From ravindra.bhatt at ecosmob.com Wed Nov 1 01:34:21 2017 From: ravindra.bhatt at ecosmob.com (Ravindrakumar Bhatt) Date: Wed, 1 Nov 2017 11:04:21 +0530 Subject: [OpenSIPS-Users] Problem in handling BYE using topology_hiding_match() In-Reply-To: References: Message-ID: Hi Bogdan, I know ACK is being properly transmitted but i have also handled ACK in main route in case it fails in match_dialog condition,but for BYE request i have not implemented any handling route. As i mentioned in earlier mails in "match_dialog" route ACK and BYE are not being relayed as in dialog messages. so do i have to handle them in main route also ? also please explain how opensips handles in dialog requests using topology_hiding and topology_hiding_match ? please find my opensips routes: https://pastebin.com/XBdWvBQW also logs with Debug mode on: https://pastebin.com/iqcEDint On Thu, Oct 26, 2017 at 3:53 PM, Ravindrakumar Bhatt < ravindra.bhatt at ecosmob.com> wrote: > Hi Bogdan, > I know ACK is being properly transmitted but i have also handled ACK in > main route in case it fails in match_dialog condition,but for BYE request i > have not implemented any handling route. > please see my main route logic as well as ACK method handling route. > route{ > > route("SANITY_CHECKS"); > route("MATCH_DIALOG"); > $avp(server_ip)="XXX.XXX.X.XXX"; > $avp(term-ip) = $Ri; > $avp(source-ip)= $si; #source ip address of message > > $avp(source-port)=$sp; # source port of message > $avp(src_ipz_ip) = $Ri; > $avp(dest-ip)=$od; > $avp(dest-port)=$op; > $avp(dst_type) = "CARRIER"; > $avp(use-rtpproxy)="YES"; > $avp(ruri)=$rU; #username in request URI > > xlog("L_INFO","[$time(%Y-%m-%d %H:%M:%S)] Incoming Request CI:$ci > RM:$rm FU:$fU TU:$tU SI:$avp(source-ip) SP:$avp(source-port) PC:$proto > TE:$avp(term-ip) \n"); > > if(method=="INVITE") { > route("PROCESS_INVITE"); > } else if(method=="CANCEL") { > route("PROCESS_CANCEL"); > } else if(method=="ACK") { > route("PROCESS_ACK"); > } else if(method=="OPTIONS") { > route("PROCESS_OPTIONS"); > } else if(method=="REGISTER"){ > route("PROCESS_REGISTER"); > } else { > sl_send_reply("501", "Method is not implemented"); ### > From here that funny 501 is coming.... > } > exit; > } > > > route[PROCESS_ACK] { > xlog("L_INFO","IN PROCESS ACK\n"); > if(t_check_trans()) { > t_relay(); > } > exit; > } > > As i mentioned in earlier mails in "match_dialog" route ACK and BYE > are not being relayed as in dialog messages. so do i have to handle them in > main route also ? also please explain how opensips handles in dialog > requests using topology_hiding and topology_hiding_match ? > link for Debug log: https://pastebin.com/iqcEDint > PFA pcap for same call done while taking debug log . > > Thanks and Regards, > Ravindrakumar Bhatt > > On Thu, Oct 26, 2017 at 3:47 PM, Ravindrakumar Bhatt < > ravindra.bhatt at ecosmob.com> wrote: > >> >> ---------- Forwarded message ---------- >> From: Ravindrakumar Bhatt >> Date: Tue, Oct 24, 2017 at 1:15 PM >> Subject: Re: [OpenSIPS-Users] Problem in handling BYE using >> topology_hiding_match() >> To: Bogdan-Andrei Iancu >> >> >> Hi Bogdan, >> I know ACK is being properly transmitted but i have also handled ACK >> in main route in case it fails in match_dialog condition,but for BYE >> request i have not implemented any handling route. >> please see my main route logic as well as ACK method handling route. >> route{ >> >> route("SANITY_CHECKS"); >> route("MATCH_DIALOG"); >> $avp(server_ip)="XXX.XXX.X.XXX"; >> $avp(term-ip) = $Ri; >> $avp(source-ip)= $si; #source ip address of message >> >> $avp(source-port)=$sp; # source port of message >> $avp(src_ipz_ip) = $Ri; >> $avp(dest-ip)=$od; >> $avp(dest-port)=$op; >> $avp(dst_type) = "CARRIER"; >> $avp(use-rtpproxy)="YES"; >> $avp(ruri)=$rU; #username in request URI >> >> xlog("L_INFO","[$time(%Y-%m-%d %H:%M:%S)] Incoming Request CI:$ci >> RM:$rm FU:$fU TU:$tU SI:$avp(source-ip) SP:$avp(source-port) PC:$proto >> TE:$avp(term-ip) \n"); >> >> if(method=="INVITE") { >> route("PROCESS_INVITE"); >> } else if(method=="CANCEL") { >> route("PROCESS_CANCEL"); >> } else if(method=="ACK") { >> route("PROCESS_ACK"); >> } else if(method=="OPTIONS") { >> route("PROCESS_OPTIONS"); >> } else if(method=="REGISTER"){ >> route("PROCESS_REGISTER"); >> } else { >> sl_send_reply("501", "Method is not implemented"); ### >> From here that funny 501 is coming.... >> } >> exit; >> } >> >> >> route[PROCESS_ACK] { >> xlog("L_INFO","IN PROCESS ACK\n"); >> if(t_check_trans()) { >> t_relay(); >> } >> exit; >> } >> >> As i mentioned in earlier mails in "match_dialog" route ACK and BYE >> are not being relayed as in dialog messages. so do i have to handle them in >> main route also ? also please explain how opensips handles in dialog >> requests using topology_hiding and topology_hiding_match ? >> link for Debug log: https://pastebin.com/iqcEDint >> PFA pcap for same call done while taking debug log . >> >> Thanks and Regards, >> Ravindrakumar Bhatt >> >> >> >> >> On Fri, Oct 20, 2017 at 9:02 PM, Bogdan-Andrei Iancu > > wrote: >> >>> Hi Ravinder, >>> >>> As per the trace, the ACK is properly routed (frames 17 & 18). Also the >>> incoming BYE (frame 19) is forwarded after a long time (see frame 22), >>> still there is a funny 501 reply for BYE (before the relay) in frame 21. >>> >>> Could you also post somewhere the OpenSIPS logs (level 4, debug) >>> corresponding to processing the incoming BYE (from 19) ? >>> >>> Best regards, >>> >>> Bogdan-Andrei Iancu >>> OpenSIPS Founder and Developer >>> http://www.opensips-solutions.com >>> >>> On 10/18/2017 11:21 AM, Ravindrakumar Bhatt wrote: >>> >>> Hi Bogdan, >>> please find my opensips call log as requested: >>> https://pastebin.com/7ypH0z0x >>> >>> as well as pcap for same call is attached with mail. >>> PFA >>> >>> Thanks and regards, >>> Ravindra Bhatt >>> >>> On Wed, Oct 18, 2017 at 12:52 PM, Bogdan-Andrei Iancu < >>> bogdan at opensips.org> wrote: >>> >>>> Hi Ravindra, >>>> >>>> Please post a link to the SIP trace/capture showing the entire SIP call >>>> (INVITE + 200 OK + ACK + BYE). The trace must cover all in and out traffic >>>> in OpenSIPS. >>>> >>>> Regards, >>>> >>>> Bogdan-Andrei Iancu >>>> OpenSIPS Founder and Developer >>>> http://www.opensips-solutions.com >>>> >>>> On 10/17/2017 12:29 PM, Ravindrakumar Bhatt wrote: >>>> >>>> Hello, >>>> i am using opensips v2.2.1 as SBC. >>>> call flow is like caller->opensips->server->opensips->callie.call is >>>> established successfully but when BYE is sent by any user it is not >>>> recognised by topology_hiding_match() also i am not getting any dialog >>>> related values(DLG_status,DLG_did etc) in ACK or BYE method. >>>> >>>> here is my logic for requset handling : >>>> >>>> route[PROCESS_INVITE] { >>>> >>>> route("SETCDR"); >>>> if(t_check_trans()) { >>>> drop(); >>>> } >>>> route("CHECK_NAT"); >>>> create_dialog(); >>>> do_accounting("aaa","cdr|failed"); >>>> route("REALY_ROUTE"); >>>> exit; >>>> } >>>> >>>> route[RELAY_ROUTE] { >>>> if(is_method("INVITE|UPDATE")) { >>>> xlog("L_INFO","CI:$ci In Relay Route RU:$ru DU:$du Tu:$tu >>>> TU:$tU CT:$ct\n"); >>>> if(!has_totag() && is_method("INVITE") && >>>> !isflagset(15)) { >>>> xlog("L_INFO","CI:$ci Topology_Hiding Call\n"); >>>> setflag(15); >>>> topology_hiding("UC"); >>>> # record_route(); >>>> >>>> >>>> } >>>> if($avp(use-rtpproxy)=="YES") { >>>> route(PROCESS_RTPPROXY_OFFER); >>>> } >>>> xlog("L_INFO","CI:$ci Routing Call IP:$avp(dest-ip) >>>> PORT:$avp(dest-port)\n"); >>>> >>>> $avp(call_status)="Failed"; >>>> t_on_branch("DEFAULT_BRANCH_ROUTE"); >>>> t_on_reply("DEFAULT_REPLY_ROUTE"); >>>> } >>>> >>>> xlog("L_INFO","DAILOG STATUS:::: $DLG_end_reason >>>> $DLG_did $DLG_status $DLG_count\n"); >>>> >>>> # record_route(); >>>> if (!t_relay()) { >>>> sl_reply_error(); >>>> } >>>> exit; >>>> } >>>> >>>> route[MATCH_DIALOG] { >>>> >>>> xlog("L_INFO",":::::DAILOG STATUS:::: $rm: $DLG_end_reason >>>> $DLG_did $DLG_status $DLG_count\n"); #HERE I AM GETTING most values >>>> null >>>> if(has_totag() && is_method("INVITE|ACK|BYE|UPDATE|REFER|PRACK|INFO") >>>> ) { >>>> xlog("L_INFO","INSIDE MATCH_DIALOG :::: $rm\n"); >>>> # loose_route(); >>>> xlog(" found match request to a dialog \n"); >>>> if(topology_hiding_match()) { #THIS >>>> CONDITIONS FAILS FOR ALL REQUESTS >>>> if(is_method("BYE")) { >>>> rtpproxy_unforce(); >>>> } >>>> route("RELAY_ROUTE"); >>>> } >>>> } >>>> } >>>> >>>> what is going wrong here ? is there any way to handle this issue? >>>> >>>> Thanks and regards , >>>> Ravindra Bhatt >>>> >>>> >>>> _______________________________________________ >>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> >>> >>> >> >> > -- *Ravindrakumar Bhatt* Jr. Software Developer Ecosmob Technologies Ltd Ahmedabad Mo:*+918460692402* -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Wed Nov 1 02:33:49 2017 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 1 Nov 2017 08:33:49 +0200 Subject: [OpenSIPS-Users] profile_count formula In-Reply-To: <03e01bcf-76ad-a499-b846-f9671ed84579@bohboh.info> References: <03e01bcf-76ad-a499-b846-f9671ed84579@bohboh.info> Message-ID: Hi, Thanks! The docs are broken, the correct formula (it's also how the code actually works) is: /max_load = (*Idle-CPU* / 100) * (*Max-Sessions* - (///*Sessions *- /profile_count))/ The "profile_count" is nothing more than the current usage of the given resource, as the load balancer sees it. As implemented, the load balancer makes use of dialog profiles to keep track of the number of calls balanced to each destination. Maybe I should also rename it to something easier to digest, such as "current_res_usage". Best regards, Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 01.11.2017 00:34, Social Boh wrote: > > Hello, > > I'm studying this formula: > > /max_load = (*Idle-CPU* / 100) * (*Max-Sessions* - (profile_count - > *Sessions*))/ > > from load_balancer module on OpenSIPs 2.3. > > Where does profile_count take his value? > > Regards/ > / > > -- > --- > I'm SoCIaL, MayBe > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Wed Nov 1 04:24:27 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Wed, 1 Nov 2017 10:24:27 +0200 Subject: [OpenSIPS-Users] Wrong Contact in location table In-Reply-To: <9eb0a075-7235-fba0-83c9-62c3ab5de93f@opensips.org> References: <59d89479-85dd-fccf-ee57-ce12296469b1@opensips.org> <222cf774-e269-9853-3436-bef40d8a410d@opensips.org> <0093287c-82e4-c11a-c933-e5e195683156@opensips.org> <741e1ed0-1831-4a1f-aa21-08ed354c2b53@opensips.org> <8045BDD6-2077-4CCC-B6C3-7E007AF631F3@tampahost.net> <9eb0a075-7235-fba0-83c9-62c3ab5de93f@opensips.org> Message-ID: <13955c88-1a4f-d106-10ec-a29525d4c39a@opensips.org> Can you print $ct.fields(uri) instead of $ct? Also, please post the output on pastebin.com. Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 10/31/2017 03:47 PM, Răzvan Crainea wrote: > Hi, Dragomir! > > Please post the logs I asked you to. > > Best regards, > Răzvan Crainea > OpenSIPS Developer > www.opensips-solutions.com > On 10/31/2017 03:45 PM, Sumit Birla wrote: >> X-Lite figures out its public IP and uses that in Via: and Contact: >> headers unless you set ‘Firewall traversal method’ to ‘None’ in the >> account settings. >> >> >> >> >>> On Oct 31, 2017, at 9:14 AM, Dragomir Haralambiev >>> > wrote: >>> >>> When REGISTER received from X-lite all is OK. Here part from >>> REGISTER message: >>> >>>     REGISTER sip:OpenSip_IP:Opensips_port;transport=UDP SIP/2.0 >>>     Via: SIP/2.0/UDP >>> 219.101.241.150:64726;branch=z9hG4bK-524287-1---39ee8718032a1a5a;rport >>>     Contact: >>> >>> >>> Here  "Via" contains real IP (this is Route IP). >>> >>> When REGISTER received from Yealink VoIP phone I have problem. Here >>> part from REGISTER message: >>> >>>     REGISTER sip:OpenSip_IP:Opensips_port SIP/2.0 >>>     Via: SIP/2.0/UDP 192.168.2.207:5062;branch=z9hG4bK77238423 >>>     Contact: >> > >>> >>> Here  "Via" contains not real IP (this is Yealink IP). >>> >>> What I do ? >>> >>> 2017-10-31 14:18 GMT+02:00 Răzvan Crainea >> >: >>> >>> So you did manage to fix your problem? >>> >>> Răzvan Crainea >>> OpenSIPS Developer >>> www.opensips-solutions.com >>> >>> On 10/31/2017 02:17 PM, Dragomir Haralambiev wrote: >>>> Hi, >>>> >>>> Thanks for your replay. >>>> >>>> I execute fix_nated_contact() after fix_nated_register(). >>>> >>>> Here part of my script: >>>> if (isflagset(NAT)) { >>>>                 if (is_method("REGISTER")) { >>>> fix_nated_register(); >>>> setbflag(NAT_BFLAG); # nat register >>>>                 } >>>> fix_nated_contact(); >>>> } >>>> >>>> In this case fix_nated_contact() will be fix contact records in >>>> location table. >>>> >>>> >>>> 2017-10-31 14:02 GMT+02:00 Răzvan Crainea >>> >: >>>> >>>> Hi, Dragomir! >>>> >>>> fix_nated_register() does not change the contact of the >>>> message, fix_nated_contact() does that. >>>> Please print the Contact header as I instructed earlier and >>>> send back the script trace. Also, please post the logs on >>>> pastebin.com or something, don't do >>>> it directly in the email. >>>> >>>> Best regards, >>>> >>>> Răzvan Crainea >>>> OpenSIPS Developer >>>> www.opensips-solutions.com >>>> >>>> On 10/31/2017 12:40 AM, Dragomir Haralambiev wrote: >>>>> What will I do to solve this problem? >>>>> >>>>> 2017-10-27 11:50 GMT+03:00 Dragomir Haralambiev >>>>> >: >>>>> >>>>> Oct 27 11:47:32 web /usr/local/sbin/opensips[30490]: >>>>> 57995010: REGISTER before fix_nated_register >>>>> >>>> >[1] >>>>> Oct 27 11:47:32 web /usr/local/sbin/opensips[30490]: >>>>> 57995010: REGISTER after fix_nated_register >>>>> >>>> >[1] >>>>> >>>>> >>>>> 2017-10-27 10:41 GMT+03:00 Răzvan Crainea >>>>> >: >>>>> >>>>> Hi, Dragomir! >>>>> >>>>> Can you also print the Contact header ($ct[1]). >>>>> >>>>> [1] >>>>> http://www.opensips.org/Documentation/Script-CoreVar-2-4#toc26 >>>>> >>>>> >>>>> Best regards, >>>>> >>>>> Răzvan Crainea >>>>> OpenSIPS Developer >>>>> 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 >>>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Nov 1 06:29:04 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 1 Nov 2017 12:29:04 +0200 Subject: [OpenSIPS-Users] Getting Wrong Time Stamp from $Tsm In-Reply-To: References: Message-ID: <70b4b6ab-f944-1397-0d34-e6928e34ed1f@opensips.org> Hi Diptesh, Unfortunately the docs are a bit misleading - the supported format for the $time() var is given by "man strftime" - and strftime has no support for time units smaller than the second. This is why the %N is not evaluated. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 10/31/2017 03:32 PM, Dipteshkumar Patel wrote: > Hello sir, > > Thanks for your prompt reply > > I used *$time(+%s%N)* but it uses only one format and give the result > "*+1509451841%N*" but i want to whole time stamp. > > This works fine from command line using *#time +%s%N.* > * > * > * > * > > > *Diptesh Patel* > Jr. Software Developer > Ecosmob Technologies Ltd > Ahmedabad > Mo:*+919898962659* > > On Tue, Oct 31, 2017 at 5:12 PM, Bogdan-Andrei Iancu > > wrote: > > Hi, > > Have you looked at the $time(format) script variable: > > http://www.opensips.org/Documentation/Script-CoreVar-2-3#toc87 > > > You can take the all the time fields at once, and use a > transformation to split them if you need the fields in separate > variables. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 10/31/2017 10:36 AM, Dipteshkumar Patel wrote: >> Dear sir, >> >> I am using Opensips version 2.3.1 >> >> I am using $Ts and $Tsm for time stamp for CDR but the >> microsecond time stamp is not same as system time stamp. To >> resolve the issue i am using exec() to get the system time but i >> takes so much time (nearly 5ms ) whenever so many calls >> concurrently running on server(We need exact time because it will >> affect the call duration). >> Can you please suggest me solution for the issue ? >> >> Thanks >> >> *Diptesh Patel* >> Jr. Software Developer >> Ecosmob Technologies Ltd >> Ahmedabad >> Mo:*+919898962659* >> >> >> _______________________________________________ >> 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 1 06:32:58 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 1 Nov 2017 12:32:58 +0200 Subject: [OpenSIPS-Users] Does the aaa_radius wait for response in async request ? In-Reply-To: References: Message-ID: <861dc75c-00e3-07b2-a2bd-659991995cbd@opensips.org> Hi Diptesh, That is the correct way of using the asyn radius interaction from OpenSIPS. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 10/31/2017 03:44 PM, Dipteshkumar Patel wrote: > Hello sir, > > I am using Opensips 2.3.1 > > I am using the launch(radius_send_acct("set2")) for sending the > accounting request to the radius server and i want to know that does > the aaa_radius module should wait for the reply or not ? where number > of calls running on the server. > (I applied the patch for the async in radiusclient source) > > > Thanks > *Diptesh Patel* > Jr. Software Developer > Ecosmob Technologies Ltd > Ahmedabad > Mo:*+919898962659* > > > _______________________________________________ > 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 diptesh.patel at ecosmob.com Wed Nov 1 06:36:58 2017 From: diptesh.patel at ecosmob.com (Dipteshkumar Patel) Date: Wed, 1 Nov 2017 16:06:58 +0530 Subject: [OpenSIPS-Users] Does the aaa_radius wait for response in async request ? In-Reply-To: <861dc75c-00e3-07b2-a2bd-659991995cbd@opensips.org> References: <861dc75c-00e3-07b2-a2bd-659991995cbd@opensips.org> Message-ID: hello sir Thanks for reply But i want to know that it will wait for the response from the radius server or not Is it create bottleneck for the system ? *Diptesh Patel* Jr. Software Developer Ecosmob Technologies Ltd Ahmedabad Mo:*+919898962659* On Wed, Nov 1, 2017 at 4:02 PM, Bogdan-Andrei Iancu wrote: > Hi Diptesh, > > That is the correct way of using the asyn radius interaction from OpenSIPS. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 10/31/2017 03:44 PM, Dipteshkumar Patel wrote: > > Hello sir, > > I am using Opensips 2.3.1 > > I am using the launch(radius_send_acct("set2")) for sending the > accounting request to the radius server and i want to know that does the > aaa_radius module should wait for the reply or not ? where number of calls > running on the server. > (I applied the patch for the async in radiusclient source) > > > Thanks > *Diptesh Patel* > Jr. Software Developer > Ecosmob Technologies Ltd > Ahmedabad > Mo:*+919898962659* > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Nov 1 07:59:46 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 1 Nov 2017 13:59:46 +0200 Subject: [OpenSIPS-Users] Does the aaa_radius wait for response in async request ? In-Reply-To: References: <861dc75c-00e3-07b2-a2bd-659991995cbd@opensips.org> Message-ID: <81913120-9403-8eef-c39a-46ce4f7f7de0@opensips.org> That is the definition of an async operation - it will wait for the reply without blocking resources. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/01/2017 12:36 PM, Dipteshkumar Patel wrote: > hello sir > > Thanks for reply > > But i want to know that it will wait for the response from the radius > server or not > > Is it create bottleneck for the system ? > > *Diptesh Patel* > Jr. Software Developer > Ecosmob Technologies Ltd > Ahmedabad > Mo:*+919898962659* > > On Wed, Nov 1, 2017 at 4:02 PM, Bogdan-Andrei Iancu > > wrote: > > Hi Diptesh, > > That is the correct way of using the asyn radius interaction from > OpenSIPS. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 10/31/2017 03:44 PM, Dipteshkumar Patel wrote: >> Hello sir, >> >> I am using Opensips 2.3.1 >> >> I am using the launch(radius_send_acct("set2")) for sending the >> accounting request to the radius server and i want to know that >> does the aaa_radius module should wait for the reply or not ? >> where number of calls running on the server. >> (I applied the patch for the async in radiusclient source) >> >> >> Thanks >> *Diptesh Patel* >> Jr. Software Developer >> Ecosmob Technologies Ltd >> Ahmedabad >> Mo:*+919898962659* >> >> >> _______________________________________________ >> 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 marketing at opensips.org Wed Nov 1 09:08:28 2017 From: marketing at opensips.org (OpenSIPS Marketing) Date: Wed, 1 Nov 2017 15:08:28 +0200 Subject: [OpenSIPS-Users] Wrong Contact in location table In-Reply-To: <13955c88-1a4f-d106-10ec-a29525d4c39a@opensips.org> References: <59d89479-85dd-fccf-ee57-ce12296469b1@opensips.org> <222cf774-e269-9853-3436-bef40d8a410d@opensips.org> <0093287c-82e4-c11a-c933-e5e195683156@opensips.org> <741e1ed0-1831-4a1f-aa21-08ed354c2b53@opensips.org> <8045BDD6-2077-4CCC-B6C3-7E007AF631F3@tampahost.net> <9eb0a075-7235-fba0-83c9-62c3ab5de93f@opensips.org> <13955c88-1a4f-d106-10ec-a29525d4c39a@opensips.org> Message-ID: OK, I managed to figure out why you get the private IP in the location table - that's because you are saving the registration in the onreply route, where you no longer have access to the fix_nated_contact() value. To fix your issue, you need to do the following: a. specify a mcontact_avp[1] in your script: modparam("registrar", "mcontact_avp", "$avp(register)") b. just after fix_nated_contact() on your REGISTER request, store the changed contact in that AVP: fix_nated_contact(); $avp(register) = $ct.fields(uri); Doing these two changes to your script should store the real IP in the location table. [1] http://www.opensips.org/html/docs/modules/2.4.x/registrar.html#idp5598160 Best regards, On 11/01/2017 10:24 AM, Răzvan Crainea wrote: > Can you print $ct.fields(uri) instead of $ct? > Also, please post the output on pastebin.com. > > Best regards, > Răzvan Crainea > OpenSIPS Developer > www.opensips-solutions.com > On 10/31/2017 03:47 PM, Răzvan Crainea wrote: >> Hi, Dragomir! >> >> Please post the logs I asked you to. >> >> Best regards, >> Răzvan Crainea >> OpenSIPS Developer >> www.opensips-solutions.com >> On 10/31/2017 03:45 PM, Sumit Birla wrote: >>> X-Lite figures out its public IP and uses that in Via: and Contact: >>> headers unless you set ‘Firewall traversal method’ to ‘None’ in the >>> account settings. >>> >>> >>> >>> >>>> On Oct 31, 2017, at 9:14 AM, Dragomir Haralambiev >>>> > wrote: >>>> >>>> When REGISTER received from X-lite all is OK. Here part from >>>> REGISTER message: >>>> >>>>     REGISTER sip:OpenSip_IP:Opensips_port;transport=UDP SIP/2.0 >>>>     Via: SIP/2.0/UDP >>>> 219.101.241.150:64726;branch=z9hG4bK-524287-1---39ee8718032a1a5a;rport >>>>     Contact: >>>> >>>> >>>> Here  "Via" contains real IP (this is Route IP). >>>> >>>> When REGISTER received from Yealink VoIP phone I have problem. Here >>>> part from REGISTER message: >>>> >>>>     REGISTER sip:OpenSip_IP:Opensips_port SIP/2.0 >>>>     Via: SIP/2.0/UDP 192.168.2.207:5062;branch=z9hG4bK77238423 >>>>     Contact: >>> > >>>> >>>> Here  "Via" contains not real IP (this is Yealink IP). >>>> >>>> What I do ? >>>> >>>> 2017-10-31 14:18 GMT+02:00 Răzvan Crainea >>> >: >>>> >>>> So you did manage to fix your problem? >>>> >>>> Răzvan Crainea >>>> OpenSIPS Developer >>>> www.opensips-solutions.com >>>> >>>> On 10/31/2017 02:17 PM, Dragomir Haralambiev wrote: >>>>> Hi, >>>>> >>>>> Thanks for your replay. >>>>> >>>>> I execute fix_nated_contact() after fix_nated_register(). >>>>> >>>>> Here part of my script: >>>>> if (isflagset(NAT)) { >>>>>                 if (is_method("REGISTER")) { >>>>> fix_nated_register(); >>>>> setbflag(NAT_BFLAG); # nat register >>>>>                 } >>>>> fix_nated_contact(); >>>>> } >>>>> >>>>> In this case fix_nated_contact() will be fix contact records >>>>> in location table. >>>>> >>>>> >>>>> 2017-10-31 14:02 GMT+02:00 Răzvan Crainea >>>> >: >>>>> >>>>> Hi, Dragomir! >>>>> >>>>> fix_nated_register() does not change the contact of the >>>>> message, fix_nated_contact() does that. >>>>> Please print the Contact header as I instructed earlier >>>>> and send back the script trace. Also, please post the logs >>>>> on pastebin.com or something, don't >>>>> do it directly in the email. >>>>> >>>>> Best regards, >>>>> >>>>> Răzvan Crainea >>>>> OpenSIPS Developer >>>>> www.opensips-solutions.com >>>>> >>>>> >>>>> On 10/31/2017 12:40 AM, Dragomir Haralambiev wrote: >>>>>> What will I do to solve this problem? >>>>>> >>>>>> 2017-10-27 11:50 GMT+03:00 Dragomir Haralambiev >>>>>> >: >>>>>> >>>>>> Oct 27 11:47:32 web /usr/local/sbin/opensips[30490]: >>>>>> 57995010: REGISTER before fix_nated_register >>>>>> >>>>> >[1] >>>>>> Oct 27 11:47:32 web /usr/local/sbin/opensips[30490]: >>>>>> 57995010: REGISTER after fix_nated_register >>>>>> >>>>> >[1] >>>>>> >>>>>> >>>>>> 2017-10-27 10:41 GMT+03:00 Răzvan Crainea >>>>>> >: >>>>>> >>>>>> Hi, Dragomir! >>>>>> >>>>>> Can you also print the Contact header ($ct[1]). >>>>>> >>>>>> [1] >>>>>> http://www.opensips.org/Documentation/Script-CoreVar-2-4#toc26 >>>>>> >>>>>> >>>>>> Best regards, >>>>>> >>>>>> Răzvan Crainea >>>>>> OpenSIPS Developer >>>>>> 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 >>>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From social at bohboh.info Wed Nov 1 10:07:21 2017 From: social at bohboh.info (Social Boh) Date: Wed, 1 Nov 2017 09:07:21 -0500 Subject: [OpenSIPS-Users] profile_count formula In-Reply-To: References: <03e01bcf-76ad-a499-b846-f9671ed84579@bohboh.info> Message-ID: <6a43bde3-4906-c318-3efa-1815007b1326@bohboh.info> Thank you for your answer, Liviu. So if I use a resource with name channels and between all servers/gateway configured I'm using 200 channels, this is the profile_count value Right? Regards --- I'm SoCIaL, MayBe El 01/11/2017 a las 01:33, Liviu Chircu escribió: > > Hi, > > Thanks! The docs are broken, the correct formula (it's also how the > code actually works) is: > > /max_load = (*Idle-CPU* / 100) * (*Max-Sessions* - (///*Sessions *- > /profile_count))/ > > The "profile_count" is nothing more than the current usage of the > given resource, as the load balancer sees it. As implemented, the load > balancer makes use of dialog profiles to keep track of the number of > calls balanced to each destination. Maybe I should also rename it to > something easier to digest, such as "current_res_usage". > > Best regards, > > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > On 01.11.2017 00:34, Social Boh wrote: >> >> Hello, >> >> I'm studying this formula: >> >> /max_load = (*Idle-CPU* / 100) * (*Max-Sessions* - (profile_count - >> *Sessions*))/ >> >> from load_balancer module on OpenSIPs 2.3. >> >> Where does profile_count take his value? >> >> Regards/ >> / >> >> -- >> --- >> I'm SoCIaL, MayBe >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Nov 1 10:16:15 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 1 Nov 2017 16:16:15 +0200 Subject: [OpenSIPS-Users] Further adventures in dual-stack ipv6/ipv4 questions In-Reply-To: <47941745-1f85-f9da-a7fa-d68f06c2bb39@street-artists.org> References: <47941745-1f85-f9da-a7fa-d68f06c2bb39@street-artists.org> Message-ID: <6501b440-e13a-b32f-fe7a-6c6742aa779b@opensips.org> Hi Daniel, Please see inline. Regards Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 10/27/2017 10:24 PM, Daniel Lakeland wrote: > So, at the moment, while topology hiding my network from ipv4 only > devices, I seem to be getting decent call performance. In fact, after > changing my create_dialog() call to not have "pPB" parameters, I'm now > not experiencing some of the random drop-outs that I experienced > before. This makes me suspect that the random drop-outs that I > experienced before enabling ipv6 were NAT related and due to pings not > reaching their intended destination. So now here's a question about > how opensips works when using TLS/TCP and how this interacts with > firewalls and NAT and ipv6. > > Suppose I have a phone that registers to opensips using TLS. This is a > TCP based socket. Suppose that the phone is ipv4, so it's behind a > NAT. Well, in truth, it's probably behind multiple layers of NAT, > potentially a carrier grade NAT somewhere deep in the bowels of an > ISP, then potentially a customer service ADSL modem or the like, and > then a router that I have control over. As I understand it after > setting TCP_PERSISTENT Opensips should leave this TLS connection open > and can then pass further reinvites, pings, and soforth down the same > socket. This is particularly true if the phone supports Sip Outbound > RFC 5626 (all ok so far?) [bogdan] The only thing required for the end point is not to close the connection. On its side, OpenSIPS will keep the TCP Conn alive for the whole duration of the registration, so the user is reachable. > > Suppose that some NAT or firewall somewhere incorrectly ages out this > TCP connection, or somewhere tcp_keepalive packets are being eaten, or > the like and when opensips tries to ping or receives a reinvite from > the far end that it tries to forward to my phone, it receives a > connection reset on the socket to my phone. Suppose further that the > tcp_no_new_conn_bflag is NOT set so Opensips is allowed to try to > reopen the connection. Does Opensips try to reach the phone with a new > connection? [bogdan] YEs, it will try to open a new TCP conn, but if the end-device is behind a NAT, it will most probably fail. > If it can't what does it do? [bogdan] The request OpenSIPS was trying to relay will be automatically be answered with 477 negative reply (as a result of the failure at the network layer) > Does it drop the call even though I don't have the dialog "B" flag? [bogdan] by itself OpenSIPS will not - it will be the decision of the end-point to keep (or not) the call if a sequential request failed. > Or, does it just wait until maybe the phone notices and re-opens the > connection itself? Suppose that I do not have the "bye" flag in my > dialog but I do have "ping" ie create_dialog("Pp"). [bogdan] The dialog will still be removed from OpenSIPS memory, but no BYE request sent. > Suppose further that after a while with Opensips in this reset state > with no open connection to my phone, my phone re-registers and a TLS > connection is available, will Opensips be able to pass reinvites and > pings for ongoing calls down this new socket? Does it understand that > this is the same phone and that this new registration+socket is > relevant to the ongoing dialog? [bogdan] jumping on new connection during a SIP session is not easy. As the TCP connections uses ephemeral ports, there are some internal aliases kept by OpenSIPS between SIP URIs and the actual TCP connections. A new connection means a new TCP port -> the existing aliases will not work. > > > Now, suppose we enable ipv6 and our phone is capable of ipv6. Under > this scenario provided firewalls are willing, a new TCP connection > could be created. Will Opensips do this if the socket drops? How does > TLS and TCP interact here? does TLS also try to open new connections > or only TCP? [bogdan] Any TCP based protocol will try to open, if tcp_no_new_conn_bflag is off. So, if the SIP URI correctly points to the end device (as IP+port), OpenSIPS should be able to reach back the end-device. > > Clearly, the connectionless UDP protocol doesn't have this connection > reset issue, and so once a phone is registered, opensips will send > things to the registration location, and if they're not received it > doesn't know except via timeout and resend. But with TCP or TLS if the > connection is broken I'd like to know how Opensips handles this case > because this could be a serious issue with reliability of phone calls > and random drops when TLS security is used with ipv4 + NAT, and I'm > wondering if ipv6 will solve this issue at least with respect to > end-to-end connections. [bogdan] All the time you need to consider both layer - the transport and the SIP layer. If the registration (at SIP layer) is not reachable anymore (based on the transport layer), the registration must be invalidated. You can do this based on registration SIP pinging in OpenSIPS. > > With the end-to-end nature of Ipv6 it seems we have a definite > possibility to avoid a lot of hassle with opensips able to directly > open a new sip connection to the phone to continue ongoing dialogs. Is > this a possibility for why I'm experiencing fewer random dropped calls? [bogdan] I wouldn't gamble on it. > > Thanks for any help! > > Dan > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From liviu at opensips.org Wed Nov 1 10:41:33 2017 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 1 Nov 2017 16:41:33 +0200 Subject: [OpenSIPS-Users] profile_count formula In-Reply-To: <6a43bde3-4906-c318-3efa-1815007b1326@bohboh.info> References: <03e01bcf-76ad-a499-b846-f9671ed84579@bohboh.info> <6a43bde3-4906-c318-3efa-1815007b1326@bohboh.info> Message-ID: Hi, The new docs should be working now. And yes, that is the "profile_count" value. However: The resources internally become normal (non-shared) dialog profiles by default. If you want to share resource usage between multiple load balancers, you must add the "/b" extension to your resource names, indicating you want them to be "binary replicated". Best regards, Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 01.11.2017 16:07, Social Boh wrote: > > Thank you for your answer, Liviu. > > So if I use a resource with name channels and between all > servers/gateway configured I'm using 200 channels, this is the > profile_count value > > Right? > > Regards > > --- > I'm SoCIaL, MayBe > El 01/11/2017 a las 01:33, Liviu Chircu escribió: >> >> Hi, >> >> Thanks! The docs are broken, the correct formula (it's also how the >> code actually works) is: >> >> /max_load = (*Idle-CPU* / 100) * (*Max-Sessions* - (///*Sessions *- >> /profile_count))/ >> >> The "profile_count" is nothing more than the current usage of the >> given resource, as the load balancer sees it. As implemented, the >> load balancer makes use of dialog profiles to keep track of the >> number of calls balanced to each destination. Maybe I should also >> rename it to something easier to digest, such as "current_res_usage". >> >> Best regards, >> >> Liviu Chircu >> OpenSIPS Developer >> http://www.opensips-solutions.com >> On 01.11.2017 00:34, Social Boh wrote: >>> >>> Hello, >>> >>> I'm studying this formula: >>> >>> /max_load = (*Idle-CPU* / 100) * (*Max-Sessions* - (profile_count - >>> *Sessions*))/ >>> >>> from load_balancer module on OpenSIPs 2.3. >>> >>> Where does profile_count take his value? >>> >>> Regards/ >>> / >>> >>> -- >>> --- >>> I'm SoCIaL, MayBe >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From social at bohboh.info Wed Nov 1 12:23:24 2017 From: social at bohboh.info (Social Boh) Date: Wed, 1 Nov 2017 11:23:24 -0500 Subject: [OpenSIPS-Users] profile_count formula In-Reply-To: References: <03e01bcf-76ad-a499-b846-f9671ed84579@bohboh.info> <6a43bde3-4906-c318-3efa-1815007b1326@bohboh.info> Message-ID: Thank you Liviu, last question, is there a way to indicate two different type of resources on FreeSWITCH Gateway? Example: resources field channels=fs://:pwd at 1.2.3.4;E1=fs://:pwd at 1.2.3.4 like other gateway: nacio=30;channels=50 Regards --- I'm SoCIaL, MayBe El 01/11/2017 a las 09:41, Liviu Chircu escribió: > > Hi, > > The new docs should be working now. And yes, that is the > "profile_count" value. However: > > The resources internally become normal (non-shared) dialog profiles by > default. If you want to share resource usage between multiple load > balancers, you must add the "/b" extension to your resource names, > indicating you want them to be "binary replicated". > > Best regards, > > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > On 01.11.2017 16:07, Social Boh wrote: >> >> Thank you for your answer, Liviu. >> >> So if I use a resource with name channels and between all >> servers/gateway configured I'm using 200 channels, this is the >> profile_count value >> >> Right? >> >> Regards >> >> --- >> I'm SoCIaL, MayBe >> El 01/11/2017 a las 01:33, Liviu Chircu escribió: >>> >>> Hi, >>> >>> Thanks! The docs are broken, the correct formula (it's also how the >>> code actually works) is: >>> >>> /max_load = (*Idle-CPU* / 100) * (*Max-Sessions* - (///*Sessions *- >>> /profile_count))/ >>> >>> The "profile_count" is nothing more than the current usage of the >>> given resource, as the load balancer sees it. As implemented, the >>> load balancer makes use of dialog profiles to keep track of the >>> number of calls balanced to each destination. Maybe I should also >>> rename it to something easier to digest, such as "current_res_usage". >>> >>> Best regards, >>> >>> Liviu Chircu >>> OpenSIPS Developer >>> http://www.opensips-solutions.com >>> On 01.11.2017 00:34, Social Boh wrote: >>>> >>>> Hello, >>>> >>>> I'm studying this formula: >>>> >>>> /max_load = (*Idle-CPU* / 100) * (*Max-Sessions* - (profile_count - >>>> *Sessions*))/ >>>> >>>> from load_balancer module on OpenSIPs 2.3. >>>> >>>> Where does profile_count take his value? >>>> >>>> Regards/ >>>> / >>>> >>>> -- >>>> --- >>>> I'm SoCIaL, MayBe >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From propalarupa at gmail.com Wed Nov 1 12:26:29 2017 From: propalarupa at gmail.com (Trupyy .) Date: Wed, 1 Nov 2017 17:26:29 +0100 Subject: [OpenSIPS-Users] AES + Opensip Message-ID: Hello Everyone, I got a quick question, hope someone can help me with it. So I got two Polycom VC devices to work with Opensips but the Calls work only when i turn off the Encryption (AES Option), when i turn it on i get an error that some security settings are not working properly. Can someone tell me if opensips is compatible with AES encryption or is the only encryption possibility, TLS? Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Nov 1 13:16:12 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 1 Nov 2017 19:16:12 +0200 Subject: [OpenSIPS-Users] Introducing OpenSIPS 2.4 Message-ID: <972d9f71-89d4-89f1-9183-f7fc0f1c506a@opensips.org> One more year, one more evolution cycle, one more OpenSIPS major release. So let me introduce you the upcoming OpenSIPS 2.4 . For the OpenSIPS 2.4 release we decided to focus on the */clustering abilities/*. Today’s VoIP world is getting more and more dynamic, services are moving into Clouds and more and more flexibility is needed for the application to fully exploit such environments. But let’s pin point the main reasons for going for a clustered approach : * scaling up with the processing/traffic load * geographical distribution * redundancy and High-Availability For the OpenSIPS 2.4 we laid down a roadmap that addresses the clustering both from the clustering engine itself (the underlayer) and from the functionalities that will perform on top of the clustering layer, to share data and state. With OpenSIPS 2.4, it will never be easier to built a consistent and powerful clustered solution: * *clustering engine* – enhances the capabilities of controlling the cluster topology, like re-routing for bypassing broken links, dynamic joining of new nodes, support for multiple capabilities per node, data syncing between nodes and many more; * *distributed user location* – this is a very complex topic as it exceeds the simple concept of data sharing. By the nature of the data (the user registrations), you may have different constraints on how data is roaming in a cluster – registrations may be tied to a node due NAT or TCP constraints. Even more, a sharding aspect must be addressed when looking at distributing the pinging effort across the cluster. So, multiple solutions are viable here, depending on what is to be achieved (scaling, redundancy) and what are the network constraints – see a detailed presentation of the available solutions; * *distributed presence server* – quite similar (but less complex) as the distributed user location, a distributed presence server provides a consistent, but distributed way of sharing presence information – SIP entities may publish data via different nodes in the SIP cluster, while the subscribers may fetch presence data via multiple various nodes. Two approaches are under work : (a) a cluster built around a noSQL DB based as primary data storage and (b) a cluster exclusively relying on OpenSIPS for data sharing; * *anycast support* – to be able to build a fully-flavored anycast support (addressing both redundancy and balancing) requires OpenSIPS to replicate/share transaction state across the nodes in the cluster (nodes sharing the same anycast IP). Depending on the nature of the replication (full transaction versus transaction meta-data) a full anycast and light anycast support will be available – here is a detailed description on the anycast support; * *clustered media relays* – as OpenSIPS has the ability to work together with several flavors of media relays (such as RTPproxy, RTPEngine, MediaProxy), the clustering support will help OpenSIPS do distributed load-balancing over the relays – even if a relay is used by multiple nodes in the cluster, all the nodes will share information on the load on the relay, to avoid overloading or idle time; * *distributed call center* – an agent is able to register with multiple queues on different nodes on a cluster. Still, all the queues do share the status / availability of the agent and its statistics for call distribution; * *custom clustering* – the OpenSIPS clustering underlayer provides at script level the ability to broadcast (in the cloud) or send to a given node a custom message/action (with replying possibility) – this is a very flexible and powerful way to build your custom distributed functionality directly at script level. And because we started on the integration path with OpenSIPS 2.3, and because we did it well, we decided to push forward on this path with the 2.4 version as well: * more *Homer integration *to be able to report TCP statistics, DB events and media relay events via HEP; * *SIPREC integration* for standard call recording. The new SIPREC module provides a standard and transparent (for the call parties) way to do call recording against an external recorder like Oreka provided by Orecx ; * more *FreeSWITCH integration* in order to capture the call-events (DTMFs, call status) from FreeSWITCH into OpenSIPS script or for being able to control a FreeSWITCH call from OpenSIPS script via ESL * *Asterisk flavored* Load-Balancing for a more realistic and accurate traffic balancing over Asterisk clusters (as the load information is fetched in realtime from Asterisk); ------------------------------------------------------------------------ The timeline for OpenSIPS 2.4 is: * Beta Release – 12-16 March 2018 * Stable Release – 23-27 April 2018 * General Availability – 1st of May 2018, during OpenSIPS Summit 2018 To talk more about the features of this new release, a public audio conference will be available on 21st of November 2017, 4 pm GMT , thanks to the kind sponsorship of UberConference . Anyone is welcome to join to find out more details or to ask questions about OpenSIPS 2.4 . This is a public and open conference, so no registration is needed, but if you want to announce your intention to participate, please let us know here: http://blog.opensips.org/2017/11/01/introducing-opensips-2-4/ Best regards, -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Wed Nov 1 13:25:57 2017 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 1 Nov 2017 19:25:57 +0200 Subject: [OpenSIPS-Users] profile_count formula In-Reply-To: References: <03e01bcf-76ad-a499-b846-f9671ed84579@bohboh.info> <6a43bde3-4906-c318-3efa-1815007b1326@bohboh.info> Message-ID: IIRC, the module will give you some warning if you attempt that, and it will only use the last resource/URL it finds in the string. Can you help me understand why you would need this feature? If you think about it, if you use 1 x "channels", FreeSWITCH will give you feedback 1 second later, and 1 will be subtracted from the maximum "E1" resource as well. This didn't make sense to me, so I did not allow it. Also, from a logical point of view: a load balancer SIP URI maps to a FS CLI interface. How could it map to two different FS CLIs? Best regards, Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 01.11.2017 18:23, Social Boh wrote: > > Thank you Liviu, > > last question, is there a way to indicate two different type of > resources on FreeSWITCH Gateway? > > Example: > > resources field > > channels=fs://:pwd at 1.2.3.4;E1=fs://:pwd at 1.2.3.4 > > like other gateway: > > nacio=30;channels=50 > > Regards > > --- > I'm SoCIaL, MayBe > El 01/11/2017 a las 09:41, Liviu Chircu escribió: >> >> Hi, >> >> The new docs should be working now. And yes, that is the >> "profile_count" value. However: >> >> The resources internally become normal (non-shared) dialog profiles >> by default. If you want to share resource usage between multiple load >> balancers, you must add the "/b" extension to your resource names, >> indicating you want them to be "binary replicated". >> >> Best regards, >> >> Liviu Chircu >> OpenSIPS Developer >> http://www.opensips-solutions.com >> On 01.11.2017 16:07, Social Boh wrote: >>> >>> Thank you for your answer, Liviu. >>> >>> So if I use a resource with name channels and between all >>> servers/gateway configured I'm using 200 channels, this is the >>> profile_count value >>> >>> Right? >>> >>> Regards >>> >>> --- >>> I'm SoCIaL, MayBe >>> El 01/11/2017 a las 01:33, Liviu Chircu escribió: >>>> >>>> Hi, >>>> >>>> Thanks! The docs are broken, the correct formula (it's also how the >>>> code actually works) is: >>>> >>>> /max_load = (*Idle-CPU* / 100) * (*Max-Sessions* - (///*Sessions *- >>>> /profile_count))/ >>>> >>>> The "profile_count" is nothing more than the current usage of the >>>> given resource, as the load balancer sees it. As implemented, the >>>> load balancer makes use of dialog profiles to keep track of the >>>> number of calls balanced to each destination. Maybe I should also >>>> rename it to something easier to digest, such as "current_res_usage". >>>> >>>> Best regards, >>>> >>>> Liviu Chircu >>>> OpenSIPS Developer >>>> http://www.opensips-solutions.com >>>> On 01.11.2017 00:34, Social Boh wrote: >>>>> >>>>> Hello, >>>>> >>>>> I'm studying this formula: >>>>> >>>>> /max_load = (*Idle-CPU* / 100) * (*Max-Sessions* - (profile_count >>>>> - *Sessions*))/ >>>>> >>>>> from load_balancer module on OpenSIPs 2.3. >>>>> >>>>> Where does profile_count take his value? >>>>> >>>>> Regards/ >>>>> / >>>>> >>>>> -- >>>>> --- >>>>> I'm SoCIaL, MayBe >>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From hunterj91 at hotmail.com Wed Nov 1 17:57:22 2017 From: hunterj91 at hotmail.com (Jonathan Hunter) Date: Wed, 1 Nov 2017 21:57:22 +0000 Subject: [OpenSIPS-Users] opensips 2.2.5 stop responding/processing requests. - Centos 6.5 In-Reply-To: <2760ebb8-7272-41f8-c813-f12f79f19688@opensips.org> References: , <2760ebb8-7272-41f8-c813-f12f79f19688@opensips.org> Message-ID: Hi Guys, Thanks for the responses,appreciate it! In the end it was an overloaded DNS server, which caused a delay in another SBC passing requests to the opensips server, so overloading it. Thanks for the advice everyone. It wasnt OpenSIPS! 