.com Tue Nov 27 07:56:14 2018 From: vishalmpai at gmail.com (Vishal Pai) Date: Tue, 27 Nov 2018 18:26:14 +0530 Subject: [OpenSIPS-Users] Replace $tU value In-Reply-To: <036240cf-e17c-3bc1-6c7e-24b11ebf50be@opensips.org> References: <1e181116-0088-83c6-0e46-17ce022e5914@opensips.org> <036240cf-e17c-3bc1-6c7e-24b11ebf50be@opensips.org> Message-ID: It worked. Thank You Bogdan On Tue, Nov 27, 2018 at 2:20 AM Bogdan-Andrei Iancu wrote: > Hi, > > uac_replace_to("sip:9876543210@$td"); > > should do it - you need to call it only for the initial INVITE. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Bootcamp 2018 > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 11/26/2018 09:01 PM, Vishal Pai wrote: > > Hi Bogdan, > > Thank you for your quick response. > > I have tried both the way still unable to overwrite the value. > > uac_replace_to("$tU","9876543210"); > uac_replace_to("9876543210","$tU"); > > Can you please let me know the right way to achieve this. > > Thanks > > > On Mon, Nov 26, 2018 at 10:59 PM Bogdan-Andrei Iancu > wrote: > >> Hi Vishal, >> >> use uac_replace_to() function: >> >> http://www.opensips.org/html/docs/modules/2.4.x/uac.html#func_uac_replace_from >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> OpenSIPS Bootcamp 2018 >> http://opensips.org/training/OpenSIPS_Bootcamp_2018/ >> >> On 11/26/2018 06:42 PM, Vishal Pai wrote: >> >> Hello Everyone >> >> Is it possible to update the $tU value to new one. Let say $tU = >> 1234567890 need to overwrite to $tU = 9876543210 how i can achieve that. >> >> >> _______________________________________________ >> 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 Nov 27 08:58:56 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 27 Nov 2018 15:58:56 +0200 Subject: [OpenSIPS-Users] OpenSIPS Summit 2019 - Amsterdam, NL Message-ID: <1a9ba488-1e8a-3b47-7b83-969da3349e82@opensips.org> OpenSIPS Summit 2019 April 30 - May 3, 2019 Amsterdam, The Netherlands Once again, the OpenSIPS Summit will take place in Amsterdam, becoming the hotspot of VoIP & RTC world for *4 full days*.**Three**days of talks, inspiring presentations, amazing demos followed by one day of trainings about *OpenSIPS* and the *Open Source* ecosystem. Join us to learn what is new & important in VoIP & RTC, to discover how technology evolves and how you can benefit of it. *Just knowledge in its most pure format*. *Some Great Reasons to Attend* * Access the latest news, knowledge and experience in the VoIP & RTC world * Learn about upcoming 3.0 OpenSIPS release and how you can leverage it * Attend unique presentations and interactive technical workshops * Meet FOSS developers and community to share experience and comments * Get solutions consultancy during the Free Design Clinics * Become an Expert attending the OpenSIPS Advanced Training *Summit Agenda* * Two full days of presentations given by key speakers * Open Discussions with key people from OpenSIPS and other OSS projects * Interactive Demos and Showcases * Design Clinics to validate your OpenSIPS deployments * One full day of OpenSIPS Training (limited seats!) * Social events in the amazing Amsterdam *Be part of it* Be part of the OpenSIPS and the Open Source community, be part of the OpenSIPS Summit 2019. *Attend to learn* - the registration process will be open in the following days, stay tuned. Nevertheless, pre-registration is available, just contact us. *Speak to share* - the Call for Papers will be announced during next week, so you can share your wisdom and experience with the world. *Sponsor to help* - we welcome any help in making the Summit such a great event. Sponsoring is a natural way of saying "Thank you" for the Open Source code you are using within your businesses. Interested? Please contact our team or email us! * * *Radisson Blu** **Rusland 17, 1012CK Amsterdam, The Netherlands* Meet us again at our familiar Venue, with even more space and comfort than ever! This year the OpenSIPS Summit expands in size and will accommodate more participants and speakers. ** -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From vitalik.voip at gmail.com Tue Nov 27 10:02:41 2018 From: vitalik.voip at gmail.com (Vitalii Aleksandrov) Date: Tue, 27 Nov 2018 17:02:41 +0200 Subject: [OpenSIPS-Users] EBR and wait_for_event() timeout In-Reply-To: <2f50553c-368b-d48e-2376-768b6e12d49c@opensips.org> References: <2f50553c-368b-d48e-2376-768b6e12d49c@opensips.org> Message-ID: Hi Bogdan, Instead of t_inject_branches I use t_relay() from wait_for_event() callback. As far as I understood the logic is the same. t_relay should just fail if transaction connected to the waiting context has been canceled by that time. Thanks for the detailed reply. > Hi Vitalii, > > The waiting for the event is not strong correlated with the > transaction. The transaction may disappear while waiting. The waiting > context is only keeping a reference to the transaction, so the > t_inject_branch() can look it up (if still exists). > So, if the transaction is canceled, the event waiting is not affected, > it will still be triggered, but the t_inject will fail (either with > transaction not found, either with not able to create more branches on > a canceled transaction). > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer >   http://www.opensips-solutions.com > OpenSIPS Bootcamp 2018 >   http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 11/16/2018 07:07 PM, Vitalii Aleksandrov wrote: >> Thanks for the information. Have one more related question. >> >> What If I call somewhere, opensips calls wait_for_event() and before >> the event happens or async timeout (will create a bug report) fired I >> CANCEL the call. >> Since async() keeps some context in transaction structure and this >> transaction is already canceled should I expect that async() task is >> also canceled and will never call a callback route? Or should I >> always check  t_was_cancelled() in the beginning of a callback route? >> >> >>> Hi Vitalii, >>> >>> For the wait_for_event(), the timeout seems to have no effect, the >>> waiting being for ever :-| . The transaction has no timeout as you >>> didn;t sent out any branch yet (the transaction timeout is for >>> waiting on replies). >>> >>> Could you please open bug report on the opensips tracker on github ? >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>>   http://www.opensips-solutions.com >>> OpenSIPS Bootcamp 2018 >>>   http://opensips.org/training/OpenSIPS_Bootcamp_2018/ >>> >>> On 11/14/2018 03:17 PM, Vitalii Aleksandrov wrote: >>>> Hi, >>>> >>>> event_routing module provides the great async function >>>> wait_for_event().  If script subscribes for a event and received it >>>> it calls some "resume_route". >>>> What I can't understand is what happens with a transaction if >>>> wait_for_event() never catches an event and reaches its timeout. >>>> Is the any way to continue script execution from the place where >>>> "wait_for_event() was called or to execute some "timeout_route" to >>>> handle transaction properly? >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> > From bogdan at opensips.org Tue Nov 27 10:34:30 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 27 Nov 2018 17:34:30 +0200 Subject: [OpenSIPS-Users] snmp stat In-Reply-To: <8B36F227BD22B041AEA7015FD914CD950381769A51@JET-EX02.jettel.ru> References: <8B36F227BD22B041AEA7015FD914CD95038176482B@JET-EX02.jettel.ru> <51e73b7c-c7fe-94f4-e755-9f10d93a8268@opensips.org> <8B36F227BD22B041AEA7015FD914CD950381769A51@JET-EX02.jettel.ru> Message-ID: Hi Yuriy, yes, you should do one stat per GW / carrier - this is not an issue as you can dynamically create as many stats you need. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 2019 https://www.opensips.org/events/Summit-2019Amsterdam/ On 11/26/2018 11:53 PM, zzz wrote: > > Hi Bogdan, > > As I understand I’ll need to update each variable individually for > each GW. > > Shell I create as many variables for all gateways I have or the one > variable would be enough ? > > Thanks, > > Yuriy > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* Monday, November 26, 2018 10:15 PM > *To:* OpenSIPS users mailling list ; zzz > > *Subject:* Re: [OpenSIPS-Users] snmp stat > > Hi Yuriy, > > By default no, but you can generate such stats by using the > "statistics" module - this module allows you to create and manage > (inc/dec) stats from the script. Second step is use AgentX to grab the > stats from OpenSIPS - via the MI interface - and publish them as mibs. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Bootcamp 2018 > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 11/24/2018 12:18 PM, zzz wrote: > > Hello, > > I’m able to collect total Opensips load with SNMP OID > .1.3.6.1.4.1.27483.3.1.3.1.3.2.1.0 > > Is it possible to get snmp load stats per gateway (carrier) which > I have in the dr_gateways (dr_carriers) table ? > > Thsnks, > > Yuriy > > > > > _______________________________________________ > > 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 Nov 27 11:55:57 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 27 Nov 2018 18:55:57 +0200 Subject: [OpenSIPS-Users] OpenSIPS Control Panel 8.2.4 was released In-Reply-To: <001001d48010$1bb732d0$53259870$@smartvox.co.uk> References: <001001d48010$1bb732d0$53259870$@smartvox.co.uk> Message-ID: <648ddd0f-12db-de37-e9d5-47aede7a0cb1@opensips.org> Hi John, I'm not excluding the issue, as the jumping between the tools in combination with the paging is a bit of a complex process. I can dive into if you could help me with a data set - an mysql dump for cdrs and siptrace to show the issue - I can import the tables and debug it (it is a bit of time consuming to try to collect the date by myself) - of course you can send the data off list :) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 2019 https://www.opensips.org/events/Summit-2019Amsterdam/ On 11/19/2018 03:59 PM, John Quick wrote: > Hello Bogdan, > > The link works okay now, but (I think) only if the selected call is on the > first page of results in the siptrace tab. > > John Quick > Smartvox Limited > > >> Hi John, >> >> I found a small typo that affected who the link was constructed . See >> > https://github.com/OpenSIPS/opensips-cp/commit/d50503123477f99b0079570361407 > 7b685ca4579 >> In order to link siptrace to cdrviewer, you need to (a) be sure homer >> tool is disabled and (b) siptrace tool is