😊 Jon ________________________________ From: Bogdan-Andrei Iancu Sent: 31 October 2017 12:04 To: OpenSIPS users mailling list; Jonathan Hunter Subject: Re: [OpenSIPS-Users] opensips 2.2.5 stop responding/processing requests. - Centos 6.5 Hi Jonathan, It looks like you have kind on task inside OpenSIPS that is using a lot of time/process resources - shortly, this translate into "there are no available worker processes to handle new jobs". * do you have any threshold set to catch any DB/DNS delays ? * have you checked the CPU usage ? * have you monitor the "load:" statistics group ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com Home — OpenSIPS Solutions www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 10/30/2017 07:33 PM, Jonathan Hunter wrote: Hi Guys, Has anyone had the following behaviour with this release of opensips? It has been running fine for a number of months, and then suddenly today the server which was running as the primary suddenly stopped responding to SIP requests, or stopped processing them, or there was a long pause in a sequence completing. So in some cases, INVITES or REGISTER messages were ignored and the initial requests werent always logged in the opensips log file (even though we do this for all messages), however the SIP requests were hitting the server as I was tracing them. In the logs I can see the following warnings we havent seen before; Oct 30 13:31:25 sgw6 VU-SIP-Proxy[11159]: WARNING:core:utimer_ticker: utimer task already scheduled for 9639891650 ms (now 9639891850 ms), it may overlap. Oct 30 13:52:48 sgw6 VU-SIP-Proxy[11159]: WARNING:core:timer_ticker: timer task already scheduled for 9640994180 ms (now 9641175280 ms), it may overlap.. Oct 30 13:52:48 sgw6 VU-SIP-Proxy[11159]: WARNING:core:timer_ticker: timer task already scheduled for 9640980270 ms (now 9641175280 ms), it may overlap.. A restart of the opensips application didnt help, and service was only restored fully when I failed over to the secondary server. Has anyone had these issues before? There are no obvious server level issues, and as I mentioned the logs dont contain anything too unusual. Any help appreciated. Many thanks Jon _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From goup2010 at gmail.com Wed Nov 1 18:26:27 2017 From: goup2010 at gmail.com (Dragomir Haralambiev) Date: Thu, 2 Nov 2017 00:26:27 +0200 Subject: [OpenSIPS-Users] Wrong Contact in location table In-Reply-To: References: <59d89479-85dd-fccf-ee57-ce12296469b1@opensips.org> <222cf774-e269-9853-3436-bef40d8a410d@opensips.org> <0093287c-82e4-c11a-c933-e5e195683156@opensips.org> <741e1ed0-1831-4a1f-aa21-08ed354c2b53@opensips.org> <8045BDD6-2077-4CCC-B6C3-7E007AF631F3@tampahost.net> <9eb0a075-7235-fba0-83c9-62c3ab5de93f@opensips.org> <13955c88-1a4f-d106-10ec-a29525d4c39a@opensips.org> Message-ID: Hi, I try this but in location table I not see real IP. Problem exists. When make save location $avp(register)=sip:55996201 at 192.27.47.139:5060 opensipsctl ul show 55996201 at 192.27.47.139 AOR:: 55996201 Contact:: sip:57996201 at 192.168.22.201:5060 Q= ContactID:: 182005870676967428 Where is problem? 2017-11-01 15:08 GMT+02:00 OpenSIPS Marketing : > OK, I managed to figure out why you get the private IP in the location > table - that's because you are saving the registration in the onreply > route, where you no longer have access to the fix_nated_contact() value. > > To fix your issue, you need to do the following: > a. specify a mcontact_avp[1] in your script: > modparam("registrar", "mcontact_avp", "$avp(register)") > > b. just after fix_nated_contact() on your REGISTER request, store the > changed contact in that AVP: > fix_nated_contact(); > $avp(register) = $ct.fields(uri); > > Doing these two changes to your script should store the real IP in the > location table. > > [1] http://www.opensips.org/html/docs/modules/2.4.x/registrar. > html#idp5598160 > > Best regards, > > On 11/01/2017 10:24 AM, Răzvan Crainea wrote: > > Can you print $ct.fields(uri) instead of $ct? > Also, please post the output on pastebin.com. > > Best regards, > > Răzvan Crainea > OpenSIPS Developerwww.opensips-solutions.com > > On 10/31/2017 03:47 PM, Răzvan Crainea wrote: > > Hi, Dragomir! > > Please post the logs I asked you to. > > Best regards, > > Răzvan Crainea > OpenSIPS Developerwww.opensips-solutions.com > > On 10/31/2017 03:45 PM, Sumit Birla wrote: > > X-Lite figures out its public IP and uses that in Via: and Contact: > headers unless you set ‘Firewall traversal method’ to ‘None’ in the account > settings. > > > > > On Oct 31, 2017, at 9:14 AM, Dragomir Haralambiev > wrote: > > When REGISTER received from X-lite all is OK. Here part from REGISTER > message: > > REGISTER sip:OpenSip_IP:Opensips_port;transport=UDP SIP/2.0 > Via: SIP/2.0/UDP 219.101.241.150:64726;branch=z9hG4bK-524287-1--- > 39ee8718032a1a5a;rport > Contact: d6122d9250e9d0bd> > > Here "Via" contains real IP (this is Route IP). > > When REGISTER received from Yealink VoIP phone I have problem. Here part > from REGISTER message: > > REGISTER sip:OpenSip_IP:Opensips_port SIP/2.0 > Via: SIP/2.0/UDP 192.168.2.207:5062;branch=z9hG4bK77238423 > Contact: > > Here "Via" contains not real IP (this is Yealink IP). > > What I do ? > > > 2017-10-31 14:18 GMT+02:00 Răzvan Crainea : > >> So you did manage to fix your problem? >> >> Răzvan Crainea >> OpenSIPS Developerwww.opensips-solutions.com >> >> On 10/31/2017 02:17 PM, Dragomir Haralambiev wrote: >> >> Hi, >> >> Thanks for your replay. >> >> I execute fix_nated_contact() after fix_nated_register(). >> >> Here part of my script: >> if (isflagset(NAT)) { >> if (is_method("REGISTER")) { >> fix_nated_register(); >> setbflag(NAT_BFLAG); # nat register >> } >> fix_nated_contact(); >> } >> >> In this case fix_nated_contact() will be fix contact records in location >> table. >> >> >> 2017-10-31 14:02 GMT+02:00 Răzvan Crainea : >> >>> Hi, Dragomir! >>> >>> fix_nated_register() does not change the contact of the message, >>> fix_nated_contact() does that. >>> Please print the Contact header as I instructed earlier and send back >>> the script trace. Also, please post the logs on pastebin.com or >>> something, don't do it directly in the email. >>> >>> Best regards, >>> >>> Răzvan Crainea >>> OpenSIPS Developerwww.opensips-solutions.com >>> >>> On 10/31/2017 12:40 AM, Dragomir Haralambiev wrote: >>> >>> What will I do to solve this problem? >>> >>> 2017-10-27 11:50 GMT+03:00 Dragomir Haralambiev : >>> >>>> Oct 27 11:47:32 web /usr/local/sbin/opensips[30490]: 57995010: >>>> REGISTER before fix_nated_register [1] >>>> Oct 27 11:47:32 web /usr/local/sbin/opensips[30490]: 57995010: >>>> REGISTER after fix_nated_register [1] >>>> >>>> >>>> 2017-10-27 10:41 GMT+03:00 Răzvan Crainea : >>>> >>>>> Hi, Dragomir! >>>>> >>>>> Can you also print the Contact header ($ct[1]). >>>>> >>>>> [1] http://www.opensips.org/Documentation/Script-CoreVar-2-4#toc26 >>>>> >>>>> Best regards, >>>>> >>>>> Răzvan Crainea >>>>> OpenSIPS Developerwww.opensips-solutions.com >>>>> >>>>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From fehmfehmfehmfehmfehm at gmail.com Thu Nov 2 05:15:37 2017 From: fehmfehmfehmfehmfehm at gmail.com (Matine Tinthaitae) Date: Thu, 2 Nov 2017 10:15:37 +0100 Subject: [OpenSIPS-Users] Differences between dialog replication and db_mode 1 Message-ID: Hi, I'm trying to set up an active/standby solution using floating ip. Only dialogs have to be shared. I've been using both the clusterer module dialog replication and the dialog module db_mode 1 with a shared database and both seem to do what i want, which is being able to kill the active node without losing call states. What's the difference/benefit of using the clusterer module over db_mode 1 in this scenario? Thank you! //Matine -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Thu Nov 2 05:31:06 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Thu, 2 Nov 2017 11:31:06 +0200 Subject: [OpenSIPS-Users] Wrong Contact in location table In-Reply-To: References: <222cf774-e269-9853-3436-bef40d8a410d@opensips.org> <0093287c-82e4-c11a-c933-e5e195683156@opensips.org> <741e1ed0-1831-4a1f-aa21-08ed354c2b53@opensips.org> <8045BDD6-2077-4CCC-B6C3-7E007AF631F3@tampahost.net> <9eb0a075-7235-fba0-83c9-62c3ab5de93f@opensips.org> <13955c88-1a4f-d106-10ec-a29525d4c39a@opensips.org> Message-ID: <58c846f0-7fe6-f687-d7aa-b505b20a41ef@opensips.org> Did you put the mcontact_avp parameter I asked you to? Can you print the value of the register just before save()? Also, can you send over some logs for the reply? Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/02/2017 12:26 AM, Dragomir Haralambiev wrote: > Hi, > > I try this but in location table I not see real IP. Problem exists. > > When make save location $avp(register)=sip:55996201 at 192.27.47.139:5060 > > > opensipsctl ul show 55996201 at 192.27.47.139 > > AOR:: 55996201 >         Contact:: sip:57996201 at 192.168.22.201:5060 > Q= >         ContactID:: 182005870676967428 > Where is problem? > > > 2017-11-01 15:08 GMT+02:00 OpenSIPS Marketing >: > > OK, I managed to figure out why you get the private IP in the > location table - that's because you are saving the registration in > the onreply route, where you no longer have access to the > fix_nated_contact() value. > > To fix your issue, you need to do the following: > a. specify a mcontact_avp[1] in your script: > modparam("registrar", "mcontact_avp", "$avp(register)") > > b. just after fix_nated_contact() on your REGISTER request, store > the changed contact in that AVP: > fix_nated_contact(); > $avp(register) = $ct.fields(uri); > > Doing these two changes to your script should store the real IP in > the location table. > > [1] > http://www.opensips.org/html/docs/modules/2.4.x/registrar.html#idp5598160 > > > Best regards, > > On 11/01/2017 10:24 AM, Răzvan Crainea wrote: >> Can you print $ct.fields(uri) instead of $ct? >> Also, please post the output on pastebin.com . >> >> Best regards, >> Răzvan Crainea >> OpenSIPS Developer >> www.opensips-solutions.com >> On 10/31/2017 03:47 PM, Răzvan Crainea wrote: >>> Hi, Dragomir! >>> >>> Please post the logs I asked you to. >>> >>> Best regards, >>> Răzvan Crainea >>> OpenSIPS Developer >>> www.opensips-solutions.com >>> On 10/31/2017 03:45 PM, Sumit Birla wrote: >>>> X-Lite figures out its public IP and uses that in Via: and >>>> Contact: headers unless you set ‘Firewall traversal method’ to >>>> ‘None’ in the account settings. >>>> >>>> >>>> >>>> >>>>> On Oct 31, 2017, at 9:14 AM, Dragomir Haralambiev >>>>> > wrote: >>>>> >>>>> When REGISTER received from X-lite all is OK. Here part from >>>>> REGISTER message: >>>>> >>>>>     REGISTER sip:OpenSip_IP:Opensips_port;transport=UDP SIP/2.0 >>>>>     Via: SIP/2.0/UDP >>>>> 219.101.241.150:64726;branch=z9hG4bK-524287-1---39ee8718032a1a5a;rport >>>>>     Contact: >>>>> >>>>> >>>>> Here  "Via" contains real IP (this is Route IP). >>>>> >>>>> When REGISTER received from Yealink VoIP phone I have problem. >>>>> Here part from REGISTER message: >>>>> >>>>>     REGISTER sip:OpenSip_IP:Opensips_port SIP/2.0 >>>>>     Via: SIP/2.0/UDP 192.168.2.207:5062;branch=z9hG4bK77238423 >>>>>     Contact: >>>> > >>>>> >>>>> Here  "Via" contains not real IP (this is Yealink IP). >>>>> >>>>> What I do ? >>>>> >>>>> 2017-10-31 14:18 GMT+02:00 Răzvan Crainea >>>> >: >>>>> >>>>> So you did manage to fix your problem? >>>>> >>>>> Răzvan Crainea >>>>> OpenSIPS Developer >>>>> www.opensips-solutions.com >>>>> >>>>> >>>>> On 10/31/2017 02:17 PM, Dragomir Haralambiev wrote: >>>>>> Hi, >>>>>> >>>>>> Thanks for your replay. >>>>>> >>>>>> I execute fix_nated_contact() after fix_nated_register(). >>>>>> >>>>>> Here part of my script: >>>>>> if (isflagset(NAT)) { >>>>>>                 if (is_method("REGISTER")) { >>>>>> fix_nated_register(); >>>>>> setbflag(NAT_BFLAG); # nat register >>>>>>                 } >>>>>> fix_nated_contact(); >>>>>> } >>>>>> >>>>>> In this case fix_nated_contact() will be fix contact >>>>>> records in location table. >>>>>> >>>>>> >>>>>> 2017-10-31 14:02 GMT+02:00 Răzvan Crainea >>>>>> >: >>>>>> >>>>>> Hi, Dragomir! >>>>>> >>>>>> fix_nated_register() does not change the contact of >>>>>> the message, fix_nated_contact() does that. >>>>>> Please print the Contact header as I instructed >>>>>> earlier and send back the script trace. Also, please >>>>>> post the logs on pastebin.com >>>>>> or something, don't do it directly in the email. >>>>>> >>>>>> Best regards, >>>>>> >>>>>> Răzvan Crainea >>>>>> OpenSIPS Developer >>>>>> www.opensips-solutions.com >>>>>> >>>>>> >>>>>> On 10/31/2017 12:40 AM, Dragomir Haralambiev wrote: >>>>>>> What will I do to solve this problem? >>>>>>> >>>>>>> 2017-10-27 11:50 GMT+03:00 Dragomir Haralambiev >>>>>>> >: >>>>>>> >>>>>>> Oct 27 11:47:32 web >>>>>>> /usr/local/sbin/opensips[30490]: 57995010: >>>>>>> REGISTER before fix_nated_register >>>>>>> >>>>>> >[1] >>>>>>> Oct 27 11:47:32 web >>>>>>> /usr/local/sbin/opensips[30490]: 57995010: >>>>>>> REGISTER after fix_nated_register >>>>>>> >>>>>> >[1] >>>>>>> >>>>>>> >>>>>>> 2017-10-27 10:41 GMT+03:00 Răzvan Crainea >>>>>>> >: >>>>>>> >>>>>>> Hi, Dragomir! >>>>>>> >>>>>>> Can you also print the Contact header ($ct[1]). >>>>>>> >>>>>>> [1] >>>>>>> http://www.opensips.org/Documentation/Script-CoreVar-2-4#toc26 >>>>>>> >>>>>>> >>>>>>> Best regards, >>>>>>> >>>>>>> Răzvan Crainea >>>>>>> OpenSIPS Developer >>>>>>> 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 >>>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From pimenta at inatel.br Thu Nov 2 08:33:34 2017 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Thu, 2 Nov 2017 12:33:34 +0000 Subject: [OpenSIPS-Users] Table location and command opensipsctl ul show. Message-ID: Hi. When I delete all registers from table location, I still can see registers via command 'opensipsctl ul show'. Why a empty table location doesn't gives a null result in 'opensipsctl ul show' ? Is it caused by the configuration from opensips.cfg ? Thanks. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Thu Nov 2 09:27:24 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Thu, 2 Nov 2017 15:27:24 +0200 Subject: [OpenSIPS-Users] Table location and command opensipsctl ul show. In-Reply-To: References: Message-ID: <6c114705-633b-00e5-b949-2741b09eee74@opensips.org> Hi, Rodrigo! The interaction between OpenSIPS memory and database is described here[1]. Depending on your db_mode configuration, you might find entries that are not yet synced in the database. [1] http://www.opensips.org/html/docs/modules/2.4.x/usrloc.html#idp5672576 Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/02/2017 02:33 PM, Rodrigo Pimenta Carvalho wrote: > > Hi. > > > When I delete all registers from table location, I still can see > registers via command 'opensipsctl ul show'. > > Why a empty table location doesn't gives a null result in 'opensipsctl > ul show' ? > > Is it caused by the configuration from opensips.cfg ? > > > Thanks. > > > RODRIGO PIMENTA CARVALHO > Inatel Competence Center > Software > Ph: +55 35 3471 9200 RAMAL 979 > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From pimenta at inatel.br Thu Nov 2 09:49:20 2017 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Thu, 2 Nov 2017 13:49:20 +0000 Subject: [OpenSIPS-Users] Table location and command opensipsctl ul show. In-Reply-To: <6c114705-633b-00e5-b949-2741b09eee74@opensips.org> References: , <6c114705-633b-00e5-b949-2741b09eee74@opensips.org> Message-ID: Ok. Thank you very much! RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________ De: Users em nome de Răzvan Crainea Enviado: quinta-feira, 2 de novembro de 2017 11:27 Para: users at lists.opensips.org Assunto: Re: [OpenSIPS-Users] Table location and command opensipsctl ul show. Hi, Rodrigo! The interaction between OpenSIPS memory and database is described here[1]. Depending on your db_mode configuration, you might find entries that are not yet synced in the database. [1] http://www.opensips.org/html/docs/modules/2.4.x/usrloc.html#idp5672576 usrloc Module - opensips.org www.opensips.org How the contacts are matched (for same AOR - Address of Record) is an important aspect of the usrloc modules, especialy in the context of NAT traversal - this raise ... Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com Home — OpenSIPS Solutions www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 11/02/2017 02:33 PM, Rodrigo Pimenta Carvalho wrote: Hi. When I delete all registers from table location, I still can see registers via command 'opensipsctl ul show'. Why a empty table location doesn't gives a null result in 'opensipsctl ul show' ? Is it caused by the configuration from opensips.cfg ? Thanks. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From loreste at vcircuitsusa.com Thu Nov 2 11:33:01 2017 From: loreste at vcircuitsusa.com (Lance Oreste) Date: Thu, 2 Nov 2017 11:33:01 -0400 Subject: [OpenSIPS-Users] jsonrpc Message-ID: How do I post to jonrpc? *We are trying to use this with an external application to add users/domains on opensips 2.3 but it does not seems to be working. We got jsonrpc setup and it is working but have no idea how to send commands to it.* curl -X POST \ http://registrar.tel.dev.overpass.com:8000/nxproxy_ proxy_mi/ \ -H 'cache-control: no-cache' \ -------------- next part -------------- An HTML attachment was scrubbed... URL: From social at bohboh.info Thu Nov 2 12:56:14 2017 From: social at bohboh.info (Social Boh) Date: Thu, 2 Nov 2017 11:56:14 -0500 Subject: [OpenSIPS-Users] ACC Extra field On OpenSIPs Console - CDR Viewer Message-ID: Hello, I'm using OpenSIPs 2.3.2 and OpenSIPs Console 7.2.3. I'd like know if is possible to see Acc Extra Fields, configured on ACC Module, on CDR Viewer menu. I tried modifying config/tools/system/cdrviewer/local.inc.php without success. Regards -- --- I'm SoCIaL, MayBe From bogdan at opensips.org Thu Nov 2 13:03:09 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 2 Nov 2017 19:03:09 +0200 Subject: [OpenSIPS-Users] Differences between dialog replication and db_mode 1 In-Reply-To: References: Message-ID: Hi Matine, When using db_mode 1 (which makes the active opensips to write the dialogs into DB in realtime), how to you load the dialogs from DB into memory on the backup opensips ? do you restart it when becoming active ? because under normal operations OpenSIPS does not read the DB dialogs. The clustered based replication for dialogs has several advantages: * you do not need the a DB at all for replication * if you use DB for restart persistence, you can use the much lighter db_mode 2 to avoid realtime and intensive DB ops * you can create active-active scenarios * it is a real-time replication Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/02/2017 11:15 AM, Matine Tinthaitae wrote: > Hi, > > I'm trying to set up an active/standby solution using floating ip. > Only dialogs have to be shared. > > I've been using both the clusterer module dialog replication and the > dialog module db_mode 1 with a shared database and both seem to do > what i want, which is being able to kill the active node without > losing call states. > > What's the difference/benefit of using the clusterer module over > db_mode 1 in this scenario? > > Thank you! > > //Matine > > > _______________________________________________ > 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 2 13:16:10 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 2 Nov 2017 19:16:10 +0200 Subject: [OpenSIPS-Users] Dispatcher module disable gateway after X failures, auto re-enable after some time In-Reply-To: References: Message-ID: Hi Adam, You can try something like this, it makes sense, considering your custom requirements. The local cache is decent as performance, no worries. In terms of changing the the destination status from script, you can do it only in the context of a call failed to that destination. If this is not good for you, let me know and I can show you some tricks ;) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 10/31/2017 02:10 PM, Adam Raszynski wrote: > My first idea was to create failure counter for each gateway (variable > in local cache) > > Than in onreply route to increment counter every time call is not > established, and reset counter if any call was successful. So we get > consecutive failures counter in cache and can make further decisions > basing on that information > > Does it make sense? Is local cache fast enough (and causes no locking > problems) to be incremented in such intensive way? > > Is it possible to change dispatcher gateway status from inside script? > Or it's possible only by using external interface? > > 2017-10-31 12:31 GMT+01:00 Bogdan-Andrei Iancu >: > > Hi Adam, > > Maybe such a custom logic is outside OpenSIPS scope - still have > you considered using the ds_set_state MI function to control from > outside the status of the destinations ? Of course you will have > to use the Event Interface too, to receive the notifications on > the destination status (when OpenSIPS is doing the state > transition) - see the E_DISPATCHER_STATUS event > (http://www.opensips.org/html/docs/modules/2.3.x/dispatcher.html#idp5746368 > ) > > So, when OpenSIPS switches a destination into probing (based on a > failure detection), you can get the event and force the > destination into Inactive ; after some custom amount of time, you > can put it back into Probing mode, to allow OpenSIPS to re-enable it. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 10/28/2017 04:13 PM, Adam Raszynski wrote: >> Hi >> >> Could you suggest some working configuration of the dispatcher >> module for the following setup: >> >> - Flag gateway as failed (even if it's still responding to >> OPTIONS) after some number of consecutive call failures. That's >> the easy part >> >> - Automatically un-flag gateway after some time, to allow >> re-checking (reset failure counter?) >> >> - Ideal solution would also include increasing re-check interval, >> ie first re-check after 1 minute, if next failure detected next >> re-check after 2 minutes, next after 4 and so on >> >> >> Best Regards >> >> >> >> _______________________________________________ >> 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 2 13:21:23 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 2 Nov 2017 19:21:23 +0200 Subject: [OpenSIPS-Users] AES + Opensip In-Reply-To: References: Message-ID: <6171011d-542f-89f8-9eaa-00f5bd2e1bdd@opensips.org> Hi, Basically your Polycom will try to do SIP over TLS with AES cipher ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/01/2017 06:26 PM, Trupyy . wrote: > Hello Everyone, > > I got a quick question, hope someone can help me with it. So I got two > Polycom VC devices to work with Opensips but the Calls work only when > i turn off the Encryption (AES Option), when i turn it on i get an > error that some security settings are not working properly. > > Can someone tell me if opensips is compatible with AES encryption or > is the only encryption possibility, TLS? > > Thank you. > > > _______________________________________________ > 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 2 13:23:26 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 2 Nov 2017 19:23:26 +0200 Subject: [OpenSIPS-Users] ACC Extra field On OpenSIPs Console - CDR Viewer In-Reply-To: References: Message-ID: Hi, I guess you mean OpenSIPS Control Panel 7.2.3 (and not OpenSIPS Console). And yes, it is possible to configure the CDRViewer to display extra DB fields. What are the changes you did in the local.inc.php file ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/02/2017 06:56 PM, Social Boh wrote: > Hello, > > I'm using OpenSIPs 2.3.2 and OpenSIPs Console 7.2.3. > > I'd like know if is possible to see Acc Extra Fields, configured on > ACC Module, on CDR Viewer menu. > > I tried modifying config/tools/system/cdrviewer/local.inc.php without > success. > > Regards > From social at bohboh.info Thu Nov 2 13:34:41 2017 From: social at bohboh.info (Social Boh) Date: Thu, 2 Nov 2017 12:34:41 -0500 Subject: [OpenSIPS-Users] ACC Extra field On OpenSIPs Console - CDR Viewer In-Reply-To: References: Message-ID: <0587738d-09a1-645b-5df4-b3d11cec75f1@bohboh.info> Sorry, OpenSIPS Control Panel 7.2.3 :) I have this line on ACC Configuration: *modparam("acc", "extra_fields", "db: gw->Gateway")* To know wich gateway OpenSIPs use to make a call. I added this field in acc table and i can see the values. My problem is I don't know how modify correctly: *config/tools/system/cdrviewer/local.inc.php* to see the values of this field on the CDR Viewer menu. With this:  // what fields to show  $show_field[0]['time'] = "Time" ;  $show_field[1]['method'] = "Method" ;  $show_field[2]['callid'] = "Sip Call ID" ;  $show_field[3]['sip_code'] = "Sip Code" ;  $show_field[4]['sip_reason'] = "Sip Reason" ;  $show_field[5]['setuptime'] = "Setup Time";  $show_field[6]['duration'] = "Duration";  $show_field[7]['from_tag'] = "Sip From Tag" ;  $show_field[8]['to_tag'] = "Sip To Tag"; * $show_field[9]['Gateway'] = "SIP Gateway";* (bold) I can see the column but not contain any value. Thank you Regards --- I'm SoCIaL, MayBe El 02/11/2017 a las 12:23, Bogdan-Andrei Iancu escribió: > Hi, > > I guess you mean OpenSIPS Control Panel 7.2.3 (and not OpenSIPS Console). > > And yes, it is possible to configure the CDRViewer to display extra DB > fields. What are the changes you did in the local.inc.php file ? > > Regards, > > Bogdan-Andrei Iancu >   OpenSIPS Founder and Developer >   http://www.opensips-solutions.com > > On 11/02/2017 06:56 PM, Social Boh wrote: >> Hello, >> >> I'm using OpenSIPs 2.3.2 and OpenSIPs Console 7.2.3. >> >> I'd like know if is possible to see Acc Extra Fields, configured on >> ACC Module, on CDR Viewer menu. >> >> I tried modifying >> p without success. >> >> Regards >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From social at bohboh.info Thu Nov 2 16:09:07 2017 From: social at bohboh.info (Social Boh) Date: Thu, 2 Nov 2017 15:09:07 -0500 Subject: [OpenSIPS-Users] OpenSIPs Control Panel Homer Menu Message-ID: <31a910a6-10e7-6aa8-cb53-31d35a6e64ba@bohboh.info> Hello, each time I try to access Homer Menu on OpenSIPs Control Panel 7.3.2, on the apache log: Call to undefined function apc_store() in /var/www/html/opensips-cp/web/tools/system/homer/homer.php on line 5 Blank page OpenSIPs Homer Configuration: $homer_URL = "http://homer.mydomain.org"; $homer_auth_method = "get"; Homer OpenSIPs configuration (preference.php) define('AUTHENTICATION',"External"); define('EXTERNAL_AUTH_URI', "http://os.mydomain.org/cp/tools/system/homer/auth.php";); define('EXTERNAL_AUTH_METHOD', "GET"); define('EXTERNAL_AUTH_PARAM', "param=KEY"); define('EXTERNAL_AUTH_REQUEST_TYPE', "get"); define('EXTERNAL_AUTH_REQUEST_KEY', "externalid"); define('EXTERNAL_AUTH_POSITIVE_REPLY', "200"); define('EXTERNAL_AUTH_REPLY_USER_INFO', "json"); Any hint Thank you in advance. Regards -- --- I'm SoCIaL, MayBe From goup2010 at gmail.com Thu Nov 2 17:43:43 2017 From: goup2010 at gmail.com (Dragomir Haralambiev) Date: Thu, 2 Nov 2017 23:43:43 +0200 Subject: [OpenSIPS-Users] Wrong Contact in location table In-Reply-To: <58c846f0-7fe6-f687-d7aa-b505b20a41ef@opensips.org> References: <222cf774-e269-9853-3436-bef40d8a410d@opensips.org> <0093287c-82e4-c11a-c933-e5e195683156@opensips.org> <741e1ed0-1831-4a1f-aa21-08ed354c2b53@opensips.org> <8045BDD6-2077-4CCC-B6C3-7E007AF631F3@tampahost.net> <9eb0a075-7235-fba0-83c9-62c3ab5de93f@opensips.org> <13955c88-1a4f-d106-10ec-a29525d4c39a@opensips.org> <58c846f0-7fe6-f687-d7aa-b505b20a41ef@opensips.org> Message-ID: Hi, Here is part of my script: .... modparam("registrar", "mcontact_avp", "$avp(register)") ..... if (t_check_status("2[0-9][0-9]")) { $log_level = 5; script_trace( 1, "$rm from $si, ruri=$ru, ct=$ct.fields(uri) avp(register)=$avp(register)", "me"); route(save_location); ....... } You can see log here: https://pastebin.com/WWQ9Mmh4 Here is the replacement contact: DBG:registrar:build_contact: created Contact HF: Contact: < sip:55595009 at 192.168.22.138:5062>;expires=360 DBG:registrar:save: replacing contact uri [sip:55595009 at 188.23.232.10:1043] with [sip:55595009 at 192.168.22.138:5062] How to stop replacing contact from 188.23.232.10:1043 to 192.168.22.138:5062 ? 2017-11-02 11:31 GMT+02:00 Răzvan Crainea : > Did you put the mcontact_avp parameter I asked you to? > Can you print the value of the register just before save()? Also, can you > send over some logs for the reply? > > Best regards, > > Răzvan Crainea > OpenSIPS Developerwww.opensips-solutions.com > > On 11/02/2017 12:26 AM, Dragomir Haralambiev wrote: > > Hi, > > I try this but in location table I not see real IP. Problem exists. > > When make save location $avp(register)=sip:55996201 at 192.27.47.139:5060 > > opensipsctl ul show 55996201 at 192.27.47.139 > > AOR:: 55996201 > Contact:: sip:57996201 at 192.168.22.201:5060 Q= > ContactID:: 182005870676967428 > > Where is problem? > > > 2017-11-01 15:08 GMT+02:00 OpenSIPS Marketing : > >> OK, I managed to figure out why you get the private IP in the location >> table - that's because you are saving the registration in the onreply >> route, where you no longer have access to the fix_nated_contact() value. >> >> To fix your issue, you need to do the following: >> a. specify a mcontact_avp[1] in your script: >> modparam("registrar", "mcontact_avp", "$avp(register)") >> >> b. just after fix_nated_contact() on your REGISTER request, store the >> changed contact in that AVP: >> fix_nated_contact(); >> $avp(register) = $ct.fields(uri); >> >> Doing these two changes to your script should store the real IP in the >> location table. >> >> [1] http://www.opensips.org/html/docs/modules/2.4.x/registrar.ht >> ml#idp5598160 >> >> Best regards, >> >> On 11/01/2017 10:24 AM, Răzvan Crainea wrote: >> >> Can you print $ct.fields(uri) instead of $ct? >> Also, please post the output on pastebin.com. >> >> Best regards, >> >> Răzvan Crainea >> OpenSIPS Developerwww.opensips-solutions.com >> >> On 10/31/2017 03:47 PM, Răzvan Crainea wrote: >> >> Hi, Dragomir! >> >> Please post the logs I asked you to. >> >> Best regards, >> >> Răzvan Crainea >> OpenSIPS Developerwww.opensips-solutions.com >> >> On 10/31/2017 03:45 PM, Sumit Birla wrote: >> >> X-Lite figures out its public IP and uses that in Via: and Contact: >> headers unless you set ‘Firewall traversal method’ to ‘None’ in the account >> settings. >> >> >> >> >> On Oct 31, 2017, at 9:14 AM, Dragomir Haralambiev >> wrote: >> >> When REGISTER received from X-lite all is OK. Here part from REGISTER >> message: >> >> REGISTER sip:OpenSip_IP:Opensips_port;transport=UDP SIP/2.0 >> Via: SIP/2.0/UDP 219.101.241.150:64726;branch=z >> 9hG4bK-524287-1---39ee8718032a1a5a;rport >> Contact: > 122d9250e9d0bd> >> >> Here "Via" contains real IP (this is Route IP). >> >> When REGISTER received from Yealink VoIP phone I have problem. Here part >> from REGISTER message: >> >> REGISTER sip:OpenSip_IP:Opensips_port SIP/2.0 >> Via: SIP/2.0/UDP 192.168.2.207:5062;branch=z9hG4bK77238423 >> Contact: >> >> Here "Via" contains not real IP (this is Yealink IP). >> >> What I do ? >> >> >> 2017-10-31 14:18 GMT+02:00 Răzvan Crainea : >> >>> So you did manage to fix your problem? >>> >>> Răzvan Crainea >>> OpenSIPS Developerwww.opensips-solutions.com >>> >>> On 10/31/2017 02:17 PM, Dragomir Haralambiev wrote: >>> >>> Hi, >>> >>> Thanks for your replay. >>> >>> I execute fix_nated_contact() after fix_nated_register(). >>> >>> Here part of my script: >>> if (isflagset(NAT)) { >>> if (is_method("REGISTER")) { >>> fix_nated_register(); >>> setbflag(NAT_BFLAG); # nat register >>> } >>> fix_nated_contact(); >>> } >>> >>> In this case fix_nated_contact() will be fix contact records in location >>> table. >>> >>> >>> 2017-10-31 14:02 GMT+02:00 Răzvan Crainea : >>> >>>> Hi, Dragomir! >>>> >>>> fix_nated_register() does not change the contact of the message, >>>> fix_nated_contact() does that. >>>> Please print the Contact header as I instructed earlier and send back >>>> the script trace. Also, please post the logs on pastebin.com or >>>> something, don't do it directly in the email. >>>> >>>> Best regards, >>>> >>>> Răzvan Crainea >>>> OpenSIPS Developerwww.opensips-solutions.com >>>> >>>> On 10/31/2017 12:40 AM, Dragomir Haralambiev wrote: >>>> >>>> What will I do to solve this problem? >>>> >>>> 2017-10-27 11:50 GMT+03:00 Dragomir Haralambiev : >>>> >>>>> Oct 27 11:47:32 web /usr/local/sbin/opensips[30490]: 57995010: >>>>> REGISTER before fix_nated_register >>>> >[1] >>>>> Oct 27 11:47:32 web /usr/local/sbin/opensips[30490]: 57995010: >>>>> REGISTER after fix_nated_register [1] >>>>> >>>>> >>>>> 2017-10-27 10:41 GMT+03:00 Răzvan Crainea : >>>>> >>>>>> Hi, Dragomir! >>>>>> >>>>>> Can you also print the Contact header ($ct[1]). >>>>>> >>>>>> [1] http://www.opensips.org/Documentation/Script-CoreVar-2-4#toc26 >>>>>> >>>>>> Best regards, >>>>>> >>>>>> Răzvan Crainea >>>>>> OpenSIPS Developerwww.opensips-solutions.com >>>>>> >>>>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>> >>> >>> _______________________________________________ >>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Fri Nov 3 05:45:45 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Fri, 3 Nov 2017 11:45:45 +0200 Subject: [OpenSIPS-Users] Wrong Contact in location table In-Reply-To: References: <0093287c-82e4-c11a-c933-e5e195683156@opensips.org> <741e1ed0-1831-4a1f-aa21-08ed354c2b53@opensips.org> <8045BDD6-2077-4CCC-B6C3-7E007AF631F3@tampahost.net> <9eb0a075-7235-fba0-83c9-62c3ab5de93f@opensips.org> <13955c88-1a4f-d106-10ec-a29525d4c39a@opensips.org> <58c846f0-7fe6-f687-d7aa-b505b20a41ef@opensips.org> Message-ID: Unfortunately I just realised that you cannot change the Contact header for this scenario. And to be honest I don't really understand why you are trying to change it - if you need the real IP and port, you can take them from the received field. Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/02/2017 11:43 PM, Dragomir Haralambiev wrote: > Hi, > > Here is part of my script: > .... > modparam("registrar", "mcontact_avp", "$avp(register)") > ..... > >     if (t_check_status("2[0-9][0-9]"))  { > $log_level = 5; > script_trace( 1, "$rm from $si, ruri=$ru, ct=$ct.fields(uri) > avp(register)=$avp(register)", "me"); > route(save_location); >                        ....... >     } > > You can see log here: > https://pastebin.com/WWQ9Mmh4 > > Here is the replacement contact: > > DBG:registrar:build_contact: created Contact HF: Contact: > >;expires=360 > DBG:registrar:save: replacing contact uri > [sip:55595009 at 188.23.232.10:1043 > ] with > [sip:55595009 at 192.168.22.138:5062 > ] > > How to stop replacing contact from 188.23.232.10:1043 > to 192.168.22.138:5062 > ? -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Fri Nov 3 05:47:07 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Fri, 3 Nov 2017 11:47:07 +0200 Subject: [OpenSIPS-Users] OpenSIPs Control Panel Homer Menu In-Reply-To: <31a910a6-10e7-6aa8-cb53-31d35a6e64ba@bohboh.info> References: <31a910a6-10e7-6aa8-cb53-31d35a6e64ba@bohboh.info> Message-ID: Hello! You should install the php apc library. On Ubuntu this is php-apcu. Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/02/2017 10:09 PM, Social Boh wrote: > Hello, > > each time I try to access Homer Menu on OpenSIPs Control Panel 7.3.2, > on the apache log: > > Call to undefined function apc_store() in > /var/www/html/opensips-cp/web/tools/system/homer/homer.php on line 5 > > Blank page > > OpenSIPs Homer Configuration: > > $homer_URL = "http://homer.mydomain.org"; > > $homer_auth_method = "get"; > > Homer OpenSIPs configuration (preference.php) > > define('AUTHENTICATION',"External"); > define('EXTERNAL_AUTH_URI', > "http://os.mydomain.org/cp/tools/system/homer/auth.php";); > define('EXTERNAL_AUTH_METHOD', "GET"); > define('EXTERNAL_AUTH_PARAM', "param=KEY"); > define('EXTERNAL_AUTH_REQUEST_TYPE', "get"); > define('EXTERNAL_AUTH_REQUEST_KEY', "externalid"); > define('EXTERNAL_AUTH_POSITIVE_REPLY', "200"); > define('EXTERNAL_AUTH_REPLY_USER_INFO', "json"); > > Any hint > > Thank you in advance. > > Regards > From bogdan at opensips.org Fri Nov 3 07:09:47 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 3 Nov 2017 13:09:47 +0200 Subject: [OpenSIPS-Users] ACC Extra field On OpenSIPs Console - CDR Viewer In-Reply-To: <0587738d-09a1-645b-5df4-b3d11cec75f1@bohboh.info> References: <0587738d-09a1-645b-5df4-b3d11cec75f1@bohboh.info> Message-ID: Hi, The change looks ok - are you sure you have data in the "Gateway" column in the DB ? Do you see any errors in the apache log when opening the CDRViewer tool ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/02/2017 07:34 PM, Social Boh wrote: > > Sorry, > > OpenSIPS Control Panel 7.2.3 :) > > I have this line on ACC Configuration: > > *modparam("acc", "extra_fields", "db: gw->Gateway")* > > To know wich gateway OpenSIPs use to make a call. > > I added this field in acc table and i can see the values. > > My problem is I don't know how modify correctly: > > *config/tools/system/cdrviewer/local.inc.php* > > to see the values of this field on the CDR Viewer menu. With this: > > // what fields to show > $show_field[0]['time'] = "Time" ; > $show_field[1]['method'] = "Method" ; > $show_field[2]['callid'] = "Sip Call ID" ; > $show_field[3]['sip_code'] = "Sip Code" ; > $show_field[4]['sip_reason'] = "Sip Reason" ; > $show_field[5]['setuptime'] = "Setup Time"; > $show_field[6]['duration'] = "Duration"; > $show_field[7]['from_tag'] = "Sip From Tag" ; > $show_field[8]['to_tag'] = "Sip To Tag"; > * $show_field[9]['Gateway'] = "SIP Gateway";* > > (bold) I can see the column but not contain any value. > > Thank you > > Regards > > --- > I'm SoCIaL, MayBe > El 02/11/2017 a las 12:23, Bogdan-Andrei Iancu escribió: >> Hi, >> >> I guess you mean OpenSIPS Control Panel 7.2.3 (and not OpenSIPS >> Console). >> >> And yes, it is possible to configure the CDRViewer to display extra >> DB fields. What are the changes you did in the local.inc.php file ? >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> On 11/02/2017 06:56 PM, Social Boh wrote: >>> Hello, >>> >>> I'm using OpenSIPs 2.3.2 and OpenSIPs Console 7.2.3. >>> >>> I'd like know if is possible to see Acc Extra Fields, configured on >>> ACC Module, on CDR Viewer menu. >>> >>> I tried modifying > >>> p without success. >>> >>> Regards >>> >> > > > > _______________________________________________ > 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 social at bohboh.info Fri Nov 3 07:44:31 2017 From: social at bohboh.info (Social Boh) Date: Fri, 3 Nov 2017 06:44:31 -0500 Subject: [OpenSIPS-Users] ACC Extra field On OpenSIPs Console - CDR Viewer In-Reply-To: References: <0587738d-09a1-645b-5df4-b3d11cec75f1@bohboh.info> Message-ID: <89b42a3b-f45b-df64-6cfa-5e489d9cb3c2@bohboh.info> Hello, I have this message:  PHP Notice:  Undefined index: Gateway in /var/www/html/opensips-cp/web/tools/system/cdrviewer/template/cdrviewer.main.php on line 236, referer: http://X.X.X.X/cp/menu.php Thank you Regards --- I'm SoCIaL, MayBe El 03/11/2017 a las 06:09, Bogdan-Andrei Iancu escribió: > Hi, > > The change looks ok - are you sure you have data in the "Gateway" > column in the DB ? Do you see any errors in the apache log when > opening the CDRViewer tool ? > > Regards, > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > On 11/02/2017 07:34 PM, Social Boh wrote: >> >> Sorry, >> >> OpenSIPS Control Panel 7.2.3 :) >> >> I have this line on ACC Configuration: >> >> *modparam("acc", "extra_fields", "db: gw->Gateway")* >> >> To know wich gateway OpenSIPs use to make a call. >> >> I added this field in acc table and i can see the values. >> >> My problem is I don't know how modify correctly: >> >> *config/tools/system/cdrviewer/local.inc.php* >> >> to see the values of this field on the CDR Viewer menu. With this: >> >>  // what fields to show >>  $show_field[0]['time'] = "Time" ; >>  $show_field[1]['method'] = "Method" ; >>  $show_field[2]['callid'] = "Sip Call ID" ; >>  $show_field[3]['sip_code'] = "Sip Code" ; >>  $show_field[4]['sip_reason'] = "Sip Reason" ; >>  $show_field[5]['setuptime'] = "Setup Time"; >>  $show_field[6]['duration'] = "Duration"; >>  $show_field[7]['from_tag'] = "Sip From Tag" ; >>  $show_field[8]['to_tag'] = "Sip To Tag"; >> * $show_field[9]['Gateway'] = "SIP Gateway";* >> >> (bold) I can see the column but not contain any value. >> >> Thank you >> >> Regards >> >> --- >> I'm SoCIaL, MayBe >> El 02/11/2017 a las 12:23, Bogdan-Andrei Iancu escribió: >>> Hi, >>> >>> I guess you mean OpenSIPS Control Panel 7.2.3 (and not OpenSIPS >>> Console). >>> >>> And yes, it is possible to configure the CDRViewer to display extra >>> DB fields. What are the changes you did in the local.inc.php file ? >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>>   OpenSIPS Founder and Developer >>> http://www.opensips-solutions.com >>> >>> On 11/02/2017 06:56 PM, Social Boh wrote: >>>> Hello, >>>> >>>> I'm using OpenSIPs 2.3.2 and OpenSIPs Console 7.2.3. >>>> >>>> I'd like know if is possible to see Acc Extra Fields, configured on >>>> ACC Module, on CDR Viewer menu. >>>> >>>> I tried modifying >> >>>> p without success. >>>> >>>> Regards >>>> >>> >> >> >> >> _______________________________________________ >> 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 social at bohboh.info Fri Nov 3 08:06:23 2017 From: social at bohboh.info (Social Boh) Date: Fri, 3 Nov 2017 07:06:23 -0500 Subject: [OpenSIPS-Users] OpenSIPs Control Panel Homer Menu In-Reply-To: References: <31a910a6-10e7-6aa8-cb53-31d35a6e64ba@bohboh.info> Message-ID: <667cccd5-ae01-9780-9c0d-eb34c39355aa@bohboh.info> Hello, Now I have this error (homer side): GET /api/v1/redirect?externalid=rx88kxQUs1DNT6es2ybwBvhevecJwjnY&url=http%3A%2F%2F1.2.3.4.0%2F HTTP/1.1. Host: 104.236.105.0. User-Agent: Mozilla/5.0 (Windows NT 6.1; Win64; x64; rv:56.0) Gecko/20100101 Firefox/56.0. Accept: text/html,application/xhtml+xml,application/xml;q=0.9,*/*;q=0.8. Accept-Language: es-ES,es;q=0.8,en-US;q=0.5,en;q=0.3. Accept-Encoding: gzip, deflate. Referer: http://5.6.7.8/cp/tools/system/homer/homer.php. Connection: keep-alive. Upgrade-Insecure-Requests: 1. HTTP/1.0 500 Internal Server Error. Date: Fri, 03 Nov 2017 11:47:36 GMT. Server: Apache/2.4.6 (CentOS) OpenSSL/1.0.2k-fips PHP/5.4.16. X-Powered-By: PHP/5.4.16. Content-Length: 0. Connection: close. Content-Type: text/html; charset=UTF-8. My homer configuration is: define('AUTHENTICATION',"External"); define('EXTERNAL_AUTH_URI', "http://5.6.7.8/cp/tools/system/homer/auth.php";); define('EXTERNAL_AUTH_METHOD', "GET"); define('EXTERNAL_AUTH_PARAM', "param=KEY"); define('EXTERNAL_AUTH_REQUEST_TYPE', "get"); define('EXTERNAL_AUTH_REQUEST_KEY', "externalid"); define('EXTERNAL_AUTH_POSITIVE_REPLY', "200"); define('EXTERNAL_AUTH_REPLY_USER_INFO', "json"); 1.2.3.4 is the Homer IP and 5.6.7.8 is the OpenSIPs Control Panel IP Thank you Regards --- I'm SoCIaL, MayBe El 03/11/2017 a las 04:47, Răzvan Crainea escribió: > Hello! > > You should install the php apc library. On Ubuntu this is php-apcu. > > Best regards, > > Răzvan Crainea > OpenSIPS Developer > www.opensips-solutions.com > > On 11/02/2017 10:09 PM, Social Boh wrote: >> Hello, >> >> each time I try to access Homer Menu on OpenSIPs Control Panel 7.3.2, >> on the apache log: >> >> Call to undefined function apc_store() in >> /var/www/html/opensips-cp/web/tools/system/homer/homer.php on line 5 >> >> Blank page >> >> OpenSIPs Homer Configuration: >> >> $homer_URL = "http://homer.mydomain.org"; >> >> $homer_auth_method = "get"; >> >> Homer OpenSIPs configuration (preference.php) >> >> define('AUTHENTICATION',"External"); >> define('EXTERNAL_AUTH_URI', >> "http://os.mydomain.org/cp/tools/system/homer/auth.php";); >> define('EXTERNAL_AUTH_METHOD', "GET"); >> define('EXTERNAL_AUTH_PARAM', "param=KEY"); >> define('EXTERNAL_AUTH_REQUEST_TYPE', "get"); >> define('EXTERNAL_AUTH_REQUEST_KEY', "externalid"); >> define('EXTERNAL_AUTH_POSITIVE_REPLY', "200"); >> define('EXTERNAL_AUTH_REPLY_USER_INFO', "json"); >> >> Any hint >> >> Thank you in advance. >> >> Regards >> > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > From bogdan at opensips.org Fri Nov 3 08:09:00 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 3 Nov 2017 14:09:00 +0200 Subject: [OpenSIPS-Users] ACC Extra field On OpenSIPs Console - CDR Viewer In-Reply-To: <89b42a3b-f45b-df64-6cfa-5e489d9cb3c2@bohboh.info> References: <0587738d-09a1-645b-5df4-b3d11cec75f1@bohboh.info> <89b42a3b-f45b-df64-6cfa-5e489d9cb3c2@bohboh.info> Message-ID: That means the "Gateway" column was *not* found in the DB query result (in the accounting table). Try to click on the CDR details icon to see the raw format for the CDR - check if your column is there. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/03/2017 01:44 PM, Social Boh wrote: > > Hello, I have this message: > > PHP Notice: Undefined index: Gateway in > /var/www/html/opensips-cp/web/tools/system/cdrviewer/template/cdrviewer.main.php > on line 236, referer: http://X.X.X.X/cp/menu.php > > Thank you > > Regards > > --- > I'm SoCIaL, MayBe > El 03/11/2017 a las 06:09, Bogdan-Andrei Iancu escribió: >> Hi, >> >> The change looks ok - are you sure you have data in the "Gateway" >> column in the DB ? Do you see any errors in the apache log when >> opening the CDRViewer tool ? >> >> Regards, >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> On 11/02/2017 07:34 PM, Social Boh wrote: >>> >>> Sorry, >>> >>> OpenSIPS Control Panel 7.2.3 :) >>> >>> I have this line on ACC Configuration: >>> >>> *modparam("acc", "extra_fields", "db: gw->Gateway")* >>> >>> To know wich gateway OpenSIPs use to make a call. >>> >>> I added this field in acc table and i can see the values. >>> >>> My problem is I don't know how modify correctly: >>> >>> *config/tools/system/cdrviewer/local.inc.php* >>> >>> to see the values of this field on the CDR Viewer menu. With this: >>> >>> // what fields to show >>> $show_field[0]['time'] = "Time" ; >>> $show_field[1]['method'] = "Method" ; >>> $show_field[2]['callid'] = "Sip Call ID" ; >>> $show_field[3]['sip_code'] = "Sip Code" ; >>> $show_field[4]['sip_reason'] = "Sip Reason" ; >>> $show_field[5]['setuptime'] = "Setup Time"; >>> $show_field[6]['duration'] = "Duration"; >>> $show_field[7]['from_tag'] = "Sip From Tag" ; >>> $show_field[8]['to_tag'] = "Sip To Tag"; >>> * $show_field[9]['Gateway'] = "SIP Gateway";* >>> >>> (bold) I can see the column but not contain any value. >>> >>> Thank you >>> >>> Regards >>> >>> --- >>> I'm SoCIaL, MayBe >>> El 02/11/2017 a las 12:23, Bogdan-Andrei Iancu escribió: >>>> Hi, >>>> >>>> I guess you mean OpenSIPS Control Panel 7.2.3 (and not OpenSIPS >>>> Console). >>>> >>>> And yes, it is possible to configure the CDRViewer to display extra >>>> DB fields. What are the changes you did in the local.inc.php file ? >>>> >>>> Regards, >>>> >>>> Bogdan-Andrei Iancu >>>> OpenSIPS Founder and Developer >>>> http://www.opensips-solutions.com >>>> >>>> On 11/02/2017 06:56 PM, Social Boh wrote: >>>>> Hello, >>>>> >>>>> I'm using OpenSIPs 2.3.2 and OpenSIPs Console 7.2.3. >>>>> >>>>> I'd like know if is possible to see Acc Extra Fields, configured >>>>> on ACC Module, on CDR Viewer menu. >>>>> >>>>> I tried modifying >>> >>>>> p without success. >>>>> >>>>> Regards >>>>> >>>> >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Fri Nov 3 08:15:00 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Fri, 3 Nov 2017 14:15:00 +0200 Subject: [OpenSIPS-Users] OpenSIPs Control Panel Homer Menu In-Reply-To: <667cccd5-ae01-9780-9c0d-eb34c39355aa@bohboh.info> References: <31a910a6-10e7-6aa8-cb53-31d35a6e64ba@bohboh.info> <667cccd5-ae01-9780-9c0d-eb34c39355aa@bohboh.info> Message-ID: <01993f73-4c11-de06-c8fe-3e5b93c62580@opensips.org> This time there seems to be an issue on the homer side. Can you check the logs to see if homer is saying anything useful? Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/03/2017 02:06 PM, Social Boh wrote: > Hello, > > Now I have this error (homer side): > > GET > /api/v1/redirect?externalid=rx88kxQUs1DNT6es2ybwBvhevecJwjnY&url=http%3A%2F%2F1.2.3.4.0%2F > HTTP/1.1. > Host: 104.236.105.0. > User-Agent: Mozilla/5.0 (Windows NT 6.1; Win64; x64; rv:56.0) > Gecko/20100101 Firefox/56.0. > Accept: text/html,application/xhtml+xml,application/xml;q=0.9,*/*;q=0.8. > Accept-Language: es-ES,es;q=0.8,en-US;q=0.5,en;q=0.3. > Accept-Encoding: gzip, deflate. > Referer: http://5.6.7.8/cp/tools/system/homer/homer.php. > Connection: keep-alive. > Upgrade-Insecure-Requests: 1. > > HTTP/1.0 500 Internal Server Error. > Date: Fri, 03 Nov 2017 11:47:36 GMT. > Server: Apache/2.4.6 (CentOS) OpenSSL/1.0.2k-fips PHP/5.4.16. > X-Powered-By: PHP/5.4.16. > Content-Length: 0. > Connection: close. > Content-Type: text/html; charset=UTF-8. > > My homer configuration is: > > define('AUTHENTICATION',"External"); > define('EXTERNAL_AUTH_URI', > "http://5.6.7.8/cp/tools/system/homer/auth.php";); > define('EXTERNAL_AUTH_METHOD', "GET"); > define('EXTERNAL_AUTH_PARAM', "param=KEY"); > define('EXTERNAL_AUTH_REQUEST_TYPE', "get"); > define('EXTERNAL_AUTH_REQUEST_KEY', "externalid"); > define('EXTERNAL_AUTH_POSITIVE_REPLY', "200"); > define('EXTERNAL_AUTH_REPLY_USER_INFO', "json"); > > 1.2.3.4 is the Homer IP and 5.6.7.8 is the OpenSIPs Control Panel IP > > Thank you > > Regards > > --- > I'm SoCIaL, MayBe > > El 03/11/2017 a las 04:47, Răzvan Crainea escribió: >> Hello! >> >> You should install the php apc library. On Ubuntu this is php-apcu. >> >> Best regards, >> >> Răzvan Crainea >> OpenSIPS Developer >> www.opensips-solutions.com >> >> On 11/02/2017 10:09 PM, Social Boh wrote: >>> Hello, >>> >>> each time I try to access Homer Menu on OpenSIPs Control Panel >>> 7.3.2, on the apache log: >>> >>> Call to undefined function apc_store() in >>> /var/www/html/opensips-cp/web/tools/system/homer/homer.php on line 5 >>> >>> Blank page >>> >>> OpenSIPs Homer Configuration: >>> >>> $homer_URL = "http://homer.mydomain.org"; >>> >>> $homer_auth_method = "get"; >>> >>> Homer OpenSIPs configuration (preference.php) >>> >>> define('AUTHENTICATION',"External"); >>> define('EXTERNAL_AUTH_URI', >>> "http://os.mydomain.org/cp/tools/system/homer/auth.php";); >>> define('EXTERNAL_AUTH_METHOD', "GET"); >>> define('EXTERNAL_AUTH_PARAM', "param=KEY"); >>> define('EXTERNAL_AUTH_REQUEST_TYPE', "get"); >>> define('EXTERNAL_AUTH_REQUEST_KEY', "externalid"); >>> define('EXTERNAL_AUTH_POSITIVE_REPLY', "200"); >>> define('EXTERNAL_AUTH_REPLY_USER_INFO', "json"); >>> >>> Any hint >>> >>> Thank you in advance. >>> >>> Regards >>> >> >> >> _______________________________________________ >> 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 social at bohboh.info Fri Nov 3 11:21:30 2017 From: social at bohboh.info (Social Boh) Date: Fri, 3 Nov 2017 10:21:30 -0500 Subject: [OpenSIPS-Users] ACC Extra field On OpenSIPs Console - CDR Viewer In-Reply-To: References: <0587738d-09a1-645b-5df4-b3d11cec75f1@bohboh.info> <89b42a3b-f45b-df64-6cfa-5e489d9cb3c2@bohboh.info> Message-ID: Hello, in Details, there is (image)... Thank you --- I'm SoCIaL, MayBe El 03/11/2017 a las 07:09, Bogdan-Andrei Iancu escribió: > That means the "Gateway" column was *not* found in the DB query result > (in the accounting table). > > Try to click on the CDR details icon to see the raw format for the CDR > - check if your column is there. > > Regards, > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > On 11/03/2017 01:44 PM, Social Boh wrote: >> >> Hello, I have this message: >> >>  PHP Notice:  Undefined index: Gateway in >> /var/www/html/opensips-cp/web/tools/system/cdrviewer/template/cdrviewer.main.php >> on line 236, referer: http://X.X.X.X/cp/menu.php >> >> Thank you >> >> Regards >> >> --- >> I'm SoCIaL, MayBe >> El 03/11/2017 a las 06:09, Bogdan-Andrei Iancu escribió: >>> Hi, >>> >>> The change looks ok - are you sure you have data in the "Gateway" >>> column in the DB ? Do you see any errors in the apache log when >>> opening the CDRViewer tool ? >>> >>> Regards, >>> Bogdan-Andrei Iancu >>> OpenSIPS Founder and Developer >>> http://www.opensips-solutions.com >>> On 11/02/2017 07:34 PM, Social Boh wrote: >>>> >>>> Sorry, >>>> >>>> OpenSIPS Control Panel 7.2.3 :) >>>> >>>> I have this line on ACC Configuration: >>>> >>>> *modparam("acc", "extra_fields", "db: gw->Gateway")* >>>> >>>> To know wich gateway OpenSIPs use to make a call. >>>> >>>> I added this field in acc table and i can see the values. >>>> >>>> My problem is I don't know how modify correctly: >>>> >>>> *config/tools/system/cdrviewer/local.inc.php* >>>> >>>> to see the values of this field on the CDR Viewer menu. With this: >>>> >>>>  // what fields to show >>>>  $show_field[0]['time'] = "Time" ; >>>>  $show_field[1]['method'] = "Method" ; >>>>  $show_field[2]['callid'] = "Sip Call ID" ; >>>>  $show_field[3]['sip_code'] = "Sip Code" ; >>>>  $show_field[4]['sip_reason'] = "Sip Reason" ; >>>>  $show_field[5]['setuptime'] = "Setup Time"; >>>>  $show_field[6]['duration'] = "Duration"; >>>>  $show_field[7]['from_tag'] = "Sip From Tag" ; >>>>  $show_field[8]['to_tag'] = "Sip To Tag"; >>>> * $show_field[9]['Gateway'] = "SIP Gateway";* >>>> >>>> (bold) I can see the column but not contain any value. >>>> >>>> Thank you >>>> >>>> Regards >>>> >>>> --- >>>> I'm SoCIaL, MayBe >>>> El 02/11/2017 a las 12:23, Bogdan-Andrei Iancu escribió: >>>>> Hi, >>>>> >>>>> I guess you mean OpenSIPS Control Panel 7.2.3 (and not OpenSIPS >>>>> Console). >>>>> >>>>> And yes, it is possible to configure the CDRViewer to display >>>>> extra DB fields. What are the changes you did in the local.inc.php >>>>> file ? >>>>> >>>>> Regards, >>>>> >>>>> Bogdan-Andrei Iancu >>>>>   OpenSIPS Founder and Developer >>>>> http://www.opensips-solutions.com >>>>> >>>>> On 11/02/2017 06:56 PM, Social Boh wrote: >>>>>> Hello, >>>>>> >>>>>> I'm using OpenSIPs 2.3.2 and OpenSIPs Console 7.2.3. >>>>>> >>>>>> I'd like know if is possible to see Acc Extra Fields, configured >>>>>> on ACC Module, on CDR Viewer menu. >>>>>> >>>>>> I tried modifying >>>> >>>>>> p without success. >>>>>> >>>>>> Regards >>>>>> >>>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> 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: details.png Type: image/png Size: 8791 bytes Desc: not available URL: From social at bohboh.info Fri Nov 3 11:22:26 2017 From: social at bohboh.info (Social Boh) Date: Fri, 3 Nov 2017 10:22:26 -0500 Subject: [OpenSIPS-Users] OpenSIPs Control Panel Homer Menu In-Reply-To: <01993f73-4c11-de06-c8fe-3e5b93c62580@opensips.org> References: <31a910a6-10e7-6aa8-cb53-31d35a6e64ba@bohboh.info> <667cccd5-ae01-9780-9c0d-eb34c39355aa@bohboh.info> <01993f73-4c11-de06-c8fe-3e5b93c62580@opensips.org> Message-ID: <32fe0423-10e7-6f23-2d01-3ebeb4f86b98@bohboh.info> The error was related with this semicolon (before the last parenthesis): define('EXTERNAL_AUTH_URI', "http://5.6.7.8/cp/tools/system/homer/auth.php"*;*); Thank you. Regards --- I'm SoCIaL, MayBe El 03/11/2017 a las 07:15, Răzvan Crainea escribió: > This time there seems to be an issue on the homer side. Can you check > the logs to see if homer is saying anything useful? > > Best regards, > > Răzvan Crainea > OpenSIPS Developer > www.opensips-solutions.com > > On 11/03/2017 02:06 PM, Social Boh wrote: >> Hello, >> >> Now I have this error (homer side): >> >> GET >> /api/v1/redirect?externalid=rx88kxQUs1DNT6es2ybwBvhevecJwjnY&url=http%3A%2F%2F1.2.3.4.0%2F >> HTTP/1.1. >> Host: 104.236.105.0. >> User-Agent: Mozilla/5.0 (Windows NT 6.1; Win64; x64; rv:56.0) >> Gecko/20100101 Firefox/56.0. >> Accept: text/html,application/xhtml+xml,application/xml;q=0.9,*/*;q=0.8. >> Accept-Language: es-ES,es;q=0.8,en-US;q=0.5,en;q=0.3. >> Accept-Encoding: gzip, deflate. >> Referer: http://5.6.7.8/cp/tools/system/homer/homer.php. >> Connection: keep-alive. >> Upgrade-Insecure-Requests: 1. >> >> HTTP/1.0 500 Internal Server Error. >> Date: Fri, 03 Nov 2017 11:47:36 GMT. >> Server: Apache/2.4.6 (CentOS) OpenSSL/1.0.2k-fips PHP/5.4.16. >> X-Powered-By: PHP/5.4.16. >> Content-Length: 0. >> Connection: close. >> Content-Type: text/html; charset=UTF-8. >> >> My homer configuration is: >> >> define('AUTHENTICATION',"External"); >> define('EXTERNAL_AUTH_URI', >> "http://5.6.7.8/cp/tools/system/homer/auth.php";); >> define('EXTERNAL_AUTH_METHOD', "GET"); >> define('EXTERNAL_AUTH_PARAM', "param=KEY"); >> define('EXTERNAL_AUTH_REQUEST_TYPE', "get"); >> define('EXTERNAL_AUTH_REQUEST_KEY', "externalid"); >> define('EXTERNAL_AUTH_POSITIVE_REPLY', "200"); >> define('EXTERNAL_AUTH_REPLY_USER_INFO', "json"); >> >> 1.2.3.4 is the Homer IP and 5.6.7.8 is the OpenSIPs Control Panel IP >> >> Thank you >> >> Regards >> >> --- >> I'm SoCIaL, MayBe >> >> El 03/11/2017 a las 04:47, Răzvan Crainea escribió: >>> Hello! >>> >>> You should install the php apc library. On Ubuntu this is php-apcu. >>> >>> Best regards, >>> >>> Răzvan Crainea >>> OpenSIPS Developer >>> www.opensips-solutions.com >>> >>> On 11/02/2017 10:09 PM, Social Boh wrote: >>>> Hello, >>>> >>>> each time I try to access Homer Menu on OpenSIPs Control Panel >>>> 7.3.2, on the apache log: >>>> >>>> Call to undefined function apc_store() in >>>> /var/www/html/opensips-cp/web/tools/system/homer/homer.php on line 5 >>>> >>>> Blank page >>>> >>>> OpenSIPs Homer Configuration: >>>> >>>> $homer_URL = "http://homer.mydomain.org"; >>>> >>>> $homer_auth_method = "get"; >>>> >>>> Homer OpenSIPs configuration (preference.php) >>>> >>>> define('AUTHENTICATION',"External"); >>>> define('EXTERNAL_AUTH_URI', >>>> "http://os.mydomain.org/cp/tools/system/homer/auth.php";); >>>> define('EXTERNAL_AUTH_METHOD', "GET"); >>>> define('EXTERNAL_AUTH_PARAM', "param=KEY"); >>>> define('EXTERNAL_AUTH_REQUEST_TYPE', "get"); >>>> define('EXTERNAL_AUTH_REQUEST_KEY', "externalid"); >>>> define('EXTERNAL_AUTH_POSITIVE_REPLY', "200"); >>>> define('EXTERNAL_AUTH_REPLY_USER_INFO', "json"); >>>> >>>> Any hint >>>> >>>> Thank you in advance. >>>> >>>> Regards >>>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From fehmfehmfehmfehmfehm at gmail.com Fri Nov 3 11:36:24 2017 From: fehmfehmfehmfehmfehm at gmail.com (Matine Tinthaitae) Date: Fri, 3 Nov 2017 16:36:24 +0100 Subject: [OpenSIPS-Users] Differences between dialog replication and db_mode 1 In-Reply-To: References: Message-ID: Hi Bogdan, Thank you for the quick reply. I had setup the cluster to start opensips on failover so i didn't realize that it only read the dialogs from database on startup. So in my case it's more or less the same functionally but i might save on performance by going with clusterer and maybe even db_mode 0. Considering the news in OpenSIPS 2.4 getting to know the clusterer module seems to be the way to go. :) Again, thank you for the help and thank you for all the great work! //Matine 2017-11-02 18:03 GMT+01:00 Bogdan-Andrei Iancu : > Hi Matine, > > When using db_mode 1 (which makes the active opensips to write the dialogs > into DB in realtime), how to you load the dialogs from DB into memory on > the backup opensips ? do you restart it when becoming active ? because > under normal operations OpenSIPS does not read the DB dialogs. > > The clustered based replication for dialogs has several advantages: > * you do not need the a DB at all for replication > * if you use DB for restart persistence, you can use the much lighter > db_mode 2 to avoid realtime and intensive DB ops > * you can create active-active scenarios > * it is a real-time replication > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 11/02/2017 11:15 AM, Matine Tinthaitae wrote: > > Hi, > > I'm trying to set up an active/standby solution using floating ip. Only > dialogs have to be shared. > > I've been using both the clusterer module dialog replication and the > dialog module db_mode 1 with a shared database and both seem to do what i > want, which is being able to kill the active node without losing call > states. > > What's the difference/benefit of using the clusterer module over db_mode 1 > in this scenario? > > Thank you! > > //Matine > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Nov 3 12:57:28 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 3 Nov 2017 18:57:28 +0200 Subject: [OpenSIPS-Users] ACC Extra field On OpenSIPs Console - CDR Viewer In-Reply-To: References: <0587738d-09a1-645b-5df4-b3d11cec75f1@bohboh.info> <89b42a3b-f45b-df64-6cfa-5e489d9cb3c2@bohboh.info> Message-ID: <616642a5-e872-cc0c-b8bd-90d022f82752@opensips.org> I see the column listed with lowercases - try to use everywhere only lowercase (including in the definition of the column in the table) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/03/2017 05:21 PM, Social Boh wrote: > > Hello, > > in Details, there is (image)... > > Thank you > > --- > I'm SoCIaL, MayBe > El 03/11/2017 a las 07:09, Bogdan-Andrei Iancu escribió: >> That means the "Gateway" column was *not* found in the DB query >> result (in the accounting table). >> >> Try to click on the CDR details icon to see the raw format for the >> CDR - check if your column is there. >> >> Regards, >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> On 11/03/2017 01:44 PM, Social Boh wrote: >>> >>> Hello, I have this message: >>> >>> PHP Notice: Undefined index: Gateway in >>> /var/www/html/opensips-cp/web/tools/system/cdrviewer/template/cdrviewer.main.php >>> on line 236, referer: http://X.X.X.X/cp/menu.php >>> >>> Thank you >>> >>> Regards >>> >>> --- >>> I'm SoCIaL, MayBe >>> El 03/11/2017 a las 06:09, Bogdan-Andrei Iancu escribió: >>>> Hi, >>>> >>>> The change looks ok - are you sure you have data in the "Gateway" >>>> column in the DB ? Do you see any errors in the apache log when >>>> opening the CDRViewer tool ? >>>> >>>> Regards, >>>> Bogdan-Andrei Iancu >>>> OpenSIPS Founder and Developer >>>> http://www.opensips-solutions.com >>>> On 11/02/2017 07:34 PM, Social Boh wrote: >>>>> >>>>> Sorry, >>>>> >>>>> OpenSIPS Control Panel 7.2.3 :) >>>>> >>>>> I have this line on ACC Configuration: >>>>> >>>>> *modparam("acc", "extra_fields", "db: gw->Gateway")* >>>>> >>>>> To know wich gateway OpenSIPs use to make a call. >>>>> >>>>> I added this field in acc table and i can see the values. >>>>> >>>>> My problem is I don't know how modify correctly: >>>>> >>>>> *config/tools/system/cdrviewer/local.inc.php* >>>>> >>>>> to see the values of this field on the CDR Viewer menu. With this: >>>>> >>>>> // what fields to show >>>>> $show_field[0]['time'] = "Time" ; >>>>> $show_field[1]['method'] = "Method" ; >>>>> $show_field[2]['callid'] = "Sip Call ID" ; >>>>> $show_field[3]['sip_code'] = "Sip Code" ; >>>>> $show_field[4]['sip_reason'] = "Sip Reason" ; >>>>> $show_field[5]['setuptime'] = "Setup Time"; >>>>> $show_field[6]['duration'] = "Duration"; >>>>> $show_field[7]['from_tag'] = "Sip From Tag" ; >>>>> $show_field[8]['to_tag'] = "Sip To Tag"; >>>>> * $show_field[9]['Gateway'] = "SIP Gateway";* >>>>> >>>>> (bold) I can see the column but not contain any value. >>>>> >>>>> Thank you >>>>> >>>>> Regards >>>>> >>>>> --- >>>>> I'm SoCIaL, MayBe >>>>> El 02/11/2017 a las 12:23, Bogdan-Andrei Iancu escribió: >>>>>> Hi, >>>>>> >>>>>> I guess you mean OpenSIPS Control Panel 7.2.3 (and not OpenSIPS >>>>>> Console). >>>>>> >>>>>> And yes, it is possible to configure the CDRViewer to display >>>>>> extra DB fields. What are the changes you did in the >>>>>> local.inc.php file ? >>>>>> >>>>>> Regards, >>>>>> >>>>>> Bogdan-Andrei Iancu >>>>>> OpenSIPS Founder and Developer >>>>>> http://www.opensips-solutions.com >>>>>> >>>>>> On 11/02/2017 06:56 PM, Social Boh wrote: >>>>>>> Hello, >>>>>>> >>>>>>> I'm using OpenSIPs 2.3.2 and OpenSIPs Console 7.2.3. >>>>>>> >>>>>>> I'd like know if is possible to see Acc Extra Fields, configured >>>>>>> on ACC Module, on CDR Viewer menu. >>>>>>> >>>>>>> I tried modifying >>>>> >>>>>>> p without success. >>>>>>> >>>>>>> Regards >>>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From social at bohboh.info Fri Nov 3 15:28:02 2017 From: social at bohboh.info (Social Boh) Date: Fri, 3 Nov 2017 14:28:02 -0500 Subject: [OpenSIPS-Users] Sending HEP Packets Message-ID: <8d9354ca-b4cd-8cf1-da2b-b0c504179f55@bohboh.info> Hello, I think I don't completely understood the logic behind proto_hep and siptrace modules on OpenSIPs 2.3 I'd like send SIP traffic from OpenSIPs to a external Homer SIP Capture Server (with Kamailio) so I know I have to configure proto_hep and siptrace modules. I think the logic is: Send SIP traffic from siptrace to hep listener: modparam("siptrace", "trace_id", "[tid]uri=sip:127.0.0.1:6060") listen=hep_udp:127.0.0.1:6060 then from the proto_hep module send the HEP packets to Node Capture: modparam("proto_hep", "hep_id", "[homer]1.2.3.4:9060; version=2") This configuration not works. This document not helping me very much: http://www.opensips.org/Documentation/Tutorials-Tracing Any hint, please? Thank you -- --- I'm SoCIaL, MayBe From goup2010 at gmail.com Fri Nov 3 15:36:38 2017 From: goup2010 at gmail.com (Dragomir Haralambiev) Date: Fri, 3 Nov 2017 21:36:38 +0200 Subject: [OpenSIPS-Users] Wrong Contact in location table In-Reply-To: References: <0093287c-82e4-c11a-c933-e5e195683156@opensips.org> <741e1ed0-1831-4a1f-aa21-08ed354c2b53@opensips.org> <8045BDD6-2077-4CCC-B6C3-7E007AF631F3@tampahost.net> <9eb0a075-7235-fba0-83c9-62c3ab5de93f@opensips.org> <13955c88-1a4f-d106-10ec-a29525d4c39a@opensips.org> <58c846f0-7fe6-f687-d7aa-b505b20a41ef@opensips.org> Message-ID: Hi, Why I need the real IP and port in location table? Now I make only outgoing call. Everything works fine on the following scenario.: User -----> Opensips ------> ITSP If I not have real IP in location table incoming calls not be implemented. ITSP ----> Opensips ---->? Opensips get IP from location table and try to send call. But in location table have not real IP. I see how the Opensips try to send call to 192.168.2.34. This is the main problem. 2017-11-03 11:45 GMT+02:00 Răzvan Crainea : > Unfortunately I just realised that you cannot change the Contact header > for this scenario. > And to be honest I don't really understand why you are trying to change it > - if you need the real IP and port, you can take them from the received > field. > > Best regards, > > Răzvan Crainea > OpenSIPS Developerwww.opensips-solutions.com > > On 11/02/2017 11:43 PM, Dragomir Haralambiev wrote: > > Hi, > > Here is part of my script: > .... > modparam("registrar", "mcontact_avp", "$avp(register)") > ..... > > if (t_check_status("2[0-9][0-9]")) { > $log_level = 5; > script_trace( 1, "$rm from $si, ruri=$ru, ct=$ct.fields(uri) > avp(register)=$avp(register)", "me"); > route(save_location); > ....... > } > > You can see log here: > https://pastebin.com/WWQ9Mmh4 > > Here is the replacement contact: > > DBG:registrar:build_contact: created Contact HF: Contact: < > sip:55595009 at 192.168.22.138:5062>;expires=360 > DBG:registrar:save: replacing contact uri [sip:55595009 at 188.23.232.10:1043] > with [sip:55595009 at 192.168.22.138:5062] > > How to stop replacing contact from 188.23.232.10:1043 to > 192.168.22.138:5062 ? > > > _______________________________________________ > 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 social at bohboh.info Fri Nov 3 16:05:56 2017 From: social at bohboh.info (Social Boh) Date: Fri, 3 Nov 2017 15:05:56 -0500 Subject: [OpenSIPS-Users] ACC Extra field On OpenSIPs Console - CDR Viewer [SOLVED] In-Reply-To: <616642a5-e872-cc0c-b8bd-90d022f82752@opensips.org> References: <0587738d-09a1-645b-5df4-b3d11cec75f1@bohboh.info> <89b42a3b-f45b-df64-6cfa-5e489d9cb3c2@bohboh.info> <616642a5-e872-cc0c-b8bd-90d022f82752@opensips.org> Message-ID: Hello, changing to lowercase solves the issue. Thank you Regards --- I'm SoCIaL, MayBe El 03/11/2017 a las 11:57, Bogdan-Andrei Iancu escribió: > I see the column listed with lowercases - try to use everywhere only > lowercase (including in the definition of the column in the table) > > Regards, > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > On 11/03/2017 05:21 PM, Social Boh wrote: >> >> Hello, >> >> in Details, there is (image)... >> >> Thank you >> >> --- >> I'm SoCIaL, MayBe >> El 03/11/2017 a las 07:09, Bogdan-Andrei Iancu escribió: >>> That means the "Gateway" column was *not* found in the DB query >>> result (in the accounting table). >>> >>> Try to click on the CDR details icon to see the raw format for the >>> CDR - check if your column is there. >>> >>> Regards, >>> Bogdan-Andrei Iancu >>> OpenSIPS Founder and Developer >>> http://www.opensips-solutions.com >>> On 11/03/2017 01:44 PM, Social Boh wrote: >>>> >>>> Hello, I have this message: >>>> >>>>  PHP Notice:  Undefined index: Gateway in >>>> /var/www/html/opensips-cp/web/tools/system/cdrviewer/template/cdrviewer.main.php >>>> on line 236, referer: http://X.X.X.X/cp/menu.php >>>> >>>> Thank you >>>> >>>> Regards >>>> >>>> --- >>>> I'm SoCIaL, MayBe >>>> El 03/11/2017 a las 06:09, Bogdan-Andrei Iancu escribió: >>>>> Hi, >>>>> >>>>> The change looks ok - are you sure you have data in the "Gateway" >>>>> column in the DB ? Do you see any errors in the apache log when >>>>> opening the CDRViewer tool ? >>>>> >>>>> Regards, >>>>> Bogdan-Andrei Iancu >>>>> OpenSIPS Founder and Developer >>>>> http://www.opensips-solutions.com >>>>> On 11/02/2017 07:34 PM, Social Boh wrote: >>>>>> >>>>>> Sorry, >>>>>> >>>>>> OpenSIPS Control Panel 7.2.3 :) >>>>>> >>>>>> I have this line on ACC Configuration: >>>>>> >>>>>> *modparam("acc", "extra_fields", "db: gw->Gateway")* >>>>>> >>>>>> To know wich gateway OpenSIPs use to make a call. >>>>>> >>>>>> I added this field in acc table and i can see the values. >>>>>> >>>>>> My problem is I don't know how modify correctly: >>>>>> >>>>>> *config/tools/system/cdrviewer/local.inc.php* >>>>>> >>>>>> to see the values of this field on the CDR Viewer menu. With this: >>>>>> >>>>>>  // what fields to show >>>>>>  $show_field[0]['time'] = "Time" ; >>>>>>  $show_field[1]['method'] = "Method" ; >>>>>>  $show_field[2]['callid'] = "Sip Call ID" ; >>>>>>  $show_field[3]['sip_code'] = "Sip Code" ; >>>>>>  $show_field[4]['sip_reason'] = "Sip Reason" ; >>>>>>  $show_field[5]['setuptime'] = "Setup Time"; >>>>>>  $show_field[6]['duration'] = "Duration"; >>>>>>  $show_field[7]['from_tag'] = "Sip From Tag" ; >>>>>>  $show_field[8]['to_tag'] = "Sip To Tag"; >>>>>> * $show_field[9]['Gateway'] = "SIP Gateway";* >>>>>> >>>>>> (bold) I can see the column but not contain any value. >>>>>> >>>>>> Thank you >>>>>> >>>>>> Regards >>>>>> >>>>>> --- >>>>>> I'm SoCIaL, MayBe >>>>>> El 02/11/2017 a las 12:23, Bogdan-Andrei Iancu escribió: >>>>>>> Hi, >>>>>>> >>>>>>> I guess you mean OpenSIPS Control Panel 7.2.3 (and not OpenSIPS >>>>>>> Console). >>>>>>> >>>>>>> And yes, it is possible to configure the CDRViewer to display >>>>>>> extra DB fields. What are the changes you did in the >>>>>>> local.inc.php file ? >>>>>>> >>>>>>> Regards, >>>>>>> >>>>>>> Bogdan-Andrei Iancu >>>>>>>   OpenSIPS Founder and Developer >>>>>>> http://www.opensips-solutions.com >>>>>>> >>>>>>> On 11/02/2017 06:56 PM, Social Boh wrote: >>>>>>>> Hello, >>>>>>>> >>>>>>>> I'm using OpenSIPs 2.3.2 and OpenSIPs Console 7.2.3. >>>>>>>> >>>>>>>> I'd like know if is possible to see Acc Extra Fields, >>>>>>>> configured on ACC Module, on CDR Viewer menu. >>>>>>>> >>>>>>>> I tried modifying >>>>>> >>>>>>>> p without success. >>>>>>>> >>>>>>>> Regards >>>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Users mailing list >>>>>> Users at lists.opensips.org >>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From goup2010 at gmail.com Sat Nov 4 08:35:29 2017 From: goup2010 at gmail.com (Dragomir Haralambiev) Date: Sat, 4 Nov 2017 14:35:29 +0200 Subject: [OpenSIPS-Users] Opensips 2.3.2 chashed Message-ID: Hello, I use latest update from git at Opensips 2.3.2. When try to use call center module Opensips crashed: 1. Log message: Nov 4 14:09:52opensips[2658]: CRITICAL:call_center:cc_queue_rmv_call: QUEUE - call not in queue l=(nil), h=(nil) Nov 4 14:09:52opensips[2666]: CRITICAL:core:receive_fd: EOF on 48 Nov 4 14:09:52opensips[2636]: INFO:core:handle_sigs: child process 2658 exited by a signal 6 Nov 4 14:09:52opensips[2636]: INFO:core:handle_sigs: core was generated Nov 4 14:09:52opensips[2636]: INFO:core:handle_sigs: terminating due to SIGCHLD Nov 4 14:09:52opensips[2655]: INFO:core:sig_usr: signal 15 received 2. Message from gdb: ... Reading symbols from /usr/local/sbin/opensips...done. [New LWP 2658] [Thread debugging using libthread_db enabled] Using host libthread_db library "/usr/lib64/libthread_db.so.1". Core was generated by `/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid -f /etc/opensips/ope'. Program terminated with signal 6, Aborted. #0 0x00007fcbf3d421f7 in raise () from /usr/lib64/libc.so.6 Missing separate debuginfos, use: debuginfo-install bzip2-libs-1.0.6-13.el7.x86_64 elfutils-libelf-0.168-8.el7.x86_64 elfutils-libs-0.168-8.el7.x86_64 glibc-2.17-196.el7.x86_64 gmp-6.0.0-15.el7.x86_64 gnutls-3.3.26-9.el7.x86_64 hiredis-0.12.1-1.el7.x86_64 json-c-0.11-4.el7_0.x86_64 keyutils-libs-1.5.8-3.el7.x86_64 krb5-libs-1.15.1-8.el7.x86_64 libattr-2.4.46-12.el7.x86_64 libcap-2.22-9.el7.x86_64 libcom_err-1.42.9-10.el7.x86_64 libffi-3.0.13-18.el7.x86_64 libgcc-4.8.5-16.el7.x86_64 libgcrypt-1.5.3-14.el7.x86_64 libgpg-error-1.12-3.el7.x86_64 libmicrohttpd-0.9.33-2.el7.x86_64 libselinux-2.5-11.el7.x86_64 libstdc++-4.8.5-16.el7.x86_64 libtasn1-4.10-1.el7.x86_64 libxml2-2.9.1-6.el7_2.3.x86_64 mariadb-libs-5.5.56-2.el7.x86_64 nettle-2.7.1-8.el7.x86_64 openssl-libs-1.0.2k-8.el7.x86_64 p11-kit-0.23.5-3.el7.x86_64 pcre-8.32-17.el7.x86_64 systemd-libs-219-42.el7_4.4.x86_64 xz-libs-5.2.2-1.el7.x86_64 zlib-1.2.7-17.el7.x86_64 3. Here is gdb bt full log: https://pastebin.com/HFkzxnty Best regards, Dragomir -------------- next part -------------- An HTML attachment was scrubbed... URL: From khorsmann at gmail.com Sun Nov 5 02:56:53 2017 From: khorsmann at gmail.com (Karsten Horsmann) Date: Sun, 5 Nov 2017 08:56:53 +0100 Subject: [OpenSIPS-Users] Sending HEP Packets In-Reply-To: <8d9354ca-b4cd-8cf1-da2b-b0c504179f55@bohboh.info> References: <8d9354ca-b4cd-8cf1-da2b-b0c504179f55@bohboh.info> Message-ID: Hi, you need also to mark packets you are interested to be sipcaptured in your route. Kind regards Karsten Horsmann Am 03.11.2017 8:32 nachm. schrieb "Social Boh" : Hello, I think I don't completely understood the logic behind proto_hep and siptrace modules on OpenSIPs 2.3 I'd like send SIP traffic from OpenSIPs to a external Homer SIP Capture Server (with Kamailio) so I know I have to configure proto_hep and siptrace modules. I think the logic is: Send SIP traffic from siptrace to hep listener: modparam("siptrace", "trace_id", "[tid]uri=sip:127.0.0.1:6060") listen=hep_udp:127.0.0.1:6060 then from the proto_hep module send the HEP packets to Node Capture: modparam("proto_hep", "hep_id", "[homer]1.2.3.4:9060; version=2") This configuration not works. This document not helping me very much: http://www.opensips.org/Documentation/Tutorials-Tracing Any hint, please? Thank you -- --- I'm SoCIaL, MayBe _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From nocbgtelcom at gmail.com Sun Nov 5 16:06:12 2017 From: nocbgtelcom at gmail.com (Hristo Donev) Date: Sun, 5 Nov 2017 23:06:12 +0200 Subject: [OpenSIPS-Users] call_center module in Opensips 2.3 Message-ID: Hello All, I try to implement call_center module in Opensips 2.3. Here are my questions? 1. From the documentation 1.3.1: "b2b_logic" and "database" must be loaded before callcenter module. I found out that "b2b_entities" should be loaded as well, otherwise the opensips does not start. Is this my mistake or a documentation error? 2. Opensips can not start if the tables "cc_agents" and "cc_flow" are empty. Why it cannot start? Should they be populated with some data? 3. Table "cc_agents", field "location" should point to the SIP agent location. If the agent's IP and port are changed dynamically - how to copy automatically "contact" field from "location" table to "cc_agents" field "location"? I would be grateful to receive a response for my questions. Thanks in advance, Hristo Donev -------------- next part -------------- An HTML attachment was scrubbed... URL: From Martin.Horne at arm.com Thu Nov 2 18:41:45 2017 From: Martin.Horne at arm.com (Martin Horne) Date: Thu, 2 Nov 2017 22:41:45 +0000 Subject: [OpenSIPS-Users] opensips on Ubuntu >= 16.04 Message-ID: Hi I have opensips built for arm64 (opensip-cp is working); has anybody got a cook book for installing opensips on Ubuntu 16.04? Would also be very interested if anybody has it building on Alpine. Thanks Martin Horne Developer Advocate | System Architect | Arm . . . . . . . . . . . . . . . . . . . . . . . . . . . m. +1 (925) 519 3056 Arm.com IMPORTANT NOTICE: The contents of this email and any attachments are confidential and may also be privileged. If you are not the intended recipient, please notify the sender immediately and do not disclose the contents to any other person, use it for any purpose, or store or copy the information in any medium. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Mon Nov 6 03:59:22 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Mon, 6 Nov 2017 10:59:22 +0200 Subject: [OpenSIPS-Users] Sending HEP Packets In-Reply-To: References: <8d9354ca-b4cd-8cf1-da2b-b0c504179f55@bohboh.info> Message-ID: <3e4a2367-fcd0-b47a-5e41-348b3d729f74@opensips.org> Hello! Shortly, you need to follow this logic: 1. Define a listener: listen=hep_udp:127.0.0.1:6060 2. Define a HEP destination to capture the traffic to[1]. modparam("proto_hep", "hep_id", "[homer_id]1.2.3.4:9060 ; version=2") 3. Next, define a tracing ID - an idea that describes where a specific package should be sent[2] modparam("siptrace", "trace_id", "[to_homer_tid]uri=hep:homer_id") 4. Start tracing the data you want by calling sip_trace() with the destination provisioned earlier and the scope you want: sip_trace("to_homer_tid", "m", "sip"); # this traces only the SIP message Hope it is clear now. [1] http://www.opensips.org/html/docs/modules/2.4.x/proto_hep.html#idp232096 [2] http://www.opensips.org/html/docs/modules/2.4.x/siptrace.html#idp5904384 [3] http://www.opensips.org/html/docs/modules/2.4.x/siptrace.html#idp5914144 Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/05/2017 09:56 AM, Karsten Horsmann wrote: > Hi, > > you need also to mark packets you are interested to be sipcaptured in > your route. > > Kind regards > Karsten Horsmann > > > Am 03.11.2017 8:32 nachm. schrieb "Social Boh" >: > > Hello, > > I think I don't completely understood the logic behind proto_hep > and siptrace modules on OpenSIPs 2.3 > > I'd like send SIP traffic from OpenSIPs to a external Homer SIP > Capture Server (with Kamailio) so I know I have to configure > proto_hep and siptrace modules. > > I think the logic is: Send SIP traffic from siptrace to hep listener: > > modparam("siptrace", "trace_id", "[tid]uri=sip:127.0.0.1:6060 > ") > > listen=hep_udp:127.0.0.1:6060 > > then from the proto_hep module send the HEP packets to Node Capture: > > modparam("proto_hep", "hep_id", "[homer]1.2.3.4:9060 > ; version=2") > > This configuration not works. > > This document not helping me very much: > > http://www.opensips.org/Documentation/Tutorials-Tracing > > > Any hint, please? > > Thank you > > -- > --- > I'm SoCIaL, MayBe > > > _______________________________________________ > 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 khamlichi.khalil at gmail.com Mon Nov 6 04:02:15 2017 From: khamlichi.khalil at gmail.com (Khalil Khamlichi) Date: Mon, 6 Nov 2017 09:02:15 +0000 Subject: [OpenSIPS-Users] opensips on Ubuntu >= 16.04 In-Reply-To: References: Message-ID: http://apt.opensips.org/packages.php?v=2.3 Sent via mobile, please forgive typos and brevity On Nov 6, 2017 8:53 AM, "Martin Horne" wrote: > Hi I have opensips built for arm64 (opensip-cp is working); has anybody > got a cook book for installing opensips on Ubuntu 16.04? > > Would also be very interested if anybody has it building on Alpine. > > Thanks > > > > > > *Martin Horne* > > Developer Advocate | > > System Architect | *Arm* > > *. . . . . . . . . . . . . . . . . . . . . . . . . . . * > > m. +1 (925) 519 3056 > > Arm.com > > > IMPORTANT NOTICE: The contents of this email and any attachments are > confidential and may also be privileged. If you are not the intended > recipient, please notify the sender immediately and do not disclose the > contents to any other person, use it for any purpose, or store or copy the > information in any medium. Thank you. > > _______________________________________________ > 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 Mon Nov 6 04:37:43 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Mon, 6 Nov 2017 11:37:43 +0200 Subject: [OpenSIPS-Users] call_center module in Opensips 2.3 In-Reply-To: References: Message-ID: Hi, Hristo! 1. You are right, b2b_entities should be loaded as a dependency for b2b_logic. You may argue that it should also be listed as a dependency to call_center as well, but indirectly. 2. Yes, you should populate the tables with the flows you want to support and the agents that are going to use the call center. 3. There is currently no way to directly set the location of the agent dynamically. However, we can do some work-arounds, for example always advertise yourself as the location of the user, and for each call you can loop the INVITE back to you and do a lookup(). Or simply use an external registrar service. Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/05/2017 11:06 PM, Hristo Donev wrote: > Hello All, > > I try to implement call_center module in Opensips 2.3. > Here are my questions? > > 1. From the documentation 1.3.1: > "b2b_logic" and "database" must be loaded before callcenter module. > > I found out that "b2b_entities" should be loaded as well, otherwise > the opensips does not start. > > Is this my mistake or a documentation error? > > 2. Opensips can not start if the tables  "cc_agents" and "cc_flow" are > empty. > Why it cannot start? Should they be populated with some data? > > 3. Table "cc_agents", field "location" should point to the SIP agent > location. > If the agent's IP and port are changed dynamically - > how to copy automatically "contact" field from "location" table to > "cc_agents" field "location"? > > I would be grateful to receive a response for my questions. > > Thanks in advance, > Hristo Donev > > > _______________________________________________ > 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 Mon Nov 6 04:42:30 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Mon, 6 Nov 2017 11:42:30 +0200 Subject: [OpenSIPS-Users] Opensips 2.3.2 chashed In-Reply-To: References: Message-ID: <1e55ada4-623a-32e6-62c4-30da0c3d381e@opensips.org> Hi, Dragomir! Can you to go in gdb top frame 3 and print call_it. Use the following snippet and attach the output in pastebin. > frame 3 > p *call_it Thanks! Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/04/2017 02:35 PM, Dragomir Haralambiev wrote: > Hello, > > I use latest update from git at Opensips 2.3.2. > When try to use call center module Opensips crashed: > > 1. Log message: > > Nov  4 14:09:52opensips[2658]: > CRITICAL:call_center:cc_queue_rmv_call:  QUEUE - call not in queue > l=(nil), h=(nil) > Nov  4 14:09:52opensips[2666]: CRITICAL:core:receive_fd: EOF on 48 > Nov  4 14:09:52opensips[2636]: INFO:core:handle_sigs: child process > 2658 exited by a signal 6 > Nov  4 14:09:52opensips[2636]: INFO:core:handle_sigs: core was generated > Nov  4 14:09:52opensips[2636]: INFO:core:handle_sigs: terminating due > to SIGCHLD > Nov  4 14:09:52opensips[2655]: INFO:core:sig_usr: signal 15 received > > 2. Message from gdb: > > ... > Reading symbols from /usr/local/sbin/opensips...done. > [New LWP 2658] > [Thread debugging using libthread_db enabled] > Using host libthread_db library "/usr/lib64/libthread_db.so.1". > Core was generated by `/usr/local/sbin/opensips -P > /var/run/opensips/opensips.pid -f /etc/opensips/ope'. > Program terminated with signal 6, Aborted. > #0  0x00007fcbf3d421f7 in raise () from /usr/lib64/libc.so.6 > Missing separate debuginfos, use: debuginfo-install > bzip2-libs-1.0.6-13.el7.x86_64 elfutils-libelf-0.168-8.el7.x86_64 > elfutils-libs-0.168-8.el7.x86_64 glibc-2.17-196.el7.x86_64 > gmp-6.0.0-15.el7.x86_64 gnutls-3.3.26-9.el7.x86_64 > hiredis-0.12.1-1.el7.x86_64 json-c-0.11-4.el7_0.x86_64 > keyutils-libs-1.5.8-3.el7.x86_64 krb5-libs-1.15.1-8.el7.x86_64 > libattr-2.4.46-12.el7.x86_64 libcap-2.22-9.el7.x86_64 > libcom_err-1.42.9-10.el7.x86_64 libffi-3.0.13-18.el7.x86_64 > libgcc-4.8.5-16.el7.x86_64 libgcrypt-1.5.3-14.el7.x86_64 > libgpg-error-1.12-3.el7.x86_64 libmicrohttpd-0.9.33-2.el7.x86_64 > libselinux-2.5-11.el7.x86_64 libstdc++-4.8.5-16.el7.x86_64 > libtasn1-4.10-1.el7.x86_64 libxml2-2.9.1-6.el7_2.3.x86_64 > mariadb-libs-5.5.56-2.el7.x86_64 nettle-2.7.1-8.el7.x86_64 > openssl-libs-1.0.2k-8.el7.x86_64 p11-kit-0.23.5-3.el7.x86_64 > pcre-8.32-17.el7.x86_64 systemd-libs-219-42.el7_4.4.x86_64 > xz-libs-5.2.2-1.el7.x86_64 zlib-1.2.7-17.el7.x86_64 > > 3. Here is gdb bt full log: > > https://pastebin.com/HFkzxnty > > 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 razvan at opensips.org Mon Nov 6 04:44:35 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Mon, 6 Nov 2017 11:44:35 +0200 Subject: [OpenSIPS-Users] Wrong Contact in location table In-Reply-To: References: <0093287c-82e4-c11a-c933-e5e195683156@opensips.org> <741e1ed0-1831-4a1f-aa21-08ed354c2b53@opensips.org> <8045BDD6-2077-4CCC-B6C3-7E007AF631F3@tampahost.net> <9eb0a075-7235-fba0-83c9-62c3ab5de93f@opensips.org> <13955c88-1a4f-d106-10ec-a29525d4c39a@opensips.org> <58c846f0-7fe6-f687-d7aa-b505b20a41ef@opensips.org> Message-ID: <541a84ad-437a-2a0a-0338-c5fca288015a@opensips.org> Hi, Dragomir! If you simply do fix_nated_register() on the REGISTER messages, all these will be sorted out. Moreover, it's actually not correct to change the contact of the user, because in the SIP message it might expect to have exactly what he sent. When using fix_nated_register(), there is another field (called Received) that stores the actual IP and port where the REGISTER came from, and when an invite comes in, it is automatically set by the lookup() function in the DST uri, without changing the contact (the contact may still be private). So simply calling fix_nated_register() should fix this issue for all scenarios. I initially though that you were using the Contact from something else and you really need that value there. Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/03/2017 09:36 PM, Dragomir Haralambiev wrote: > Hi, > > Why I need the real IP and port in location table? > > Now I make only outgoing call. Everything works fine on the following > scenario.: > User -----> Opensips ------> ITSP > > If I not have real IP in location table incoming calls not be implemented. > ITSP ----> Opensips ---->? > > Opensips get IP from location table and try to send call. But in > location table have not real IP. > I see how the Opensips try to send call to 192.168.2.34. > > This is the main problem. > > 2017-11-03 11:45 GMT+02:00 Răzvan Crainea >: > > Unfortunately I just realised that you cannot change the Contact > header for this scenario. > And to be honest I don't really understand why you are trying to > change it - if you need the real IP and port, you can take them > from the received field. > > Best regards, > > Răzvan Crainea > OpenSIPS Developer > www.opensips-solutions.com > > On 11/02/2017 11:43 PM, Dragomir Haralambiev wrote: >> Hi, >> >> Here is part of my script: >> .... >> modparam("registrar", "mcontact_avp", "$avp(register)") >> ..... >> >>     if (t_check_status("2[0-9][0-9]"))  { >> $log_level = 5; >> script_trace( 1, "$rm from $si, ruri=$ru, ct=$ct.fields(uri) >> avp(register)=$avp(register)", "me"); >> route(save_location); >>                        ....... >>     } >> >> You can see log here: >> https://pastebin.com/WWQ9Mmh4 >> >> Here is the replacement contact: >> >> DBG:registrar:build_contact: created Contact HF: Contact: >> > >;expires=360 >> DBG:registrar:save: replacing contact uri >> [sip:55595009 at 188.23.232.10:1043 >> ] with >> [sip:55595009 at 192.168.22.138:5062 >> ] >> >> How to stop replacing contact from 188.23.232.10:1043 >> to 192.168.22.138:5062 >> ? > > _______________________________________________ > 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 eahaselhoff at gmail.com Mon Nov 6 05:29:42 2017 From: eahaselhoff at gmail.com (Edwin) Date: Mon, 6 Nov 2017 03:29:42 -0700 (MST) Subject: [OpenSIPS-Users] b2bua To header format In-Reply-To: <1509008552952-0.post@n2.nabble.com> References: <1509005288912-0.post@n2.nabble.com> <4734ae5a-49c0-06a2-0647-d91d0faa3101@opensips.org> <1509008552952-0.post@n2.nabble.com> Message-ID: <1509964182808-0.post@n2.nabble.com> Hi Răzvan, After the change to b2bua we present the To header like this:"To: sip:+31123456789 at domain.com;user=phone After migration we still have problems with certain sip clients. In some cases a software update did the trick. I will make a github feature request to add the <> symbols anyway, even if it is following RFC. In case of adding the ;user=phone, i'm in doubt if you don't have to enclose the URI because of the semicolon. See RFC 3261: " The Contact, From, and To header fields contain a URI. If the URI contains a comma, question mark or semicolon, the URI MUST be enclosed in angle brackets (< and >). Any URI parameters are contained within these brackets. If the URI is not enclosed in angle brackets, any semicolon-delimited parameters are header-parameters, not URI parameters. -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html From razvan at opensips.org Mon Nov 6 05:53:40 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Mon, 6 Nov 2017 12:53:40 +0200 Subject: [OpenSIPS-Users] b2bua To header format In-Reply-To: <1509964182808-0.post@n2.nabble.com> References: <1509005288912-0.post@n2.nabble.com> <4734ae5a-49c0-06a2-0647-d91d0faa3101@opensips.org> <1509008552952-0.post@n2.nabble.com> <1509964182808-0.post@n2.nabble.com> Message-ID: Hi, Edwin! In this case I think you are right, because the ;user=phone should be an URI parameter, not a header parameter. Although the syntax is fine, the semantics change - not sure if that bothers you though. Please open a bug report to keep track of this issue and point it to this discussion :). Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/06/2017 12:29 PM, Edwin wrote: > Hi Răzvan, > > After the change to b2bua we present the To header like this:"To: > sip:+31123456789 at domain.com;user=phone > > After migration we still have problems with certain sip clients. In some > cases a software update did the trick. I will make a github feature request > to add the <> symbols anyway, even if it is following RFC. > > In case of adding the ;user=phone, i'm in doubt if you don't have to enclose > the URI because of the semicolon. See RFC 3261: > > " The Contact, From, and To header fields contain a URI. If the URI > contains a comma, question mark or semicolon, the URI MUST be enclosed in > angle brackets (< and >). Any URI parameters are contained within these > brackets. If the URI is not enclosed in angle brackets, any > semicolon-delimited parameters are header-parameters, not URI parameters. > > > > > > -- > 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 rmundkowsky at ets.org Mon Nov 6 10:14:44 2017 From: rmundkowsky at ets.org (Mundkowsky, Robert) Date: Mon, 6 Nov 2017 15:14:44 +0000 Subject: [OpenSIPS-Users] opensips on Ubuntu >= 16.04 In-Reply-To: References: , Message-ID: I installed it on Ubuntu 16.04.2. The normal installation worked fine. Only think I did manually was to create a boot script for it. ________________________________ From: Users on behalf of Khalil Khamlichi Sent: Monday, November 6, 2017 4:02 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] opensips on Ubuntu >= 16.04 http://apt.opensips.org/packages.php?v=2.3 Sent via mobile, please forgive typos and brevity On Nov 6, 2017 8:53 AM, "Martin Horne" > wrote: Hi I have opensips built for arm64 (opensip-cp is working); has anybody got a cook book for installing opensips on Ubuntu 16.04? Would also be very interested if anybody has it building on Alpine. Thanks Martin Horne Developer Advocate | System Architect | Arm . . . . . . . . . . . . . . . . . . . . . . . . . . . m. +1 (925) 519 3056 Arm.com IMPORTANT NOTICE: The contents of this email and any attachments are confidential and may also be privileged. If you are not the intended recipient, please notify the sender immediately and do not disclose the contents to any other person, use it for any purpose, or store or copy the information in any medium. Thank you. _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: From goup2010 at gmail.com Mon Nov 6 11:18:26 2017 From: goup2010 at gmail.com (Dragomir Haralambiev) Date: Mon, 6 Nov 2017 18:18:26 +0200 Subject: [OpenSIPS-Users] Wrong Contact in location table In-Reply-To: <541a84ad-437a-2a0a-0338-c5fca288015a@opensips.org> References: <0093287c-82e4-c11a-c933-e5e195683156@opensips.org> <741e1ed0-1831-4a1f-aa21-08ed354c2b53@opensips.org> <8045BDD6-2077-4CCC-B6C3-7E007AF631F3@tampahost.net> <9eb0a075-7235-fba0-83c9-62c3ab5de93f@opensips.org> <13955c88-1a4f-d106-10ec-a29525d4c39a@opensips.org> <58c846f0-7fe6-f687-d7aa-b505b20a41ef@opensips.org> <541a84ad-437a-2a0a-0338-c5fca288015a@opensips.org> Message-ID: Hi, Thanks for your email. Here is part of my location table: contact_id username domain contact received path expires q callid cseq last_modified flags cflags user_agent socket methods sip_instance attr ------------------- --------- ------ -------------------------------------------------------------------------- -------- ------ ------------------- ------ ------------------------------------------------ ------ ------------------- ------ --------- -------------------------------- ----------------------- ------- ------------ -------- 181494352482801881 57996206 (NULL) sip:57996206 at 192.168.22.206:5062 (NULL) (NULL) 2017-11-06 17:58:48 -1.00 1162502851 at 192.168.22.206 162 2017-11-06 17:52:48 0 NAT_BFLAG Yealink SIP-T19P 31.72.0.75 udp:OpenSips_IP:5060 16383 (NULL) (NULL) 181654460760464436 57996204 (NULL) sip:57996204 at 192.168.22.204:5060 (NULL) (NULL) 2017-11-06 17:59:02 -1.00 0_1763370066 at 192.168.22.204 179 2017-11-06 17:53:02 0 NAT_BFLAG Yealink SIP-T21P_E2 52.81.0.25 udp:OpenSips_IP:5060 16383 (NULL) (NULL) Why "received" field is blank? Where could be the problem? 