enabled. >> >> Let me know if this fix does the trick for you. >> >> Best regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com From johan at democon.be Tue Nov 27 14:13:59 2018 From: johan at democon.be (johan de clercq) Date: Tue, 27 Nov 2018 20:13:59 +0100 Subject: [OpenSIPS-Users] FW: segfault in opensips 2.4 In-Reply-To: <000f01d48685$30fb3990$92f1acb0$@democon.be> References: <000f01d48685$30fb3990$92f1acb0$@democon.be> Message-ID: <002d01d48685$59e6fe70$0db4fb50$@democon.be> Hello, I have received the following segfault in opensips kernel: [71330.742487] opensips[1268]: segfault at f0 ip 00007fc617abad0e sp 00007ffdcaa3c090 error 4 in registrar.so[7fc617aa9000+23000] Can somebody please explain what logs are needed ? Johan De Clercq, Managing Director Democon bvba - Ooigemstraat 41 - 8780 Oostrozebeke Tel +3256980990 - GSM +32478720104 -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 15602 bytes Desc: not available URL: From z at startit.ru Tue Nov 27 15:12:29 2018 From: z at startit.ru (zzz) Date: Tue, 27 Nov 2018 20:12:29 +0000 Subject: [OpenSIPS-Users] snmp stat In-Reply-To: References: <8B36F227BD22B041AEA7015FD914CD95038176482B@JET-EX02.jettel.ru> <51e73b7c-c7fe-94f4-e755-9f10d93a8268@opensips.org> <8B36F227BD22B041AEA7015FD914CD950381769A51@JET-EX02.jettel.ru> Message-ID: <8B36F227BD22B041AEA7015FD914CD95038176E01E@JET-EX02.jettel.ru> Thank you very much Bogdan, I’ll try to do it as you said. Thanks, Yuriy From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Tuesday, November 27, 2018 4:35 PM To: zzz ; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] snmp stat Hi Yuriy, yes, you should do one stat per GW / carrier - this is not an issue as you can dynamically create as many stats you need. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 2019 https://www.opensips.org/events/Summit-2019Amsterdam/ On 11/26/2018 11:53 PM, zzz wrote: Hi Bogdan, As I understand I’ll need to update each variable individually for each GW. Shell I create as many variables for all gateways I have or the one variable would be enough ? Thanks, Yuriy From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Monday, November 26, 2018 10:15 PM To: OpenSIPS users mailling list ; zzz Subject: Re: [OpenSIPS-Users] snmp stat Hi Yuriy, By default no, but you can generate such stats by using the "statistics" module - this module allows you to create and manage (inc/dec) stats from the script. Second step is use AgentX to grab the stats from OpenSIPS - via the MI interface - and publish them as mibs. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 11/24/2018 12:18 PM, zzz wrote: Hello, I’m able to collect total Opensips load with SNMP OID .1.3.6.1.4.1.27483.3.1.3.1.3.2.1.0 Is it possible to get snmp load stats per gateway (carrier) which I have in the dr_gateways (dr_carriers) table ? Thsnks, Yuriy _______________________________________________ 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 z at startit.ru Tue Nov 27 15:27:36 2018 From: z at startit.ru (zzz) Date: Tue, 27 Nov 2018 20:27:36 +0000 Subject: [OpenSIPS-Users] OpenSIPS Summit 2019 - Amsterdam, NL In-Reply-To: <1a9ba488-1e8a-3b47-7b83-969da3349e82@opensips.org> References: <1a9ba488-1e8a-3b47-7b83-969da3349e82@opensips.org> Message-ID: <8B36F227BD22B041AEA7015FD914CD95038176E073@JET-EX02.jettel.ru> Hello Bogdan, Couldn’t find the price if I’d like to visit this event. This event is very close to ITW in Chicago so unfortunately I couldn’t visit your summit last year. BR Yuriy From: Users [mailto:users-bounces at lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Tuesday, November 27, 2018 2:59 PM To: users at lists.opensips.org; developensips ; business at lists.opensips.org Subject: [OpenSIPS-Users] OpenSIPS Summit 2019 - Amsterdam, NL [http://www.opensips.org/events/Summit-2019Amsterdam/Amsterdam2019_round.png] OpenSIPS Summit 2019 April 30 - May 3, 2019 Amsterdam, The Netherlands Once again, the OpenSIPS Summit will take place in Amsterdam, becoming the hotspot of VoIP & RTC world for 4 full days. Three days of talks, inspiring presentations, amazing demos followed by one day of trainings about OpenSIPS and the Open Source ecosystem. Join us to learn what is new & important in VoIP & RTC, to discover how technology evolves and how you can benefit of it. Just knowledge in its most pure format. Some Great Reasons to Attend * Access the latest news, knowledge and experience in the VoIP & RTC world * Learn about upcoming 3.0 OpenSIPS release and how you can leverage it * Attend unique presentations and interactive technical workshops * Meet FOSS developers and community to share experience and comments * Get solutions consultancy during the Free Design Clinics * Become an Expert attending the OpenSIPS Advanced Training Summit Agenda * Two full days of presentations given by key speakers * Open Discussions with key people from OpenSIPS and other OSS projects * Interactive Demos and Showcases * Design Clinics to validate your OpenSIPS deployments * One full day of OpenSIPS Training (limited seats!) * Social events in the amazing Amsterdam Be part of it Be part of the OpenSIPS and the Open Source community, be part of the OpenSIPS Summit 2019. Attend to learn - the registration process will be open in the following days, stay tuned. Nevertheless, pre-registration is available, just contact us. Speak to share - the Call for Papers will be announced during next week, so you can share your wisdom and experience with the world. Sponsor to help - we welcome any help in making the Summit such a great event. Sponsoring is a natural way of saying "Thank you" for the Open Source code you are using within your businesses. Interested? Please contact our team or email us! [http://www.opensips.org/events/Summit-2018Amsterdam/Amsterdam_Radisson.png] Radisson Blu Rusland 17, 1012CK Amsterdam, The Netherlands Meet us again at our familiar Venue, with even more space and comfort than ever! This year the OpenSIPS Summit expands in size and will accommodate more participants and speakers. -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From mark at sowegatel.com Tue Nov 27 17:01:46 2018 From: mark at sowegatel.com (Mark Thomas) Date: Tue, 27 Nov 2018 17:01:46 -0500 Subject: [OpenSIPS-Users] Opensips Twilio Trunk Message-ID: <5bfdbeca.1c69fb81.ccbfb.1800@mx.google.com> I have been working with Opensips for a few weeks now. I’ve got everything going pretty good except for the fact that connection through a trunk is dropping after 32 seconds. Twilio claimed that this was due to the ack r-uri not containing the address in the 200 ok contact header. I examined it and found that to be the case. The Contact header in the 200 ok is sip:172.18.46.46:5060. After it comes into contact with opensips it changes to the ip address of the trunk. So whenever it’s sending the ack back it’s not traversing on the other end. I’ve run into a brick wall here and I really would be thrilled if someone could help me to get some insight on this issue. -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Wed Nov 28 02:59:58 2018 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Wed, 28 Nov 2018 09:59:58 +0200 Subject: [OpenSIPS-Users] Opensips Twilio Trunk In-Reply-To: <5bfdbeca.1c69fb81.ccbfb.1800@mx.google.com> References: <5bfdbeca.1c69fb81.ccbfb.1800@mx.google.com> Message-ID: Hi, Mark! Make sure you call fix_nated_contact(); on the 200 OK going to Twilio. Best regards, Răzvan On 11/28/18 12:01 AM, Mark Thomas wrote: > I have been working with Opensips for a few weeks now. I’ve got > everything going pretty good except for the fact that connection through > a trunk is dropping after 32 seconds. Twilio claimed that this was due > to the ack r-uri not containing the address in the 200 ok contact > header. I examined it and found that to be the case. The Contact header > in the 200 ok is sip:172.18.46.46:5060. After it comes into contact with > opensips it changes to the ip address of the trunk. So whenever it’s > sending the ack back it’s not traversing on the other end. I’ve run into > a brick wall here and I really would be thrilled if someone could help > me to get some insight on this issue. > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com Meet the OpenSIPS team at the next OpenSIPS Summit: https://www.opensips.org/events From bogdan at opensips.org Wed Nov 28 05:10:43 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 28 Nov 2018 12:10:43 +0200 Subject: [OpenSIPS-Users] OpenSIPS Summit 2019 - Amsterdam, NL In-Reply-To: <8B36F227BD22B041AEA7015FD914CD95038176E073@JET-EX02.jettel.ru> References: <1a9ba488-1e8a-3b47-7b83-969da3349e82@opensips.org> <8B36F227BD22B041AEA7015FD914CD95038176E073@JET-EX02.jettel.ru> Message-ID: <82917a88-a4e5-b2b9-3ce1-873468379dff@opensips.org> Hi Yuriy, The registration process (and fees) are not yet ready - we are still crouching data from last year in order to make a budgetary calculation :) Nevertheless, last year the Summit only was $299 and Summit + Training was $549. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 2019 https://www.opensips.org/events/Summit-2019Amsterdam/ On 11/27/2018 10:27 PM, zzz wrote: > > Hello Bogdan, > > Couldn’t find the price if I’d like to visit this event. > > This event is very close to ITW in Chicago so unfortunately I couldn’t > visit your summit last year. > > BR > > Yuriy > > *From:*Users [mailto:users-bounces at lists.opensips.org] *On Behalf Of > *Bogdan-Andrei Iancu > *Sent:* Tuesday, November 27, 2018 2:59 PM > *To:* users at lists.opensips.org; developensips > ; business at lists.opensips.org > *Subject:* [OpenSIPS-Users] OpenSIPS Summit 2019 - Amsterdam, NL > > > > > OpenSIPS Summit 2019 > > April 30 - May 3, 2019 > > Amsterdam, The Netherlands > > Once again, the OpenSIPS Summit > will take > place in Amsterdam, becoming the hotspot of VoIP & RTC world for *4 > full days*.