2017-11-06 11:44 GMT+02:00 Răzvan Crainea : > Hi, Dragomir! > > If you simply do fix_nated_register() on the REGISTER messages, all these > will be sorted out. Moreover, it's actually not correct to change the > contact of the user, because in the SIP message it might expect to have > exactly what he sent. > When using fix_nated_register(), there is another field (called Received) > that stores the actual IP and port where the REGISTER came from, and when > an invite comes in, it is automatically set by the lookup() function in the > DST uri, without changing the contact (the contact may still be private). > > So simply calling fix_nated_register() should fix this issue for all > scenarios. > I initially though that you were using the Contact from something else and > you really need that value there. > > Best regards, > > Răzvan Crainea > OpenSIPS Developerwww.opensips-solutions.com > > On 11/03/2017 09:36 PM, Dragomir Haralambiev wrote: > > Hi, > > Why I need the real IP and port in location table? > > Now I make only outgoing call. Everything works fine on the following > scenario.: > User -----> Opensips ------> ITSP > > If I not have real IP in location table incoming calls not be implemented. > ITSP ----> Opensips ---->? > > Opensips get IP from location table and try to send call. But in location > table have not real IP. > I see how the Opensips try to send call to 192.168.2.34. > > This is the main problem. > > 2017-11-03 11:45 GMT+02:00 Răzvan Crainea : > >> Unfortunately I just realised that you cannot change the Contact header >> for this scenario. >> And to be honest I don't really understand why you are trying to change >> it - if you need the real IP and port, you can take them from the received >> field. >> >> Best regards, >> >> Răzvan Crainea >> OpenSIPS Developerwww.opensips-solutions.com >> >> On 11/02/2017 11:43 PM, Dragomir Haralambiev wrote: >> >> Hi, >> >> Here is part of my script: >> .... >> modparam("registrar", "mcontact_avp", "$avp(register)") >> ..... >> >> if (t_check_status("2[0-9][0-9]")) { >> $log_level = 5; >> script_trace( 1, "$rm from $si, ruri=$ru, ct=$ct.fields(uri) >> avp(register)=$avp(register)", "me"); >> route(save_location); >> ....... >> } >> >> You can see log here: >> https://pastebin.com/WWQ9Mmh4 >> >> Here is the replacement contact: >> >> DBG:registrar:build_contact: created Contact HF: Contact: < >> sip:55595009 at 192.168.22.138:5062>;expires=360 >> DBG:registrar:save: replacing contact uri [sip:55595009 at 188.23.232.10:10 >> 43] with [sip:55595009 at 192.168.22.138:5062] >> >> How to stop replacing contact from 188.23.232.10:1043 to >> 192.168.22.138:5062 ? >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From goup2010 at gmail.com Mon Nov 6 14:43:56 2017 From: goup2010 at gmail.com (Dragomir Haralambiev) Date: Mon, 6 Nov 2017 21:43:56 +0200 Subject: [OpenSIPS-Users] Opensips 2.3.2 chashed In-Reply-To: <1e55ada4-623a-32e6-62c4-30da0c3d381e@opensips.org> References: <1e55ada4-623a-32e6-62c4-30da0c3d381e@opensips.org> Message-ID: Hello, 1. Openips log Nov 6 20:01:21 /opensips[3358]: ERROR:call_center:set_call_leg: failed to init new b2bua call (empty ID received) Nov 6 20:01:21 /opensips[3358]: ERROR:call_center:w_handle_call: failed to set new destination for call 2. gdb [Thread debugging using libthread_db enabled] Using host libthread_db library "/usr/lib64/libthread_db.so.1". Core was generated by `/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid -f /etc/opensips/ope'. Program terminated with signal 11, Segmentation fault. #0 0x00007f666940eccb in cc_queue_pop_call_for_agent (data=0x7f6672e73a88, agent=agent at entry=0x7f6672e74b30) at cc_data.c:1021 1021 if (call_it->b2bua_id.len!=0 && call_it->flow->skill==agent->skills[i]) { Missing separate debuginfos, use: debuginfo-install bzip2-libs-1.0.6-13.el7.x86_64 elfutils-libelf-0.168-8.el7.x86_64 elfutils-libs-0.168-8.el7.x86_64 glibc-2.17-196.el7.x86_64 gmp-6.0.0-15.el7.x86_64 gnutls-3.3.26-9.el7.x86_64 hiredis-0.12.1-1.el7.x86_64 json-c-0.11-4.el7_0.x86_64 keyutils-libs-1.5.8-3.el7.x86_64 krb5-libs-1.15.1-8.el7.x86_64 libattr-2.4.46-12.el7.x86_64 libcap-2.22-9.el7.x86_64 libcom_err-1.42.9-10.el7.x86_64 libffi-3.0.13-18.el7.x86_64 libgcc-4.8.5-16.el7.x86_64 libgcrypt-1.5.3-14.el7.x86_64 libgpg-error-1.12-3.el7.x86_64 libmicrohttpd-0.9.33-2.el7.x86_64 libselinux-2.5-11.el7.x86_64 libstdc++-4.8.5-16.el7.x86_64 libtasn1-4.10-1.el7.x86_64 libxml2-2.9.1-6.el7_2.3.x86_64 mariadb-libs-5.5.56-2.el7.x86_64 nettle-2.7.1-8.el7.x86_64 openssl-libs-1.0.2k-8.el7.x86_64 p11-kit-0.23.5-3.el7.x86_64 pcre-8.32-17.el7.x86_64 systemd-libs-219-42.el7_4.4.x86_64 xz-libs-5.2.2-1.el7.x86_64 zlib-1.2.7-17.el7.x86_64 3. bt full https://pastebin.com/FrnRDcaK 2017-11-06 11:42 GMT+02:00 Răzvan Crainea : > Hi, Dragomir! > > Can you to go in gdb top frame 3 and print call_it. Use the following > snippet and attach the output in pastebin. > > > frame 3 > > p *call_it > > Thanks! > > Răzvan Crainea > OpenSIPS Developerwww.opensips-solutions.com > > On 11/04/2017 02:35 PM, Dragomir Haralambiev wrote: > > Hello, > > I use latest update from git at Opensips 2.3.2. > When try to use call center module Opensips crashed: > > 1. Log message: > > Nov 4 14:09:52opensips[2658]: CRITICAL:call_center:cc_queue_rmv_call: > QUEUE - call not in queue l=(nil), h=(nil) > Nov 4 14:09:52opensips[2666]: CRITICAL:core:receive_fd: EOF on 48 > Nov 4 14:09:52opensips[2636]: INFO:core:handle_sigs: child process 2658 > exited by a signal 6 > Nov 4 14:09:52opensips[2636]: INFO:core:handle_sigs: core was generated > Nov 4 14:09:52opensips[2636]: INFO:core:handle_sigs: terminating due to > SIGCHLD > Nov 4 14:09:52opensips[2655]: INFO:core:sig_usr: signal 15 received > > 2. Message from gdb: > > ... > Reading symbols from /usr/local/sbin/opensips...done. > [New LWP 2658] > [Thread debugging using libthread_db enabled] > Using host libthread_db library "/usr/lib64/libthread_db.so.1". > Core was generated by `/usr/local/sbin/opensips -P > /var/run/opensips/opensips.pid -f /etc/opensips/ope'. > Program terminated with signal 6, Aborted. > #0 0x00007fcbf3d421f7 in raise () from /usr/lib64/libc.so.6 > Missing separate debuginfos, use: debuginfo-install > bzip2-libs-1.0.6-13.el7.x86_64 elfutils-libelf-0.168-8.el7.x86_64 > elfutils-libs-0.168-8.el7.x86_64 glibc-2.17-196.el7.x86_64 > gmp-6.0.0-15.el7.x86_64 gnutls-3.3.26-9.el7.x86_64 > hiredis-0.12.1-1.el7.x86_64 json-c-0.11-4.el7_0.x86_64 > keyutils-libs-1.5.8-3.el7.x86_64 krb5-libs-1.15.1-8.el7.x86_64 > libattr-2.4.46-12.el7.x86_64 libcap-2.22-9.el7.x86_64 > libcom_err-1.42.9-10.el7.x86_64 libffi-3.0.13-18.el7.x86_64 > libgcc-4.8.5-16.el7.x86_64 libgcrypt-1.5.3-14.el7.x86_64 > libgpg-error-1.12-3.el7.x86_64 libmicrohttpd-0.9.33-2.el7.x86_64 > libselinux-2.5-11.el7.x86_64 libstdc++-4.8.5-16.el7.x86_64 > libtasn1-4.10-1.el7.x86_64 libxml2-2.9.1-6.el7_2.3.x86_64 > mariadb-libs-5.5.56-2.el7.x86_64 nettle-2.7.1-8.el7.x86_64 > openssl-libs-1.0.2k-8.el7.x86_64 p11-kit-0.23.5-3.el7.x86_64 > pcre-8.32-17.el7.x86_64 systemd-libs-219-42.el7_4.4.x86_64 > xz-libs-5.2.2-1.el7.x86_64 zlib-1.2.7-17.el7.x86_64 > > 3. Here is gdb bt full log: > > https://pastebin.com/HFkzxnty > > Best regards, > Dragomir > > > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From goup2010 at gmail.com Mon Nov 6 15:06:38 2017 From: goup2010 at gmail.com (Dragomir Haralambiev) Date: Mon, 6 Nov 2017 22:06:38 +0200 Subject: [OpenSIPS-Users] Opensips 2.3.2 chashed In-Reply-To: <1e55ada4-623a-32e6-62c4-30da0c3d381e@opensips.org> References: <1e55ada4-623a-32e6-62c4-30da0c3d381e@opensips.org> Message-ID: Hi, Sorry the last email is other Opensips crash. This is the info you want: https://pastebin.com/ejNr9de0 Best regards, Dragomir 2017-11-06 11:42 GMT+02:00 Răzvan Crainea : > Hi, Dragomir! > > Can you to go in gdb top frame 3 and print call_it. Use the following > snippet and attach the output in pastebin. > > > frame 3 > > p *call_it > > Thanks! > > Răzvan Crainea > OpenSIPS Developerwww.opensips-solutions.com > > On 11/04/2017 02:35 PM, Dragomir Haralambiev wrote: > > Hello, > > I use latest update from git at Opensips 2.3.2. > When try to use call center module Opensips crashed: > > 1. Log message: > > Nov 4 14:09:52opensips[2658]: CRITICAL:call_center:cc_queue_rmv_call: > QUEUE - call not in queue l=(nil), h=(nil) > Nov 4 14:09:52opensips[2666]: CRITICAL:core:receive_fd: EOF on 48 > Nov 4 14:09:52opensips[2636]: INFO:core:handle_sigs: child process 2658 > exited by a signal 6 > Nov 4 14:09:52opensips[2636]: INFO:core:handle_sigs: core was generated > Nov 4 14:09:52opensips[2636]: INFO:core:handle_sigs: terminating due to > SIGCHLD > Nov 4 14:09:52opensips[2655]: INFO:core:sig_usr: signal 15 received > > 2. Message from gdb: > > ... > Reading symbols from /usr/local/sbin/opensips...done. > [New LWP 2658] > [Thread debugging using libthread_db enabled] > Using host libthread_db library "/usr/lib64/libthread_db.so.1". > Core was generated by `/usr/local/sbin/opensips -P > /var/run/opensips/opensips.pid -f /etc/opensips/ope'. > Program terminated with signal 6, Aborted. > #0 0x00007fcbf3d421f7 in raise () from /usr/lib64/libc.so.6 > Missing separate debuginfos, use: debuginfo-install > bzip2-libs-1.0.6-13.el7.x86_64 elfutils-libelf-0.168-8.el7.x86_64 > elfutils-libs-0.168-8.el7.x86_64 glibc-2.17-196.el7.x86_64 > gmp-6.0.0-15.el7.x86_64 gnutls-3.3.26-9.el7.x86_64 > hiredis-0.12.1-1.el7.x86_64 json-c-0.11-4.el7_0.x86_64 > keyutils-libs-1.5.8-3.el7.x86_64 krb5-libs-1.15.1-8.el7.x86_64 > libattr-2.4.46-12.el7.x86_64 libcap-2.22-9.el7.x86_64 > libcom_err-1.42.9-10.el7.x86_64 libffi-3.0.13-18.el7.x86_64 > libgcc-4.8.5-16.el7.x86_64 libgcrypt-1.5.3-14.el7.x86_64 > libgpg-error-1.12-3.el7.x86_64 libmicrohttpd-0.9.33-2.el7.x86_64 > libselinux-2.5-11.el7.x86_64 libstdc++-4.8.5-16.el7.x86_64 > libtasn1-4.10-1.el7.x86_64 libxml2-2.9.1-6.el7_2.3.x86_64 > mariadb-libs-5.5.56-2.el7.x86_64 nettle-2.7.1-8.el7.x86_64 > openssl-libs-1.0.2k-8.el7.x86_64 p11-kit-0.23.5-3.el7.x86_64 > pcre-8.32-17.el7.x86_64 systemd-libs-219-42.el7_4.4.x86_64 > xz-libs-5.2.2-1.el7.x86_64 zlib-1.2.7-17.el7.x86_64 > > 3. Here is gdb bt full log: > > https://pastebin.com/HFkzxnty > > Best regards, > Dragomir > > > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From alexanderhenryperkins at gmail.com Mon Nov 6 20:35:26 2017 From: alexanderhenryperkins at gmail.com (Alexander Perkins) Date: Mon, 6 Nov 2017 19:35:26 -0600 Subject: [OpenSIPS-Users] Getting RateLimit Config From Database Message-ID: Hi All. I have been researching OpenSIPs for quite some time now and have implemented a deployment successfully with it. However, I was never able to successfully pull the RateLimit configuration from a database. For example, if I want to limit a SIP client to only one call per second and have that information pull from MySQL, how would I do that? I have tried searching Google, but cannot find much information. If I cannot pull it in from MySQL, can I reload the RateLimit config without restarting OpenSIPs? Any help is appreciated, Alex -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Tue Nov 7 03:55:16 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Tue, 7 Nov 2017 10:55:16 +0200 Subject: [OpenSIPS-Users] Wrong Contact in location table In-Reply-To: References: <0093287c-82e4-c11a-c933-e5e195683156@opensips.org> <741e1ed0-1831-4a1f-aa21-08ed354c2b53@opensips.org> <8045BDD6-2077-4CCC-B6C3-7E007AF631F3@tampahost.net> <9eb0a075-7235-fba0-83c9-62c3ab5de93f@opensips.org> <13955c88-1a4f-d106-10ec-a29525d4c39a@opensips.org> <58c846f0-7fe6-f687-d7aa-b505b20a41ef@opensips.org> <541a84ad-437a-2a0a-0338-c5fca288015a@opensips.org> Message-ID: <355520de-3005-380e-a7f6-76e149793017@opensips.org> Hi, Dragomir! This is something that I noticed from the first email you have sent - the fix_nated_register() function is not called, or does not work properly. Can you also print the avp you are setting in the received_avp[1]. Also, call script trace for the reply too. [1] http://www.opensips.org/html/docs/modules/2.4.x/nathelper.html#idp5510048 Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/06/2017 06:18 PM, Dragomir Haralambiev wrote: > Hi, > > Thanks for your email. > > Here is part of my location table: > >          contact_id  username   domain  contact  received  path    > expires                   q  callid                                   >       cseq  last_modified        flags  cflags     user_agent socket  >                  methods  sip_instance  attr > -------------------  ---------  ------ > -------------------------------------------------------------------------- > --------  ------  -------------------  ------ > ------------------------------------------------  ------ > -------------------  ------  --------- > --------------------------------  ----------------------- -------  > ------------  -------- >  181494352482801881  57996206   (NULL) > sip:57996206 at 192.168.22.206:5062 >                             >               (NULL)    (NULL) 2017-11-06 17:58:48   -1.00 > 1162502851 at 192.168.22.206           >             162  2017-11-06 17:52:48       0 NAT_BFLAG  Yealink > SIP-T19P 31.72.0.75  udp:OpenSips_IP:5060    16383  (NULL)        (NULL) >  181654460760464436  57996204   (NULL) > sip:57996204 at 192.168.22.204:5060 >                             >               (NULL)    (NULL) 2017-11-06 17:59:02   -1.00 > 0_1763370066 at 192.168.22.204       >                 179  2017-11-06 17:53:02       0 NAT_BFLAG  Yealink > SIP-T21P_E2 52.81.0.25 udp:OpenSips_IP:5060    16383  (NULL)        > (NULL) > Why "received" field is blank? > Where could be the problem? > > > 2017-11-06 11:44 GMT+02:00 Răzvan Crainea >: > > Hi, Dragomir! > > If you simply do fix_nated_register() on the REGISTER messages, > all these will be sorted out. Moreover, it's actually not correct > to change the contact of the user, because in the SIP message it > might expect to have exactly what he sent. > When using fix_nated_register(), there is another field (called > Received) that stores the actual IP and port where the REGISTER > came from, and when an invite comes in, it is automatically set by > the lookup() function in the DST uri, without changing the contact > (the contact may still be private). > > So simply calling fix_nated_register() should fix this issue for > all scenarios. > I initially though that you were using the Contact from something > else and you really need that value there. > > Best regards, > > Răzvan Crainea > OpenSIPS Developer > www.opensips-solutions.com > > On 11/03/2017 09:36 PM, Dragomir Haralambiev wrote: >> Hi, >> >> Why I need the real IP and port in location table? >> >> Now I make only outgoing call. Everything works fine on the >> following scenario.: >> User -----> Opensips ------> ITSP >> >> If I not have real IP in location table incoming calls not be >> implemented. >> ITSP ----> Opensips ---->? >> >> Opensips get IP from location table and try to send call. But in >> location table have not real IP. >> I see how the Opensips try to send call to 192.168.2.34. >> >> This is the main problem. >> >> 2017-11-03 11:45 GMT+02:00 Răzvan Crainea > >: >> >> Unfortunately I just realised that you cannot change the >> Contact header for this scenario. >> And to be honest I don't really understand why you are trying >> to change it - if you need the real IP and port, you can take >> them from the received field. >> >> Best regards, >> >> Răzvan Crainea >> OpenSIPS Developer >> www.opensips-solutions.com >> >> On 11/02/2017 11:43 PM, Dragomir Haralambiev wrote: >>> Hi, >>> >>> Here is part of my script: >>> .... >>> modparam("registrar", "mcontact_avp", "$avp(register)") >>> ..... >>> >>>     if (t_check_status("2[0-9][0-9]")) { >>> $log_level = 5; >>> script_trace( 1, "$rm from $si, ruri=$ru, ct=$ct.fields(uri) >>> avp(register)=$avp(register)", "me"); >>> route(save_location); >>>                        ....... >>>     } >>> >>> You can see log here: >>> https://pastebin.com/WWQ9Mmh4 >>> >>> Here is the replacement contact: >>> >>> DBG:registrar:build_contact: created Contact HF: Contact: >>> >> >;expires=360 >>> DBG:registrar:save: replacing contact uri >>> [sip:55595009 at 188.23.232.10:1043 >>> ] with >>> [sip:55595009 at 192.168.22.138:5062 >>> ] >>> >>> How to stop replacing contact from 188.23.232.10:1043 >>> to 192.168.22.138:5062 >>> ? >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Tue Nov 7 04:04:32 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Tue, 7 Nov 2017 11:04:32 +0200 Subject: [OpenSIPS-Users] Getting RateLimit Config From Database In-Reply-To: References: Message-ID: Hi, Alexander! You can pull the limits per user either by using the load_credentials mechanism[1], which automatically pulls user information after authentication, or by doing a "manual" DB query using the avp_db_query() function[2]. After you have the limit value in an AVP, you can pass that pseudo-variable to the rl_check() function[3]. [1] http://www.opensips.org/html/docs/modules/2.4.x/auth_db#idp5578480 [2] http://www.opensips.org/html/docs/modules/2.4.x/avpops.html#idp5619104 [3] http://www.opensips.org/html/docs/modules/2.4.x/ratelimit#idp5577136 Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/07/2017 03:35 AM, Alexander Perkins wrote: > Hi All.  I have been researching OpenSIPs for quite some time now and > have implemented a deployment successfully with it.  However, I was > never able to successfully pull the RateLimit configuration from a > database.  For example, if I want to limit a SIP client to only one > call per second and have that information pull from MySQL, how would I > do that?  I have tried searching Google, but cannot find much > information.  If I cannot pull it in from MySQL, can I reload the > RateLimit config without restarting OpenSIPs? > > Any help is appreciated, > Alex > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From nocbgtelcom at gmail.com Tue Nov 7 05:25:10 2017 From: nocbgtelcom at gmail.com (Hristo Donev) Date: Tue, 7 Nov 2017 12:25:10 +0200 Subject: [OpenSIPS-Users] Wrong Contact in location table In-Reply-To: <355520de-3005-380e-a7f6-76e149793017@opensips.org> References: <0093287c-82e4-c11a-c933-e5e195683156@opensips.org> <741e1ed0-1831-4a1f-aa21-08ed354c2b53@opensips.org> <8045BDD6-2077-4CCC-B6C3-7E007AF631F3@tampahost.net> <9eb0a075-7235-fba0-83c9-62c3ab5de93f@opensips.org> <13955c88-1a4f-d106-10ec-a29525d4c39a@opensips.org> <58c846f0-7fe6-f687-d7aa-b505b20a41ef@opensips.org> <541a84ad-437a-2a0a-0338-c5fca288015a@opensips.org> <355520de-3005-380e-a7f6-76e149793017@opensips.org> Message-ID: YES !!! Here is problem: I have follow line: modparam("nathelper|registrar","received_avp", "$avp(42)") This not working. All is OK if I use: modparam("nathelper","received_avp", "$avp(42)") modparam("registrar", "received_avp", "$avp(42)") 2017-11-07 10:55 GMT+02:00 Răzvan Crainea : > Hi, Dragomir! > > This is something that I noticed from the first email you have sent - the > fix_nated_register() function is not called, or does not work properly. Can > you also print the avp you are setting in the received_avp[1]. Also, call > script trace for the reply too. > > [1] http://www.opensips.org/html/docs/modules/2.4.x/nathelper. > html#idp5510048 > > Best regards, > > Răzvan Crainea > OpenSIPS Developerwww.opensips-solutions.com > > On 11/06/2017 06:18 PM, Dragomir Haralambiev wrote: > > Hi, > > Thanks for your email. > > Here is part of my location table: > > contact_id username domain contact > received path expires > q callid cseq > last_modified flags cflags user_agent > socket methods sip_instance attr > ------------------- --------- ------ ------------------------------ > -------------------------------------------- -------- ------ > ------------------- ------ ------------------------------------------------ > ------ ------------------- ------ --------- > -------------------------------- ----------------------- ------- > ------------ -------- > 181494352482801881 57996206 (NULL) sip:57996206 at 192.168.22.206:5062 > (NULL) (NULL) 2017-11-06 > 17:58:48 -1.00 1162502851 at 192.168.22.206 > 162 2017-11-06 17:52:48 0 NAT_BFLAG Yealink SIP-T19P 31.72.0.75 > udp:OpenSips_IP:5060 16383 (NULL) (NULL) > 181654460760464436 57996204 (NULL) sip:57996204 at 192.168.22.204:5060 > (NULL) (NULL) 2017-11-06 > 17:59:02 -1.00 0_1763370066 at 192.168.22.204 > 179 2017-11-06 17:53:02 0 NAT_BFLAG Yealink SIP-T21P_E2 > 52.81.0.25 udp:OpenSips_IP:5060 16383 (NULL) (NULL) > > Why "received" field is blank? > Where could be the problem? > > > 2017-11-06 11:44 GMT+02:00 Răzvan Crainea : > >> Hi, Dragomir! >> >> If you simply do fix_nated_register() on the REGISTER messages, all these >> will be sorted out. Moreover, it's actually not correct to change the >> contact of the user, because in the SIP message it might expect to have >> exactly what he sent. >> When using fix_nated_register(), there is another field (called Received) >> that stores the actual IP and port where the REGISTER came from, and when >> an invite comes in, it is automatically set by the lookup() function in the >> DST uri, without changing the contact (the contact may still be private). >> >> So simply calling fix_nated_register() should fix this issue for all >> scenarios. >> I initially though that you were using the Contact from something else >> and you really need that value there. >> >> Best regards, >> >> Răzvan Crainea >> OpenSIPS Developerwww.opensips-solutions.com >> >> On 11/03/2017 09:36 PM, Dragomir Haralambiev wrote: >> >> Hi, >> >> Why I need the real IP and port in location table? >> >> Now I make only outgoing call. Everything works fine on the following >> scenario.: >> User -----> Opensips ------> ITSP >> >> If I not have real IP in location table incoming calls not be implemented. >> ITSP ----> Opensips ---->? >> >> Opensips get IP from location table and try to send call. But in location >> table have not real IP. >> I see how the Opensips try to send call to 192.168.2.34. >> >> This is the main problem. >> >> 2017-11-03 11:45 GMT+02:00 Răzvan Crainea : >> >>> Unfortunately I just realised that you cannot change the Contact header >>> for this scenario. >>> And to be honest I don't really understand why you are trying to change >>> it - if you need the real IP and port, you can take them from the received >>> field. >>> >>> Best regards, >>> >>> Răzvan Crainea >>> OpenSIPS Developerwww.opensips-solutions.com >>> >>> On 11/02/2017 11:43 PM, Dragomir Haralambiev wrote: >>> >>> Hi, >>> >>> Here is part of my script: >>> .... >>> modparam("registrar", "mcontact_avp", "$avp(register)") >>> ..... >>> >>> if (t_check_status("2[0-9][0-9]")) { >>> $log_level = 5; >>> script_trace( 1, "$rm from $si, ruri=$ru, ct=$ct.fields(uri) >>> avp(register)=$avp(register)", "me"); >>> route(save_location); >>> ....... >>> } >>> >>> You can see log here: >>> https://pastebin.com/WWQ9Mmh4 >>> >>> Here is the replacement contact: >>> >>> DBG:registrar:build_contact: created Contact HF: Contact: < >>> sip:55595009 at 192.168.22.138:5062>;expires=360 >>> DBG:registrar:save: replacing contact uri [sip:55595009 at 188.23.232.10:10 >>> 43] with [sip:55595009 at 192.168.22.138:5062] >>> >>> How to stop replacing contact from 188.23.232.10:1043 to >>> 192.168.22.138:5062 ? >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From goup2010 at gmail.com Tue Nov 7 08:55:58 2017 From: goup2010 at gmail.com (Dragomir Haralambiev) Date: Tue, 7 Nov 2017 15:55:58 +0200 Subject: [OpenSIPS-Users] Wrong Contact in location table In-Reply-To: References: <0093287c-82e4-c11a-c933-e5e195683156@opensips.org> <741e1ed0-1831-4a1f-aa21-08ed354c2b53@opensips.org> <8045BDD6-2077-4CCC-B6C3-7E007AF631F3@tampahost.net> <9eb0a075-7235-fba0-83c9-62c3ab5de93f@opensips.org> <13955c88-1a4f-d106-10ec-a29525d4c39a@opensips.org> <58c846f0-7fe6-f687-d7aa-b505b20a41ef@opensips.org> <541a84ad-437a-2a0a-0338-c5fca288015a@opensips.org> <355520de-3005-380e-a7f6-76e149793017@opensips.org> Message-ID: Thanks Razvan and Hristo, Add in my script follow lines and incoming calls worked fine: modparam("registrar", "received_avp", "$avp(received)") modparam("nathelper","received_avp", "$avp(received)") Thank you! 2017-11-07 12:25 GMT+02:00 Hristo Donev : > YES !!! > > Here is problem: > > I have follow line: > modparam("nathelper|registrar","received_avp", "$avp(42)") > > This not working. > > All is OK if I use: > modparam("nathelper","received_avp", "$avp(42)") > modparam("registrar", "received_avp", "$avp(42)") > > > > > > 2017-11-07 10:55 GMT+02:00 Răzvan Crainea : > >> Hi, Dragomir! >> >> This is something that I noticed from the first email you have sent - the >> fix_nated_register() function is not called, or does not work properly. Can >> you also print the avp you are setting in the received_avp[1]. Also, call >> script trace for the reply too. >> >> [1] http://www.opensips.org/html/docs/modules/2.4.x/nathelper.ht >> ml#idp5510048 >> >> Best regards, >> >> Răzvan Crainea >> OpenSIPS Developerwww.opensips-solutions.com >> >> On 11/06/2017 06:18 PM, Dragomir Haralambiev wrote: >> >> Hi, >> >> Thanks for your email. >> >> Here is part of my location table: >> >> contact_id username domain contact >> received path expires >> q callid cseq >> last_modified flags cflags user_agent >> socket methods sip_instance attr >> ------------------- --------- ------ ------------------------------ >> -------------------------------------------- -------- ------ >> ------------------- ------ ------------------------------------------------ >> ------ ------------------- ------ --------- >> -------------------------------- ----------------------- ------- >> ------------ -------- >> 181494352482801881 57996206 (NULL) sip:57996206 at 192.168.22.206:5062 >> (NULL) (NULL) 2017-11-06 >> 17:58:48 -1.00 1162502851 at 192.168.22.206 >> 162 2017-11-06 17:52:48 0 NAT_BFLAG Yealink SIP-T19P 31.72.0.75 >> udp:OpenSips_IP:5060 16383 (NULL) (NULL) >> 181654460760464436 57996204 (NULL) sip:57996204 at 192.168.22.204:5060 >> (NULL) (NULL) 2017-11-06 >> 17:59:02 -1.00 0_1763370066 at 192.168.22.204 >> 179 2017-11-06 17:53:02 0 NAT_BFLAG Yealink SIP-T21P_E2 >> 52.81.0.25 udp:OpenSips_IP:5060 16383 (NULL) (NULL) >> >> Why "received" field is blank? >> Where could be the problem? >> >> >> 2017-11-06 11:44 GMT+02:00 Răzvan Crainea : >> >>> Hi, Dragomir! >>> >>> If you simply do fix_nated_register() on the REGISTER messages, all >>> these will be sorted out. Moreover, it's actually not correct to change the >>> contact of the user, because in the SIP message it might expect to have >>> exactly what he sent. >>> When using fix_nated_register(), there is another field (called >>> Received) that stores the actual IP and port where the REGISTER came from, >>> and when an invite comes in, it is automatically set by the lookup() >>> function in the DST uri, without changing the contact (the contact may >>> still be private). >>> >>> So simply calling fix_nated_register() should fix this issue for all >>> scenarios. >>> I initially though that you were using the Contact from something else >>> and you really need that value there. >>> >>> Best regards, >>> >>> Răzvan Crainea >>> OpenSIPS Developerwww.opensips-solutions.com >>> >>> On 11/03/2017 09:36 PM, Dragomir Haralambiev wrote: >>> >>> Hi, >>> >>> Why I need the real IP and port in location table? >>> >>> Now I make only outgoing call. Everything works fine on the following >>> scenario.: >>> User -----> Opensips ------> ITSP >>> >>> If I not have real IP in location table incoming calls not be >>> implemented. >>> ITSP ----> Opensips ---->? >>> >>> Opensips get IP from location table and try to send call. But in >>> location table have not real IP. >>> I see how the Opensips try to send call to 192.168.2.34. >>> >>> This is the main problem. >>> >>> 2017-11-03 11:45 GMT+02:00 Răzvan Crainea : >>> >>>> Unfortunately I just realised that you cannot change the Contact header >>>> for this scenario. >>>> And to be honest I don't really understand why you are trying to change >>>> it - if you need the real IP and port, you can take them from the received >>>> field. >>>> >>>> Best regards, >>>> >>>> Răzvan Crainea >>>> OpenSIPS Developerwww.opensips-solutions.com >>>> >>>> On 11/02/2017 11:43 PM, Dragomir Haralambiev wrote: >>>> >>>> Hi, >>>> >>>> Here is part of my script: >>>> .... >>>> modparam("registrar", "mcontact_avp", "$avp(register)") >>>> ..... >>>> >>>> if (t_check_status("2[0-9][0-9]")) { >>>> $log_level = 5; >>>> script_trace( 1, "$rm from $si, ruri=$ru, ct=$ct.fields(uri) >>>> avp(register)=$avp(register)", "me"); >>>> route(save_location); >>>> ....... >>>> } >>>> >>>> You can see log here: >>>> https://pastebin.com/WWQ9Mmh4 >>>> >>>> Here is the replacement contact: >>>> >>>> DBG:registrar:build_contact: created Contact HF: Contact: < >>>> sip:55595009 at 192.168.22.138:5062>;expires=360 >>>> DBG:registrar:save: replacing contact uri [ >>>> sip:55595009 at 188.23.232.10:1043] with [sip:55595009 at 192.168.22.138:5062 >>>> ] >>>> >>>> How to stop replacing contact from 188.23.232.10:1043 to >>>> 192.168.22.138:5062 ? >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>> >>> >>> _______________________________________________ >>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Tue Nov 7 09:54:53 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Tue, 7 Nov 2017 16:54:53 +0200 Subject: [OpenSIPS-Users] Wrong Contact in location table In-Reply-To: References: <8045BDD6-2077-4CCC-B6C3-7E007AF631F3@tampahost.net> <9eb0a075-7235-fba0-83c9-62c3ab5de93f@opensips.org> <13955c88-1a4f-d106-10ec-a29525d4c39a@opensips.org> <58c846f0-7fe6-f687-d7aa-b505b20a41ef@opensips.org> <541a84ad-437a-2a0a-0338-c5fca288015a@opensips.org> <355520de-3005-380e-a7f6-76e149793017@opensips.org> Message-ID: <7b019424-ed67-5d64-f6e1-897b98bffe1c@opensips.org> Hi, Dragomir! So you did not have the parameter provisioned at all? Or what was the initial issue? Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/07/2017 03:55 PM, Dragomir Haralambiev wrote: > Thanks Razvan and Hristo, > > Add in my script follow lines and incoming calls worked fine: > > modparam("registrar", "received_avp", "$avp(received)") > modparam("nathelper","received_avp", "$avp(received)") > > Thank you! > > 2017-11-07 12:25 GMT+02:00 Hristo Donev >: > > YES !!! > > Here is problem: > > I have follow line: > modparam("nathelper|registrar","received_avp", "$avp(42)") > > This not working. > > All is OK if I use: > modparam("nathelper","received_avp", "$avp(42)") > modparam("registrar", "received_avp", "$avp(42)") > > > > > > 2017-11-07 10:55 GMT+02:00 Răzvan Crainea >: > > Hi, Dragomir! > > This is something that I noticed from the first email you have > sent - the fix_nated_register() function is not called, or > does not work properly. Can you also print the avp you are > setting in the received_avp[1]. Also, call script trace for > the reply too. > > [1] > http://www.opensips.org/html/docs/modules/2.4.x/nathelper.html#idp5510048 > > > Best regards, > > Răzvan Crainea > OpenSIPS Developer > www.opensips-solutions.com > > On 11/06/2017 06:18 PM, Dragomir Haralambiev wrote: >> Hi, >> >> Thanks for your email. >> >> Here is part of my location table: >> >>          contact_id  username  domain  contact             >>  received  path expires                   q  callid         >> cseq  last_modified  flags  cflags     user_agent             >>     socket      methods  sip_instance  attr >> -------------------  --------- ------ >> -------------------------------------------------------------------------- >> --------  ------ -------------------  ------ >> ------------------------------------------------ ------  >> -------------------  ------ --------- >> -------------------------------- -----------------------  >> ------- ------------  -------- >>  181494352482801881  57996206  (NULL) >> sip:57996206 at 192.168.22.206:5062 >>       (NULL)    >> (NULL)  2017-11-06 17:58:48   -1.00 1162502851 at 192.168.22.206 >>                           >> 162 2017-11-06 17:52:48       0 NAT_BFLAG  Yealink SIP-T19P >> 31.72.0.75  udp:OpenSips_IP:5060    16383 (NULL)        (NULL) >>  181654460760464436  57996204  (NULL) >> sip:57996204 at 192.168.22.204:5060 >>       (NULL)    >> (NULL)  2017-11-06 17:59:02   -1.00 >> 0_1763370066 at 192.168.22.204 >>                         >> 179 2017-11-06 17:53:02       0 NAT_BFLAG  Yealink >> SIP-T21P_E2 52.81.0.25    udp:OpenSips_IP:5060   16383  >> (NULL)        (NULL) >> Why "received" field is blank? >> Where could be the problem? >> >> >> 2017-11-06 11:44 GMT+02:00 Răzvan Crainea >> >: >> >> Hi, Dragomir! >> >> If you simply do fix_nated_register() on the REGISTER >> messages, all these will be sorted out. Moreover, it's >> actually not correct to change the contact of the user, >> because in the SIP message it might expect to have >> exactly what he sent. >> When using fix_nated_register(), there is another field >> (called Received) that stores the actual IP and port >> where the REGISTER came from, and when an invite comes >> in, it is automatically set by the lookup() function in >> the DST uri, without changing the contact (the contact >> may still be private). >> >> So simply calling fix_nated_register() should fix this >> issue for all scenarios. >> I initially though that you were using the Contact from >> something else and you really need that value there. >> >> Best regards, >> >> Răzvan Crainea >> OpenSIPS Developer >> www.opensips-solutions.com >> >> >> On 11/03/2017 09:36 PM, Dragomir Haralambiev wrote: >>> Hi, >>> >>> Why I need the real IP and port in location table? >>> >>> Now I make only outgoing call. Everything works fine on >>> the following scenario.: >>> User -----> Opensips ------> ITSP >>> >>> If I not have real IP in location table incoming calls >>> not be implemented. >>> ITSP ----> Opensips ---->? >>> >>> Opensips get IP from location table and try to send >>> call. But in location table have not real IP. >>> I see how the Opensips try to send call to 192.168.2.34. >>> >>> This is the main problem. >>> >>> 2017-11-03 11:45 GMT+02:00 Răzvan Crainea >>> >: >>> >>> Unfortunately I just realised that you cannot change >>> the Contact header for this scenario. >>> And to be honest I don't really understand why you >>> are trying to change it - if you need the real IP >>> and port, you can take them from the received field. >>> >>> Best regards, >>> >>> Răzvan Crainea >>> OpenSIPS Developer >>> www.opensips-solutions.com >>> >>> >>> On 11/02/2017 11:43 PM, Dragomir Haralambiev wrote: >>>> Hi, >>>> >>>> Here is part of my script: >>>> .... >>>> modparam("registrar", "mcontact_avp", "$avp(register)") >>>> ..... >>>> >>>>     if (t_check_status("2[0-9][0-9]")) { >>>> $log_level = 5; >>>> script_trace( 1, "$rm from $si, ruri=$ru, >>>> ct=$ct.fields(uri) avp(register)=$avp(register)", >>>> "me"); >>>> route(save_location); >>>>  ....... >>>>     } >>>> >>>> You can see log here: >>>> https://pastebin.com/WWQ9Mmh4 >>>> >>>> Here is the replacement contact: >>>> >>>> DBG:registrar:build_contact: created Contact HF: >>>> Contact: >>> >;expires=360 >>>> DBG:registrar:save: replacing contact uri >>>> [sip:55595009 at 188.23.232.10:1043 >>>> ] with >>>> [sip:55595009 at 192.168.22.138:5062 >>>> ] >>>> >>>> How to stop replacing contact from >>>> 188.23.232.10:1043 to >>>> 192.168.22.138:5062 ? >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From goup2010 at gmail.com Tue Nov 7 10:21:28 2017 From: goup2010 at gmail.com (Dragomir Haralambiev) Date: Tue, 7 Nov 2017 17:21:28 +0200 Subject: [OpenSIPS-Users] Wrong Contact in location table In-Reply-To: <7b019424-ed67-5d64-f6e1-897b98bffe1c@opensips.org> References: <8045BDD6-2077-4CCC-B6C3-7E007AF631F3@tampahost.net> <9eb0a075-7235-fba0-83c9-62c3ab5de93f@opensips.org> <13955c88-1a4f-d106-10ec-a29525d4c39a@opensips.org> <58c846f0-7fe6-f687-d7aa-b505b20a41ef@opensips.org> <541a84ad-437a-2a0a-0338-c5fca288015a@opensips.org> <355520de-3005-380e-a7f6-76e149793017@opensips.org> <7b019424-ed67-5d64-f6e1-897b98bffe1c@opensips.org> Message-ID: I had the following line in my script: modparam("nathelper|registrar", "received_avp", "$avp(received)") I delete it, and put new modparam("registrar", "received_avp", "$avp(received)") modparam("nathelper","received_avp", "$avp(received)") 2017-11-07 16:54 GMT+02:00 Răzvan Crainea : > Hi, Dragomir! > > So you did not have the parameter provisioned at all? Or what was the > initial issue? > > Best regards, > > Răzvan Crainea > OpenSIPS Developerwww.opensips-solutions.com > > On 11/07/2017 03:55 PM, Dragomir Haralambiev wrote: > > Thanks Razvan and Hristo, > > Add in my script follow lines and incoming calls worked fine: > > modparam("registrar", "received_avp", "$avp(received)") > modparam("nathelper","received_avp", "$avp(received)") > > Thank you! > > 2017-11-07 12:25 GMT+02:00 Hristo Donev : > >> YES !!! >> >> Here is problem: >> >> I have follow line: >> modparam("nathelper|registrar","received_avp", "$avp(42)") >> >> This not working. >> >> All is OK if I use: >> modparam("nathelper","received_avp", "$avp(42)") >> modparam("registrar", "received_avp", "$avp(42)") >> >> >> >> >> >> 2017-11-07 10:55 GMT+02:00 Răzvan Crainea : >> >>> Hi, Dragomir! >>> >>> This is something that I noticed from the first email you have sent - >>> the fix_nated_register() function is not called, or does not work properly. >>> Can you also print the avp you are setting in the received_avp[1]. Also, >>> call script trace for the reply too. >>> >>> [1] http://www.opensips.org/html/docs/modules/2.4.x/nathelper.ht >>> ml#idp5510048 >>> >>> Best regards, >>> >>> Răzvan Crainea >>> OpenSIPS Developerwww.opensips-solutions.com >>> >>> On 11/06/2017 06:18 PM, Dragomir Haralambiev wrote: >>> >>> Hi, >>> >>> Thanks for your email. >>> >>> Here is part of my location table: >>> >>> contact_id username domain contact >>> received path expires >>> q callid cseq >>> last_modified flags cflags user_agent >>> socket methods sip_instance attr >>> ------------------- --------- ------ ------------------------------ >>> -------------------------------------------- -------- ------ >>> ------------------- ------ ------------------------------------------------ >>> ------ ------------------- ------ --------- >>> -------------------------------- ----------------------- ------- >>> ------------ -------- >>> 181494352482801881 57996206 (NULL) sip:57996206 at 192.168.22.206:5062 >>> (NULL) (NULL) 2017-11-06 >>> 17:58:48 -1.00 1162502851 at 192.168.22.206 >>> 162 2017-11-06 17:52:48 0 NAT_BFLAG Yealink SIP-T19P 31.72.0.75 >>> udp:OpenSips_IP:5060 16383 (NULL) (NULL) >>> 181654460760464436 57996204 (NULL) sip:57996204 at 192.168.22.204:5060 >>> (NULL) (NULL) 2017-11-06 >>> 17:59:02 -1.00 0_1763370066 at 192.168.22.204 >>> 179 2017-11-06 17:53:02 0 NAT_BFLAG Yealink SIP-T21P_E2 >>> 52.81.0.25 udp:OpenSips_IP:5060 16383 (NULL) (NULL) >>> >>> Why "received" field is blank? >>> Where could be the problem? >>> >>> >>> 2017-11-06 11:44 GMT+02:00 Răzvan Crainea : >>> >>>> Hi, Dragomir! >>>> >>>> If you simply do fix_nated_register() on the REGISTER messages, all >>>> these will be sorted out. Moreover, it's actually not correct to change the >>>> contact of the user, because in the SIP message it might expect to have >>>> exactly what he sent. >>>> When using fix_nated_register(), there is another field (called >>>> Received) that stores the actual IP and port where the REGISTER came from, >>>> and when an invite comes in, it is automatically set by the lookup() >>>> function in the DST uri, without changing the contact (the contact may >>>> still be private). >>>> >>>> So simply calling fix_nated_register() should fix this issue for all >>>> scenarios. >>>> I initially though that you were using the Contact from something else >>>> and you really need that value there. >>>> >>>> Best regards, >>>> >>>> Răzvan Crainea >>>> OpenSIPS Developerwww.opensips-solutions.com >>>> >>>> On 11/03/2017 09:36 PM, Dragomir Haralambiev wrote: >>>> >>>> Hi, >>>> >>>> Why I need the real IP and port in location table? >>>> >>>> Now I make only outgoing call. Everything works fine on the following >>>> scenario.: >>>> User -----> Opensips ------> ITSP >>>> >>>> If I not have real IP in location table incoming calls not be >>>> implemented. >>>> ITSP ----> Opensips ---->? >>>> >>>> Opensips get IP from location table and try to send call. But in >>>> location table have not real IP. >>>> I see how the Opensips try to send call to 192.168.2.34. >>>> >>>> This is the main problem. >>>> >>>> 2017-11-03 11:45 GMT+02:00 Răzvan Crainea : >>>> >>>>> Unfortunately I just realised that you cannot change the Contact >>>>> header for this scenario. >>>>> And to be honest I don't really understand why you are trying to >>>>> change it - if you need the real IP and port, you can take them from the >>>>> received field. >>>>> >>>>> Best regards, >>>>> >>>>> Răzvan Crainea >>>>> OpenSIPS Developerwww.opensips-solutions.com >>>>> >>>>> On 11/02/2017 11:43 PM, Dragomir Haralambiev wrote: >>>>> >>>>> Hi, >>>>> >>>>> Here is part of my script: >>>>> .... >>>>> modparam("registrar", "mcontact_avp", "$avp(register)") >>>>> ..... >>>>> >>>>> if (t_check_status("2[0-9][0-9]")) { >>>>> $log_level = 5; >>>>> script_trace( 1, "$rm from $si, ruri=$ru, ct=$ct.fields(uri) >>>>> avp(register)=$avp(register)", "me"); >>>>> route(save_location); >>>>> ....... >>>>> } >>>>> >>>>> You can see log here: >>>>> https://pastebin.com/WWQ9Mmh4 >>>>> >>>>> Here is the replacement contact: >>>>> >>>>> DBG:registrar:build_contact: created Contact HF: Contact: < >>>>> sip:55595009 at 192.168.22.138:5062>;expires=360 >>>>> DBG:registrar:save: replacing contact uri [ >>>>> sip:55595009 at 188.23.232.10:1043] with [sip:55595009 at 192.168.22.138:5 >>>>> 062] >>>>> >>>>> How to stop replacing contact from 188.23.232.10:1043 to >>>>> 192.168.22.138:5062 ? >>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>> >>> >>> _______________________________________________ >>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From propalarupa at gmail.com Tue Nov 7 11:37:36 2017 From: propalarupa at gmail.com (Trupyy .) Date: Tue, 7 Nov 2017 17:37:36 +0100 Subject: [OpenSIPS-Users] AES + Opensip In-Reply-To: <6171011d-542f-89f8-9eaa-00f5bd2e1bdd@opensips.org> References: <6171011d-542f-89f8-9eaa-00f5bd2e1bdd@opensips.org> Message-ID: Hi, Thanks for the answer, sry im not that strong in encryption. I know opensip uses tls and it makes sense to me that tls + something else on top of it would not work, i was interested if it is possible to make opensips work with aes, im guessing not? Thanks! On Nov 2, 2017 6:21 PM, "Bogdan-Andrei Iancu" wrote: > Hi, > > Basically your Polycom will try to do SIP over TLS with AES cipher ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 11/01/2017 06:26 PM, Trupyy . wrote: > > Hello Everyone, > > I got a quick question, hope someone can help me with it. So I got two > Polycom VC devices to work with Opensips but the Calls work only when i > turn off the Encryption (AES Option), when i turn it on i get an error that > some security settings are not working properly. > > Can someone tell me if opensips is compatible with AES encryption or is > the only encryption possibility, TLS? > > Thank you. > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From dlakelan at street-artists.org Tue Nov 7 12:01:46 2017 From: dlakelan at street-artists.org (Daniel Lakeland) Date: Tue, 7 Nov 2017 09:01:46 -0800 Subject: [OpenSIPS-Users] AES + Opensip In-Reply-To: References: <6171011d-542f-89f8-9eaa-00f5bd2e1bdd@opensips.org> Message-ID: On 11/07/2017 08:37 AM, Trupyy . wrote: > Hi, > > Thanks for the answer, sry im not that strong in encryption. I know > opensip uses tls and it makes sense to me that tls + something else on > top of it would not work, i was interested if it is possible to make > opensips work with aes, im guessing not? > > Thanks! TLS is a method for setting up an encrypted connection, AES is a kind of encryption algorithm (a cipher). So TLS can set up encrypted sessions with many different algorithms/ciphers, what you probably need is to configure your TLS settings such that the ciphers that are allowed include the AES cipher that the phone wants to use. At least this should point you in the direction of what to look at. I can't help much more than that because I don't know a thing about polycom phones nor do I know where to configure the allowed ciphers, would take a bunch of googling to figure those out, but at least now you maybe know what to google ;-) From dlakelan at street-artists.org Tue Nov 7 12:22:02 2017 From: dlakelan at street-artists.org (Daniel Lakeland) Date: Tue, 7 Nov 2017 09:22:02 -0800 Subject: [OpenSIPS-Users] Further adventures in dual-stack ipv6/ipv4 questions In-Reply-To: <6501b440-e13a-b32f-fe7a-6c6742aa779b@opensips.org> References: <47941745-1f85-f9da-a7fa-d68f06c2bb39@street-artists.org> <6501b440-e13a-b32f-fe7a-6c6742aa779b@opensips.org> Message-ID: <0103f9fc-3e0e-0b39-37fb-c0de024d2b41@street-artists.org> On 11/01/2017 07:16 AM, Bogdan-Andrei Iancu wrote: Bogdan: thanks so much for your detailed info on what happens. I think this will help me debug stuff. I have some further questions below if you have time. > >> Suppose I have a phone that registers to opensips using TLS. This is >> a TCP based socket. Suppose that the phone is ipv4, so it's behind a >> NAT. Well, in truth, it's probably behind multiple layers of NAT, >> potentially a carrier grade NAT somewhere deep in the bowels of an >> ISP, then potentially a customer service ADSL modem or the like, and >> then a router that I have control over. As I understand it after >> setting TCP_PERSISTENT Opensips should leave this TLS connection open >> and can then pass further reinvites, pings, and soforth down the same >> socket. This is particularly true if the phone supports Sip Outbound >> RFC 5626 (all ok so far?) > [bogdan] > The only thing required for the end point is not to close the > connection. On its side, OpenSIPS will keep the TCP Conn alive for the > whole duration of the registration, so the user is reachable. My impression is that over WiFi if there are momentary lapses in connectivity, Android may sometimes decide that it needs to reassociate to a new network, scan, then reassociate to the same network (this is different from a simple roaming between APs and very different from an 802.11r fast transition). Even though it's associated to the same network, it still resets TCP connections. This means the TCP connection to OpenSIPS drops. If this occurs in the middle of a call, it sounds like I have to get lucky and hope that OpenSIPS reopens the connection to my phone via its reachable IPv6 address. In the absence of that, my call is probably going to drop. It also sounds like this might rarely work due to ephemeral ports on my client. >> Suppose further that after a while with Opensips in this reset state >> with no open connection to my phone, my phone re-registers and a TLS >> connection is available, will Opensips be able to pass reinvites and >> pings for ongoing calls down this new socket? Does it understand that >> this is the same phone and that this new registration+socket is >> relevant to the ongoing dialog? > [bogdan] > jumping on new connection during a SIP session is not easy. As the TCP > connections uses ephemeral ports, there are some internal aliases kept > by OpenSIPS between SIP URIs and the actual TCP connections. A new > connection means a new TCP port -> the existing aliases will not work. If the client sends some kind of unique ID, such as a standard Sip_instance (for example used by OBI ATAs and in a UUID format) or an rinstance attribute (used by Bria on Android and just in ;rinstance=.........; with a big hex number in there or whatever (admittedly it seems like clients do a lot of different stuff here). Can / Does OpenSIPS try to reassociate its connection to the same instance? [bogdan] > All the time you need to consider both layer - the transport and the > SIP layer. If the registration (at SIP layer) is not reachable anymore > (based on the transport layer), the registration must be invalidated. > You can do this based on registration SIP pinging in OpenSIPS. >> >> With the end-to-end nature of Ipv6 it seems we have a definite >> possibility to avoid a lot of hassle with opensips able to directly >> open a new sip connection to the phone to continue ongoing dialogs. >> Is this a possibility for why I'm experiencing fewer random dropped >> calls? > [bogdan] > I wouldn't gamble on it. This seems to be a major bottleneck for reliable calls over TLS via WiFi. If the TLS connection can't be maintained because momentary WiFi droppage causes reset packets to be sent by Android OS or other OS then the registration is invalidated and the call will drop as soon as the remote party tries to do anything (such as reINVITE or the like). This will be true even though the RTP is traveling around just fine. Is that more or less the sum of it? Are there things I could do to improve this situation? Suppose far end wants to reINVITE either as a kind of pinging, or for a hold or something. It sends SIP message to OpenSIPS, now over the last few minutes OpenSIPS lost the specific registration from my client, and then regained it when my client re-opened the connection. Is there a way I can make OpenSIPS know what happened and forward my phone the INVITE on the new socket? What things need to be supported on the client? What needs configuration in the server? Thanks again. Dan From dlakelan at street-artists.org Tue Nov 7 12:31:33 2017 From: dlakelan at street-artists.org (Daniel Lakeland) Date: Tue, 7 Nov 2017 09:31:33 -0800 Subject: [OpenSIPS-Users] Further adventures in dual-stack ipv6/ipv4 questions In-Reply-To: <0103f9fc-3e0e-0b39-37fb-c0de024d2b41@street-artists.org> References: <47941745-1f85-f9da-a7fa-d68f06c2bb39@street-artists.org> <6501b440-e13a-b32f-fe7a-6c6742aa779b@opensips.org> <0103f9fc-3e0e-0b39-37fb-c0de024d2b41@street-artists.org> Message-ID: <982c89a7-3be6-e882-9d88-8f595151d8e4@street-artists.org> On 11/07/2017 09:22 AM, Daniel Lakeland wrote: > > This seems to be a major bottleneck for reliable calls over TLS via > WiFi. If the TLS connection can't be maintained because momentary WiFi > droppage causes reset packets to be sent by Android OS or other OS > then the registration is invalidated and the call will drop as soon as > the remote party tries to do anything (such as reINVITE or the like). > This will be true even though the RTP is traveling around just fine. > Is that more or less the sum of it? > > Are there things I could do to improve this situation? Suppose far end > wants to reINVITE either as a kind of pinging, or for a hold or > something. It sends SIP message to OpenSIPS, now over the last few > minutes OpenSIPS lost the specific registration from my client, and > then regained it when my client re-opened the connection. Is there a > way I can make OpenSIPS know what happened and forward my phone the > INVITE on the new socket? What things need to be supported on the > client? What needs configuration in the server? It may help to know that I did the following test just now: Bria 3.9.6.99409 on Android registered to OpenSIPS via ipv6 which then talks to Asterisk on same server. Connect to OpenSIPS via Wifi using TLS on ipv6 only network, call asterisk to extension that reads the time for several minutes While listening to the time, turn off Wifi entirely and wait about 4 seconds... phone now switches to T-Mobile ipv6 mobile data network, magically audio returns after a few seconds. Now, put phone into airplane mode, then turn on just wifi. Magically after several seconds, audio returns... I didn't do packet captures etc, so I'm not sure what SIP messages were sent. But my guess is that when connection dropped, Bria was re-connecting, and then re-inviting the call to its new address. Bria is probably sending the Call-ID so the momentary losses of connections/registrations were OK for this call. Now, if OpenSIPS had sent a ping, or the remote end had reINVITED during the period after I turned off wifi and before Bria got itself connected to OpenSIPS on the mobile network... or the like... I imagine I might have had a different experience. From adrian.fretwell at topgreen.co.uk Tue Nov 7 13:59:32 2017 From: adrian.fretwell at topgreen.co.uk (Adrian Fretwell) Date: Tue, 7 Nov 2017 18:59:32 +0000 Subject: [OpenSIPS-Users] Tearing down the RTPProxy session for the current call Message-ID: <75c75c8d-c0d8-d52c-5815-1edf2c00200a@topgreen.co.uk> Hello, I was just looking at some statistics from my rtpproxy today and noticed that "RTP sessions created", "RTP sessions destroyed" AND "RTP sessions ended due to media timeout" all showed the same number. Although I am aware that the function rtpproxy_unforce() is available, in example routing scripts, I have never seen it used. Should I be calling rtpproxy_unforce() in my routing script at the end of a call? My assumption is that if I do, then the "RTP sessions ended due to media timeout" will be much lower. As always any advice will be gratefully received. Kind regards, Adrian Fretwell -------------- next part -------------- An HTML attachment was scrubbed... URL: From mas_66 at tiscali.co.uk Tue Nov 7 14:31:35 2017 From: mas_66 at tiscali.co.uk (mas_66 at tiscali.co.uk) Date: Tue, 7 Nov 2017 19:31:35 +0000 (GMT) Subject: [OpenSIPS-Users] Opensips v2.3.2 Message-ID: <13583756.2208531510083095821.JavaMail.defaultUser@defaultHost> Hello Users I have an issue with seeing calls within control panel either in Dialog(current calls) or anything in SIP trace Any experience with what may be causing this before i try to reinvent the wheel? Thanks in advance Mas From Rajesh.Govindaraj at ipc.com Tue Nov 7 14:58:07 2017 From: Rajesh.Govindaraj at ipc.com (Govindaraj, Rajesh) Date: Tue, 7 Nov 2017 19:58:07 +0000 Subject: [OpenSIPS-Users] opensips default tls port in route header Message-ID: Hi, Opensips version 1.11.5_tls is being used. We are getting a route header in INVITE from Avaya without any port but transport as tls. We expect Opensips use default port of 5061 in this case, but it uses 5060 and call fails. Wanted to know if this issue is known and/or fixed in any latest versions of opensips. Thanks, Route: 2017-11-06T12:21:46.000-05:00 [local2] [debug] ccm-1.ipc.com /usr/sbin/opensipsExternal[6687]: DBG:core:parse_uri: uname=[0x844a06]-> uval=[0x844a0d]-> 2017-11-06T12:21:46.000-05:00 [local2] [debug] ccm-1.ipc.com /usr/sbin/opensipsExternal[6687]: DBG:core:grep_sock_info: checking if host==us: 10==10 && [10.64.49.2] == [10.64.49.2] 2017-11-06T12:21:46.000-05:00 [local2] [debug] ccm-1.ipc.com /usr/sbin/opensipsExternal[6687]: DBG:core:grep_sock_info: checking if port 5066 matches port 5060 2017-11-06T12:21:46.000-05:00 [local2] [debug] ccm-1.ipc.com /usr/sbin/opensipsExternal[6687]: DBG:core:grep_sock_info: checking if host==us: 10==10 && [10.64.49.2] == [10.64.49.2] 2017-11-06T12:21:46.000-05:00 [local2] [debug] ccm-1.ipc.com /usr/sbin/opensipsExternal[6687]: DBG:core:grep_sock_info: checking if port 5066 matches port 5060 2017-11-06T12:21:46.000-05:00 [local2] [debug] ccm-1.ipc.com /usr/sbin/opensipsExternal[6687]: DBG:core:grep_sock_info: checking if host==us: 10==10 && [10.64.49.2] == [10.64.49.2] 2017-11-06T12:21:46.000-05:00 [local2] [debug] ccm-1.ipc.com /usr/sbin/opensipsExternal[6687]: DBG:core:grep_sock_info: checking if port 5061 matches port 5060 2017-11-06T12:21:46.000-05:00 [local2] [debug] ccm-1.ipc.com /usr/sbin/opensipsExternal[6687]: DBG:core:check_self: host != me 2017-11-06T12:21:46.000-05:00 [local2] [debug] ccm-1.ipc.com /usr/sbin/opensipsExternal[6687]: DBG:rr:after_loose: Topmost URI is NOT myself 2017-11-06T12:21:46.000-05:00 [local2] [err] ccm-1.ipc.com /usr/sbin/opensipsExternal[6687]: Attempt to route with preloaded Route's [sip:3035380101 at avaya.com/sip:72051 at 10.64.110.12/sip:72051 at avaya.com/5526f8c8c31641e7a41005056abff8] Rajeshkumar Govindaraj Software Engineer 777 Commerce Drive, Fairfield, CT-06825 T +1 201 253 7803 * M +1 475 439 9918 * E Rajesh.Govindaraj at ipc.com Follow us on twitter: @ipc_Systems_Inc www.ipc.com [cid:image006.jpg at 01D1940F.3E021840] DISCLAIMER: This e-mail may contain information that is confidential, privileged or otherwise protected from disclosure. If you are not an intended recipient of this e-mail, do not duplicate or redistribute it by any means. Please delete it and any attachments and notify the sender that you have received it in error. Unintended recipients are prohibited from taking action on the basis of information in this e-mail. E-mail messages may contain computer viruses or other defects, may not be accurately replicated on other systems, or may be intercepted, deleted or interfered with without the knowledge of the sender or the intended recipient. If you are not comfortable with the risks associated with e-mail messages, you may decide not to use e-mail to communicate with IPC. IPC reserves the right, to the extent and under circumstances permitted by applicable law, to retain, monitor and intercept e-mail messages to and from its systems. -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 4676 bytes Desc: image001.jpg URL: From eahaselhoff at gmail.com Tue Nov 7 17:26:38 2017 From: eahaselhoff at gmail.com (Edwin) Date: Tue, 7 Nov 2017 15:26:38 -0700 (MST) Subject: [OpenSIPS-Users] replace-origin flag error Message-ID: <1510093598685-0.post@n2.nabble.com> I get an error on opensips 2.3.2 (Debian) when I run rtpengine_manage("replace-origin ICE=remove"); ERROR:rtpengine:parse_flags: error processing flag `replace-origin': cannot add to bson item Any idea what is the cause or even better, the solution? -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html From razvan at opensips.org Wed Nov 8 05:12:32 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Wed, 8 Nov 2017 12:12:32 +0200 Subject: [OpenSIPS-Users] replace-origin flag error In-Reply-To: <1510093598685-0.post@n2.nabble.com> References: <1510093598685-0.post@n2.nabble.com> Message-ID: <5e44a837-9055-b9aa-00d9-21b32ae05e2c@opensips.org> Hi, Edwin! What message is the ERROR triggered on? Is it on the INVITE, ACK, BYE? Can you post the entire DEBUG log? Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/08/2017 12:26 AM, Edwin wrote: > I get an error on opensips 2.3.2 (Debian) when I run > rtpengine_manage("replace-origin ICE=remove"); > ERROR:rtpengine:parse_flags: error processing flag `replace-origin': cannot > add to bson item > > Any idea what is the cause or even better, the solution? > > > > -- > 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 nferreira at fuze.com Wed Nov 8 05:17:07 2017 From: nferreira at fuze.com (Nuno Ferreira) Date: Wed, 8 Nov 2017 10:17:07 +0000 Subject: [OpenSIPS-Users] Introducing OpenSIPS 2.4 In-Reply-To: <972d9f71-89d4-89f1-9183-f7fc0f1c506a@opensips.org> References: <972d9f71-89d4-89f1-9183-f7fc0f1c506a@opensips.org> Message-ID: Hi Bogdan, Do you have further details to share about the "*clustered media relays*" and "*Asterisk flavored* Load-Balancing" features? Thanks, Nuno Ferreira On Wed, Nov 1, 2017 at 5:16 PM, Bogdan-Andrei Iancu wrote: > One more year, one more evolution cycle, one more OpenSIPS major release. > So let me introduce you the upcoming OpenSIPS 2.4 . > > For the OpenSIPS 2.4 release we decided to focus on the *clustering > abilities*. Today’s VoIP world is getting more and more dynamic, services > are moving into Clouds and more and more flexibility is needed for the > application to fully exploit such environments. But let’s pin point the > main reasons for going for a clustered approach : > > - scaling up with the processing/traffic load > - geographical distribution > - redundancy and High-Availability > > For the OpenSIPS 2.4 we laid down a roadmap that addresses the clustering > both from the clustering engine itself (the underlayer) and from the > functionalities that will perform on top of the clustering layer, to share > data and state. > > With OpenSIPS 2.4, it will never be easier to built a consistent and > powerful clustered solution: > > - *clustering engine* – enhances the capabilities of controlling the > cluster topology, like re-routing for bypassing broken links, dynamic > joining of new nodes, support for multiple capabilities per node, data > syncing between nodes and many more; > - *distributed user location* – this is a very complex topic as it > exceeds the simple concept of data sharing. By the nature of the data (the > user registrations), you may have different constraints on how data is > roaming in a cluster – registrations may be tied to a node due NAT or TCP > constraints. Even more, a sharding aspect must be addressed when looking at > distributing the pinging effort across the cluster. So, multiple solutions > are viable here, depending on what is to be achieved (scaling, redundancy) > and what are the network constraints – see a detailed presentation > > of the available solutions; > - *distributed presence server* – quite similar (but less complex) as > the distributed user location, a distributed presence server provides a > consistent, but distributed way of sharing presence information – SIP > entities may publish data via different nodes in the SIP cluster, while the > subscribers may fetch presence data via multiple various nodes. Two > approaches are under work : (a) a cluster built around a noSQL DB based as > primary data storage and (b) a cluster exclusively relying on OpenSIPS for > data sharing; > - *anycast support* – to be able to build a fully-flavored anycast > support (addressing both redundancy and balancing) requires OpenSIPS to > replicate/share transaction state across the nodes in the cluster (nodes > sharing the same anycast IP). Depending on the nature of the replication > (full transaction versus transaction meta-data) a full anycast and light > anycast support will be available – here is a detailed description on > the anycast > > support; > - *clustered media relays* – as OpenSIPS has the ability to work > together with several flavors of media relays (such as RTPproxy, RTPEngine, > MediaProxy), the clustering support will help OpenSIPS do distributed > load-balancing over the relays – even if a relay is used by multiple nodes > in the cluster, all the nodes will share information on the load on the > relay, to avoid overloading or idle time; > - *distributed call center* – an agent is able to register with > multiple queues on different nodes on a cluster. Still, all the queues do > share the status / availability of the agent and its statistics for call > distribution; > - *custom clustering* – the OpenSIPS clustering underlayer provides at > script level the ability to broadcast (in the cloud) or send to a given > node a custom message/action (with replying possibility) – this is a very > flexible and powerful way to build your custom distributed functionality > directly at script level. > > And because we started on the integration path with OpenSIPS 2.3, and > because we did it well, we decided to push forward on this path with the > 2.4 version as well: > > - more *Homer integration *to be able to report TCP statistics, DB > events and media relay events via HEP; > - *SIPREC integration* for standard call recording. The new SIPREC > module > provides a standard and transparent (for the call parties) way to do call > recording against an external recorder like Oreka > provided by Orecx > ; > - more *FreeSWITCH integration* in order to capture the call-events > (DTMFs, call status) from FreeSWITCH into OpenSIPS script or for being able > to control a FreeSWITCH call from OpenSIPS script via ESL > - *Asterisk flavored* Load-Balancing for a more realistic and accurate > traffic balancing over Asterisk clusters (as the load information is > fetched in realtime from Asterisk); > > ------------------------------ > > The timeline for OpenSIPS 2.4 is: > > - Beta Release – 12-16 March 2018 > - Stable Release – 23-27 April 2018 > - General Availability – 1st of May 2018, during OpenSIPS Summit 2018 > > > To talk more about the features of this new release, a public audio > conference will be available on 21st > of November 2017, 4 pm GMT > , thanks to > the kind sponsorship of UberConference . > Anyone is welcome to join to find out more details or to ask questions > about OpenSIPS 2.4 . > > This is a public and open conference, so no registration is needed, but if > you want to announce your intention to participate, please let us know here: > > http://blog.opensips.org/2017/11/01/introducing-opensips-2-4/ > > > Best regards, > > -- > 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 > > -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.* -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Wed Nov 8 05:22:37 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Wed, 8 Nov 2017 12:22:37 +0200 Subject: [OpenSIPS-Users] opensips default tls port in route header In-Reply-To: References: Message-ID: Yes, this should be solved in the latest 1.11 release. Note that this release is no longer maintained, so if 1.11 still does not work, we suggest you to upgrade to a newer stable version. Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/07/2017 09:58 PM, Govindaraj, Rajesh wrote: > > Hi, > > Opensips version 1.11.5_tls is being used. > > We are getting a route header in INVITE from Avaya without any port > but transport as tls. We expect Opensips use default port of 5061 in > this case, but it uses 5060 and call fails. > > > Wanted to know if this issue is known and/or fixed in any latest > versions of opensips. > > Thanks, > > Route: > > > > 2017-11-06T12:21:46.000-05:00 [local2] [debug] ccm-1.ipc.com > /usr/sbin/opensipsExternal[6687]: DBG:core:parse_uri: > uname=[0x844a06]-> uval=[0x844a0d]-> > 2017-11-06T12:21:46.000-05:00 [local2] [debug] ccm-1.ipc.com > /usr/sbin/opensipsExternal[6687]: DBG:core:grep_sock_info: checking if > host==us: 10==10 &&  [10.64.49.2] == [10.64.49.2] > 2017-11-06T12:21:46.000-05:00 [local2] [debug] ccm-1.ipc.com > /usr/sbin/opensipsExternal[6687]: DBG:core:grep_sock_info: checking if > port 5066 matches port 5060 > 2017-11-06T12:21:46.000-05:00 [local2] [debug] ccm-1.ipc.com > /usr/sbin/opensipsExternal[6687]: DBG:core:grep_sock_info: checking if > host==us: 10==10 &&  [10.64.49.2] == [10.64.49.2] > 2017-11-06T12:21:46.000-05:00 [local2] [debug] ccm-1.ipc.com > /usr/sbin/opensipsExternal[6687]: DBG:core:grep_sock_info: checking if > port 5066 matches port 5060 > 2017-11-06T12:21:46.000-05:00 [local2] [debug] ccm-1.ipc.com > /usr/sbin/opensipsExternal[6687]: DBG:core:grep_sock_info: checking if > host==us: 10==10 &&  [10.64.49.2] == [10.64.49.2] > 2017-11-06T12:21:46.000-05:00 [local2] [debug] ccm-1.ipc.com > /usr/sbin/opensipsExternal[6687]: DBG:core:grep_sock_info: checking if > port 5061 matches port 5060 > 2017-11-06T12:21:46.000-05:00 [local2] [debug] ccm-1.ipc.com > /usr/sbin/opensipsExternal[6687]: DBG:core:check_self: host != me > 2017-11-06T12:21:46.000-05:00 [local2] [debug] ccm-1.ipc.com > /usr/sbin/opensipsExternal[6687]: DBG:rr:after_loose: Topmost URI is > NOT myself > > 2017-11-06T12:21:46.000-05:00 [local2] [err] ccm-1.ipc.com > /usr/sbin/opensipsExternal[6687]: Attempt to route with preloaded > Route's > [sip:3035380101 at avaya.com/sip:72051 at 10.64.110.12/sip:72051 at avaya.com/5526f8c8c31641e7a41005056abff8] > > > *Rajeshkumar Govindaraj* > > Software Engineer > 777 Commerce Drive, > Fairfield, CT-06825 > *T*+1 201 253 7803 |*M* +1 475 439 9918 |*E* Rajesh.Govindaraj at ipc.com > > > Follow us on twitter: @ipc_Systems_Inc www.ipc.com > > cid:image006.jpg at 01D1940F.3E021840 > > > > > DISCLAIMER: This e-mail may contain information that is confidential, > privileged or otherwise protected from disclosure. If you are not an > intended recipient of this e-mail, do not duplicate or redistribute it > by any means. Please delete it and any attachments and notify the > sender that you have received it in error. Unintended recipients are > prohibited from taking action on the basis of information in this > e-mail. E-mail messages may contain computer viruses or other defects, > may not be accurately replicated on other systems, or may be > intercepted, deleted or interfered with without the knowledge of the > sender or the intended recipient. If you are not comfortable with the > risks associated with e-mail messages, you may decide not to use > e-mail to communicate with IPC. IPC reserves the right, to the extent > and under circumstances permitted by applicable law, to retain, > monitor and intercept e-mail messages to and from its systems. > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 4676 bytes Desc: not available URL: From razvan at opensips.org Wed Nov 8 05:23:44 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Wed, 8 Nov 2017 12:23:44 +0200 Subject: [OpenSIPS-Users] Opensips v2.3.2 In-Reply-To: <13583756.2208531510083095821.JavaMail.defaultUser@defaultHost> References: <13583756.2208531510083095821.JavaMail.defaultUser@defaultHost> Message-ID: <4c123c0b-4727-f242-eab6-02ee445cc93a@opensips.org> Hi, Mas! Are you seeing the dialogs in the database? Also, what db_mode[1] are you using? [1] http://www.opensips.org/html/docs/modules/2.4.x/dialog#idp5562032 Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/07/2017 09:31 PM, mas_66--- via Users wrote: > Hello Users > I have an issue with seeing calls within control panel either in Dialog(current > calls) or anything in SIP trace > Any experience with what may be causing this before i try to reinvent the > wheel? > Thanks in advance > Mas > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From razvan at opensips.org Wed Nov 8 05:47:09 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Wed, 8 Nov 2017 12:47:09 +0200 Subject: [OpenSIPS-Users] Tearing down the RTPProxy session for the current call In-Reply-To: <75c75c8d-c0d8-d52c-5815-1edf2c00200a@topgreen.co.uk> References: <75c75c8d-c0d8-d52c-5815-1edf2c00200a@topgreen.co.uk> Message-ID: <6923b55a-26cf-e8b3-37ab-4d908b294bd2@opensips.org> Hi, Adrian! Yes, you should be calling rtpproxy_unforce() when a BYE is received. You are right, if you do that, you won't be seeing too many "RTP sessions ended due to media timeout". Best regardsm Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/07/2017 08:59 PM, Adrian Fretwell wrote: > > Hello, > > I was just looking at some statistics from my rtpproxy today and > noticed that "RTP sessions created", "RTP sessions destroyed" AND "RTP > sessions ended due to media timeout" all showed the same number. > > Although I am aware that the function rtpproxy_unforce() is available, > in example routing scripts, I have never seen it used. > > Should I be calling rtpproxy_unforce() in my routing script at the end > of a call? > > My assumption is that if I do, then the "RTP sessions ended due to > media timeout" will be much lower. > > As always any advice will be gratefully received. > > 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 wilhelm.lundgren at gmail.com Wed Nov 8 07:09:53 2017 From: wilhelm.lundgren at gmail.com (Wilhelm Lundgren) Date: Wed, 8 Nov 2017 13:09:53 +0100 Subject: [OpenSIPS-Users] avp_db_query in opensips.cfg Message-ID: Hi, Im running opensips 2.2 (built on the platform) on an arm platform (wheezy). In the config i use several avp_db_query against the sqlite db. This works fine. However, when using the same config on the same linux (but for amd64, or i386), using the same opensips version, i cant start opensips, i get: ERROR:avpops:avpops_db_bind: driver for DB URL [default] does not support raw queries! when starting opensips. Removing all the avp_db_query from the cfg will make opensips start ok, but i need those queries. Why does those queries work on arm and not on x86? I dont understand why there should be any difference. /Wilhelm -------------- next part -------------- An HTML attachment was scrubbed... URL: From goup2010 at gmail.com Wed Nov 8 07:23:57 2017 From: goup2010 at gmail.com (Dragomir Haralambiev) Date: Wed, 8 Nov 2017 14:23:57 +0200 Subject: [OpenSIPS-Users] outbound call center services Message-ID: Hello all, The Call Center module is develop for inbound call center service. Does anyone have any idea how to develop (auto dialer) for outbound call center services with call distribution to agents. Best regards, Dragomir -------------- next part -------------- An HTML attachment was scrubbed... URL: From khamlichi.khalil at gmail.com Wed Nov 8 07:36:35 2017 From: khamlichi.khalil at gmail.com (Khalil Khamlichi) Date: Wed, 8 Nov 2017 12:36:35 +0000 Subject: [OpenSIPS-Users] outbound call center services In-Reply-To: References: Message-ID: You will surely need answering machine detection at a certain point of your developement for this kind of job (dialer) which is not possible because opensips does not handle media. So your best bet is freeswitch or asterisk. Freeswitch is optimal but lacks a free answering machine detection module. Asterisk less programatic but has it for free. Sent via mobile, please forgive typos and brevity On Nov 8, 2017 12:31 PM, "Dragomir Haralambiev" wrote: Hello all, The Call Center module is develop for inbound call center service. Does anyone have any idea how to develop (auto dialer) for outbound call center services with call distribution to agents. 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 khamlichi.khalil at gmail.com Wed Nov 8 07:37:47 2017 From: khamlichi.khalil at gmail.com (Khalil Khamlichi) Date: Wed, 8 Nov 2017 12:37:47 +0000 Subject: [OpenSIPS-Users] outbound call center services In-Reply-To: References: Message-ID: You can still use opensips callcenter module for your agents though. As only dialer function will be handled by freeswitch. Sent via mobile, please forgive typos and brevity On Nov 8, 2017 12:36 PM, "Khalil Khamlichi" wrote: > You will surely need answering machine detection at a certain point of > your developement for this kind of job (dialer) which is not possible > because opensips does not handle media. So your best bet is freeswitch or > asterisk. Freeswitch is optimal but lacks a free answering machine > detection module. Asterisk less programatic but has it for free. > > Sent via mobile, please forgive typos and brevity > > On Nov 8, 2017 12:31 PM, "Dragomir Haralambiev" > wrote: > > Hello all, > > The Call Center module is develop for inbound call center service. > > Does anyone have any idea how to develop (auto dialer) for outbound call > center services with call distribution to agents. > > 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 eahaselhoff at gmail.com Wed Nov 8 08:54:42 2017 From: eahaselhoff at gmail.com (Edwin) Date: Wed, 8 Nov 2017 06:54:42 -0700 (MST) Subject: [OpenSIPS-Users] replace-origin flag error In-Reply-To: <5e44a837-9055-b9aa-00d9-21b32ae05e2c@opensips.org> References: <1510093598685-0.post@n2.nabble.com> <5e44a837-9055-b9aa-00d9-21b32ae05e2c@opensips.org> Message-ID: <1510149282639-0.post@n2.nabble.com> Hi Răzvan, It happens in my case when I call to mobile en reject the message, a 486 Busy Here is sent, debug doesn't tell that much: +++++ Reply route route_to_gateway +++++ INVITE 486 Busy here 1.2.3.4 Busy here DBG:core:parse_headers: flags=2000 DBG:core:parse_headers: flags=ffffffffffffffff DBG:core:parse_sdp: message body has length zero DBG:core:parse_headers: flags=40 ERROR:rtpengine:parse_flags: error processing flag `replace-origin': cannot add to bson item Nov 8 14:44:40 back-1-staging opensips[19089]: ... DBG:tm:t_should_relay_response: T_code=180, new_code=486 DBG:tm:t_pick_branch: picked branch 0, code 486 (prio=686) DBG:core:parse_headers: flags=ffffffffffffffff I just call a 'rtpengine_manage("replace-origin ICE=remove")' in the reply route, but maybe I have to only do this for Invite, ack etc? Gr. Edwin -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html From razvan at opensips.org Wed Nov 8 09:14:59 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Wed, 8 Nov 2017 16:14:59 +0200 Subject: [OpenSIPS-Users] avp_db_query in opensips.cfg In-Reply-To: References: Message-ID: <45756329-a60d-7c65-f85f-65e1b4b05439@opensips.org> Hi, Wilhelm! I don't think the problem is the linux version, but OpenSIPS version. Can you check if the two installs are using different versions of OpenSIPS (try ./opensips -V). Lately there was a commit which might trigger this error and I want to double check if that's the case. Can you send the version of OpenSIPS you are using? Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/08/2017 02:09 PM, Wilhelm Lundgren wrote: > Hi, > > Im running opensips 2.2 (built on the platform) on an arm platform > (wheezy). In the config i use several avp_db_query against the sqlite > db. This works fine. > > However, when using the same config on the same linux (but for amd64, > or i386), using the same opensips version,  i cant start opensips, i get: > > ERROR:avpops:avpops_db_bind: driver for DB URL [default] does not > support raw queries! when starting opensips. > > Removing all the avp_db_query from the cfg will make opensips start > ok, but i need those queries. > > Why does those queries work on arm and not on x86? I dont understand > why there should be any difference. > > /Wilhelm > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Wed Nov 8 09:16:50 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Wed, 8 Nov 2017 16:16:50 +0200 Subject: [OpenSIPS-Users] replace-origin flag error In-Reply-To: <1510149282639-0.post@n2.nabble.com> References: <1510093598685-0.post@n2.nabble.com> <5e44a837-9055-b9aa-00d9-21b32ae05e2c@opensips.org> <1510149282639-0.post@n2.nabble.com> Message-ID: <56163b8e-72ef-916a-e56d-a7eb94e85034@opensips.org> Hi, Edwin! I think you are right - when deleting the session (a negative reply is received), that flag should no longer be used. I will push a fix now. What version of OpenSIPS are you using, and how did you install it? Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/08/2017 03:54 PM, Edwin wrote: > Hi Răzvan, > > It happens in my case when I call to mobile en reject the message, a 486 > Busy Here is sent, debug doesn't tell that much: > > +++++ Reply route route_to_gateway +++++ INVITE 486 Busy here 1.2.3.4 > Busy here > DBG:core:parse_headers: flags=2000 > DBG:core:parse_headers: flags=ffffffffffffffff > DBG:core:parse_sdp: message body has length zero > DBG:core:parse_headers: flags=40 > ERROR:rtpengine:parse_flags: error processing flag `replace-origin': cannot > add to bson item > Nov 8 14:44:40 back-1-staging opensips[19089]: ... > DBG:tm:t_should_relay_response: T_code=180, new_code=486 > DBG:tm:t_pick_branch: picked branch 0, code 486 (prio=686) > DBG:core:parse_headers: flags=ffffffffffffffff > > I just call a 'rtpengine_manage("replace-origin ICE=remove")' in the reply > route, but maybe I have to only do this for Invite, ack etc? > > Gr. > > Edwin > > > > -- > 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 wilhelm.lundgren at gmail.com Wed Nov 8 11:04:34 2017 From: wilhelm.lundgren at gmail.com (Wilhelm Lundgren) Date: Wed, 8 Nov 2017 17:04:34 +0100 Subject: [OpenSIPS-Users] avp_db_query in opensips.cfg In-Reply-To: <45756329-a60d-7c65-f85f-65e1b4b05439@opensips.org> References: <45756329-a60d-7c65-f85f-65e1b4b05439@opensips.org> Message-ID: Ahh, thats probably it: Working version: version: opensips 2.2.1 (arm/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. git revision: 68ace2e main.c compiled on 14:42:20 Sep 14 2016 with gcc 4.6 None working version: version: opensips 2.2.5 (i386/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. git revision: 4d0cc20 main.c compiled on 09:52:01 Nov 8 2017 with gcc 4.7 On 8 November 2017 at 15:14, Răzvan Crainea wrote: > Hi, Wilhelm! > > I don't think the problem is the linux version, but OpenSIPS version. Can > you check if the two installs are using different versions of OpenSIPS (try > ./opensips -V). > Lately there was a commit which might trigger this error and I want to > double check if that's the case. Can you send the version of OpenSIPS you > are using? > > Best regards, > > Răzvan Crainea > OpenSIPS Developerwww.opensips-solutions.com > > On 11/08/2017 02:09 PM, Wilhelm Lundgren wrote: > > Hi, > > Im running opensips 2.2 (built on the platform) on an arm platform > (wheezy). In the config i use several avp_db_query against the sqlite db. > This works fine. > > However, when using the same config on the same linux (but for amd64, or > i386), using the same opensips version, i cant start opensips, i get: > > ERROR:avpops:avpops_db_bind: driver for DB URL [default] does not support > raw queries! when starting opensips. > > Removing all the avp_db_query from the cfg will make opensips start ok, > but i need those queries. > > Why does those queries work on arm and not on x86? I dont understand why > there should be any difference. > > /Wilhelm > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Wed Nov 8 11:32:17 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Wed, 8 Nov 2017 18:32:17 +0200 Subject: [OpenSIPS-Users] [BLOG] Call Recording in OpenSIPS 2.4 using SIPREC Message-ID: Hi, Everyone! Find out how you can do call recording in OpenSIPS using the SIPREC protocol: https://blog.opensips.org/2017/11/08/call-recording-in-opensips-2-4-using-siprec/ Best regards, -- Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com From eahaselhoff at gmail.com Wed Nov 8 12:04:01 2017 From: eahaselhoff at gmail.com (Edwin) Date: Wed, 8 Nov 2017 10:04:01 -0700 (MST) Subject: [OpenSIPS-Users] replace-origin flag error In-Reply-To: <56163b8e-72ef-916a-e56d-a7eb94e85034@opensips.org> References: <1510093598685-0.post@n2.nabble.com> <5e44a837-9055-b9aa-00d9-21b32ae05e2c@opensips.org> <1510149282639-0.post@n2.nabble.com> <56163b8e-72ef-916a-e56d-a7eb94e85034@opensips.org> Message-ID: <1510160641879-0.post@n2.nabble.com> Hi Răzvan, I'm using version 2.3.2-1 from apt.opensips.org repository, on a Debian stretch release. Is there a workaround for now, something like: if(t_check_status("403|404|480|484|486|487") rtpengine_manage("ICE=remove"); } else { rtpengine_manage("replace-origin ICE=remove"); } -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html From bogdan at opensips.org Wed Nov 8 12:31:18 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 8 Nov 2017 19:31:18 +0200 Subject: [OpenSIPS-Users] Introducing OpenSIPS 2.4 In-Reply-To: References: <972d9f71-89d4-89f1-9183-f7fc0f1c506a@opensips.org> Message-ID: <703b5e19-8d2d-84a9-9dc3-a8c576d67e0e@opensips.org> Hi Nuno, On the Asterisk part, the plan is to do exactly what we already have for FreeSWITCH (see https://blog.opensips.org/2017/03/01/freeswitch-driven-routing-in-opensips-2-3/) In terms of clustering media relays, is about the ability to share the state (enabled/disabled) between all the cluster nodes using the media relays. Optionally, we are looking in adding the ability to balance the traffic between the relays, in a cluster-level aware (all the nodes in the cluster will share the information on the load of the media relays ) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/08/2017 12:17 PM, Nuno Ferreira wrote: > Hi Bogdan, > > Do you have further details to share about the "*clustered media > relays*" and "*Asterisk flavored* Load-Balancing" features? > > Thanks, > > Nuno Ferreira > > On Wed, Nov 1, 2017 at 5:16 PM, Bogdan-Andrei Iancu > > wrote: > > One more year, one more evolution cycle, one more OpenSIPS major > release. So let me introduce you the upcoming OpenSIPS 2.4 . > > For the OpenSIPS 2.4 release we decided to focus on the > */clustering abilities/*. Today’s VoIP world is getting more and > more dynamic, services are moving into Clouds and more and more > flexibility is needed for the application to fully exploit such > environments. But let’s pin point the main reasons for going for a > clustered approach : > > * scaling up with the processing/traffic load > * geographical distribution > * redundancy and High-Availability > > For the OpenSIPS 2.4 we laid down a roadmap that addresses the > clustering both from the clustering engine itself (the underlayer) > and from the functionalities that will perform on top of the > clustering layer, to share data and state. > > With OpenSIPS 2.4, it will never be easier to built a consistent > and powerful clustered solution: > > * *clustering engine* – enhances the capabilities of controlling > the cluster topology, like re-routing for bypassing broken > links, dynamic joining of new nodes, support for multiple > capabilities per node, data syncing between nodes and many more; > * *distributed user location* – this is a very complex topic as > it exceeds the simple concept of data sharing. By the nature > of the data (the user registrations), you may have different > constraints on how data is roaming in a cluster – > registrations may be tied to a node due NAT or TCP > constraints. Even more, a sharding aspect must be addressed > when looking at distributing the pinging effort across the > cluster. So, multiple solutions are viable here, depending on > what is to be achieved (scaling, redundancy) and what are the > network constraints – see a detailed presentation > > of the available solutions; > * *distributed presence server* – quite similar (but less > complex) as the distributed user location, a distributed > presence server provides a consistent, but distributed way of > sharing presence information – SIP entities may publish data > via different nodes in the SIP cluster, while the subscribers > may fetch presence data via multiple various nodes. Two > approaches are under work : (a) a cluster built around a noSQL > DB based as primary data storage and (b) a cluster exclusively > relying on OpenSIPS for data sharing; > * *anycast support* – to be able to build a fully-flavored > anycast support (addressing both redundancy and balancing) > requires OpenSIPS to replicate/share transaction state across > the nodes in the cluster (nodes sharing the same anycast IP). > Depending on the nature of the replication (full transaction > versus transaction meta-data) a full anycast and light anycast > support will be available – here is a detailed description on > the anycast > > support; > * *clustered media relays* – as OpenSIPS has the ability to work > together with several flavors of media relays (such as > RTPproxy, RTPEngine, MediaProxy), the clustering support will > help OpenSIPS do distributed load-balancing over the relays – > even if a relay is used by multiple nodes in the cluster, all > the nodes will share information on the load on the relay, to > avoid overloading or idle time; > * *distributed call center* – an agent is able to register with > multiple queues on different nodes on a cluster. Still, all > the queues do share the status / availability of the agent and > its statistics for call distribution; > * *custom clustering* – the OpenSIPS clustering underlayer > provides at script level the ability to broadcast (in the > cloud) or send to a given node a custom message/action (with > replying possibility) – this is a very flexible and powerful > way to build your custom distributed functionality directly at > script level. > > And because we started on the integration path with OpenSIPS 2.3, > and because we did it well, we decided to push forward on this > path with the 2.4 version as well: > > * more *Homer integration *to be able to report TCP statistics, > DB events and media relay events via HEP; > * *SIPREC integration* for standard call recording. The new > SIPREC module > > provides a standard and transparent (for the call parties) way > to do call recording against an external recorder like Oreka > provided by Orecx > ; > * more *FreeSWITCH integration* in order to capture the > call-events (DTMFs, call status) from FreeSWITCH into OpenSIPS > script or for being able to control a FreeSWITCH call from > OpenSIPS script via ESL > * *Asterisk flavored* Load-Balancing for a more realistic and > accurate traffic balancing over Asterisk clusters (as the load > information is fetched in realtime from Asterisk); > > ------------------------------------------------------------------------ > > The timeline for OpenSIPS 2.4 is: > > * Beta Release – 12-16 March 2018 > * Stable Release – 23-27 April 2018 > * General Availability – 1st of May 2018, during OpenSIPS Summit > 2018 > > To talk more about the features of this new release, a public > audio conference will be > available on 21st of November 2017, 4 pm GMT > , thanks to > the kind sponsorship of UberConference > . Anyone is welcome to join to > find out more details or to ask questions about OpenSIPS > 2.4 . > > This is a public and open conference, so no registration is > needed, but if you want to announce your intention to participate, > please let us know here: > > http://blog.opensips.org/2017/11/01/introducing-opensips-2-4/ > > > > Best regards, > > -- > 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 > > > > > > *Confidentiality Notice: The information contained in this e-mail and any > attachments may be confidential. If you are not an intended recipient, you > are hereby notified that any dissemination, distribution or copying of > this > e-mail is strictly prohibited. If you have received this e-mail in error, > please notify the sender and permanently delete the e-mail and any > attachments immediately. You should not retain, copy or use this e-mail or > any attachment for any purpose, nor disclose all or any part of the > contents to any other person. Thank you.* > > > _______________________________________________ > 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 Thu Nov 9 03:12:02 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Thu, 9 Nov 2017 10:12:02 +0200 Subject: [OpenSIPS-Users] replace-origin flag error In-Reply-To: <1510160641879-0.post@n2.nabble.com> References: <1510093598685-0.post@n2.nabble.com> <5e44a837-9055-b9aa-00d9-21b32ae05e2c@opensips.org> <1510149282639-0.post@n2.nabble.com> <56163b8e-72ef-916a-e56d-a7eb94e85034@opensips.org> <1510160641879-0.post@n2.nabble.com> Message-ID: <7247d3fd-b266-f109-0cf2-51e64c10c2e9@opensips.org> Yes, that should work too! Nevertheless I will fix the code as well, because it's not such a big change. Thanks for reporting it! Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/08/2017 07:04 PM, Edwin wrote: > Hi Răzvan, > > I'm using version 2.3.2-1 from apt.opensips.org repository, on a Debian > stretch release. > > Is there a workaround for now, something like: > > if(t_check_status("403|404|480|484|486|487") > rtpengine_manage("ICE=remove"); > } > else { > rtpengine_manage("replace-origin ICE=remove"); > } > > > > -- > 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 Thu Nov 9 03:18:39 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Thu, 9 Nov 2017 10:18:39 +0200 Subject: [OpenSIPS-Users] avp_db_query in opensips.cfg In-Reply-To: References: <45756329-a60d-7c65-f85f-65e1b4b05439@opensips.org> Message-ID: <8ae0725d-cb2a-739a-8ef2-b10c69ccbda1@opensips.org> Yes, that's probably it. I've just pushed a fix on all supported branches that should sort this out. Please test it and let me know if there are any other issues. Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/08/2017 06:04 PM, Wilhelm Lundgren wrote: > Ahh, thats probably it: > > Working version: > version: opensips 2.2.1 (arm/linux) > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > MAX_URI_SIZE 1024, BUF_SIZE 65535 > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. > git revision: 68ace2e > main.c compiled on 14:42:20 Sep 14 2016 with gcc 4.6 > > > None working version: > version: opensips 2.2.5 (i386/linux) > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > MAX_URI_SIZE 1024, BUF_SIZE 65535 > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. > git revision: 4d0cc20 > main.c compiled on 09:52:01 Nov  8 2017 with gcc 4.7 > > > > > On 8 November 2017 at 15:14, Răzvan Crainea > wrote: > > Hi, Wilhelm! > > I don't think the problem is the linux version, but OpenSIPS > version. Can you check if the two installs are using different > versions of OpenSIPS (try ./opensips -V). > Lately there was a commit which might trigger this error and I > want to double check if that's the case. Can you send the version > of OpenSIPS you are using? > > Best regards, > > Răzvan Crainea > OpenSIPS Developer > www.opensips-solutions.com > > On 11/08/2017 02:09 PM, Wilhelm Lundgren wrote: >> Hi, >> >> Im running opensips 2.2 (built on the platform) on an arm >> platform (wheezy). In the config i use several avp_db_query >> against the sqlite db. This works fine. >> >> However, when using the same config on the same linux (but for >> amd64, or i386), using the same opensips version,  i cant start >> opensips, i get: >> >> ERROR:avpops:avpops_db_bind: driver for DB URL [default] does not >> support raw queries! when starting opensips. >> >> Removing all the avp_db_query from the cfg will make opensips >> start ok, but i need those queries. >> >> Why does those queries work on arm and not on x86? I dont >> understand why there should be any difference. >> >> /Wilhelm >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Thu Nov 9 03:29:10 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Thu, 9 Nov 2017 10:29:10 +0200 Subject: [OpenSIPS-Users] replace-origin flag error In-Reply-To: <7247d3fd-b266-f109-0cf2-51e64c10c2e9@opensips.org> References: <1510093598685-0.post@n2.nabble.com> <5e44a837-9055-b9aa-00d9-21b32ae05e2c@opensips.org> <1510149282639-0.post@n2.nabble.com> <56163b8e-72ef-916a-e56d-a7eb94e85034@opensips.org> <1510160641879-0.post@n2.nabble.com> <7247d3fd-b266-f109-0cf2-51e64c10c2e9@opensips.org> Message-ID: Done, fixed on all supported branches. Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/09/2017 10:12 AM, Răzvan Crainea wrote: > Yes, that should work too! Nevertheless I will fix the code as well, > because it's not such a big change. > > Thanks for reporting it! > > Best regards, > > Răzvan Crainea > OpenSIPS Developer > www.opensips-solutions.com > > On 11/08/2017 07:04 PM, Edwin wrote: >> Hi Răzvan, >> >> I'm using version 2.3.2-1 from apt.opensips.org repository, on a Debian >> stretch release. >> >> Is there a workaround for now, something like: >> >> if(t_check_status("403|404|480|484|486|487") >> rtpengine_manage("ICE=remove"); >> } >> else { >> rtpengine_manage("replace-origin ICE=remove"); >> } >> >> >> >> -- >> 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 > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From schoolhouse at a2es.co.uk Thu Nov 9 04:15:32 2017 From: schoolhouse at a2es.co.uk (Schoolhouse Filing) Date: Thu, 9 Nov 2017 09:15:32 +0000 Subject: [OpenSIPS-Users] Tearing down the RTPProxy session for the current call In-Reply-To: <6923b55a-26cf-e8b3-37ab-4d908b294bd2@opensips.org> References: <75c75c8d-c0d8-d52c-5815-1edf2c00200a@topgreen.co.uk> <6923b55a-26cf-e8b3-37ab-4d908b294bd2@opensips.org> Message-ID: Hi Razvan, Thankyou for clarifying this, I was a little confused because the Opensips generated scripts that use rtpproxy for NAT do not call the rtpproxy_unforce() function. I will add a call to trpproxy_unforce() in my route that handles BYEs and in the missed call (failure) routes. Many Thanks, Adrian. On 08/11/17 10:47, Răzvan Crainea wrote: > Hi, Adrian! > > Yes, you should be calling rtpproxy_unforce() when a BYE is received. > You are right, if you do that, you won't be seeing too many "RTP > sessions ended due to media timeout". > > Best regardsm > Răzvan Crainea > OpenSIPS Developer > www.opensips-solutions.com > On 11/07/2017 08:59 PM, Adrian Fretwell wrote: >> >> Hello, >> >> I was just looking at some statistics from my rtpproxy today and >> noticed that "RTP sessions created", "RTP sessions destroyed" AND >> "RTP sessions ended due to media timeout" all showed the same number. >> >> Although I am aware that the function rtpproxy_unforce() is >> available, in example routing scripts, I have never seen it used. >> >> Should I be calling rtpproxy_unforce() in my routing script at the >> end of a call? >> >> My assumption is that if I do, then the "RTP sessions ended due to >> media timeout" will be much lower. >> >> As always any advice will be gratefully received. >> >> 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 diptesh.patel at ecosmob.com Thu Nov 9 08:40:56 2017 From: diptesh.patel at ecosmob.com (Dipteshkumar Patel) Date: Thu, 9 Nov 2017 19:10:56 +0530 Subject: [OpenSIPS-Users] radius authentication message failed with ERROR Message-ID: Hello sir, I am using opensips 2.3.1 and i am using freeradius-server 2.2.0 for accounting When i am stress testing with 100cps and 1500 cc then the following error produced by opensips ERROR:aaa_radius:resume_send_acct: radius authentication message failed with ERROR Nov 9 13:19:14 OS /usr/local/sbin/opensips[12910]: rc_send_server: recvfrom: XXX.XXX.X.XX:1813: reply is too short Nov 9 13:19:14 OS /usr/local/sbin/opensips[12910]: ERROR:acc:acc_aaa_request: Radius accounting request failed for status: 'Start' Call-Id: '97844-1608 at XXX.XXX.XXX.XXX' what is the cause for this ?? *Diptesh Patel* Jr. Software Developer Ecosmob Technologies Ltd Ahmedabad Mo:*+919898962659* -------------- next part -------------- An HTML attachment was scrubbed... URL: From eahaselhoff at gmail.com Thu Nov 9 09:33:26 2017 From: eahaselhoff at gmail.com (Edwin) Date: Thu, 9 Nov 2017 07:33:26 -0700 (MST) Subject: [OpenSIPS-Users] replace-origin flag error In-Reply-To: References: <1510093598685-0.post@n2.nabble.com> <5e44a837-9055-b9aa-00d9-21b32ae05e2c@opensips.org> <1510149282639-0.post@n2.nabble.com> <56163b8e-72ef-916a-e56d-a7eb94e85034@opensips.org> <1510160641879-0.post@n2.nabble.com> <7247d3fd-b266-f109-0cf2-51e64c10c2e9@opensips.org> Message-ID: <1510238006163-0.post@n2.nabble.com> Hi Răzvan, Great job, thanks for the quick fix. How do I install the update from Debian repository, wil there be a 2.3.2-2 update? thx, Edwin -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html From mas_66 at tiscali.co.uk Thu Nov 9 14:03:12 2017 From: mas_66 at tiscali.co.uk (mas_66) Date: Thu, 09 Nov 2017 19:03:12 +0000 Subject: [OpenSIPS-Users] Opensips v2.3.2 Message-ID: Hi RazvanThanks for the reply.Nothing in db, and i am using db_mode 1So i presume the call data is not being fed to sql. Any good tips on what logs i can look at to see at which level its falling?ThanksMas Sent from Samsung Mobile on O2 -------- Original message --------From: Răzvan Crainea Date: 08/11/2017 10:23 (GMT+00:00) To: users at lists.opensips.org Subject: Re: [OpenSIPS-Users] Opensips v2.3.2 Hi, Mas! Are you seeing the dialogs in the database? Also, what db_mode[1] are you using? [1] http://www.opensips.org/html/docs/modules/2.4.x/dialog#idp5562032 Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/07/2017 09:31 PM, mas_66--- via Users wrote: > Hello Users > I have an issue with seeing calls within control panel either in Dialog(current > calls) or anything in SIP trace > Any experience with what may be causing this before i try to reinvent the > wheel? > Thanks in advance > Mas > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Fri Nov 10 05:21:47 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Fri, 10 Nov 2017 12:21:47 +0200 Subject: [OpenSIPS-Users] Opensips v2.3.2 In-Reply-To: References: Message-ID: You should check the logs to see if there are any problems with the dialog module while inserting the dialogs in the DB. Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/09/2017 09:03 PM, mas_66 via Users wrote: > > Hi Razvan > Thanks for the reply. > Nothing in db, and i am using db_mode 1 > So i presume the call data is not being fed to sql. > Any good tips on what logs i can look at to see at which level its > falling? > Thanks > Mas > > > Sent from Samsung Mobile on O2 > > -------- Original message -------- > From: Răzvan Crainea > Date: 08/11/2017 10:23 (GMT+00:00) > To: users at lists.opensips.org > Subject: Re: [OpenSIPS-Users] Opensips v2.3.2 > > Hi, Mas! > > Are you seeing the dialogs in the database? Also, what db_mode[1] are > you using? > > [1] http://www.opensips.org/html/docs/modules/2.4.x/dialog#idp5562032 > > Best regards, > > Răzvan Crainea > OpenSIPS Developer > www.opensips-solutions.com > > On 11/07/2017 09:31 PM, mas_66--- via Users wrote: > > Hello Users > > I have an issue with seeing calls within control panel either in > Dialog(current > > calls) or anything in SIP trace > > Any experience with what may be causing this before i try to > reinvent the > > wheel? > > Thanks in advance > > Mas > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From goup2010 at gmail.com Fri Nov 10 09:21:05 2017 From: goup2010 at gmail.com (Dragomir Haralambiev) Date: Fri, 10 Nov 2017 16:21:05 +0200 Subject: [OpenSIPS-Users] call_center module in Opensips 2.3 In-Reply-To: References: Message-ID: Hi, I try your idea to loop the INVITE back to Opensips and do a lookup(). I pit into "location" field from cc_agents table this one "sip:localnumber at Opensips_IP:Opensips_port". Call is connected but I receive follow log message: ERROR:b2b_entities:b2b_prescript_f: No dialog found, callid= [B2B.337.5360955.1510320840], method=OPTIONS When call is finished record from dialog tables is not deleted. Where is problem? Regards, Dragomir 2017-11-06 11:37 GMT+02:00 Răzvan Crainea : > Hi, Hristo! > > 1. You are right, b2b_entities should be loaded as a dependency for > b2b_logic. You may argue that it should also be listed as a dependency to > call_center as well, but indirectly. > > 2. Yes, you should populate the tables with the flows you want to support > and the agents that are going to use the call center. > > 3. There is currently no way to directly set the location of the agent > dynamically. However, we can do some work-arounds, for example always > advertise yourself as the location of the user, and for each call you can > loop the INVITE back to you and do a lookup(). Or simply use an external > registrar service. > > Best regards, > > Răzvan Crainea > OpenSIPS Developerwww.opensips-solutions.com > > On 11/05/2017 11:06 PM, Hristo Donev wrote: > > Hello All, > > I try to implement call_center module in Opensips 2.3. > Here are my questions? > > 1. From the documentation 1.3.1: > "b2b_logic" and "database" must be loaded before callcenter module. > > I found out that "b2b_entities" should be loaded as well, otherwise the > opensips does not start. > > Is this my mistake or a documentation error? > > 2. Opensips can not start if the tables "cc_agents" and "cc_flow" are > empty. > Why it cannot start? Should they be populated with some data? > > 3. Table "cc_agents", field "location" should point to the SIP agent > location. > If the agent's IP and port are changed dynamically - > how to copy automatically "contact" field from "location" table to > "cc_agents" field "location"? > > I would be grateful to receive a response for my questions. > > Thanks in advance, > Hristo Donev > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From sobomax at sippysoft.com Sat Nov 11 00:01:35 2017 From: sobomax at sippysoft.com (Maxim Sobolev) Date: Fri, 10 Nov 2017 21:01:35 -0800 Subject: [OpenSIPS-Users] Introducing OpenSIPS 2.4 In-Reply-To: <703b5e19-8d2d-84a9-9dc3-a8c576d67e0e@opensips.org> References: <972d9f71-89d4-89f1-9183-f7fc0f1c506a@opensips.org> <703b5e19-8d2d-84a9-9dc3-a8c576d67e0e@opensips.org> Message-ID: Bogdan, with regards to the media relays clustering what's the advantage of sharing that load info between each signalling node versus having each node tracking this independently? In my view the latter could be more reliable and much less complicated construct. The only disadvantage is that you'd get more command load on relays, however at least with the RTPproxy pulling load stats is very lightweight operation so even with tens of signalling nodes pulling single media relay every 1-10 seconds it won't cause any noticeable performance degradation on the relay. On the flip side, each signalling node would get accurate view from its vantage PoV, so in the case of geographically distributed system when signalling node can only see subset of all media nodes it would still be able to make proper decisions. This is the approach we use in the rtp_cluster and it works pretty well with cluster size of up to 5 signalling and 10 RTP handling nodes, 40-50K media sessions in total. It can also give you accurate RTT information, so your signalling node can not only factor in the load but also proximity or each and every media relay. As far as the load tracking is concerned, I think the approach to implement "b2b-driven routing" using API that is specific to each particular b2b is somewhat wasteful and is not very future-proof. What we would like to see instead, is for opensips to publish some kind of API (preferably SIP-based, using OPTIONS or SUBSCRIBE/NOTIFY mechanism) to pull this information out and let each b2b vendor to implement proper hooks. Then it can go as far as making this info some king of RFC. Anyhow, just my $0.02c. Not volunteering to do opensips side (ENOTIME), but if opensips project comes up with the reasonable b2bua-agnostic load query API to use we might look at implementing it in the sippy [py/go]-B2BUAs. -Max On Wed, Nov 8, 2017 at 9:31 AM, Bogdan-Andrei Iancu wrote: > Hi Nuno, > > On the Asterisk part, the plan is to do exactly what we already have for > FreeSWITCH (see https://blog.opensips.org/2017/03/01/freeswitch-driven- > routing-in-opensips-2-3/) > > In terms of clustering media relays, is about the ability to share the > state (enabled/disabled) between all the cluster nodes using the media > relays. Optionally, we are looking in adding the ability to balance the > traffic between the relays, in a cluster-level aware (all the nodes in the > cluster will share the information on the load of the media relays ) > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From wilhelm.lundgren at gmail.com Sat Nov 11 05:32:10 2017 From: wilhelm.lundgren at gmail.com (Wilhelm Lundgren) Date: Sat, 11 Nov 2017 11:32:10 +0100 Subject: [OpenSIPS-Users] avp_db_query in opensips.cfg In-Reply-To: <8ae0725d-cb2a-739a-8ef2-b10c69ccbda1@opensips.org> References: <45756329-a60d-7c65-f85f-65e1b4b05439@opensips.org> <8ae0725d-cb2a-739a-8ef2-b10c69ccbda1@opensips.org> Message-ID: I have tested it on 2.2 branch and everything works fine! Thanks! Wilhelm On 9 November 2017 at 09:18, Răzvan Crainea wrote: > Yes, that's probably it. > I've just pushed a fix on all supported branches that should sort this > out. Please test it and let me know if there are any other issues. > > Best regards, > > Răzvan Crainea > OpenSIPS Developerwww.opensips-solutions.com > > On 11/08/2017 06:04 PM, Wilhelm Lundgren wrote: > > Ahh, thats probably it: > > Working version: > version: opensips 2.2.1 (arm/linux) > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > MAX_URI_SIZE 1024, BUF_SIZE 65535 > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. > git revision: 68ace2e > main.c compiled on 14:42:20 Sep 14 2016 with gcc 4.6 > > > None working version: > version: opensips 2.2.5 (i386/linux) > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > MAX_URI_SIZE 1024, BUF_SIZE 65535 > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. > git revision: 4d0cc20 > main.c compiled on 09:52:01 Nov 8 2017 with gcc 4.7 > > > > > On 8 November 2017 at 15:14, Răzvan Crainea wrote: > >> Hi, Wilhelm! >> >> I don't think the problem is the linux version, but OpenSIPS version. Can >> you check if the two installs are using different versions of OpenSIPS (try >> ./opensips -V). >> Lately there was a commit which might trigger this error and I want to >> double check if that's the case. Can you send the version of OpenSIPS you >> are using? >> >> Best regards, >> >> Răzvan Crainea >> OpenSIPS Developerwww.opensips-solutions.com >> >> On 11/08/2017 02:09 PM, Wilhelm Lundgren wrote: >> >> Hi, >> >> Im running opensips 2.2 (built on the platform) on an arm platform >> (wheezy). In the config i use several avp_db_query against the sqlite db. >> This works fine. >> >> However, when using the same config on the same linux (but for amd64, or >> i386), using the same opensips version, i cant start opensips, i get: >> >> ERROR:avpops:avpops_db_bind: driver for DB URL [default] does not support >> raw queries! when starting opensips. >> >> Removing all the avp_db_query from the cfg will make opensips start ok, >> but i need those queries. >> >> Why does those queries work on arm and not on x86? I dont understand why >> there should be any difference. >> >> /Wilhelm >> >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From social at bohboh.info Sun Nov 12 17:20:29 2017 From: social at bohboh.info (Social Boh) Date: Sun, 12 Nov 2017 17:20:29 -0500 Subject: [OpenSIPS-Users] Sending HEP Packets In-Reply-To: <3e4a2367-fcd0-b47a-5e41-348b3d729f74@opensips.org> References: <8d9354ca-b4cd-8cf1-da2b-b0c504179f55@bohboh.info> <3e4a2367-fcd0-b47a-5e41-348b3d729f74@opensips.org> Message-ID: In my case don't work... My configuration (OpenSIPs 2.3.2): listen=hep_udp:127.0.0.1:6060 #### PROTO_HEP Module loadmodule "proto_hep.so" modparam("proto_hep", "hep_id", "[homer] 1.2.3.4:9060; version=2") modparam("proto_hep", "homer5_on", 1) modparam("proto_hep", "homer5_delim", "#") modparam("proto_hep", "hep_capture_id", 12345) #### SIPTRACE Module loadmodule "siptrace.so" modparam("siptrace", "trace_on", 1) modparam("siptrace", "trace_id", "[traceid]uri=hep:homer") in the script for initial INVITE: sip_trace("traceid","M","sip"); the Error: ERROR:proto_hep:hep_udp_send: sendto(sock,0x7fb60874b710,1101,0,0x7fb608749bd0,16): Invalid argument(22) CRITICAL:proto_hep:hep_udp_send: invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net ERROR:proto_hep:msg_send: send() to 1.2.3.4:9060 for proto hep_udp/8 failed ERROR:proto_hep:send_hep_message: Cannot send hep message! ERROR:siptrace:save_siptrace: Failed to duplicate with hep to <1.2.3.4:9060> 1.2.3.4 is the Homer SIP Capture Server remote IP Any hint? --- I'm SoCIaL, MayBe El 06/11/2017 a las 03:59, Răzvan Crainea escribió: > Hello! > > Shortly, you need to follow this logic: > 1. Define a listener: > listen=hep_udp:127.0.0.1:6060 > > 2. Define a HEP destination to capture the traffic to[1]. > modparam("proto_hep", "hep_id", "[homer_id]1.2.3.4:9060 > ; version=2") > > 3. Next, define a tracing ID - an idea that describes where a specific > package should be sent[2] > modparam("siptrace", "trace_id", "[to_homer_tid]uri=hep:homer_id") > > 4. Start tracing the data you want by calling sip_trace() with the > destination provisioned earlier and the scope you want: > sip_trace("to_homer_tid", "m", "sip"); # this traces only the SIP message > > Hope it is clear now. > > [1] > http://www.opensips.org/html/docs/modules/2.4.x/proto_hep.html#idp232096 > [2] > http://www.opensips.org/html/docs/modules/2.4.x/siptrace.html#idp5904384 > [3] > http://www.opensips.org/html/docs/modules/2.4.x/siptrace.html#idp5914144 > > Best regards, > Răzvan Crainea > OpenSIPS Developer > www.opensips-solutions.com > On 11/05/2017 09:56 AM, Karsten Horsmann wrote: >> Hi, >> >> you need also to mark packets you are interested to be sipcaptured in >> your route. >> >> Kind regards >> Karsten Horsmann >> >> >> Am 03.11.2017 8:32 nachm. schrieb "Social Boh" > >: >> >> Hello, >> >> I think I don't completely understood the logic behind proto_hep >> and siptrace modules on OpenSIPs 2.3 >> >> I'd like send SIP traffic from OpenSIPs to a external Homer SIP >> Capture Server (with Kamailio) so I know I have to configure >> proto_hep and siptrace modules. >> >> I think the logic is: Send SIP traffic from siptrace to hep listener: >> >> modparam("siptrace", "trace_id", "[tid]uri=sip:127.0.0.1:6060 >> ") >> >> listen=hep_udp:127.0.0.1:6060 >> >> then from the proto_hep module send the HEP packets to Node Capture: >> >> modparam("proto_hep", "hep_id", "[homer]1.2.3.4:9060 >> ; version=2") >> >> This configuration not works. >> >> This document not helping me very much: >> >> http://www.opensips.org/Documentation/Tutorials-Tracing >> >> >> Any hint, please? >> >> Thank you >> >> -- >> --- >> I'm SoCIaL, MayBe >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Mon Nov 13 03:55:00 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Mon, 13 Nov 2017 10:55:00 +0200 Subject: [OpenSIPS-Users] Sending HEP Packets In-Reply-To: References: <8d9354ca-b4cd-8cf1-da2b-b0c504179f55@bohboh.info> <3e4a2367-fcd0-b47a-5e41-348b3d729f74@opensips.org> Message-ID: <5aefcd46-6881-b398-d115-16f4dd6d96c3@opensips.org> Hello! The error is pretty self-explanatory: "one possible reason is the server is bound to localhost and attempts to send to the net" - you have the hep bound to localhost and you are trying to send messages to an external host - from 127.0.0.1 you can only communicate with 127.0.0.1. If you want to send to an external host, you should use a different interface. Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/13/2017 12:20 AM, Social Boh wrote: > > In my case don't work... My configuration (OpenSIPs 2.3.2): > > listen=hep_udp:127.0.0.1:6060 > > #### PROTO_HEP Module > loadmodule "proto_hep.so" > modparam("proto_hep", "hep_id", "[homer] 1.2.3.4:9060; version=2") > modparam("proto_hep", "homer5_on", 1) > modparam("proto_hep", "homer5_delim", "#") > modparam("proto_hep", "hep_capture_id", 12345) > > #### SIPTRACE Module > loadmodule "siptrace.so" > modparam("siptrace", "trace_on", 1) > modparam("siptrace", "trace_id", "[traceid]uri=hep:homer") > > in the script for initial INVITE: > > sip_trace("traceid","M","sip"); > > the Error: > > ERROR:proto_hep:hep_udp_send: > sendto(sock,0x7fb60874b710,1101,0,0x7fb608749bd0,16): Invalid argument(22) > CRITICAL:proto_hep:hep_udp_send: invalid sendtoparameters#012one > possible reason is the server is bound to localhost and#012attempts to > send to the net > ERROR:proto_hep:msg_send: send() to 1.2.3.4:9060 for proto hep_udp/8 > failed > ERROR:proto_hep:send_hep_message: Cannot send hep message! > ERROR:siptrace:save_siptrace: Failed to duplicate with hep to > <1.2.3.4:9060> > > 1.2.3.4 is the Homer SIP Capture Server remote IP > > Any hint? > > --- > I'm SoCIaL, MayBe > El 06/11/2017 a las 03:59, Răzvan Crainea escribió: >> Hello! >> >> Shortly, you need to follow this logic: >> 1. Define a listener: >> listen=hep_udp:127.0.0.1:6060 >> >> 2. Define a HEP destination to capture the traffic to[1]. >> modparam("proto_hep", "hep_id", "[homer_id]1.2.3.4:9060 >> ; version=2") >> >> 3. Next, define a tracing ID - an idea that describes where a >> specific package should be sent[2] >> modparam("siptrace", "trace_id", "[to_homer_tid]uri=hep:homer_id") >> >> 4. Start tracing the data you want by calling sip_trace() with the >> destination provisioned earlier and the scope you want: >> sip_trace("to_homer_tid", "m", "sip"); # this traces only the SIP message >> >> Hope it is clear now. >> >> [1] >> http://www.opensips.org/html/docs/modules/2.4.x/proto_hep.html#idp232096 >> [2] >> http://www.opensips.org/html/docs/modules/2.4.x/siptrace.html#idp5904384 >> [3] >> http://www.opensips.org/html/docs/modules/2.4.x/siptrace.html#idp5914144 >> >> Best regards, >> Răzvan Crainea >> OpenSIPS Developer >> www.opensips-solutions.com >> On 11/05/2017 09:56 AM, Karsten Horsmann wrote: >>> Hi, >>> >>> you need also to mark packets you are interested to be sipcaptured >>> in your route. >>> >>> Kind regards >>> Karsten Horsmann >>> >>> >>> Am 03.11.2017 8:32 nachm. schrieb "Social Boh" >> >: >>> >>> Hello, >>> >>> I think I don't completely understood the logic behind proto_hep >>> and siptrace modules on OpenSIPs 2.3 >>> >>> I'd like send SIP traffic from OpenSIPs to a external Homer SIP >>> Capture Server (with Kamailio) so I know I have to configure >>> proto_hep and siptrace modules. >>> >>> I think the logic is: Send SIP traffic from siptrace to hep >>> listener: >>> >>> modparam("siptrace", "trace_id", "[tid]uri=sip:127.0.0.1:6060 >>> ") >>> >>> listen=hep_udp:127.0.0.1:6060 >>> >>> then from the proto_hep module send the HEP packets to Node Capture: >>> >>> modparam("proto_hep", "hep_id", "[homer]1.2.3.4:9060 >>> ; version=2") >>> >>> This configuration not works. >>> >>> This document not helping me very much: >>> >>> http://www.opensips.org/Documentation/Tutorials-Tracing >>> >>> >>> Any hint, please? >>> >>> Thank you >>> >>> -- >>> --- >>> I'm SoCIaL, MayBe >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From social at bohboh.info Mon Nov 13 11:19:39 2017 From: social at bohboh.info (Social Boh) Date: Mon, 13 Nov 2017 11:19:39 -0500 Subject: [OpenSIPS-Users] OpenSIPs Control Panel Homer Menu In-Reply-To: <32fe0423-10e7-6f23-2d01-3ebeb4f86b98@bohboh.info> References: <31a910a6-10e7-6aa8-cb53-31d35a6e64ba@bohboh.info> <667cccd5-ae01-9780-9c0d-eb34c39355aa@bohboh.info> <01993f73-4c11-de06-c8fe-3e5b93c62580@opensips.org> <32fe0423-10e7-6f23-2d01-3ebeb4f86b98@bohboh.info> Message-ID: <13d3238b-79c4-5904-a8b9-f00cf68f145c@bohboh.info> Hello, if I access OpenSIPs Control Panel over https, with a Let's Encrypt Certificate, I can't see Homer page (blank page). The only error/notice on the ssl_error.log is: PHP Notice:  A session had already been started - ignoring session_start() in /var/www/html/opensips-cp/config/session.inc.php on line 23, referer: https://opensipscontrolpanel.org/cp/tools/system/homer/homer.php If I use OpenSIPs Control Panel over httpd all working fine. Regards --- I'm SoCIaL, MayBe El 03/11/2017 a las 10:22, Social Boh escribió: > > The error was related with this semicolon (before the last parenthesis): > > define('EXTERNAL_AUTH_URI', > "http://5.6.7.8/cp/tools/system/homer/auth.php"*;*); > > Thank you. > > Regards > > --- > I'm SoCIaL, MayBe > El 03/11/2017 a las 07:15, Răzvan Crainea escribió: >> This time there seems to be an issue on the homer side. Can you check >> the logs to see if homer is saying anything useful? >> >> Best regards, >> >> Răzvan Crainea >> OpenSIPS Developer >> www.opensips-solutions.com >> >> On 11/03/2017 02:06 PM, Social Boh wrote: >>> Hello, >>> >>> Now I have this error (homer side): >>> >>> GET >>> /api/v1/redirect?externalid=rx88kxQUs1DNT6es2ybwBvhevecJwjnY&url=http%3A%2F%2F1.2.3.4.0%2F >>> HTTP/1.1. >>> Host: 104.236.105.0. >>> User-Agent: Mozilla/5.0 (Windows NT 6.1; Win64; x64; rv:56.0) >>> Gecko/20100101 Firefox/56.0. >>> Accept: >>> text/html,application/xhtml+xml,application/xml;q=0.9,*/*;q=0.8. >>> Accept-Language: es-ES,es;q=0.8,en-US;q=0.5,en;q=0.3. >>> Accept-Encoding: gzip, deflate. >>> Referer: http://5.6.7.8/cp/tools/system/homer/homer.php. >>> Connection: keep-alive. >>> Upgrade-Insecure-Requests: 1. >>> >>> HTTP/1.0 500 Internal Server Error. >>> Date: Fri, 03 Nov 2017 11:47:36 GMT. >>> Server: Apache/2.4.6 (CentOS) OpenSSL/1.0.2k-fips PHP/5.4.16. >>> X-Powered-By: PHP/5.4.16. >>> Content-Length: 0. >>> Connection: close. >>> Content-Type: text/html; charset=UTF-8. >>> >>> My homer configuration is: >>> >>> define('AUTHENTICATION',"External"); >>> define('EXTERNAL_AUTH_URI', >>> "http://5.6.7.8/cp/tools/system/homer/auth.php";); >>> define('EXTERNAL_AUTH_METHOD', "GET"); >>> define('EXTERNAL_AUTH_PARAM', "param=KEY"); >>> define('EXTERNAL_AUTH_REQUEST_TYPE', "get"); >>> define('EXTERNAL_AUTH_REQUEST_KEY', "externalid"); >>> define('EXTERNAL_AUTH_POSITIVE_REPLY', "200"); >>> define('EXTERNAL_AUTH_REPLY_USER_INFO', "json"); >>> >>> 1.2.3.4 is the Homer IP and 5.6.7.8 is the OpenSIPs Control Panel IP >>> >>> Thank you >>> >>> Regards >>> >>> --- >>> I'm SoCIaL, MayBe >>> >>> El 03/11/2017 a las 04:47, Răzvan Crainea escribió: >>>> Hello! >>>> >>>> You should install the php apc library. On Ubuntu this is php-apcu. >>>> >>>> Best regards, >>>> >>>> Răzvan Crainea >>>> OpenSIPS Developer >>>> www.opensips-solutions.com >>>> >>>> On 11/02/2017 10:09 PM, Social Boh wrote: >>>>> Hello, >>>>> >>>>> each time I try to access Homer Menu on OpenSIPs Control Panel >>>>> 7.3.2, on the apache log: >>>>> >>>>> Call to undefined function apc_store() in >>>>> /var/www/html/opensips-cp/web/tools/system/homer/homer.php on line 5 >>>>> >>>>> Blank page >>>>> >>>>> OpenSIPs Homer Configuration: >>>>> >>>>> $homer_URL = "http://homer.mydomain.org"; >>>>> >>>>> $homer_auth_method = "get"; >>>>> >>>>> Homer OpenSIPs configuration (preference.php) >>>>> >>>>> define('AUTHENTICATION',"External"); >>>>> define('EXTERNAL_AUTH_URI', >>>>> "http://os.mydomain.org/cp/tools/system/homer/auth.php";); >>>>> define('EXTERNAL_AUTH_METHOD', "GET"); >>>>> define('EXTERNAL_AUTH_PARAM', "param=KEY"); >>>>> define('EXTERNAL_AUTH_REQUEST_TYPE', "get"); >>>>> define('EXTERNAL_AUTH_REQUEST_KEY', "externalid"); >>>>> define('EXTERNAL_AUTH_POSITIVE_REPLY', "200"); >>>>> define('EXTERNAL_AUTH_REPLY_USER_INFO', "json"); >>>>> >>>>> Any hint >>>>> >>>>> Thank you in advance. >>>>> >>>>> Regards >>>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From y.kirsanov at gmail.com Mon Nov 13 21:05:33 2017 From: y.kirsanov at gmail.com (Yury Kirsanov) Date: Tue, 14 Nov 2017 13:05:33 +1100 Subject: [OpenSIPS-Users] fix_nated_sdp and rtpproxy Message-ID: Hi, I have a problem with remote client who can't properly configure NAT and is sending private IP in all SDP and SIP packets. I've resolved issues with registration but can't resolve the issue with SDP. My topology is as following: ----- ------- ---------- -------- ----------- So, I need to simultaneously engage RTP Proxy to bridge external Internet and our local LAN and need to rewrite SDP body of original Client's packet with received IP address. If I'm doing fix_nated_sdp and then rtpproxy_offer - I'm getting doubled IPs in SDP message, like this: 100.11.100.200192.168.20.200 Is there any way to update SDP of client's packet and then command RTP Proxy to use updated IP? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: From diptesh.patel at ecosmob.com Tue Nov 14 09:47:47 2017 From: diptesh.patel at ecosmob.com (Dipteshkumar Patel) Date: Tue, 14 Nov 2017 20:17:47 +0530 Subject: [OpenSIPS-Users] Getting Wrong Time Stamp from $Tsm In-Reply-To: <70b4b6ab-f944-1397-0d34-e6928e34ed1f@opensips.org> References: <70b4b6ab-f944-1397-0d34-e6928e34ed1f@opensips.org> Message-ID: hello all, I got the option to get right time stamp both in seconds and microseconds and that is get_timestamp(sec_avp,usec_avp); which uses gettimeofday() system call internally. refer following link https://www.opensips.org/Documentation/Script-CoreFunctions-2-3 *Diptesh Patel* Jr. Software Developer Ecosmob Technologies Ltd Ahmedabad Mo:*+919898962659* On Wed, Nov 1, 2017 at 3:59 PM, Bogdan-Andrei Iancu wrote: > Hi Diptesh, > > Unfortunately the docs are a bit misleading - the supported format for the > $time() var is given by "man strftime" - and strftime has no support for > time units smaller than the second. This is why the %N is not evaluated. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 10/31/2017 03:32 PM, Dipteshkumar Patel wrote: > > Hello sir, > > Thanks for your prompt reply > > I used *$time(+%s%N)* but it uses only one format and give the result " > *+1509451841%N*" but i want to whole time stamp. > > This works fine from command line using *#time +%s%N.* > > > > > *Diptesh Patel* > Jr. Software Developer > Ecosmob Technologies Ltd > Ahmedabad > Mo:*+919898962659* > > On Tue, Oct 31, 2017 at 5:12 PM, Bogdan-Andrei Iancu > wrote: > >> Hi, >> >> Have you looked at the $time(format) script variable: >> >> http://www.opensips.org/Documentation/Script-CoreVar-2-3#toc87 >> >> You can take the all the time fields at once, and use a transformation to >> split them if you need the fields in separate variables. >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> On 10/31/2017 10:36 AM, Dipteshkumar Patel wrote: >> >> Dear sir, >> >> I am using Opensips version 2.3.1 >> >> I am using $Ts and $Tsm for time stamp for CDR but the microsecond time >> stamp is not same as system time stamp. To resolve the issue i am using >> exec() to get the system time but i takes so much time (nearly 5ms ) >> whenever so many calls concurrently running on server(We need exact time >> because it will affect the call duration). >> >> Can you please suggest me solution for the issue ? >> >> Thanks >> >> *Diptesh Patel* >> Jr. Software Developer >> Ecosmob Technologies Ltd >> Ahmedabad >> Mo:*+919898962659* >> >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From k.galinurov at gmail.com Tue Nov 14 10:49:45 2017 From: k.galinurov at gmail.com (Kirill Galinurov) Date: Tue, 14 Nov 2017 18:49:45 +0300 Subject: [OpenSIPS-Users] topology_hiding Message-ID: Hi. All we try topology hiding module. We have opensips 2.3/2 like sbc with to interfaces. And ip addresses are X.X.X.X and y.y.y.y. We have to softswitch with ip addreses z.z.z.1 and z.z.z.2 Also we loadbalancing calls between softswitch . My config if(is_method("INVITE")) { create_dialog(); topology_hiding(); if (check_address("1","$si","$sp","any","$avp(ctx)")){ force_send_socket(udp:10.240.250.121:5126); setflag(CALL_SW); do_accounting("log", "cdr|failed"); $dlg_val(callee) = $rU; $dlg_val(calling) =$fU; $dlg_val(callcount) = $var(size); xlog("L_INFO","+++++Incoming call from $fU $sp $proto \n"); $acc_extra(src_ip) = $si; $acc_leg(caller) = $fU; $acc_leg(callee) = $rU; if(!lb_start_or_next("1","st","s")){ switch($rc){ case -1: xlog("L_INFO","error 500 General dispatch error\n"); send_reply("503", "No resource avaiable"); acc_log_request("503 General dispatch error"); break; case -2: xlog("L_INFO", "503 No free resources on nodes\n"); send_reply("503", "No resource avaible"); acc_log_request("503 No free resource on nodes"); break; case -3: xlog("L_INFO","606 No nodes avaible\n"); send_reply("503", "No resource avaible"); acc_log_request("606 No nodes avaible"); break; } } else { xlog("L_INFO","Dispatch request to softswitch $du\n"); $avp(node) = $du; $acc_extra(dst_node) = $du; t_on_failure("fail_softswitch"); route(relay); } } else if (check_address("2","$Ri","$Rp","any","$avp(ctx)")){ setflag(CALL_FROM_SW); do_accounting("log", "cdr|failed"); $dlg_val(callee) = $rU; $dlg_val(calling) = $fU; $dlg_val(callcount) = $var(size); xlog("L_INFO","+++++Outgouing call from $fU $sp $proto \n"); $acc_extra(src_ip) = $si; $acc_leg(caller) = $fU; $acc_leg(callee) = $rU; if(!lb_start_or_next("$avp(ctx)","st","s")){ switch($rc){ case -1: xlog("L_INFO","error 500 General dispatch error\n"); send_reply("503", "No resource avaiable"); acc_log_request("503 General dispatch error"); break; case -2: xlog("L_INFO", "503 No free resources on operator\n"); send_reply("503", "No resource avaible"); acc_log_request("503 No free resource on operator"); break; case -3: xlog("L_INFO","606 No resources avaible\n"); send_reply("503", "No resource avaible"); acc_log_request("606 No nodes avaible"); break; } } else { xlog("L_INFO","Dispatch request to operator sbc $du\n"); $avp(node) = $du; $acc_extra(dst_node) = $du; t_on_failure("fail_operator"); route(relay); } } else sl_send_reply("403", "Forbidden posible wrong port"); } } In invite on softswitch i see NVITE sip:12345 at XX.XX.XX.XX:5126 SIP/2.0 Via: SIP/2.0/UDP YY.YY.YY.YY:5126;branch=z9hG4bKe4d9.14b51d57.0 Max-Forwards: 69 From: "Anonymous" ;tag=as59cc9e35 To: Contact: Call-ID: 0068dba856ae05e8406504ab71fc9900 at CC.CC.CC.CC.CC CSeq: 102 INVITE Where CC is client ip. I thithk it must be opensips ip adress 10.240.250.121:5126 What is wrong? -------------- next part -------------- An HTML attachment was scrubbed... URL: From propalarupa at gmail.com Tue Nov 14 14:52:14 2017 From: propalarupa at gmail.com (Eust) Date: Tue, 14 Nov 2017 12:52:14 -0700 (MST) Subject: [OpenSIPS-Users] AES + Opensip In-Reply-To: References: <6171011d-542f-89f8-9eaa-00f5bd2e1bdd@opensips.org> Message-ID: <1510689134709-0.post@n2.nabble.com> Hi, thanks for the explanation, il see what can be done, sadly there is not much information on OpenSips + AES but il try and see if I can figure it out. If someone has done some research before or tried it out id be grateful for some info. :) Thanks. Eust -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html From razvan at opensips.org Wed Nov 15 04:43:16 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Wed, 15 Nov 2017 11:43:16 +0200 Subject: [OpenSIPS-Users] AES + Opensip In-Reply-To: <1510689134709-0.post@n2.nabble.com> References: <6171011d-542f-89f8-9eaa-00f5bd2e1bdd@opensips.org> <1510689134709-0.post@n2.nabble.com> Message-ID: <34e8b4f4-490c-1ab1-b5f1-995efcfdb2af@opensips.org> Hi, Eust! As Daniel already pointed out, AES is a cipher, not an encryption mechanism. If you want to use AES encryption for your traffic, simply configure TLS to use an AES cipher by setting the ciphers_list parameter[1]. To find out the available AES ciphers, run on your machine: openssl ciphers Then choose one (or more) ciphers and feed them to the ciphers_list parameters. For example: modparam("tls_mgm", "ciphers_list", "ECDHE-ECDSA-AES256-GCM-SHA384:ECDHE-RSA-AES256-GCM-SHA384:DHE-RSA-AES256-GCM-SHA384") The previous configuration specifies 3 AES ciphers that can be used for TLS encryption. In my opinion I don't see how further this could get, unless you provide us a more detailed requirement or something specific you want to do. PS: my system uses 50 AES based ciphers and only 7 other ciphers, so I am pretty sure that any TLS session is most likely using an AES based cipher. [1] http://www.opensips.org/html/docs/modules/2.4.x/tls_mgm#idp5669152 Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/14/2017 09:52 PM, Eust wrote: > Hi, > > thanks for the explanation, il see what can be done, sadly there is not much > information on OpenSips + AES but il try and see if I can figure it out. > > If someone has done some research before or tried it out id be grateful for > some info. :) > > Thanks. > > Eust > > > > -- > 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 miha at softnet.si Wed Nov 15 06:52:22 2017 From: miha at softnet.si (Miha) Date: Wed, 15 Nov 2017 12:52:22 +0100 Subject: [OpenSIPS-Users] issue with compiling opensips Message-ID: <44c9cc15-817d-0bd7-9189-ceaab5aee979@softnet.si> Hello which deb pack should I install regarding this error: https://pastebin.com/c4RHMbcT I installed this package " libjson-c-dev" but it is not ok. tnx miha -------------- next part -------------- An HTML attachment was scrubbed... URL: From igor.pavlov1987 at gmail.com Thu Nov 16 00:52:51 2017 From: igor.pavlov1987 at gmail.com (Igor Pavlov) Date: Thu, 16 Nov 2017 09:52:51 +0400 Subject: [OpenSIPS-Users] Opensips 2.2.5 - tls_mgm unable to set the memory allocation functions Message-ID: <8ca0f44c-7f87-31b1-adb2-a81f86f13136@gmail.com> Hi all, I have a trouble with opensips 2.2.5 TLS, hackneyed problem but I can't understand the reason. > Nov 16 08:23:13 r01 opensips[25496]: ERROR:tls_mgm:mod_init: unable to > set the memory allocation functions > Nov 16 08:23:13 r01 opensips[25496]: ERROR:tls_mgm:mod_init: NOTE: > check if you have openssl 1.0.1e-fips, as this version is know to be > broken; if so, you need to upgr > ade or downgrade to a differen openssl version !! > Nov 16 08:23:13 r01 opensips[25496]: ERROR:core:init_mod: failed to > initialize module tls_mgm > Nov 16 08:23:13 r01 opensips[25496]: ERROR:core:main: error while > initializing modules Opensips was installed from repos on CentOS 7, openssl version 1.0.2k: > openssl version -a > OpenSSL 1.0.2k-fips  26 Jan 2017 > built on: reproducible build, date unspecified > platform: linux-x86_64 > options:  bn(64,64) md2(int) rc4(16x,int) des(idx,cisc,16,int) > idea(int) blowfish(idx) > compiler: gcc -I. -I.. -I../include  -fPIC -DOPENSSL_PIC -DZLIB > -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -DKRB5_MIT > -m64 -DL_ENDIAN -Wall -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 > -fexceptions -fstack-protector-strong --param=ssp-buffer-size=4 > -grecord-gcc-switches   -m64 -mtune=generic -Wa,--noexecstack -DPURIFY > -DOPENSSL_IA32_SSE2 -DOPENSSL_BN_ASM_MONT -DOPENSSL_BN_ASM_MONT5 > -DOPENSSL_BN_ASM_GF2m -DRC4_ASM -DSHA1_ASM -DSHA256_ASM -DSHA512_ASM > -DMD5_ASM -DAES_ASM -DVPAES_ASM -DBSAES_ASM -DWHIRLPOOL_ASM > -DGHASH_ASM -DECP_NISTZ256_ASM > OPENSSLDIR: "/etc/pki/tls" > engines:  dynamic > version: opensips 2.2.5 (x86_64/linux) > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > MAX_URI_SIZE 1024, BUF_SIZE 65535 > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. > main.c compiled on 03:00:56 Jul  6 2017 with gcc 4.8.5 -- ____________ Best regards, Igor Pavlov -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Thu Nov 16 05:22:40 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Thu, 16 Nov 2017 12:22:40 +0200 Subject: [OpenSIPS-Users] Opensips 2.2.5 - tls_mgm unable to set the memory allocation functions In-Reply-To: <8ca0f44c-7f87-31b1-adb2-a81f86f13136@gmail.com> References: <8ca0f44c-7f87-31b1-adb2-a81f86f13136@gmail.com> Message-ID: This issue has already been documented here[1]. There is no fix at the moment, but there are some work-arounds that can be done. [1] https://github.com/OpenSIPS/opensips/issues/1114 Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/16/2017 07:52 AM, Igor Pavlov wrote: > > Hi all, > > I have a trouble with opensips 2.2.5 TLS, hackneyed problem but I > can't understand the reason. > >> Nov 16 08:23:13 r01 opensips[25496]: ERROR:tls_mgm:mod_init: unable >> to set the memory allocation functions >> Nov 16 08:23:13 r01 opensips[25496]: ERROR:tls_mgm:mod_init: NOTE: >> check if you have openssl 1.0.1e-fips, as this version is know to be >> broken; if so, you need to upgr >> ade or downgrade to a differen openssl version !! >> Nov 16 08:23:13 r01 opensips[25496]: ERROR:core:init_mod: failed to >> initialize module tls_mgm >> Nov 16 08:23:13 r01 opensips[25496]: ERROR:core:main: error while >> initializing modules > Opensips was installed from repos on CentOS 7, openssl version 1.0.2k: > >> openssl version -a >> OpenSSL 1.0.2k-fips  26 Jan 2017 >> built on: reproducible build, date unspecified >> platform: linux-x86_64 >> options:  bn(64,64) md2(int) rc4(16x,int) des(idx,cisc,16,int) >> idea(int) blowfish(idx) >> compiler: gcc -I. -I.. -I../include  -fPIC -DOPENSSL_PIC -DZLIB >> -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -DKRB5_MIT >> -m64 -DL_ENDIAN -Wall -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 >> -fexceptions -fstack-protector-strong --param=ssp-buffer-size=4 >> -grecord-gcc-switches   -m64 -mtune=generic -Wa,--noexecstack >> -DPURIFY -DOPENSSL_IA32_SSE2 -DOPENSSL_BN_ASM_MONT >> -DOPENSSL_BN_ASM_MONT5 -DOPENSSL_BN_ASM_GF2m -DRC4_ASM -DSHA1_ASM >> -DSHA256_ASM -DSHA512_ASM -DMD5_ASM -DAES_ASM -DVPAES_ASM -DBSAES_ASM >> -DWHIRLPOOL_ASM -DGHASH_ASM -DECP_NISTZ256_ASM >> OPENSSLDIR: "/etc/pki/tls" >> engines:  dynamic >> version: opensips 2.2.5 (x86_64/linux) >> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, >> F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT >> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, >> MAX_URI_SIZE 1024, BUF_SIZE 65535 >> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. >> main.c compiled on 03:00:56 Jul  6 2017 with gcc 4.8.5 > > > -- > ____________ > Best regards, > Igor Pavlov > > > _______________________________________________ > 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 Thu Nov 16 05:23:40 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Thu, 16 Nov 2017 12:23:40 +0200 Subject: [OpenSIPS-Users] issue with compiling opensips In-Reply-To: <44c9cc15-817d-0bd7-9189-ceaab5aee979@softnet.si> References: <44c9cc15-817d-0bd7-9189-ceaab5aee979@softnet.si> Message-ID: Are you using an older version of OpenSIPS? This should have been fixed in all supported versions. Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/15/2017 01:52 PM, Miha wrote: > Hello > > which deb pack should I install regarding this error: > https://pastebin.com/c4RHMbcT > > I installed this package " libjson-c-dev" but it is not ok. > > > tnx > miha > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From miha at softnet.si Thu Nov 16 06:49:52 2017 From: miha at softnet.si (Miha) Date: Thu, 16 Nov 2017 12:49:52 +0100 Subject: [OpenSIPS-Users] issue with compiling opensips In-Reply-To: References: <44c9cc15-817d-0bd7-9189-ceaab5aee979@softnet.si> Message-ID: hello Razvan no, i downloaded it from opensips on git, i did this yesterday (2.3) br miha On 16/11/2017 11:23, Răzvan Crainea wrote: > Are you using an older version of OpenSIPS? This should have been > fixed in all supported versions. > > Best regards, > Răzvan Crainea > OpenSIPS Developer > www.opensips-solutions.com > On 11/15/2017 01:52 PM, Miha wrote: >> Hello >> >> which deb pack should I install regarding this error: >> https://pastebin.com/c4RHMbcT >> >> I installed this package " libjson-c-dev" but it is not ok. >> >> >> tnx >> miha >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Nov 16 07:03:15 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 16 Nov 2017 14:03:15 +0200 Subject: [OpenSIPS-Users] radius authentication message failed with ERROR In-Reply-To: References: Message-ID: <297d1279-aae3-58ed-4945-81b4e1672c58@opensips.org> Hi Diptesh, It seems that the error is generated by the libradius, not at OpenSIPS level. Maybe it is a communication error between the radius server and the libradiusclient - could you do pcap for the RADIUS traffic and validate it with wireshark for example ? Maybe indeed, the server is sending bogus replies. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/09/2017 03:40 PM, Dipteshkumar Patel wrote: > Hello sir, > > I am using opensips 2.3.1 and i am using freeradius-server 2.2.0 for > accounting > > When i am stress testing with 100cps and 1500 cc then the following > error produced by opensips > > ERROR:aaa_radius:resume_send_acct: radius authentication message > failed with ERROR > Nov 9 13:19:14 OS /usr/local/sbin/opensips[12910]: rc_send_server: > recvfrom: XXX.XXX.X.XX:1813: reply is too short > Nov 9 13:19:14 OS /usr/local/sbin/opensips[12910]: > ERROR:acc:acc_aaa_request: Radius accounting request failed for > status: 'Start' Call-Id: '97844-1608 at XXX.XXX.XXX.XXX' > > what is the cause for this ?? > > *Diptesh Patel* > Jr. Software Developer > Ecosmob Technologies Ltd > Ahmedabad > Mo:*+919898962659* > > > _______________________________________________ > 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 16 07:10:51 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 16 Nov 2017 14:10:51 +0200 Subject: [OpenSIPS-Users] call_center module in Opensips 2.3 In-Reply-To: References: Message-ID: Dragomir, The Call Center module handles the agents as simple SIP URIs. if in your case the agents are actually registering devices, this is a different matter. What I suggest, for simplicity reasons is to use 2 opensips instances. The first to be a dedicated call center seeing the agents as AoRs (sip:user at domain) where the domain points to the IP of the second server The second instance to be simple registrar server for the "domain" where the agents are registering their devices. In terms of call you will have: ---(call to queue)----> OPENSIPS1 -----(call to agent user at domain)----> OPENSIPS2 -----(call to registered device)-----> AGENT DEVICE Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/10/2017 04:21 PM, Dragomir Haralambiev wrote: > Hi, > > I try your idea to loop the INVITE back to Opensips and do a lookup(). > I pit into "location" field from cc_agents table this one > "sip:localnumber at Opensips_IP:Opensips_port". > > Call is connected but I receive follow log message: > > ERROR:b2b_entities:b2b_prescript_f: No dialog found, callid= > [B2B.337.5360955.1510320840], method=OPTIONS > > When call is finished record from dialog tables is not deleted. > > Where is problem? > > Regards, > Dragomir > > 2017-11-06 11:37 GMT+02:00 Răzvan Crainea >: > > Hi, Hristo! > > 1. You are right, b2b_entities should be loaded as a dependency > for b2b_logic. You may argue that it should also be listed as a > dependency to call_center as well, but indirectly. > > 2. Yes, you should populate the tables with the flows you want to > support and the agents that are going to use the call center. > > 3. There is currently no way to directly set the location of the > agent dynamically. However, we can do some work-arounds, for > example always advertise yourself as the location of the user, and > for each call you can loop the INVITE back to you and do a > lookup(). Or simply use an external registrar service. > > Best regards, > > Răzvan Crainea > OpenSIPS Developer > www.opensips-solutions.com > > On 11/05/2017 11:06 PM, Hristo Donev wrote: >> Hello All, >> >> I try to implement call_center module in Opensips 2.3. >> Here are my questions? >> >> 1. From the documentation 1.3.1: >> "b2b_logic" and "database" must be loaded before callcenter module. >> >> I found out that "b2b_entities" should be loaded as well, >> otherwise the opensips does not start. >> >> Is this my mistake or a documentation error? >> >> 2. Opensips can not start if the tables "cc_agents" and "cc_flow" >> are empty. >> Why it cannot start? Should they be populated with some data? >> >> 3. Table "cc_agents", field "location" should point to the SIP >> agent location. >> If the agent's IP and port are changed dynamically - >> how to copy automatically "contact" field from "location" table >> to "cc_agents" field "location"? >> >> I would be grateful to receive a response for my questions. >> >> Thanks in advance, >> Hristo Donev >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Nov 16 07:24:15 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 16 Nov 2017 14:24:15 +0200 Subject: [OpenSIPS-Users] Introducing OpenSIPS 2.4 In-Reply-To: References: <972d9f71-89d4-89f1-9183-f7fc0f1c506a@opensips.org> <703b5e19-8d2d-84a9-9dc3-a8c576d67e0e@opensips.org> Message-ID: <2f3e6172-3501-4cfc-75b5-f84f84035e3d@opensips.org> Hi Maxim, Thank you for the valuable input - it is true what you are saying IF the media relay has the ability to publish (or report) its internal load. What you are describing here is what we did in terms of SIP call balancing, with FreeSWITCH - local call counting versus pulling call counters from FS. In regards to the second topic, that is true also, let me give it some thoughts. The only issue I see here is with "let each vendor to implement proper hooks" :). But as concept it is something definitely interesting. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/11/2017 07:01 AM, Maxim Sobolev wrote: > Bogdan, with regards to the media relays clustering what's the > advantage of sharing that load info between each signalling node > versus having each node tracking this independently? In my view the > latter could be more reliable and much less complicated construct. The > only disadvantage is that you'd get more command load on relays, > however at least with the RTPproxy pulling load stats is very > lightweight operation so even with tens of signalling nodes pulling > single media relay every 1-10 seconds it won't cause any noticeable > performance degradation on the relay. On the flip side, each > signalling node would get accurate view from its vantage PoV, so in > the case of geographically distributed system when signalling node can > only see subset of all media nodes it would still be able to make > proper decisions. This is the approach we use in the rtp_cluster and > it works pretty well with cluster size of up to 5 signalling and 10 > RTP handling nodes, 40-50K media sessions in total. It can also give > you accurate RTT information, so your signalling node can not only > factor in the load but also proximity or each and every media relay. > > As far as the load tracking is concerned, I think the approach to > implement "b2b-driven routing" using API that is specific to each > particular b2b is somewhat wasteful and is not very future-proof. What > we would like to see instead, is for opensips to publish some kind of > API (preferably SIP-based, using OPTIONS or SUBSCRIBE/NOTIFY > mechanism) to pull this information out and let each b2b vendor to > implement proper hooks. Then it can go as far as making this info some > king of RFC. > > Anyhow, just my $0.02c. Not volunteering to do opensips side > (ENOTIME), but if opensips project comes up with the reasonable > b2bua-agnostic load query API to use we might look at implementing it > in the sippy [py/go]-B2BUAs. > > -Max > > On Wed, Nov 8, 2017 at 9:31 AM, Bogdan-Andrei Iancu > > wrote: > > Hi Nuno, > > On the Asterisk part, the plan is to do exactly what we already > have for FreeSWITCH (see > https://blog.opensips.org/2017/03/01/freeswitch-driven-routing-in-opensips-2-3/ > ) > > In terms of clustering media relays, is about the ability to share > the state (enabled/disabled) between all the cluster nodes using > the media relays. Optionally, we are looking in adding the ability > to balance the traffic between the relays, in a cluster-level > aware (all the nodes in the cluster will share the information on > the load of the media relays ) > > Regards, > > 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 Thu Nov 16 07:27:12 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 16 Nov 2017 14:27:12 +0200 Subject: [OpenSIPS-Users] OpenSIPs Control Panel Homer Menu In-Reply-To: <13d3238b-79c4-5904-a8b9-f00cf68f145c@bohboh.info> References: <31a910a6-10e7-6aa8-cb53-31d35a6e64ba@bohboh.info> <667cccd5-ae01-9780-9c0d-eb34c39355aa@bohboh.info> <01993f73-4c11-de06-c8fe-3e5b93c62580@opensips.org> <32fe0423-10e7-6f23-2d01-3ebeb4f86b98@bohboh.info> <13d3238b-79c4-5904-a8b9-f00cf68f145c@bohboh.info> Message-ID: Hi, in the home tool config file ( config/tools/system/homer/local.inc.php) , how the $homer_URL is defined ? with HTTP or HTTPS ? Have you read the note from the file in regards to HTTP versus HTTPS ?? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/13/2017 06:19 PM, Social Boh wrote: > > Hello, > > if I access OpenSIPs Control Panel over https, with a Let's Encrypt > Certificate, I can't see Homer page (blank page). > > The only error/notice on the ssl_error.log is: > > PHP Notice: A session had already been started - ignoring > session_start() in /var/www/html/opensips-cp/config/session.inc.php on > line 23, referer: > https://opensipscontrolpanel.org/cp/tools/system/homer/homer.php > > If I use OpenSIPs Control Panel over httpd all working fine. > > Regards > > --- > I'm SoCIaL, MayBe > El 03/11/2017 a las 10:22, Social Boh escribió: >> >> The error was related with this semicolon (before the last parenthesis): >> >> define('EXTERNAL_AUTH_URI', >> "http://5.6.7.8/cp/tools/system/homer/auth.php"*;*); >> >> Thank you. >> >> Regards >> >> --- >> I'm SoCIaL, MayBe >> El 03/11/2017 a las 07:15, Răzvan Crainea escribió: >>> This time there seems to be an issue on the homer side. Can you >>> check the logs to see if homer is saying anything useful? >>> >>> Best regards, >>> >>> Răzvan Crainea >>> OpenSIPS Developer >>> www.opensips-solutions.com >>> >>> On 11/03/2017 02:06 PM, Social Boh wrote: >>>> Hello, >>>> >>>> Now I have this error (homer side): >>>> >>>> GET >>>> /api/v1/redirect?externalid=rx88kxQUs1DNT6es2ybwBvhevecJwjnY&url=http%3A%2F%2F1.2.3.4.0%2F >>>> HTTP/1.1. >>>> Host: 104.236.105.0. >>>> User-Agent: Mozilla/5.0 (Windows NT 6.1; Win64; x64; rv:56.0) >>>> Gecko/20100101 Firefox/56.0. >>>> Accept: >>>> text/html,application/xhtml+xml,application/xml;q=0.9,*/*;q=0.8. >>>> Accept-Language: es-ES,es;q=0.8,en-US;q=0.5,en;q=0.3. >>>> Accept-Encoding: gzip, deflate. >>>> Referer: http://5.6.7.8/cp/tools/system/homer/homer.php. >>>> Connection: keep-alive. >>>> Upgrade-Insecure-Requests: 1. >>>> >>>> HTTP/1.0 500 Internal Server Error. >>>> Date: Fri, 03 Nov 2017 11:47:36 GMT. >>>> Server: Apache/2.4.6 (CentOS) OpenSSL/1.0.2k-fips PHP/5.4.16. >>>> X-Powered-By: PHP/5.4.16. >>>> Content-Length: 0. >>>> Connection: close. >>>> Content-Type: text/html; charset=UTF-8. >>>> >>>> My homer configuration is: >>>> >>>> define('AUTHENTICATION',"External"); >>>> define('EXTERNAL_AUTH_URI', >>>> "http://5.6.7.8/cp/tools/system/homer/auth.php";); >>>> define('EXTERNAL_AUTH_METHOD', "GET"); >>>> define('EXTERNAL_AUTH_PARAM', "param=KEY"); >>>> define('EXTERNAL_AUTH_REQUEST_TYPE', "get"); >>>> define('EXTERNAL_AUTH_REQUEST_KEY', "externalid"); >>>> define('EXTERNAL_AUTH_POSITIVE_REPLY', "200"); >>>> define('EXTERNAL_AUTH_REPLY_USER_INFO', "json"); >>>> >>>> 1.2.3.4 is the Homer IP and 5.6.7.8 is the OpenSIPs Control Panel IP >>>> >>>> Thank you >>>> >>>> Regards >>>> >>>> --- >>>> I'm SoCIaL, MayBe >>>> >>>> El 03/11/2017 a las 04:47, Răzvan Crainea escribió: >>>>> Hello! >>>>> >>>>> You should install the php apc library. On Ubuntu this is php-apcu. >>>>> >>>>> Best regards, >>>>> >>>>> Răzvan Crainea >>>>> OpenSIPS Developer >>>>> www.opensips-solutions.com >>>>> >>>>> On 11/02/2017 10:09 PM, Social Boh wrote: >>>>>> Hello, >>>>>> >>>>>> each time I try to access Homer Menu on OpenSIPs Control Panel >>>>>> 7.3.2, on the apache log: >>>>>> >>>>>> Call to undefined function apc_store() in >>>>>> /var/www/html/opensips-cp/web/tools/system/homer/homer.php on line 5 >>>>>> >>>>>> Blank page >>>>>> >>>>>> OpenSIPs Homer Configuration: >>>>>> >>>>>> $homer_URL = "http://homer.mydomain.org"; >>>>>> >>>>>> $homer_auth_method = "get"; >>>>>> >>>>>> Homer OpenSIPs configuration (preference.php) >>>>>> >>>>>> define('AUTHENTICATION',"External"); >>>>>> define('EXTERNAL_AUTH_URI', >>>>>> "http://os.mydomain.org/cp/tools/system/homer/auth.php";); >>>>>> define('EXTERNAL_AUTH_METHOD', "GET"); >>>>>> define('EXTERNAL_AUTH_PARAM', "param=KEY"); >>>>>> define('EXTERNAL_AUTH_REQUEST_TYPE', "get"); >>>>>> define('EXTERNAL_AUTH_REQUEST_KEY', "externalid"); >>>>>> define('EXTERNAL_AUTH_POSITIVE_REPLY', "200"); >>>>>> define('EXTERNAL_AUTH_REPLY_USER_INFO', "json"); >>>>>> >>>>>> Any hint >>>>>> >>>>>> Thank you in advance. >>>>>> >>>>>> Regards >>>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Nov 16 07:32:30 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 16 Nov 2017 14:32:30 +0200 Subject: [OpenSIPS-Users] fix_nated_sdp and rtpproxy In-Reply-To: References: Message-ID: Hi Yury, Why do you need to update the incoming SDP before engaging RTPproxy ? Just pass to it whatever you get into SDP (even if private) and let rtpproxy to "learn" the correct IP:port of the media end points based on the incoming traffic. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/14/2017 04:05 AM, Yury Kirsanov wrote: > Hi, > I have a problem with remote client who can't properly configure NAT > and is sending private IP in all SDP and SIP packets. I've resolved > issues with registration but can't resolve the issue with SDP. My > topology is as following: > > ----- ------- ---------- > -------- ----------- > > So, I need to simultaneously engage RTP Proxy to bridge external > Internet and our local LAN and need to rewrite SDP body of original > Client's packet with received IP address. If I'm doing fix_nated_sdp > and then rtpproxy_offer - I'm getting doubled IPs in SDP message, like > this: > > 100.11.100.200192.168.20.200 > > > Is there any way to update SDP of client's packet and then command RTP > Proxy to use updated IP? 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 Thu Nov 16 07:36:49 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 16 Nov 2017 14:36:49 +0200 Subject: [OpenSIPS-Users] topology_hiding In-Reply-To: References: Message-ID: <9ad38fc0-d509-ab86-c021-605809f4ff27@opensips.org> Hi, First, note that "2" is written "two", not "to" - just to be sure people do understand your explanations :). Now, you say you still see the Client IP in the Call-ID and in the FROM headers ?? If that is the case: 1) to change the call-id, see the "C" option in the topology_hiding() call 2) The FROM hdr is outside the topology-hiding scope, but if you want to change it see the uac_replace_from() function from the the uac module. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/14/2017 05:49 PM, Kirill Galinurov wrote: > Hi. All we try topology hiding module. > We have opensips 2.3/2 like sbc with to interfaces. > And ip addresses are X.X.X.X and y.y.y.y. > We have to softswitch with ip addreses z.z.z.1 and z.z.z.2 > Also we loadbalancing calls between softswitch . > My config > if(is_method("INVITE")) { > create_dialog(); > topology_hiding(); > if (check_address("1","$si","$sp","any","$avp(ctx)")){ > force_send_socket(udp:10.240.250.121:5126 > ); > setflag(CALL_SW); > do_accounting("log", "cdr|failed"); > $dlg_val(callee) = $rU; > $dlg_val(calling) =$fU; > $dlg_val(callcount) = $var(size); > xlog("L_INFO","+++++Incoming call from $fU $sp $proto > \n"); > $acc_extra(src_ip) = $si; > $acc_leg(caller) = $fU; > $acc_leg(callee) = $rU; > if(!lb_start_or_next("1","st","s")){ > switch($rc){ > case -1: > xlog("L_INFO","error 500 General dispatch > error\n"); > send_reply("503", "No resource avaiable"); > acc_log_request("503 General dispatch error"); > break; > > case -2: > xlog("L_INFO", "503 No free resources on > nodes\n"); > send_reply("503", "No resource avaible"); > acc_log_request("503 No free resource on > nodes"); > break; > case -3: > xlog("L_INFO","606 No nodes avaible\n"); > send_reply("503", "No resource avaible"); > acc_log_request("606 No nodes avaible"); > break; > } > } else { > xlog("L_INFO","Dispatch request to softswitch > $du\n"); > $avp(node) = $du; > $acc_extra(dst_node) = $du; > t_on_failure("fail_softswitch"); > route(relay); > } > } > else > if (check_address("2","$Ri","$Rp","any","$avp(ctx)")){ > setflag(CALL_FROM_SW); > do_accounting("log", "cdr|failed"); > $dlg_val(callee) = $rU; > $dlg_val(calling) = $fU; > $dlg_val(callcount) = $var(size); > xlog("L_INFO","+++++Outgouing call from $fU $sp $proto > \n"); > $acc_extra(src_ip) = $si; > $acc_leg(caller) = $fU; > $acc_leg(callee) = $rU; > if(!lb_start_or_next("$avp(ctx)","st","s")){ > switch($rc){ > case -1: > xlog("L_INFO","error 500 General dispatch > error\n"); > send_reply("503", "No resource avaiable"); > acc_log_request("503 General dispatch error"); > break; > > case -2: > xlog("L_INFO", "503 No free resources on > operator\n"); > send_reply("503", "No resource avaible"); > acc_log_request("503 No free resource on > operator"); > break; > case -3: > xlog("L_INFO","606 No resources avaible\n"); > send_reply("503", "No resource avaible"); > acc_log_request("606 No nodes avaible"); > break; > } > } else { > xlog("L_INFO","Dispatch request to operator sbc > $du\n"); > $avp(node) = $du; > $acc_extra(dst_node) = $du; > t_on_failure("fail_operator"); > route(relay); > } > } > else > sl_send_reply("403", "Forbidden posible wrong port"); > > } > } > In invite on softswitch i see > NVITE sip:12345 at XX.XX.XX.XX:5126 SIP/2.0 > Via: SIP/2.0/UDP YY.YY.YY.YY:5126;branch=z9hG4bKe4d9.14b51d57.0 > Max-Forwards: 69 > From: "Anonymous" >;tag=as59cc9e35 > To: > Contact: > Call-ID: 0068dba856ae05e8406504ab71fc9900 at CC.CC.CC.CC.CC > > CSeq: 102 INVITE > > Where CC is client ip. I thithk it must be opensips ip adress > 10.240.250.121:5126 > What is wrong? > > > _______________________________________________ > 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 16 08:01:31 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 16 Nov 2017 15:01:31 +0200 Subject: [OpenSIPS-Users] fix_nated_sdp and rtpproxy In-Reply-To: References: Message-ID: Hi Yury, RTPproxy learns not only from SDP, but based on the incoming RTP traffic. Your Client will send RTP to RTPproxy (as the Client will receive back a valid IP in the SDP) and RTPproxy will see where the RTP traffic comes from and starts sending back RTP to that IP:port. Just be sure you do not use the "a" and "r" flags when calling rtpproxy engage. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/16/2017 02:48 PM, Yury Kirsanov wrote: > Hi Bogdan, > Thanks for your answer and support! But the issue is that there is a > private IP address in SDP coming from the client so if I just pass it > via RTP proxy - it would learn that incorrect IP and would try to > forward packets to it causing one-way audio. That's why I first wanted > to overstamp that incorrect IP with received IP and then pass it to > the RTP Proxy. Hope this makes my idea clear. > > Regards, > Yury. > > 2017-11-16 23:32 GMT+11:00 Bogdan-Andrei Iancu >: > > Hi Yury, > > Why do you need to update the incoming SDP before engaging > RTPproxy ? Just pass to it whatever you get into SDP (even if > private) and let rtpproxy to "learn" the correct IP:port of the > media end points based on the incoming traffic. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 11/14/2017 04:05 AM, Yury Kirsanov wrote: >> Hi, >> I have a problem with remote client who can't properly configure >> NAT and is sending private IP in all SDP and SIP packets. I've >> resolved issues with registration but can't resolve the issue >> with SDP. My topology is as following: >> >> ----- ------- ---------- >> -------- ----------- >> >> So, I need to simultaneously engage RTP Proxy to bridge external >> Internet and our local LAN and need to rewrite SDP body of >> original Client's packet with received IP address. If I'm doing >> fix_nated_sdp and then rtpproxy_offer - I'm getting doubled IPs >> in SDP message, like this: >> >> 100.11.100.200192.168.20.200 >> >> >> Is there any way to update SDP of client's packet and then >> command RTP Proxy to use updated IP? 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 razvan at opensips.org Thu Nov 16 10:47:12 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Thu, 16 Nov 2017 17:47:12 +0200 Subject: [OpenSIPS-Users] issue with compiling opensips In-Reply-To: References: <44c9cc15-817d-0bd7-9189-ceaab5aee979@softnet.si> Message-ID: What version of libjson are you using, and what OS are you running? Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/16/2017 01:49 PM, Miha wrote: > hello Razvan > > no, i downloaded it from opensips on git, i did this yesterday (2.3) > > > br > miha > > On 16/11/2017 11:23, Răzvan Crainea wrote: >> Are you using an older version of OpenSIPS? This should have been >> fixed in all supported versions. >> >> Best regards, >> Răzvan Crainea >> OpenSIPS Developer >> www.opensips-solutions.com >> On 11/15/2017 01:52 PM, Miha wrote: >>> Hello >>> >>> which deb pack should I install regarding this error: >>> https://pastebin.com/c4RHMbcT >>> >>> I installed this package " libjson-c-dev" but it is not ok. >>> >>> >>> tnx >>> miha >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Nov 16 11:26:47 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 16 Nov 2017 18:26:47 +0200 Subject: [OpenSIPS-Users] fix_nated_sdp and rtpproxy In-Reply-To: References: Message-ID: <268801e7-54c7-2693-c756-efd01dcd0717@opensips.org> Using the "i" and "e" flags to control the RTP interfaces does not affect the "learning" mode. You should you "r" flag only if the received IP in SDP is public. Otherwise do not use it. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/16/2017 03:07 PM, Yury Kirsanov wrote: > Hi Bogdan, > Thanks for clarification, I didn't know that RTP Proxy learns from > source packets, it's quite hard to find documentation on it. The > problem here is that I'm using RTP Proxy in bridge mode, connecting > public interface with internal LAN and my upstream providers are using > sets of their own proxies too, besides customers connecting to this > OpenSIPS server. So I have to use flag "r" (and, of course, flags of > direction - "i" and "e") or else there would be no audio on calls when > SDP in provider's answer is different from SIP packet source IP. Of > course, I can analyze where the initial INVITE packet is coming from > and enable different options but that would be more complex rather > than just have SDP modified before passing it to RTP Proxy. > Regards, > Yury. > > 2017-11-17 0:01 GMT+11:00 Bogdan-Andrei Iancu >: > > Hi Yury, > > RTPproxy learns not only from SDP, but based on the incoming RTP > traffic. Your Client will send RTP to RTPproxy (as the Client will > receive back a valid IP in the SDP) and RTPproxy will see where > the RTP traffic comes from and starts sending back RTP to that > IP:port. > > Just be sure you do not use the "a" and "r" flags when calling > rtpproxy engage. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 11/16/2017 02:48 PM, Yury Kirsanov wrote: >> Hi Bogdan, >> Thanks for your answer and support! But the issue is that there >> is a private IP address in SDP coming from the client so if I >> just pass it via RTP proxy - it would learn that incorrect IP and >> would try to forward packets to it causing one-way audio. That's >> why I first wanted to overstamp that incorrect IP with received >> IP and then pass it to the RTP Proxy. Hope this makes my idea clear. >> >> Regards, >> Yury. >> >> 2017-11-16 23:32 GMT+11:00 Bogdan-Andrei Iancu >> >: >> >> Hi Yury, >> >> Why do you need to update the incoming SDP before engaging >> RTPproxy ? Just pass to it whatever you get into SDP (even if >> private) and let rtpproxy to "learn" the correct IP:port of >> the media end points based on the incoming traffic. >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> >> On 11/14/2017 04:05 AM, Yury Kirsanov wrote: >>> Hi, >>> I have a problem with remote client who can't properly >>> configure NAT and is sending private IP in all SDP and SIP >>> packets. I've resolved issues with registration but can't >>> resolve the issue with SDP. My topology is as following: >>> >>> ----- ------- ---------- >>> -------- ----------- >>> >>> So, I need to simultaneously engage RTP Proxy to bridge >>> external Internet and our local LAN and need to rewrite SDP >>> body of original Client's packet with received IP address. >>> If I'm doing fix_nated_sdp and then rtpproxy_offer - I'm >>> getting doubled IPs in SDP message, like this: >>> >>> 100.11.100.200192.168.20.200 >>> >>> >>> Is there any way to update SDP of client's packet and then >>> command RTP Proxy to use updated IP? 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 bakern at gmail.com Fri Nov 17 01:13:02 2017 From: bakern at gmail.com (Nathan Baker) Date: Fri, 17 Nov 2017 01:13:02 -0500 Subject: [OpenSIPS-Users] Topology Hiding Presence Proxy Message-ID: Hello, I have been looking for some examples on how to use the topology hiding module and proxying presence (SUBSCRIBE, NOTIFY, etc.) together, similar to how the mid_registrar module would handle registrations. Does anyone have any examples or suggestions on how to do this? I have the clients subscribing to the server (through OpenSIPS proxy to another presence server), but I'm not sure how to store the subscriptions and how to route the presence messages. For registrations you can just do save("location"), but is there an equivalent for subscriptions/presence? It seems like the handle_subscribe() function from the presence module wouldn't apply because it's part of a presence server, or can you just use it for storing subscriptions but never publish anything? I don't see a function to look up these stored subscriptions. Would it be better to just assume that the subscriptions are valid and relay the messages? If so, what is the best way to handle the routing? Any help or insight would be greatly appreciated! Thanks, Nate -------------- next part -------------- An HTML attachment was scrubbed... URL: From miha at softnet.si Fri Nov 17 02:28:46 2017 From: miha at softnet.si (Miha) Date: Fri, 17 Nov 2017 08:28:46 +0100 Subject: [OpenSIPS-Users] issue with compiling opensips In-Reply-To: References: <44c9cc15-817d-0bd7-9189-ceaab5aee979@softnet.si> Message-ID: <319fde66-a20d-f15a-adab-8bf7885ea8f1@softnet.si> hey this is what i have: cat /etc/issue Debian GNU/Linux 9 \n \ apt list libjson* Listing... Done libjson-any-perl/stable 1.39-1 all libjson-c-dev/stable,now 0.12.1-1.1 amd64 [installed] libjson-c-doc/stable 0.12.1-1.1 all libjson-c3/stable,now 0.12.1-1.1 amd64 [installed,automatic] libjson-glib-1.0-0/stable,now 1.2.6-1 amd64 [installed,automatic] libjson-glib-1.0-common/stable,now 1.2.6-1 all [installed,automatic] libjson-glib-dev/stable 1.2.6-1 amd64 libjson-glib-doc/stable 1.2.6-1 all libjson-java/stable 2.4-3 all libjson-maybexs-perl/stable 1.003008-1 all libjson-multivalueordered-perl/stable 0.005-1 all libjson-perl/stable 2.90-1 all libjson-pointer-perl/stable 0.07-1 all libjson-pp-perl/stable 2.27400-1 all libjson-rpc-perl/stable 1.06-2 all libjson-simple-doc/stable 1.1.1-4 all libjson-simple-java/stable 1.1.1-4 all libjson-smart-java/stable 2.2-1 all libjson-types-perl/stable 0.05-1 all libjson-validator-perl/stable 0.92+dfsg-1 all libjson-webtoken-perl/stable 0.10-2 all libjson-xs-perl/stable 3.030-1 amd64 libjsoncpp-dev/stable,now 1.7.4-3 amd64 [installed] libjsoncpp-doc/stable 1.7.4-3 all libjsoncpp1/stable,now 1.7.4-3 amd64 [installed,automatic] libjsonm-ocaml/stable 0.9.1-2 amd64 libjsonm-ocaml-dev/stable 0.9.1-2 amd64 libjsonm-ocaml-doc/stable 0.9.1-2 all libjsonp-java/stable 1.0.4-1 all libjsonp-java-doc/stable 1.0.4-1 all libjsonpath-java/stable 2.0.0-3 all libjsonrpccpp-client0/stable 0.7.0-1+b2 amd64 libjsonrpccpp-client0-dbg/stable 0.7.0-1+b2 amd64 libjsonrpccpp-common0/stable 0.7.0-1+b2 amd64 libjsonrpccpp-common0-dbg/stable 0.7.0-1+b2 amd64 libjsonrpccpp-dev/stable 0.7.0-1+b2 amd64 libjsonrpccpp-server0/stable 0.7.0-1+b2 amd64 libjsonrpccpp-server0-dbg/stable 0.7.0-1+b2 amd64 libjsonrpccpp-stub0/stable 0.7.0-1+b2 amd64 libjsonrpccpp-stub0-dbg/stable 0.7.0-1+b2 amd64 libjsonrpccpp-tools/stable 0.7.0-1+b2 amd64 On 16/11/2017 16:47, Răzvan Crainea wrote: > What version of libjson are you using, and what OS are you running? > Răzvan Crainea > OpenSIPS Developer > www.opensips-solutions.com > On 11/16/2017 01:49 PM, Miha wrote: >> hello Razvan >> >> no, i downloaded it from opensips on git, i did this yesterday (2.3) >> >> >> br >> miha >> >> On 16/11/2017 11:23, Răzvan Crainea wrote: >>> Are you using an older version of OpenSIPS? This should have been >>> fixed in all supported versions. >>> >>> Best regards, >>> Răzvan Crainea >>> OpenSIPS Developer >>> www.opensips-solutions.com >>> On 11/15/2017 01:52 PM, Miha wrote: >>>> Hello >>>> >>>> which deb pack should I install regarding this error: >>>> https://pastebin.com/c4RHMbcT >>>> >>>> I installed this package " libjson-c-dev" but it is not ok. >>>> >>>> >>>> tnx >>>> miha >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From kkothari157 at gmail.com Fri Nov 17 02:42:42 2017 From: kkothari157 at gmail.com (Ketan Kothari) Date: Fri, 17 Nov 2017 13:12:42 +0530 Subject: [OpenSIPS-Users] issue with compiling opensips In-Reply-To: <319fde66-a20d-f15a-adab-8bf7885ea8f1@softnet.si> References: <44c9cc15-817d-0bd7-9189-ceaab5aee979@softnet.si> <319fde66-a20d-f15a-adab-8bf7885ea8f1@softnet.si> Message-ID: Hello Miha, Please verify below dependency installed on your server or not. libjson-c-dev libjson-c-doc libjson-c2 libjson-c2-dbg libjson0 libjson0-dev On Fri, Nov 17, 2017 at 12:58 PM, Miha wrote: > hey > > this is what i have: > > cat /etc/issue > Debian GNU/Linux 9 \n \ > > > apt list libjson* > Listing... Done > libjson-any-perl/stable 1.39-1 all > libjson-c-dev/stable,now 0.12.1-1.1 amd64 [installed] > libjson-c-doc/stable 0.12.1-1.1 all > libjson-c3/stable,now 0.12.1-1.1 amd64 [installed,automatic] > libjson-glib-1.0-0/stable,now 1.2.6-1 amd64 [installed,automatic] > libjson-glib-1.0-common/stable,now 1.2.6-1 all [installed,automatic] > libjson-glib-dev/stable 1.2.6-1 amd64 > libjson-glib-doc/stable 1.2.6-1 all > libjson-java/stable 2.4-3 all > libjson-maybexs-perl/stable 1.003008-1 all > libjson-multivalueordered-perl/stable 0.005-1 all > libjson-perl/stable 2.90-1 all > libjson-pointer-perl/stable 0.07-1 all > libjson-pp-perl/stable 2.27400-1 all > libjson-rpc-perl/stable 1.06-2 all > libjson-simple-doc/stable 1.1.1-4 all > libjson-simple-java/stable 1.1.1-4 all > libjson-smart-java/stable 2.2-1 all > libjson-types-perl/stable 0.05-1 all > libjson-validator-perl/stable 0.92+dfsg-1 all > libjson-webtoken-perl/stable 0.10-2 all > libjson-xs-perl/stable 3.030-1 amd64 > libjsoncpp-dev/stable,now 1.7.4-3 amd64 [installed] > libjsoncpp-doc/stable 1.7.4-3 all > libjsoncpp1/stable,now 1.7.4-3 amd64 [installed,automatic] > libjsonm-ocaml/stable 0.9.1-2 amd64 > libjsonm-ocaml-dev/stable 0.9.1-2 amd64 > libjsonm-ocaml-doc/stable 0.9.1-2 all > libjsonp-java/stable 1.0.4-1 all > libjsonp-java-doc/stable 1.0.4-1 all > libjsonpath-java/stable 2.0.0-3 all > libjsonrpccpp-client0/stable 0.7.0-1+b2 amd64 > libjsonrpccpp-client0-dbg/stable 0.7.0-1+b2 amd64 > libjsonrpccpp-common0/stable 0.7.0-1+b2 amd64 > libjsonrpccpp-common0-dbg/stable 0.7.0-1+b2 amd64 > libjsonrpccpp-dev/stable 0.7.0-1+b2 amd64 > libjsonrpccpp-server0/stable 0.7.0-1+b2 amd64 > libjsonrpccpp-server0-dbg/stable 0.7.0-1+b2 amd64 > libjsonrpccpp-stub0/stable 0.7.0-1+b2 amd64 > libjsonrpccpp-stub0-dbg/stable 0.7.0-1+b2 amd64 > libjsonrpccpp-tools/stable 0.7.0-1+b2 amd64 > > > > > On 16/11/2017 16:47, Răzvan Crainea wrote: > > What version of libjson are you using, and what OS are you running? > > Răzvan Crainea > OpenSIPS Developerwww.opensips-solutions.com > > On 11/16/2017 01:49 PM, Miha wrote: > > hello Razvan > > no, i downloaded it from opensips on git, i did this yesterday (2.3) > > > br > miha > > On 16/11/2017 11:23, Răzvan Crainea wrote: > > Are you using an older version of OpenSIPS? This should have been fixed in > all supported versions. > > Best regards, > > Răzvan Crainea > OpenSIPS Developerwww.opensips-solutions.com > > On 11/15/2017 01:52 PM, Miha wrote: > > Hello > > which deb pack should I install regarding this error: > https://pastebin.com/c4RHMbcT > > I installed this package " libjson-c-dev" but it is not ok. > > > tnx > miha > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From k.galinurov at gmail.com Fri Nov 17 03:26:42 2017 From: k.galinurov at gmail.com (Kirill Galinurov) Date: Fri, 17 Nov 2017 11:26:42 +0300 Subject: [OpenSIPS-Users] topology_hiding In-Reply-To: <9ad38fc0-d509-ab86-c021-605809f4ff27@opensips.org> References: <9ad38fc0-d509-ab86-c021-605809f4ff27@opensips.org> Message-ID: Hi Bogdan. Thanks you for the answer. I don't want mislead you. I resolve my problem. I use uac_replace_from() uac_replace_to() fo rewrite to and from headers. if(is_method("INVITE")) { if (check_address("1","$si","$sp","any","$avp(ctx)")){ force_send_socket(udp:10.240.250.121:9918); create_dialog(); topology_hiding(); setflag(CALL_SW); do_accounting("log", "cdr|failed"); $dlg_val(callee) = $rU; $dlg_val(calling) =$fU; $dlg_val(callcount) = $var(size); xlog("L_INFO","+++++Incoming call from $fU $sp $proto \n"); $acc_extra(src_ip) = $si; $acc_leg(caller) = $fU; $acc_leg(callee) = $rU; if(!lb_start_or_next("1","st","s")){ switch($rc){ case -1: xlog("L_INFO","error 500 General dispatch error\n"); send_reply("503", "No resource avaiable"); acc_log_request("503 General dispatch error"); break; case -2: xlog("L_INFO", "503 No free resources on nodes\n"); send_reply("503", "No resource avaible"); acc_log_request("503 No free resource on nodes"); break; case -3: xlog("L_INFO","606 No nodes avaible\n"); send_reply("503", "No resource avaible"); acc_log_request("606 No nodes avaible"); break; } } else { xlog("L_INFO","Dispatch request to softswitch $du\n"); $avp(node) = $du; $acc_extra(dst_node) = $du; $ru = "sip:" + $rU + "@" + $dd + ":"+$rp +";transport=udp"; uac_replace_from("",'sip:$fU at 10.240.250.121'); uac_replace_to("sip:$tU at 10.240.250.121:$rp"); t_on_failure("fail_softswitch"); route(relay); } But I doubt that it correct way. Also i want to avp_db_query fo get ip from db. Can i use variables in force_send_socket() function ? Regards Kirill Galinurov 2017-11-16 15:36 GMT+03:00 Bogdan-Andrei Iancu : > Hi, > > First, note that "2" is written "two", not "to" - just to be sure people > do understand your explanations :). > > Now, you say you still see the Client IP in the Call-ID and in the FROM > headers ?? If that is the case: > > 1) to change the call-id, see the "C" option in the topology_hiding() call > > 2) The FROM hdr is outside the topology-hiding scope, but if you want to > change it see the uac_replace_from() function from the the uac module. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 11/14/2017 05:49 PM, Kirill Galinurov wrote: > > Hi. All we try topology hiding module. > We have opensips 2.3/2 like sbc with to interfaces. > And ip addresses are X.X.X.X and y.y.y.y. > We have to softswitch with ip addreses z.z.z.1 and z.z.z.2 > Also we loadbalancing calls between softswitch . > My config > if(is_method("INVITE")) { > create_dialog(); > topology_hiding(); > if (check_address("1","$si","$sp","any","$avp(ctx)")){ > force_send_socket(udp:10.240.250.121:5126); > setflag(CALL_SW); > do_accounting("log", "cdr|failed"); > $dlg_val(callee) = $rU; > $dlg_val(calling) =$fU; > $dlg_val(callcount) = $var(size); > xlog("L_INFO","+++++Incoming call from $fU $sp $proto > \n"); > $acc_extra(src_ip) = $si; > $acc_leg(caller) = $fU; > $acc_leg(callee) = $rU; > if(!lb_start_or_next("1","st","s")){ > switch($rc){ > case -1: > xlog("L_INFO","error 500 General dispatch > error\n"); > send_reply("503", "No resource avaiable"); > acc_log_request("503 General dispatch error"); > break; > > case -2: > xlog("L_INFO", "503 No free resources on > nodes\n"); > send_reply("503", "No resource avaible"); > acc_log_request("503 No free resource on > nodes"); > break; > case -3: > xlog("L_INFO","606 No nodes avaible\n"); > send_reply("503", "No resource avaible"); > acc_log_request("606 No nodes avaible"); > break; > } > } else { > xlog("L_INFO","Dispatch request to softswitch > $du\n"); > $avp(node) = $du; > $acc_extra(dst_node) = $du; > t_on_failure("fail_softswitch"); > route(relay); > } > } > else > if (check_address("2","$Ri","$Rp","any","$avp(ctx)")){ > setflag(CALL_FROM_SW); > do_accounting("log", "cdr|failed"); > $dlg_val(callee) = $rU; > $dlg_val(calling) = $fU; > $dlg_val(callcount) = $var(size); > xlog("L_INFO","+++++Outgouing call from $fU $sp $proto > \n"); > $acc_extra(src_ip) = $si; > $acc_leg(caller) = $fU; > $acc_leg(callee) = $rU; > if(!lb_start_or_next("$avp(ctx)","st","s")){ > switch($rc){ > case -1: > xlog("L_INFO","error 500 General dispatch > error\n"); > send_reply("503", "No resource avaiable"); > acc_log_request("503 General dispatch error"); > break; > > case -2: > xlog("L_INFO", "503 No free resources on > operator\n"); > send_reply("503", "No resource avaible"); > acc_log_request("503 No free resource on > operator"); > break; > case -3: > xlog("L_INFO","606 No resources avaible\n"); > send_reply("503", "No resource avaible"); > acc_log_request("606 No nodes avaible"); > break; > } > } else { > xlog("L_INFO","Dispatch request to operator sbc > $du\n"); > $avp(node) = $du; > $acc_extra(dst_node) = $du; > t_on_failure("fail_operator"); > route(relay); > } > } > else > sl_send_reply("403", "Forbidden posible wrong port"); > > } > } > In invite on softswitch i see > NVITE sip:12345 at XX.XX.XX.XX:5126 SIP/2.0 > Via: SIP/2.0/UDP YY.YY.YY.YY:5126;branch=z9hG4bKe4d9.14b51d57.0 > Max-Forwards: 69 > From: "Anonymous" ;tag=as59cc9e35 > To: > Contact: > Call-ID: 0068dba856ae05e8406504ab71fc9900 at CC.CC.CC.CC.CC > CSeq: 102 INVITE > > Where CC is client ip. I thithk it must be opensips ip adress > 10.240.250.121:5126 > What is wrong? > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Nov 17 04:09:35 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 17 Nov 2017 11:09:35 +0200 Subject: [OpenSIPS-Users] call_center module in Opensips 2.3 In-Reply-To: References: Message-ID: <7aa9ba5f-6fc3-6caa-1e6f-74c8494f26f2@opensips.org> Hi Dragomir, There is a difference between a Call Center service and the Call Center module - the Call Center module in OpenSIPs is a call queuing with distribution. And I didn't mention adding more servers, I said to use 2 different opensips instances (on same server, different port), just for a better script separation and to simplify your work. Of course, you can still do it with a single instance, as Razvan suggested, if you have some good scripting skills. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/16/2017 04:57 PM, Dragomir Haralambiev wrote: > Hi Bogdan, > > Thanks for your replay. > > When speek about callcenter service I think the basic functionality > must be: outgoing calls, incoming calls, voice recording and > transcoding (change the voice codec). > > How to solve this case? > We have the option to choose one of the following two solutions. > > 1. Increase the number of servers (what you offer). > > To do this I must to add other one server(service) for recording and > transcoding like Asterisk or Freeswitch. > > Opensips_1 <-----> Opensips_2 <------> Asterisk or FreeSwitch > > Total Efficiency = > Efficiency(Opensips_1).Efficiency(Opensips_2).Efficiency(Asterisk)=0,9 > . 0,9 . 0,9 =0.72 (72%). > > This shows theoretically that a horizontal increase in servers will > result in a loss of 25% working capacity. > > 2. To use one server (service) to solve this case. > > Until now I have been using Opensipts, but he can not solve these > problems. > Unfortunately I have to use Asterisk or Freeswich for callcenter service. > > Best regards, > Dragomir > > 2017-11-16 14:10 GMT+02:00 Bogdan-Andrei Iancu >: > > Dragomir, > > The Call Center module handles the agents as simple SIP URIs. if > in your case the agents are actually registering devices, this is > a different matter. What I suggest, for simplicity reasons is to > use 2 opensips instances. > > The first to be a dedicated call center seeing the agents as AoRs > (sip:user at domain) where the domain points to the IP of the second > server > > The second instance to be simple registrar server for the "domain" > where the agents are registering their devices. > > In terms of call you will have: > > ---(call to queue)----> OPENSIPS1 -----(call to agent > user at domain)----> OPENSIPS2 -----(call to registered device)-----> > AGENT DEVICE > > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 11/10/2017 04:21 PM, Dragomir Haralambiev wrote: >> Hi, >> >> I try your idea to loop the INVITE back to Opensips and do a >> lookup(). I pit into "location" field from cc_agents table this >> one "sip:localnumber at Opensips_IP:Opensips_port". >> >> Call is connected but I receive follow log message: >> >> ERROR:b2b_entities:b2b_prescript_f: No dialog found, callid= >> [B2B.337.5360955.1510320840], method=OPTIONS >> >> When call is finished record from dialog tables is not deleted. >> >> Where is problem? >> >> Regards, >> Dragomir >> >> 2017-11-06 11:37 GMT+02:00 Răzvan Crainea > >: >> >> Hi, Hristo! >> >> 1. You are right, b2b_entities should be loaded as a >> dependency for b2b_logic. You may argue that it should also >> be listed as a dependency to call_center as well, but indirectly. >> >> 2. Yes, you should populate the tables with the flows you >> want to support and the agents that are going to use the call >> center. >> >> 3. There is currently no way to directly set the location of >> the agent dynamically. However, we can do some work-arounds, >> for example always advertise yourself as the location of the >> user, and for each call you can loop the INVITE back to you >> and do a lookup(). Or simply use an external registrar service. >> >> Best regards, >> >> Răzvan Crainea >> OpenSIPS Developer >> www.opensips-solutions.com >> >> On 11/05/2017 11:06 PM, Hristo Donev wrote: >>> Hello All, >>> >>> I try to implement call_center module in Opensips 2.3. >>> Here are my questions? >>> >>> 1. From the documentation 1.3.1: >>> "b2b_logic" and "database" must be loaded before callcenter >>> module. >>> >>> I found out that "b2b_entities" should be loaded as well, >>> otherwise the opensips does not start. >>> >>> Is this my mistake or a documentation error? >>> >>> 2. Opensips can not start if the tables "cc_agents" and >>> "cc_flow" are empty. >>> Why it cannot start? Should they be populated with some data? >>> >>> 3. Table "cc_agents", field "location" should point to the >>> SIP agent location. >>> If the agent's IP and port are changed dynamically - >>> how to copy automatically "contact" field from "location" >>> table to "cc_agents" field "location"? >>> >>> I would be grateful to receive a response for my questions. >>> >>> Thanks in advance, >>> Hristo Donev >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Nov 17 04:28:18 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 17 Nov 2017 11:28:18 +0200 Subject: [OpenSIPS-Users] Topology Hiding Presence Proxy In-Reply-To: References: Message-ID: <68aeb282-9df4-fcb9-a006-cd9895ed195b@opensips.org> Hi Nathan, Based on your description you do not want topology-hiding (to simply hide IPs from the end-points), but a mid-presence agent, right ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/17/2017 08:13 AM, Nathan Baker wrote: > Hello, > > I have been looking for some examples on how to use the topology > hiding module and proxying presence (SUBSCRIBE, NOTIFY, etc.) > together, similar to how the mid_registrar module would handle > registrations. Does anyone have any examples or suggestions on how to > do this? > > I have the clients subscribing to the server (through OpenSIPS proxy > to another presence server), but I'm not sure how to store the > subscriptions and how to route the presence messages. For > registrations you can just do save("location"), but is there an > equivalent for subscriptions/presence? It seems like the > handle_subscribe() function from the presence module wouldn't apply > because it's part of a presence server, or can you just use it for > storing subscriptions but never publish anything? I don't see a > function to look up these stored subscriptions. > > Would it be better to just assume that the subscriptions are valid and > relay the messages? If so, what is the best way to handle the routing? > > Any help or insight would be greatly appreciated! > > Thanks, > Nate > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Nov 17 04:33:01 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 17 Nov 2017 11:33:01 +0200 Subject: [OpenSIPS-Users] Getting Wrong Time Stamp from $Tsm In-Reply-To: References: <70b4b6ab-f944-1397-0d34-e6928e34ed1f@opensips.org> Message-ID: <058a0751-9b28-3aa8-7767-8f5af15778e4@opensips.org> Hi Diptesh, Good catch - I almost forgot about that function :). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/14/2017 04:47 PM, Dipteshkumar Patel wrote: > hello all, > > I got the option to get right time stamp both in seconds and > microseconds and that is > > > get_timestamp(sec_avp,usec_avp); > > which uses gettimeofday() system call internally. > > refer following link > https://www.opensips.org/Documentation/Script-CoreFunctions-2-3 > > *Diptesh Patel* > Jr. Software Developer > Ecosmob Technologies Ltd > Ahmedabad > Mo:*+919898962659* > > On Wed, Nov 1, 2017 at 3:59 PM, Bogdan-Andrei Iancu > > wrote: > > Hi Diptesh, > > Unfortunately the docs are a bit misleading - the supported format > for the $time() var is given by "man strftime" - and strftime has > no support for time units smaller than the second. This is why the > %N is not evaluated. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 10/31/2017 03:32 PM, Dipteshkumar Patel wrote: >> Hello sir, >> >> Thanks for your prompt reply >> >> I used *$time(+%s%N)* but it uses only one format and give the >> result "*+1509451841%N*" but i want to whole time stamp. >> >> This works fine from command line using *#time +%s%N.