**Three**days of talks, inspiring presentations, amazing > demos followed by one day of trainings about *OpenSIPS* and the *Open > Source* ecosystem. > > Join us to learn what is new & important in VoIP & RTC, to discover > how technology evolves and how you can benefit of it. *Just knowledge > in its most pure format*. > > *Some Great Reasons to Attend* > > * Access the latest news, knowledge and experience in the VoIP & RTC > world > * Learn about upcoming 3.0 OpenSIPS release and how you can leverage it > * Attend unique presentations and interactive technical workshops > * Meet FOSS developers and community to share experience and comments > * Get solutions consultancy during the Free Design Clinics > * Become an Expert attending the OpenSIPS Advanced Training > > *Summit Agenda* > > * Two full days of presentations given by key speakers > * Open Discussions with key people from OpenSIPS and other OSS projects > * Interactive Demos and Showcases > * Design Clinics to validate your OpenSIPS deployments > * One full day of OpenSIPS Training (limited seats!) > * Social events in the amazing Amsterdam > > *Be part of it* > > Be part of the OpenSIPS and the Open Source community, be part of the > OpenSIPS Summit 2019. > > *Attend to learn* - the registration process will be open in the > following days, stay tuned. Nevertheless, pre-registration is > available, just contact us. > > *Speak to share* - the Call for Papers will be announced during next > week, so you can share your wisdom and experience with the world. > > *Sponsor to help* - we welcome any help in making the Summit such a > great event. Sponsoring is a natural way of saying "Thank you" for the > Open Source code you are using within your businesses. > > Interested? Please contact > > our team or email > us! > > > > *Radisson Blu > Rusland 17, 1012CK Amsterdam, The Netherlands* > > Meet us again at our familiar Venue, with even more space and comfort > than ever! This year the OpenSIPS Summit expands in size and will > accommodate more participants and speakers. > > > > > > > > > > > -- > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.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 Wed Nov 28 05:11:59 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 28 Nov 2018 12:11:59 +0200 Subject: [OpenSIPS-Users] FW: segfault in opensips 2.4 In-Reply-To: <002d01d48685$59e6fe70$0db4fb50$@democon.be> References: <000f01d48685$30fb3990$92f1acb0$@democon.be> <002d01d48685$59e6fe70$0db4fb50$@democon.be> Message-ID: <627a4ccb-36dc-8755-74b0-555e031b0844@opensips.org> Hello Johan, Ideally we will need the backtrace from the corefile, so first check if your crash produced a core file and if yes, use gdb to extract the backtrace (let me know if you need instructions). Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 2019 https://www.opensips.org/events/Summit-2019Amsterdam/ On 11/27/2018 09:13 PM, johan de clercq wrote: > > Hello, I have received the following segfault in opensips > > kernel: [71330.742487] opensips[1268]: segfault at f0 ip > 00007fc617abad0e sp 00007ffdcaa3c090 error 4 in > registrar.so[7fc617aa9000+23000] > > Can somebody please explain what logs are needed ? > > cid:F3100D46-F00D-4610-87ED-3E91DA790A82 > > Johan De Clercq, Managing Director > Democon bvba - Ooigemstraat 41 - 8780 Oostrozebeke > > Tel +3256980990– GSM +32478720104 > > > > _______________________________________________ > 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: image001.png Type: image/png Size: 15602 bytes Desc: not available URL: From Johan at democon.be Wed Nov 28 05:21:39 2018 From: Johan at democon.be (Johan De Clercq) Date: Wed, 28 Nov 2018 11:21:39 +0100 Subject: [OpenSIPS-Users] FW: segfault in opensips 2.4 In-Reply-To: <627a4ccb-36dc-8755-74b0-555e031b0844@opensips.org> References: <000f01d48685$30fb3990$92f1acb0$@democon.be> <002d01d48685$59e6fe70$0db4fb50$@democon.be> <627a4ccb-36dc-8755-74b0-555e031b0844@opensips.org> Message-ID: Yes bogdan, i do need instructions :-) On Wed, 28 Nov 2018, 11:12 Bogdan-Andrei Iancu, wrote: > Hello Johan, > > Ideally we will need the backtrace from the corefile, so first check if > your crash produced a core file and if yes, use gdb to extract the > backtrace (let me know if you need instructions). > > Best regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Summit 2019 > https://www.opensips.org/events/Summit-2019Amsterdam/ > > On 11/27/2018 09:13 PM, johan de clercq wrote: > > > > Hello, I have received the following segfault in opensips > > > > kernel: [71330.742487] opensips[1268]: segfault at f0 ip 00007fc617abad0e > sp 00007ffdcaa3c090 error 4 in registrar.so[7fc617aa9000+23000] > > > > Can somebody please explain what logs are needed ? > > > > > > > > [image: cid:F3100D46-F00D-4610-87ED-3E91DA790A82] > > Johan De Clercq, Managing Director > Democon bvba - Ooigemstraat 41 - 8780 Oostrozebeke > > Tel +3256980990 – GSM +32478720104 > > > > > _______________________________________________ > 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: image001.png Type: image/png Size: 15602 bytes Desc: not available URL: From sagarmalam at gmail.com Wed Nov 28 05:40:31 2018 From: sagarmalam at gmail.com (sagar malam) Date: Wed, 28 Nov 2018 16:10:31 +0530 Subject: [OpenSIPS-Users] OpenSIPS : XMPP TO SIP AND VISE VERSA Message-ID: Hello, We have a system which using FreeSWITCH for calls and Ejabberd( XMPP) for chat.Now we want to support SIP SIMPLE as well.But we are not going to move old users to SIP SIMPLE but only new users will use SIP SIMPLE.This leads to requirement of having SIP-XMPP gateway for which are are considering OpenSIPS server. I have gone through documentation about OpenSIPS XMPP module, it is