* >> * >> * >> * >> * >> >> >> *Diptesh Patel* >> Jr. Software Developer >> Ecosmob Technologies Ltd >> Ahmedabad >> Mo:*+919898962659* >> >> On Tue, Oct 31, 2017 at 5:12 PM, Bogdan-Andrei Iancu >> > wrote: >> >> Hi, >> >> Have you looked at the $time(format) script variable: >> >> http://www.opensips.org/Documentation/Script-CoreVar-2-3#toc87 >> >> >> You can take the all the time fields at once, and use a >> transformation to split them if you need the fields in >> separate variables. >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> >> On 10/31/2017 10:36 AM, Dipteshkumar Patel wrote: >>> Dear sir, >>> >>> I am using Opensips version 2.3.1 >>> >>> I am using $Ts and $Tsm for time stamp for CDR but the >>> microsecond time stamp is not same as system time stamp. To >>> resolve the issue i am using exec() to get the system time >>> but i takes so much time (nearly 5ms ) whenever so many >>> calls concurrently running on server(We need exact time >>> because it will affect the call duration). >>> Can you please suggest me solution for the issue ? >>> >>> Thanks >>> >>> *Diptesh Patel* >>> Jr. Software Developer >>> Ecosmob Technologies Ltd >>> Ahmedabad >>> Mo:*+919898962659* >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Nov 17 05:55:05 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 17 Nov 2017 12:55:05 +0200 Subject: [OpenSIPS-Users] [OpenSIPS-News] Introducing OpenSIPS 2.4 In-Reply-To: <972d9f71-89d4-89f1-9183-f7fc0f1c506a@opensips.org> References: <972d9f71-89d4-89f1-9183-f7fc0f1c506a@opensips.org> Message-ID: Heads up, on next Tuesday, 21st of November 2017, 4 pm GMT , we have the audio conference on OpenSIPS 2.4, hosted on https://www.uberconference.com/opensips . Be there, so you will be up2date !! Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/01/2017 07:16 PM, Bogdan-Andrei Iancu wrote: > One more year, one more evolution cycle, one more OpenSIPS major > release. So let me introduce you the upcoming OpenSIPS 2.4 . > > For the OpenSIPS 2.4 release we decided to focus on the */clustering > abilities/*. Today’s VoIP world is getting more and more dynamic, > services are moving into Clouds and more and more flexibility is > needed for the application to fully exploit such environments. But > let’s pin point the main reasons for going for a clustered approach : > > * scaling up with the processing/traffic load > * geographical distribution > * redundancy and High-Availability > > For the OpenSIPS 2.4 we laid down a roadmap that addresses the > clustering both from the clustering engine itself (the underlayer) and > from the functionalities that will perform on top of the clustering > layer, to share data and state. > > With OpenSIPS 2.4, it will never be easier to built a consistent and > powerful clustered solution: > > * *clustering engine* – enhances the capabilities of controlling the > cluster topology, like re-routing for bypassing broken links, > dynamic joining of new nodes, support for multiple capabilities > per node, data syncing between nodes and many more; > * *distributed user location* – this is a very complex topic as it > exceeds the simple concept of data sharing. By the nature of the > data (the user registrations), you may have different constraints > on how data is roaming in a cluster – registrations may be tied to > a node due NAT or TCP constraints. Even more, a sharding aspect > must be addressed when looking at distributing the pinging effort > across the cluster. So, multiple solutions are viable here, > depending on what is to be achieved (scaling, redundancy) and what > are the network constraints – see a detailed presentation > > of the available solutions; > * *distributed presence server* – quite similar (but less complex) > as the distributed user location, a distributed presence server > provides a consistent, but distributed way of sharing presence > information – SIP entities may publish data via different nodes in > the SIP cluster, while the subscribers may fetch presence data via > multiple various nodes. Two approaches are under work : (a) a > cluster built around a noSQL DB based as primary data storage and > (b) a cluster exclusively relying on OpenSIPS for data sharing; > * *anycast support* – to be able to build a fully-flavored anycast > support (addressing both redundancy and balancing) requires > OpenSIPS to replicate/share transaction state across the nodes in > the cluster (nodes sharing the same anycast IP). Depending on the > nature of the replication (full transaction versus transaction > meta-data) a full anycast and light anycast support will be > available – here is a detailed description on the anycast > > support; > * *clustered media relays* – as OpenSIPS has the ability to work > together with several flavors of media relays (such as RTPproxy, > RTPEngine, MediaProxy), the clustering support will help OpenSIPS > do distributed load-balancing over the relays – even if a relay is > used by multiple nodes in the cluster, all the nodes will share > information on the load on the relay, to avoid overloading or idle > time; > * *distributed call center* – an agent is able to register with > multiple queues on different nodes on a cluster. Still, all the > queues do share the status / availability of the agent and its > statistics for call distribution; > * *custom clustering* – the OpenSIPS clustering underlayer provides > at script level the ability to broadcast (in the cloud) or send to > a given node a custom message/action (with replying possibility) – > this is a very flexible and powerful way to build your custom > distributed functionality directly at script level. > > And because we started on the integration path with OpenSIPS 2.3, and > because we did it well, we decided to push forward on this path with > the 2.4 version as well: > > * more *Homer integration *to be able to report TCP statistics, DB > events and media relay events via HEP; > * *SIPREC integration* for standard call recording. The new SIPREC > module > > provides a standard and transparent (for the call parties) way to > do call recording against an external recorder like Oreka > provided by Orecx > ; > * more *FreeSWITCH integration* in order to capture the call-events > (DTMFs, call status) from FreeSWITCH into OpenSIPS script or for > being able to control a FreeSWITCH call from OpenSIPS script via ESL > * *Asterisk flavored* Load-Balancing for a more realistic and > accurate traffic balancing over Asterisk clusters (as the load > information is fetched in realtime from Asterisk); > > ------------------------------------------------------------------------ > > The timeline for OpenSIPS 2.4 is: > > * Beta Release – 12-16 March 2018 > * Stable Release – 23-27 April 2018 > * General Availability – 1st of May 2018, during OpenSIPS Summit > 2018 > > To talk more about the features of this new release, a public audio > conference will be > available on 21st of November 2017, 4 pm GMT > , thanks to > the kind sponsorship of UberConference > . Anyone is welcome to join to find > out more details or to ask questions about OpenSIPS > 2.4 . > > This is a public and open conference, so no registration is needed, > but if you want to announce your intention to participate, please let > us know here: > > http://blog.opensips.org/2017/11/01/introducing-opensips-2-4/ > > > Best regards, > > -- > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > > _______________________________________________ > News mailing list > News at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/news -------------- next part -------------- An HTML attachment was scrubbed... URL: From dangngocvu4 at gmail.com Wed Nov 15 22:46:26 2017 From: dangngocvu4 at gmail.com (=?utf-8?B?Tmfhu41jIFbFqSDEkOG6t25n?=) Date: Thu, 16 Nov 2017 10:46:26 +0700 Subject: [OpenSIPS-Users] Module Redis in Opensips 2.3 Version Message-ID: <460A0AB0-29C9-47A6-BE65-D3C73F0591BF@gmail.com> Dear all! i’m Vu, recently i have upgrade my project from opensips 2.2.2 to opensips 2.3 but i research opensips 2.3 don’t have file CACHEDB_REDIS.SO also packed: opensips-redis.x86_64 like opensips 2.2.2 . So i want to know how to install module redis in opensips 2.3, please! Thanks for read! -------------- next part -------------- An HTML attachment was scrubbed... URL: From bakern at gmail.com Fri Nov 17 08:57:21 2017 From: bakern at gmail.com (Nathan Baker) Date: Fri, 17 Nov 2017 08:57:21 -0500 Subject: [OpenSIPS-Users] Topology Hiding Presence Proxy In-Reply-To: <68aeb282-9df4-fcb9-a006-cd9895ed195b@opensips.org> References: <68aeb282-9df4-fcb9-a006-cd9895ed195b@opensips.org> Message-ID: Hi Bogdan, Yes, I think that's correct, although I didn't see any module that would implement a mid-presence agent. I don't so much need to hide IPs, but I do need to change the Contact header, so I started using topology_hiding for that since it has it built in. I originally tried to use mid_registrar but stopped for now because it doesn't store locations in the database and doesn't support presence (yet). Any insight or suggestions would be greatly appreciated, I would basically be fine with a B2BUA if there was one that did forward registrations and presence. Thanks, Nate On Nov 17, 2017 4:28 AM, "Bogdan-Andrei Iancu" wrote: Hi Nathan, Based on your description you do not want topology-hiding (to simply hide IPs from the end-points), but a mid-presence agent, right ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/17/2017 08:13 AM, Nathan Baker wrote: Hello, I have been looking for some examples on how to use the topology hiding module and proxying presence (SUBSCRIBE, NOTIFY, etc.) together, similar to how the mid_registrar module would handle registrations. Does anyone have any examples or suggestions on how to do this? I have the clients subscribing to the server (through OpenSIPS proxy to another presence server), but I'm not sure how to store the subscriptions and how to route the presence messages. For registrations you can just do save("location"), but is there an equivalent for subscriptions/presence? It seems like the handle_subscribe() function from the presence module wouldn't apply because it's part of a presence server, or can you just use it for storing subscriptions but never publish anything? I don't see a function to look up these stored subscriptions. Would it be better to just assume that the subscriptions are valid and relay the messages? If so, what is the best way to handle the routing? Any help or insight would be greatly appreciated! Thanks, Nate _______________________________________________ Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Fri Nov 17 09:25:25 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Fri, 17 Nov 2017 16:25:25 +0200 Subject: [OpenSIPS-Users] Module Redis in Opensips 2.3 Version In-Reply-To: <460A0AB0-29C9-47A6-BE65-D3C73F0591BF@gmail.com> References: <460A0AB0-29C9-47A6-BE65-D3C73F0591BF@gmail.com> Message-ID: <60dab731-b47b-b5f8-f1bb-92731d6c4451@opensips.org> Hi, Vu! You should have packages for redis. What OS are you using? Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/16/2017 05:46 AM, Ngọc Vũ Đặng wrote: > Dear all! > i’m Vu,    recently i have upgrade my project from opensips 2.2.2 to > opensips 2.3 but i research opensips 2.3 don’t have file CACHEDB_REDIS.SO > also packed:  opensips-redis.x86_64 like opensips 2.2.2 . So i want to > know how to install module redis in opensips 2.3, please! > Thanks for read! > > > _______________________________________________ > 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 yukko.sergey at gmail.com Sat Nov 18 16:30:50 2017 From: yukko.sergey at gmail.com (SK) Date: Sat, 18 Nov 2017 23:30:50 +0200 Subject: [OpenSIPS-Users] high availability rtpproxy with opensips Message-ID: Hello I have configured simple AOR and dialogs replication using a clusterer module. 2 boxes with opensips (SRV1 and SRV2) work as active-standby sharing one IP-address (keepalived). Now I want to enable high availability for rtpproxy. How can I send commands from SRV1 to both SRV1 and SRV2 rtpproxy instance, so SRV2 immediately starts to forward RTP packets of the active call in case SRV1 fails inside the active call and IP is moved from SRV1 to SRV2? Does dialog mirroring done by clusterer module makes this for me automatically in case both servers have identical nathelper module setup? -------------- next part -------------- An HTML attachment was scrubbed... URL: From callum.guy at x-on.co.uk Mon Nov 20 04:09:20 2017 From: callum.guy at x-on.co.uk (Callum Guy) Date: Mon, 20 Nov 2017 09:09:20 +0000 Subject: [OpenSIPS-Users] high availability rtpproxy with opensips In-Reply-To: References: Message-ID: My understanding is that this is not straightforward (or even possible?) with RTPProxy as it does not share any state with additional instances meaning calls could use either instance from either server however they cannot switch back and forward between proxies during a call session. You may find it worthwhile to investigate a solution which provides shared state such as RTPEngine which can use Redis to maintain a shared state between instances. http://www.opensips.org/html/docs/modules/2.1.x/rtpengine On Sat, Nov 18, 2017 at 9:34 PM SK wrote: > Hello > > I have configured simple AOR and dialogs replication using a clusterer > module. 2 boxes with opensips (SRV1 and SRV2) work as active-standby > sharing one IP-address (keepalived). > > Now I want to enable high availability for rtpproxy. > > How can I send commands from SRV1 to both SRV1 and SRV2 rtpproxy instance, > so SRV2 immediately starts to forward RTP packets of the active call in > case SRV1 fails inside the active call and IP is moved from SRV1 to SRV2? > > Does dialog mirroring done by clusterer module makes this for me > automatically in case both servers have identical nathelper module setup? > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Callum Guy Head of Information Security X-on -- *0333 332 0000 | www.x-on.co.uk | ** * X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales. Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD. Company Registration No. 2578478. The information in this e-mail is confidential and for use by the addressee(s) only. If you are not the intended recipient, please notify X-on immediately on +44(0)333 332 0000 and delete the message from your computer. If you are not a named addressee you must not use, disclose, disseminate, distribute, copy, print or reply to this email. Views or opinions expressed by an individual within this email may not necessarily reflect the views of X-on or its associated companies. Although X-on routinely screens for viruses, addressees should scan this email and any attachments for viruses. X-on makes no representation or warranty as to the absence of viruses in this email or any attachments. -------------- next part -------------- An HTML attachment was scrubbed... URL: From nferreira at fuze.com Mon Nov 20 04:22:22 2017 From: nferreira at fuze.com (Nuno Ferreira) Date: Mon, 20 Nov 2017 09:22:22 +0000 Subject: [OpenSIPS-Users] high availability rtpproxy with opensips In-Reply-To: References: Message-ID: You can use rtp_cluster (https://github.com/sippy/rtp_cluster) to create a cluster of rtpproxy servers/instances. From an OpenSIPS point of view it's the same of using a single rtpproxy instance. A seg, 20/11/2017, 09:11, Callum Guy escreveu: > My understanding is that this is not straightforward (or even possible?) > with RTPProxy as it does not share any state with additional instances > meaning calls could use either instance from either server however they > cannot switch back and forward between proxies during a call session. > > You may find it worthwhile to investigate a solution which provides shared > state such as RTPEngine which can use Redis to maintain a shared state > between instances. > http://www.opensips.org/html/docs/modules/2.1.x/rtpengine > > On Sat, Nov 18, 2017 at 9:34 PM SK wrote: > >> Hello >> >> I have configured simple AOR and dialogs replication using a clusterer >> module. 2 boxes with opensips (SRV1 and SRV2) work as active-standby >> sharing one IP-address (keepalived). >> >> Now I want to enable high availability for rtpproxy. >> >> How can I send commands from SRV1 to both SRV1 and SRV2 rtpproxy >> instance, so SRV2 immediately starts to forward RTP packets of the active >> call in case SRV1 fails inside the active call and IP is moved from SRV1 to >> SRV2? >> >> Does dialog mirroring done by clusterer module makes this for me >> automatically in case both servers have identical nathelper module setup? >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > -- > Callum Guy > Head of Information Security > X-on > > > *0333 332 0000 | www.x-on.co.uk | ** > > * > X-on is a trading name of Storacall Technology Ltd a limited company > registered in England and Wales. > Registered Office : Avaland House, 110 London Road, Apsley, Hemel > Hempstead, Herts, HP3 9SD. Company Registration No. 2578478. > The information in this e-mail is confidential and for use by the > addressee(s) only. If you are not the intended recipient, please notify > X-on immediately on +44(0)333 332 0000 and delete the > message from your computer. If you are not a named addressee you must not > use, disclose, disseminate, distribute, copy, print or reply to this email. Views > or opinions expressed by an individual > within this email may not necessarily reflect the views of X-on or its > associated companies. Although X-on routinely screens for viruses, > addressees should scan this email and any attachments > for viruses. X-on makes no representation or warranty as to the absence of > viruses in this email or any attachments. > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- *Nuno Ferreira* | Architect, CoreUC | nferreira at fuze.com | +351 308805903 Rua Carlos Silva Melo Magalhães 23, 3800-126 Aveiro, Portugal -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.* -------------- next part -------------- An HTML attachment was scrubbed... URL: From marketing at opensips.org Mon Nov 20 09:25:44 2017 From: marketing at opensips.org (OpenSIPS Team) Date: Mon, 20 Nov 2017 16:25:44 +0200 Subject: [OpenSIPS-Users] [BLOG] Deep Dive into the OpenSIPS Statistics Support Message-ID: <6347cef8-03fd-69b6-da2d-17860b07fa58@opensips.org> Interested in learning about concepts such as architecture and advanced uses for the OpenSIPS statistics? Then make sure to check out our latest blog post: https://blog.opensips.org/2017/11/20/deep-dive-into-the-opensips-statistics-support/ -- OpenSIPS Team https://opensips.org From miha at softnet.si Mon Nov 20 09:38:13 2017 From: miha at softnet.si (Miha) Date: Mon, 20 Nov 2017 15:38:13 +0100 Subject: [OpenSIPS-Users] issue with compiling opensips In-Reply-To: References: <44c9cc15-817d-0bd7-9189-ceaab5aee979@softnet.si> <319fde66-a20d-f15a-adab-8bf7885ea8f1@softnet.si> Message-ID: <6f0f49de-cbd1-5dca-2827-d8444804b965@softnet.si> hello is there on www where written which libraries are all mandatory (debian)? tnx miha On 17/11/2017 08:42, Ketan Kothari wrote: > Hello Miha, > > Please verify below dependency installed on your server or not. > > libjson-c-dev > libjson-c-doc > libjson-c2 > libjson-c2-dbg > libjson0 > libjson0-dev > > > > On Fri, Nov 17, 2017 at 12:58 PM, Miha > wrote: > > hey > > this is what i have: > > cat /etc/issue > Debian GNU/Linux 9 \n \ > > > apt list libjson* > Listing... Done > libjson-any-perl/stable 1.39-1 all > libjson-c-dev/stable,now 0.12.1-1.1 amd64 [installed] > libjson-c-doc/stable 0.12.1-1.1 all > libjson-c3/stable,now 0.12.1-1.1 amd64 [installed,automatic] > libjson-glib-1.0-0/stable,now 1.2.6-1 amd64 [installed,automatic] > libjson-glib-1.0-common/stable,now 1.2.6-1 all [installed,automatic] > libjson-glib-dev/stable 1.2.6-1 amd64 > libjson-glib-doc/stable 1.2.6-1 all > libjson-java/stable 2.4-3 all > libjson-maybexs-perl/stable 1.003008-1 all > libjson-multivalueordered-perl/stable 0.005-1 all > libjson-perl/stable 2.90-1 all > libjson-pointer-perl/stable 0.07-1 all > libjson-pp-perl/stable 2.27400-1 all > libjson-rpc-perl/stable 1.06-2 all > libjson-simple-doc/stable 1.1.1-4 all > libjson-simple-java/stable 1.1.1-4 all > libjson-smart-java/stable 2.2-1 all > libjson-types-perl/stable 0.05-1 all > libjson-validator-perl/stable 0.92+dfsg-1 all > libjson-webtoken-perl/stable 0.10-2 all > libjson-xs-perl/stable 3.030-1 amd64 > libjsoncpp-dev/stable,now 1.7.4-3 amd64 [installed] > libjsoncpp-doc/stable 1.7.4-3 all > libjsoncpp1/stable,now 1.7.4-3 amd64 [installed,automatic] > libjsonm-ocaml/stable 0.9.1-2 amd64 > libjsonm-ocaml-dev/stable 0.9.1-2 amd64 > libjsonm-ocaml-doc/stable 0.9.1-2 all > libjsonp-java/stable 1.0.4-1 all > libjsonp-java-doc/stable 1.0.4-1 all > libjsonpath-java/stable 2.0.0-3 all > libjsonrpccpp-client0/stable 0.7.0-1+b2 amd64 > libjsonrpccpp-client0-dbg/stable 0.7.0-1+b2 amd64 > libjsonrpccpp-common0/stable 0.7.0-1+b2 amd64 > libjsonrpccpp-common0-dbg/stable 0.7.0-1+b2 amd64 > libjsonrpccpp-dev/stable 0.7.0-1+b2 amd64 > libjsonrpccpp-server0/stable 0.7.0-1+b2 amd64 > libjsonrpccpp-server0-dbg/stable 0.7.0-1+b2 amd64 > libjsonrpccpp-stub0/stable 0.7.0-1+b2 amd64 > libjsonrpccpp-stub0-dbg/stable 0.7.0-1+b2 amd64 > libjsonrpccpp-tools/stable 0.7.0-1+b2 amd64 > > > > > On 16/11/2017 16:47, Răzvan Crainea wrote: >> What version of libjson are you using, and what OS are you running? >> Răzvan Crainea >> OpenSIPS Developer >> www.opensips-solutions.com >> On 11/16/2017 01:49 PM, Miha wrote: >>> hello Razvan >>> >>> no, i downloaded it from opensips on git, i did this yesterday (2.3) >>> >>> >>> br >>> miha >>> >>> On 16/11/2017 11:23, Răzvan Crainea wrote: >>>> Are you using an older version of OpenSIPS? This should have >>>> been fixed in all supported versions. >>>> >>>> Best regards, >>>> Răzvan Crainea >>>> OpenSIPS Developer >>>> www.opensips-solutions.com >>>> On 11/15/2017 01:52 PM, Miha wrote: >>>>> Hello >>>>> >>>>> which deb pack should I install regarding this error: >>>>> https://pastebin.com/c4RHMbcT >>>>> >>>>> I installed this package " libjson-c-dev" but it is not ok. >>>>> >>>>> >>>>> tnx >>>>> miha >>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From xaled at web.de Mon Nov 20 09:46:16 2017 From: xaled at web.de (xaled) Date: Mon, 20 Nov 2017 15:46:16 +0100 Subject: [OpenSIPS-Users] call_center module and loose routing Message-ID: <010301d3620e$526d8610$f7489230$@web.de> Hi I want to use opensips with call_center module in a loose route scenario where the INVITE has to be routed by opensips to the next hop (sipserver) based on what is left from Route header after loose_route() is performed. The call_center module has to update the RURI with agent location and let the next hop from the route header perform the actual routing to the agent. Here are the expected route and RURI header values at the opensips, sipserver and agent: opensips sipserver agent RURI: 0800CallCenter agent agent Route: opensips, sipserver sipserver Will something like this work as described above: loose_route(); cc_handle_call(test_flow); Thanks, xaled -------------- next part -------------- An HTML attachment was scrubbed... URL: From bakern at gmail.com Mon Nov 20 13:57:45 2017 From: bakern at gmail.com (Nathan Baker) Date: Mon, 20 Nov 2017 13:57:45 -0500 Subject: [OpenSIPS-Users] Topology Hiding Presence Proxy In-Reply-To: References: <68aeb282-9df4-fcb9-a006-cd9895ed195b@opensips.org> Message-ID: I just wanted to follow up on this one more time with some additional details. In the topology hiding tutorial ( http://www.opensips.org/Documentation/Tutorials-Topology-Hiding) it mentions that presence dialogs should be supported on top of the TM module: "When running strictly on top of the TM module, the topology hiding SIP messages will be bigger when compared to the initial requests ( since OpenSIPS will encode all the needed information in a parameter of the Contact header ), but all type of SIP requests and dialogs will be supported ( INVITE dialogs, Presence dialogs, SIP MESSAGE, etc )." When I attempt to do this, the SUBSCRIBE works fine, but when the NOTIFY comes back from the server I get errors when it does the topology_hiding_match(), usually something like: ERROR:core:parse_params: invalid character, ; expected, found o ERROR:core:do_parse_rr_body: failed to parse params ERROR:core:do_parse_rr_body: failed to parse RR headers ERROR:topology_hiding:topo_no_dlg_seq_handling: failed parsing route set I can see the th_contact_encode_param value in the request URI, but there is no route set in the NOTIFY message. Is it expecting that there is one? I guess I need to turn on debugging or dig into the topology hiding module code to see what's causing the errors. Any help would be greatly appreciated! Thanks, Nate On Fri, Nov 17, 2017 at 8:57 AM, Nathan Baker wrote: > Hi Bogdan, > > Yes, I think that's correct, although I didn't see any module that would > implement a mid-presence agent. I don't so much need to hide IPs, but I do > need to change the Contact header, so I started using topology_hiding for > that since it has it built in. I originally tried to use mid_registrar but > stopped for now because it doesn't store locations in the database and > doesn't support presence (yet). Any insight or suggestions would be > greatly appreciated, I would basically be fine with a B2BUA if there was > one that did forward registrations and presence. > > Thanks, > Nate > > > On Nov 17, 2017 4:28 AM, "Bogdan-Andrei Iancu" > wrote: > > Hi Nathan, > > Based on your description you do not want topology-hiding (to simply hide > IPs from the end-points), but a mid-presence agent, right ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 11/17/2017 08:13 AM, Nathan Baker wrote: > > Hello, > > I have been looking for some examples on how to use the topology hiding > module and proxying presence (SUBSCRIBE, NOTIFY, etc.) together, similar to > how the mid_registrar module would handle registrations. Does anyone have > any examples or suggestions on how to do this? > > I have the clients subscribing to the server (through OpenSIPS proxy to > another presence server), but I'm not sure how to store the subscriptions > and how to route the presence messages. For registrations you can just do > save("location"), but is there an equivalent for subscriptions/presence? > It seems like the handle_subscribe() function from the presence module > wouldn't apply because it's part of a presence server, or can you just use > it for storing subscriptions but never publish anything? I don't see a > function to look up these stored subscriptions. > > Would it be better to just assume that the subscriptions are valid and > relay the messages? If so, what is the best way to handle the routing? > > Any help or insight would be greatly appreciated! > > Thanks, > Nate > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From kurgan-rus at inbox.ru Tue Nov 21 03:36:16 2017 From: kurgan-rus at inbox.ru (=?UTF-8?B?QWxleGV5IEsu?=) Date: Tue, 21 Nov 2017 11:36:16 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?setflag_parameters?= Message-ID: <1511253376.959267934@f170.i.mail.ru> Hello all. It's about 'setflag' core function's documentation. For releases 2.3 and 2.4 it's written " The value of the parameter can be in range of 0..31 ". // setflag(13); But as I know, we can also use a string parameter. // setflag(TCP_PERSISTENT); So, it's like an error in documentation. It would be nice of you if you fix the documentation. Best regards, Alexey http://alexeyka.zantsev.com/   From marco.hierl at next-id.de Tue Nov 21 04:07:12 2017 From: marco.hierl at next-id.de (Marco Hierl) Date: Tue, 21 Nov 2017 09:07:12 +0000 Subject: [OpenSIPS-Users] rtp port changed in REPLY's SDP after announcement Message-ID: <58ac999d138842ec933c290b76ac188a@next-id.de> Dear all, I want to play an early announcement before connecting a call. The announcement can be heard by the caller, but the callee can't hear the caller after the call is connected! To be more precise: The openSIPS proxy receives a call from an external interconnection partner (public IP on external interface) and sends it on to an announcement server (private IP on internal interface). After REPLY 183 the announcement starts and the caller can hear the announcement. It finishes with REPLY603 that comes to the openSIPS failure route. Some stuff is done (e.g. changing R-URI), a new t_relay will be done and an INVITE is send to the callee (public IP on external interface). The callee sends 200ok, it's forwarded to the caller, that sends ACK like usual. But the caller does not send RTP to the IP/PORT indicated in the 200OK. The interconnection partner said that the problem is, that the RTP port in REPLY 183 is different to that in REPLY200 (not allowed, see RFC6337). (just to mention: the to-tag is different too) I'm using openSIPS 2.3 and in order to overcome the issue with internal and external addresses and also to keep the rtp streams I'm using rtpengine 5.4 (on a different machine) Is the change of the RTP Port allowed in this way? If not, what I can do better? Or is it a fault from rtp-engine, should it keep the first port? Any help is very appreciated! I tried to make a picture... maybe it helps a bit to understand what I was not able to explain above... No From To Message 1 external ownProxy Request: INVITE sip:+49xxx at us.de;user=phone 2 ownProxy external Status: 100 Giving a try rtpengine_offer(external internal replace-session-connection ICE=remove trust-address replace-origin) 3 to announcement server Request: INVITE sip:ANNC at mediaserver:5060;user=phone | 4 from announcement server Status: 100 Trying | 5 from announcement server Status: 183 Session Progress | o=root 1135152900 1135152900 IN IP4 mediaserver m=audio 13788 RTP/AVP 8 101 rtpengine_answer(internal external replace-session-connection ICE=remove trust-address replace-origin) 6 ownProxy external Status: 183 Session Progress o=root 1135152900 1135152900 IN IP4 ownProxy m=audio 23024 RTP/AVP 8 101 < ========== ANNOUNCEMENT ========= > 7 from announcement server Status: 603 Declined openSIPS failure route and rtpengine_offer(external external replace-session-connection ICE=remove trust-address replace-origin) and new t_relay 8 to callee Request: INVITE sip:bob at externalcarrier:5060;user=phone 9 from callee Status: 100 Trying 10 from callee Status: 200 OK o=- 239036129 823630844 IN IP4 10.0.203.7 m=audio 1044 RTP/AVP 8 101 rtpengine_answer(external external replace-session-connection ICE=remove trust-address replace-origin) 11 ownProxy external Status: 200 OK o=- 239036129 823630844 IN IP4 ownProxy m=audio 23056 RTP/AVP 8 101 < ========== RTP from B to A is ok, A send NO RTP to B ========= > 12 external ownProxy Request: ACK sip:ownProxy;did=5db.a2833e17 13 external ownProxy Request: BYE sip:ownProxy;did=5db.a2833e17 14 ownProxy external Status: 200 OK version: opensips 2.3.0 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. git revision: fe0ebcb45 main.c compiled on 09:02:02 Sep 14 2017 with gcc 6.3.0 rtpengine version 5.4.0.0+0~mr5.4.0.0 git-master-a9d55c6 Thanks and best regards!!! Marco -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Tue Nov 21 04:21:19 2017 From: liviu at opensips.org (Liviu Chircu) Date: Tue, 21 Nov 2017 11:21:19 +0200 Subject: [OpenSIPS-Users] setflag parameters In-Reply-To: <1511253376.959267934@f170.i.mail.ru> References: <1511253376.959267934@f170.i.mail.ru> Message-ID: <757be89f-79b5-abbc-44bb-727d7c11724c@opensips.org> Hi Alexey, Thank you for the hint - I just fixed the 2.4, 2.3 and 2.2 docs. Cheers, Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 21.11.2017 10:36, Alexey K. via Users wrote: > Hello all. > > It's about 'setflag' core function's documentation. > For releases 2.3 and 2.4 it's written " The value of the parameter can be in range of 0..31 ". // setflag(13); > But as I know, we can also use a string parameter. // setflag(TCP_PERSISTENT); > > So, it's like an error in documentation. > It would be nice of you if you fix the documentation. > > > Best regards, Alexey > http://alexeyka.zantsev.com/ > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From kurgan-rus at inbox.ru Tue Nov 21 04:24:20 2017 From: kurgan-rus at inbox.ru (=?UTF-8?B?QWxleGV5IEsu?=) Date: Tue, 21 Nov 2017 12:24:20 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?setflag_parameters?= In-Reply-To: <757be89f-79b5-abbc-44bb-727d7c11724c@opensips.org> References: <1511253376.959267934@f170.i.mail.ru> <757be89f-79b5-abbc-44bb-727d7c11724c@opensips.org> Message-ID: <1511256260.97723026@f175.i.mail.ru> Thank you. -- Best regards, Alexey http://alexeyka.zantsev.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Nov 21 04:34:09 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 21 Nov 2017 11:34:09 +0200 Subject: [OpenSIPS-Users] fix_nated_sdp and rtpproxy In-Reply-To: References: <268801e7-54c7-2693-c756-efd01dcd0717@opensips.org> Message-ID: Hi Yury, OK, so SIP goes like 192.168.1.1 / 1.1.1.1 -> 2.2.2.2 -> 3.3.3.3 , while RTP 192.168.1.1 / 1.1.1.1 -> 3.3.3.4/5 . or ? what is the full path for SIP and RTP ? Without the "r" flag, rtpproxy will stay and listen for incoming traffic in order to learn the end points, it will not use the SIP signaling IP for anything (rtpproxy makes no assumptions based on the SIP ips) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/16/2017 10:25 PM, Yury Kirsanov wrote: > Bogdan, > But in SIP packets from upstream carriers there's a public IP address > in SDP header, pointing to different IP than SIP packet is coming > from. Both of these IP addresses are public. But customers who connect > via public Internet to OpenSIPS server sitting on public interface are > actually behind NAT and that's why packets from them contain private > IP address in SDP header. Example, to clarify things up: > > Client has a PBX behind a router that's doing NAT. Internal IP of PBX > is 192.168.1.1, public IP of client is 1.1.1.1, router is doing just > NAT without SIP ALG. Customer connects over the Internet to OpenSIPS > at 2.2.2.2, so packets are reaching OpenSIPS. But in SDP of customer's > SIP packets there's c=192.168.1.1 and all the rest of descriptors are > with the same IP. > > Now, provider is at IP of 3.3.3.3 having RTP proxies at 3.3.3.4 and > 3.3.3.5. So, packets from provider are reaching OpenSIPS at 2.2.2.2 > via the Internet and contain different IP address in SDP from SIP > packet (3.3.3.4 or 3.3.3.5 in SDP, SIP has 3.3.3.3) > > RTP Proxy at my side is operating in bridge mode, one side is > connected to public interface with IP of 4.4.4.4 and 4.4.4.5 (two > proxies), and another side is connected to DMZ internal LAN, IPs > 10.0.0.4 and 10.0.0.5. > > So, in order to use fix_nated_sdp() AND engage RTP proxy when the call > is coming from customer - I should NOT use flag "r". But if the call > is coming from provider - I must be using that flag or otherwise RTP > Proxy would be expecting packets to be coming from 3.3.3.3 rather than > 3.3.3.4 or 3.3.3.5. That's why my idea was first to update customer's > part of SDP from 192.168.1.1 to 1.1.1.1 (where packets from customer > are coming from) and then pass this modified SIP packet with new SDP > to RTP proxy, so it would have flag "r" enabled and start sending > packets to IP of 1.1.1.1 straight away. > > Thanks and best regards, > Yury. > > > 2017-11-17 3:26 GMT+11:00 Bogdan-Andrei Iancu >: > > Using the "i" and "e" flags to control the RTP interfaces does not > affect the "learning" mode. You should you "r" flag only if the > received IP in SDP is public. Otherwise do not use it. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 11/16/2017 03:07 PM, Yury Kirsanov wrote: >> Hi Bogdan, >> Thanks for clarification, I didn't know that RTP Proxy learns >> from source packets, it's quite hard to find documentation on it. >> The problem here is that I'm using RTP Proxy in bridge mode, >> connecting public interface with internal LAN and my upstream >> providers are using sets of their own proxies too, besides >> customers connecting to this OpenSIPS server. So I have to use >> flag "r" (and, of course, flags of direction - "i" and "e") or >> else there would be no audio on calls when SDP in provider's >> answer is different from SIP packet source IP. Of course, I can >> analyze where the initial INVITE packet is coming from and enable >> different options but that would be more complex rather than just >> have SDP modified before passing it to RTP Proxy. >> Regards, >> Yury. >> >> 2017-11-17 0:01 GMT+11:00 Bogdan-Andrei Iancu >> >: >> >> Hi Yury, >> >> RTPproxy learns not only from SDP, but based on the incoming >> RTP traffic. Your Client will send RTP to RTPproxy (as the >> Client will receive back a valid IP in the SDP) and RTPproxy >> will see where the RTP traffic comes from and starts sending >> back RTP to that IP:port. >> >> Just be sure you do not use the "a" and "r" flags when >> calling rtpproxy engage. >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> >> On 11/16/2017 02:48 PM, Yury Kirsanov wrote: >>> Hi Bogdan, >>> Thanks for your answer and support! But the issue is that >>> there is a private IP address in SDP coming from the client >>> so if I just pass it via RTP proxy - it would learn that >>> incorrect IP and would try to forward packets to it causing >>> one-way audio. That's why I first wanted to overstamp that >>> incorrect IP with received IP and then pass it to the RTP >>> Proxy. Hope this makes my idea clear. >>> >>> Regards, >>> Yury. >>> >>> 2017-11-16 23:32 GMT+11:00 Bogdan-Andrei Iancu >>> >: >>> >>> Hi Yury, >>> >>> Why do you need to update the incoming SDP before >>> engaging RTPproxy ? Just pass to it whatever you get >>> into SDP (even if private) and let rtpproxy to "learn" >>> the correct IP:port of the media end points based on the >>> incoming traffic. >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> OpenSIPS Founder and Developer >>> http://www.opensips-solutions.com >>> >>> >>> On 11/14/2017 04:05 AM, Yury Kirsanov wrote: >>>> Hi, >>>> I have a problem with remote client who can't properly >>>> configure NAT and is sending private IP in all SDP and >>>> SIP packets. I've resolved issues with registration but >>>> can't resolve the issue with SDP. My topology is as >>>> following: >>>> >>>> ----- ------- >>>> ---------- -------- ----------- >>>> >>>> So, I need to simultaneously engage RTP Proxy to bridge >>>> external Internet and our local LAN and need to rewrite >>>> SDP body of original Client's packet with received IP >>>> address. If I'm doing fix_nated_sdp and then >>>> rtpproxy_offer - I'm getting doubled IPs in SDP >>>> message, like this: >>>> >>>> 100.11.100.200192.168.20.200 >>>> >>>> >>>> Is there any way to update SDP of client's packet and >>>> then command RTP Proxy to use updated IP? 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 Tue Nov 21 04:36:05 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 21 Nov 2017 11:36:05 +0200 Subject: [OpenSIPS-Users] Topology Hiding Presence Proxy In-Reply-To: References: <68aeb282-9df4-fcb9-a006-cd9895ed195b@opensips.org> Message-ID: Hi Nate, If you just need to change the Contact hdr, the topology-hiding will do the job - have you tried to simply use the the topology_hiding() function when routing the initial SUBSCRIBE request ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/17/2017 03:57 PM, Nathan Baker wrote: > Hi Bogdan, > > Yes, I think that's correct, although I didn't see any module that > would implement a mid-presence agent. I don't so much need to hide > IPs, but I do need to change the Contact header, so I started using > topology_hiding for that since it has it built in. I originally tried > to use mid_registrar but stopped for now because it doesn't store > locations in the database and doesn't support presence (yet). Any > insight or suggestions would be greatly appreciated, I would basically > be fine with a B2BUA if there was one that did forward registrations > and presence. > > Thanks, > Nate > > > On Nov 17, 2017 4:28 AM, "Bogdan-Andrei Iancu" > wrote: > > Hi Nathan, > > Based on your description you do not want topology-hiding (to > simply hide IPs from the end-points), but a mid-presence agent, > right ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 11/17/2017 08:13 AM, Nathan Baker wrote: >> Hello, >> >> I have been looking for some examples on how to use the topology >> hiding module and proxying presence (SUBSCRIBE, NOTIFY, etc.) >> together, similar to how the mid_registrar module would handle >> registrations. Does anyone have any examples or suggestions on >> how to do this? >> >> I have the clients subscribing to the server (through OpenSIPS >> proxy to another presence server), but I'm not sure how to store >> the subscriptions and how to route the presence messages. For >> registrations you can just do save("location"), but is there an >> equivalent for subscriptions/presence? It seems like the >> handle_subscribe() function from the presence module wouldn't >> apply because it's part of a presence server, or can you just use >> it for storing subscriptions but never publish anything? I don't >> see a function to look up these stored subscriptions. >> >> Would it be better to just assume that the subscriptions are >> valid and relay the messages? If so, what is the best way to >> handle the routing? >> >> Any help or insight would be greatly appreciated! >> >> Thanks, >> Nate >> >> >> _______________________________________________ >> 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 21 04:55:05 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 21 Nov 2017 11:55:05 +0200 Subject: [OpenSIPS-Users] call_center module and loose routing In-Reply-To: <010301d3620e$526d8610$f7489230$@web.de> References: <010301d3620e$526d8610$f7489230$@web.de> Message-ID: Hi, The routing in Call Center module is not compatible with the loose_route(). The loose_route is a mechanism used by a SIP proxy staying in the middle of a SIP session. On the other hand, the Call Center is using a SIP Back-2-Back engine in order to implement the queuing. Better try to explain what are you trying to achieve here. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/20/2017 04:46 PM, xaled wrote: > > Hi > > I want to use opensips with call_center module in a loose route > scenario where the INVITE has to be routed by opensips to the next hop > (sipserver) based on what is left from Route header after > loose_route() is performed. The call_center module has to update the > RURI with agent location and let the next hop from the route header > perform the actual routing to the agent. > > Here are the expected route and RURI header values at the opensips, > sipserver and agent: > > opensips sipserver agent > > RURI: 0800CallCenter agent agent > > Route: opensips, sipserver sipserver > > Will something like this work as described above: > > loose_route(); > > cc_handle_call(test_flow); > > Thanks, > > xaled > > > > _______________________________________________ > 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 21 05:09:50 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 21 Nov 2017 12:09:50 +0200 Subject: [OpenSIPS-Users] rtp port changed in REPLY's SDP after announcement In-Reply-To: <58ac999d138842ec933c290b76ac188a@next-id.de> References: <58ac999d138842ec933c290b76ac188a@next-id.de> Message-ID: <70fdf3f1-9752-69e4-d5fc-96212d72f4ae@opensips.org> Hello Marco, Did the interconnection partner mentioned the chapter/paragraph in RFC6337 where the restriction on the port is mentioned (I was not able to find it on a quick scan) ? it is the first time I hear about anything like this. Now, if you use rtpproxy (I do not know about rtpengine) on your side and you do rtpproxy_offer only once (when getting the call from the carrier) , you can do rtpproxy_answer multiple time (during the serial forking). In this case, the RTP IP:port towards the carrier will never change, but the RTP callee leg will be changed. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/21/2017 11:07 AM, Marco Hierl wrote: > > Dear all, > > I want to play an early announcement before connecting a call. The > announcement can be heard by the caller, but the callee can't hear the > caller after the call is connected! > > To be more precise: > > The openSIPS proxy receives a call from an external interconnection > partner (public IP on external interface) and sends it on to an > announcement server (private IP on internal interface). > > After REPLY 183 the announcement starts and the caller can hear the > announcement. It finishes with REPLY603 that comes to the openSIPS > failure route. Some stuff is done (e.g. changing R-URI), > > a new t_relay will be done and an INVITE is send to the callee (public > IP on external interface). The callee sends 200ok, it's forwarded to > the caller, that sends ACK like usual. > > But the caller does not send RTP to the IP/PORT indicated in the 200OK. > > The interconnection partner said that the problem is, that the RTP > port in REPLY 183 is different to that in REPLY200 (not allowed, see > RFC6337). (just to mention: the to-tag is different too) > > I'm using openSIPS 2.3 and in order to overcome the issue with > internal and external addresses and also to keep the rtp streams I'm > using rtpengine 5.4 (on a different machine) > > Is the change of the RTP Port allowed in this way? If not, what I can > do better? Or is it a fault from rtp-engine, should it keep the first > port? > > Any help is very appreciated! > > I tried to make a picture... maybe it helps a bit to understand what I > was not able to explain above... > > No From To Message > > 1 external ownProxy Request: INVITE sip:+49xxx at us.de;user=phone > > 2 ownProxy external Status: 100 Giving a try > > rtpengine_offer(external > internal replace-session-connection ICE=remove trust-address > replace-origin) > > 3 to announcement > server Request: INVITE sip:ANNC at mediaserver:5060;user=phone | > > 4 from announcement server Status: 100 Trying | > > 5 from announcement server Status: 183 > Session Progress | > > o=root > 1135152900 1135152900 IN IP4 mediaserver > > m=audio 13788 RTP/AVP 8 101 > > rtpengine_answer(internal external replace-session-connection > ICE=remove trust-address replace-origin) > > 6 ownProxy external Status: 183 Session Progress > > o=root 1135152900 1135152900 IN > IP4 ownProxy > > m=audio 23024 RTP/AVP 8 101 > > < ========== ANNOUNCEMENT ========= > > > 7 from announcement server Status: 603 Declined > > openSIPS failure route > > and > rtpengine_offer(external external replace-session-connection > ICE=remove trust-address replace-origin) > > and new t_relay > > 8 to callee Request: INVITE > sip:bob at externalcarrier:5060;user=phone > > 9 from callee Status: 100 Trying > > 10 from callee Status: 200 OK > > o=- > 239036129 823630844 IN IP4 10.0.203.7 > > m=audio 1044 RTP/AVP 8 101 > > rtpengine_answer(external external replace-session-connection > ICE=remove trust-address replace-origin) > > 11 ownProxy external Status: 200 OK > > o=- 239036129 823630844 IN IP4 ownProxy > > m=audio 23056 RTP/AVP 8 101 > > < ========== RTP from B to A is ok, A send NO RTP to B > ========= > > > 12 external ownProxy Request: ACK sip:ownProxy