clear that OpenSIPS can convert SIP MESSAGE to XMPP so SIP to XMPP will work but what about XMPP to SIP MESSAGE ? Does that work as well ? -- Thanks, Sagar -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Nov 28 06:17:39 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 28 Nov 2018 13:17:39 +0200 Subject: [OpenSIPS-Users] OpenSIPS Control Panel 8.2.4 was released In-Reply-To: <001701d48673$9cda4be0$d68ee3a0$@smartvox.co.uk> References: <001001d48010$1bb732d0$53259870$@smartvox.co.uk> <648ddd0f-12db-de37-e9d5-47aede7a0cb1@opensips.org> <001701d48673$9cda4be0$d68ee3a0$@smartvox.co.uk> Message-ID: <81a7415a-765e-2118-3102-50fb6ce73c9b@opensips.org> Hi John, Thanks for the testing data, I managed to reproduce and fix it - the fix is a bit different from the original intention, but IMO it does a better job. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 2019 https://www.opensips.org/events/Summit-2019Amsterdam/ On 11/27/2018 07:07 PM, John Quick wrote: > Please find an SQL dump attached for the acc and sip_trace tables. > When I view this using control panel, the call that is at the top of the CDR table is on page 6 of the sip trace table. > The link always sends you to page 1 in sip trace, not to page 6. > > John Quick > Smartvox Limited > Tel: 01727-221221 > > > -----Original Message----- > From: Bogdan-Andrei Iancu > Sent: 27 November 2018 16:56 > To: john.quick at smartvox.co.uk; users at lists.opensips.org > Subject: Re: [OpenSIPS-Users] OpenSIPS Control Panel 8.2.4 was released > > Hi John, > > I'm not excluding the issue, as the jumping between the tools in combination with the paging is a bit of a complex process. I can dive into if you could help me with a data set - an mysql dump for cdrs and siptrace to show the issue - I can import the tables and debug it (it is a bit of time consuming to try to collect the date by myself) - of course you can send the data off list :) > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Summit 2019 > https://www.opensips.org/events/Summit-2019Amsterdam/ > > On 11/19/2018 03:59 PM, John Quick wrote: >> Hello Bogdan, >> >> The link works okay now, but (I think) only if the selected call is on >> the first page of results in the siptrace tab. >> >> John Quick >> Smartvox Limited >> >> >>> Hi John, >>> >>> I found a small typo that affected who the link was constructed . See >>> >> https://github.com/OpenSIPS/opensips-cp/commit/d50503123477f99b0079570 >> 361407 >> 7b685ca4579 >>> In order to link siptrace to cdrviewer, you need to (a) be sure homer >>> tool is disabled and (b) siptrace tool is enabled. >>> >>> Let me know if this fix does the trick for you. >>> >>> Best regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> http://www.opensips-solutions.com From bogdan at opensips.org Wed Nov 28 07:26:49 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 28 Nov 2018 14:26:49 +0200 Subject: [OpenSIPS-Users] FW: segfault in opensips 2.4 In-Reply-To: References: <000f01d48685$30fb3990$92f1acb0$@democon.be> <002d01d48685$59e6fe70$0db4fb50$@democon.be> <627a4ccb-36dc-8755-74b0-555e031b0844@opensips.org> Message-ID: Once you get the core file on your hands, do : gdb /path/to/opensips/binary /path/to/corefile after that, in gdb, do "bt full" Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 2019 https://www.opensips.org/events/Summit-2019Amsterdam/ On 11/28/2018 12:21 PM, Johan De Clercq wrote: > Yes bogdan, i do need instructions :-) > > On Wed, 28 Nov 2018, 11:12 Bogdan-Andrei Iancu, > wrote: > > Hello Johan, > > Ideally we will need the backtrace from the corefile, so first > check if your crash produced a core file and if yes, use gdb to > extract the backtrace (let me know if you need instructions). > > Best regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Summit 2019 > https://www.opensips.org/events/Summit-2019Amsterdam/ > > On 11/27/2018 09:13 PM, johan de clercq wrote: >> >> Hello, I have received the following segfault in opensips >> >> kernel: [71330.742487] opensips[1268]: segfault at f0 ip >> 00007fc617abad0e sp 00007ffdcaa3c090 error 4 in >> registrar.so[7fc617aa9000+23000] >> >> Can somebody please explain what logs are needed ? >> >> cid:F3100D46-F00D-4610-87ED-3E91DA790A82 >> >> Johan De Clercq, Managing Director >> Democon bvba - Ooigemstraat 41 - 8780 Oostrozebeke >> >> Tel +3256980990– GSM +32478720104 >> >> >> >> _______________________________________________ >> 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 Nov 28 14:18:34 2018 From: johan at democon.be (johan de clercq) Date: Wed, 28 Nov 2018 20:18:34 +0100 Subject: [OpenSIPS-Users] FW: segfault in opensips 2.4 In-Reply-To: References: <000f01d48685$30fb3990$92f1acb0$@democon.be> <002d01d48685$59e6fe70$0db4fb50$@democon.be> <627a4ccb-36dc-8755-74b0-555e031b0844@opensips.org> Message-ID: <005501d4874f$284dcae0$78e960a0$@democon.be> Sorry, Bogdan , ulimit -c outputted 0, so no core dump. Anyway, I did now ulimit -c unlimited, so let’s see if it reproduces itself. Thanks for the instructions. From: Bogdan-Andrei Iancu Sent: Wednesday, November 28, 2018 1:27 PM To: Johan De Clercq Cc: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] FW: segfault in opensips 2.4 Once you get the core file on your hands, do : gdb /path/to/opensips/binary /path/to/corefile after that, in gdb, do "bt full" Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 2019 https://www.opensips.org/events/Summit-2019Amsterdam/ On 11/28/2018 12:21 PM, Johan De Clercq wrote: Yes bogdan, i do need instructions :-) On Wed, 28 Nov 2018, 11:12 Bogdan-Andrei Iancu, > wrote: Hello Johan, Ideally we will need the backtrace from the corefile, so first check if your crash produced a core file and if yes, use gdb to extract the backtrace (let me know if you need instructions). Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 2019 https://www.opensips.org/events/Summit-2019Amsterdam/ On 11/27/2018 09:13 PM, johan de clercq wrote: Hello, I have received the following segfault in opensips kernel: [71330.742487] opensips[1268]: segfault at f0 ip 00007fc617abad0e sp 00007ffdcaa3c090 error 4 in registrar.so[7fc617aa9000+23000] Can somebody please explain what logs are needed ? Johan De Clercq, Managing Director Democon bvba - Ooigemstraat 41 - 8780 Oostrozebeke Tel +3256980990 – GSM +32478720104 _______________________________________________ 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 jock.mckechnie at gmail.com Wed Nov 28 15:59:18 2018 From: jock.mckechnie at gmail.com (Jock McKechnie) Date: Wed, 28 Nov 2018 14:59:18 -0600 Subject: [OpenSIPS-Users] "bad uri" response header collisions bug in OpenSIPS 2.4.3 Message-ID: Afternoon folks; I recently upgraded one of our production boxes to 2.4.3-1.el7 out of the OpenSIPS yum repo and within a few minutes I started seeing errors like this in the OpenSIPS logging: ERROR:core:parse_uri: bad uri, state 0 parsed: < 183> (4) / < 183 Session Progress#015#012Via: SIP/2.0/UDP 20> (42) ERROR:core:parse_sip_msg_uri: bad uri < 183 Session Progress#015#012Via: SIP/2.0/UDP 20> ERROR:perl:perl_exec2: failed to parse Request-URI ERROR:core:parse_uri: bad uri, state 0 parsed: < 183> (4) / < 183 Session Progress#015#012Via: SIP/2.0/UDP 20> (42) ERROR:core:parse_sip_msg_uri: bad uri < 183 Session Progress#015#012Via: SIP/2.0/UDP 20> ERROR:perl:perl_exec2: failed to parse Request-URI ERROR:tm:_reply_light: failed to generate 400 reply when a final 400 was sent out ERROR:signaling:sig_send_reply_mod: failed to send reply with tm module ERROR:perl:perl_exec2: failed to send reply ERROR:tm:_reply_light: failed to generate 500 reply when a final 400 was sent out I trapped SIP for a while and OpenSIPS appears to be periodically responding to messages with impressively malformed response type/headers: ACK 183 Session Progress Via: SIP/2.0/UDP 20 SIP/2.0 This belongs to the end of the call flow below (from 192.168.80.13 through our 2.4.3 proxy 192.168.93.214, to the destination server 192.168.80.56). You'll see the .56 returns a 403 Forbidden which the OpenSIPS is supposed to be ACKing, but instead it does something Very Odd with the header - and then proceeds to throw the logged errors above and returns a 400 to the requesting server. U 2018/11/28 10:18:22.758859 192.168.80.56:5060 -> 192.168.93.214:5060 SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.93.214:5060;received=192.168.93.214;branch=z9hG4bKcb74.9a10e3a3.0 Via: SIP/2.0/UDP 192.168.80.13:5060;rport=5060;received=192.168.80.13;branch=z9hG4bKPjdb6efcd9-373b-4122-8bde-9d92a608d6ee Record-Route: Call-ID: e33b7043-ef96-4102-9f6a-bcb9c97ac4da From: ;tag=c0d1ec70-fbaa-429c-acc9-17dec131459e To: ;tag=f9e1dd31-b586-4f47-9fb8-3d541b658a67 CSeq: 5413 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0 U 2018/11/28 10:18:22.759216 192.168.93.214:5060 -> 192.168.80.56:5060 ACK 183 Session Progress Via: SIP/2.0/UDP 20 SIP/2.0 Via: SIP/2.0/UDP 192.168.93.214:5060;branch=z9hG4bKcb74.9a10e3a3.0 From: ;tag=c0d1ec70-fbaa-429c-acc9-17dec131459e i: e33b7043-ef96-4102-9f6a-bcb9c97ac4da To: ;tag=f9e1dd31-b586-4f47-9fb8-3d541b658a67 CSeq: 5413 ACK Max-Forwards: 70 User-Agent: OpenSIPS (2.4.3 (x86_64/linux)) Content-Length: 0 U 2018/11/28 10:18:22.759666 192.168.93.214:5060 -> 192.168.80.13:5060 SIP/2.0 400 Bad Request-URI v: SIP/2.0/UDP 192.168.80.13:5060;received=192.168.80.13;rport=5060;branch=z9hG4bKPjdb6efcd9-373b-4122-8bde-9d92a608d6ee f: sip:+16465551212 at 192.168.80.13;tag=c0d1ec70-fbaa-429c-acc9-17dec131459e t: sip:+13235551212 at 192.168.93.214;tag=155c340f586c28d0300cf5a6ccf90d99-03d7 i: e33b7043-ef96-4102-9f6a-bcb9c97ac4da CSeq: 5413 INVITE Server: OpenSIPS (2.4.3 (x86_64/linux)) Content-Length: 0 I'm going to try and get more detailed debug dumps from OpenSIPS but it's only happening on a subset of calls and, of course, this was a production box (I didn't notice this in my limited testing) so I'm not able to safely live debug this box. I wanted to pitch it over the fence as soon as I could to get things rolling in case you guys might have an idea what's going on. Thanks much, - Jock From razvan at opensips.org Thu Nov 29 03:37:56 2018 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Thu, 29 Nov 2018 10:37:56 +0200 Subject: [OpenSIPS-Users] "bad uri" response header collisions bug in OpenSIPS 2.4.3 In-Reply-To: References: Message-ID: <75ba90a0-60e9-5be0-0ef3-ae8497f625b4@opensips.org> Hi, Jock! The errors you dumped below are found in the proxy's logs (192.168.93.214)? I don't actually understand how OpenSIPS triggers the ACK parsing error if he's the one generating it. Is OpenSIPS somehow looping the ACK to itself to re-parse it? Are you also tracking the internal interface? Best regards, Răzvan On 11/28/18 10:59 PM, Jock McKechnie wrote: > Afternoon folks; > > I recently upgraded one of our production boxes to 2.4.3-1.el7 out of > the OpenSIPS yum repo and within a few minutes I started seeing errors > like this in the OpenSIPS logging: > ERROR:core:parse_uri: bad uri, state 0 parsed: < 183> (4) / < 183 > Session Progress#015#012Via: SIP/2.0/UDP 20> (42) > ERROR:core:parse_sip_msg_uri: bad uri < 183 Session > Progress#015#012Via: SIP/2.0/UDP 20> > ERROR:perl:perl_exec2: failed to parse Request-URI > ERROR:core:parse_uri: bad uri, state 0 parsed: < 183> (4) / < 183 > Session Progress#015#012Via: SIP/2.0/UDP 20> (42) > ERROR:core:parse_sip_msg_uri: bad uri < 183 Session > Progress#015#012Via: SIP/2.0/UDP 20> > ERROR:perl:perl_exec2: failed to parse Request-URI > ERROR:tm:_reply_light: failed to generate 400 reply when a final 400 > was sent out > ERROR:signaling:sig_send_reply_mod: failed to send reply with tm module > ERROR:perl:perl_exec2: failed to send reply > ERROR:tm:_reply_light: failed to generate 500 reply when a final 400 > was sent out > > I trapped SIP for a while and OpenSIPS appears to be periodically > responding to messages with impressively malformed response > type/headers: > ACK 183 Session Progress > Via: SIP/2.0/UDP 20 SIP/2.0 > > This belongs to the end of the call flow below (from 192.168.80.13 > through our 2.4.3 proxy 192.168.93.214, to the destination server > 192.168.80.56). You'll see the .56 returns a 403 Forbidden which the > OpenSIPS is supposed to be ACKing, but instead it does something Very > Odd with the header - and then proceeds to throw the logged errors > above and returns a 400 to the requesting server. > > U 2018/11/28 10:18:22.758859 192.168.80.56:5060 -> 192.168.93.214:5060 > SIP/2.0 403 Forbidden > Via: SIP/2.0/UDP > 192.168.93.214:5060;received=192.168.93.214;branch=z9hG4bKcb74.9a10e3a3.0 > Via: SIP/2.0/UDP > 192.168.80.13:5060;rport=5060;received=192.168.80.13;branch=z9hG4bKPjdb6efcd9-373b-4122-8bde-9d92a608d6ee > Record-Route: > Call-ID: e33b7043-ef96-4102-9f6a-bcb9c97ac4da > From: ;tag=c0d1ec70-fbaa-429c-acc9-17dec131459e > To: ;tag=f9e1dd31-b586-4f47-9fb8-3d541b658a67 > CSeq: 5413 INVITE > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, > NOTIFY, REFER, MESSAGE, OPTIONS > Content-Length: 0 > > U 2018/11/28 10:18:22.759216 192.168.93.214:5060 -> 192.168.80.56:5060 > ACK 183 Session Progress > Via: SIP/2.0/UDP 20 SIP/2.0 > Via: SIP/2.0/UDP 192.168.93.214:5060;branch=z9hG4bKcb74.9a10e3a3.0 > From: ;tag=c0d1ec70-fbaa-429c-acc9-17dec131459e > i: e33b7043-ef96-4102-9f6a-bcb9c97ac4da > To: ;tag=f9e1dd31-b586-4f47-9fb8-3d541b658a67 > CSeq: 5413 ACK > Max-Forwards: 70 > User-Agent: OpenSIPS (2.4.3 (x86_64/linux)) > Content-Length: 0 > > U 2018/11/28 10:18:22.759666 192.168.93.214:5060 -> 192.168.80.13:5060 > SIP/2.0 400 Bad Request-URI > v: SIP/2.0/UDP 192.168.80.13:5060;received=192.168.80.13;rport=5060;branch=z9hG4bKPjdb6efcd9-373b-4122-8bde-9d92a608d6ee > f: sip:+16465551212 at 192.168.80.13;tag=c0d1ec70-fbaa-429c-acc9-17dec131459e > t: sip:+13235551212 at 192.168.93.214;tag=155c340f586c28d0300cf5a6ccf90d99-03d7 > i: e33b7043-ef96-4102-9f6a-bcb9c97ac4da > CSeq: 5413 INVITE > Server: OpenSIPS (2.4.3 (x86_64/linux)) > Content-Length: 0 > > I'm going to try and get more detailed debug dumps from OpenSIPS but > it's only happening on a subset of calls and, of course, this was a > production box (I didn't notice this in my limited testing) so I'm not > able to safely live debug this box. > > I wanted to pitch it over the fence as soon as I could to get things > rolling in case you guys might have an idea what's going on. > > Thanks much, > > - Jock > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com Meet the OpenSIPS team at the next OpenSIPS Summit: https://www.opensips.org/events From aronp at guaranteedplus.com Thu Nov 29 19:51:33 2018 From: aronp at guaranteedplus.com (Podrigal, Aron) Date: Thu, 29 Nov 2018 18:51:33 -0600 Subject: [OpenSIPS-Users] async() with postgres Message-ID: Hi Seems like the db_postgres module does not support async operations. Any plans for that? -- - Aron Podrigal -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Nov 30 03:28:19 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 30 Nov 2018 10:28:19 +0200 Subject: [OpenSIPS-Users] async() with postgres In-Reply-To: References: Message-ID: Hi Aron, There is no plan for the moment - not even sure if the pg lib does offer any support to implement async ops (like access to the FD of the connection). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 2019 https://www.opensips.org/events/Summit-2019Amsterdam/ On 11/30/2018 02:51 AM, Podrigal, Aron wrote: > Hi > > Seems like the db_postgres module does not support async operations. > Any plans for that? > > -- > > - > Aron Podrigal > > > > _______________________________________________ > 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: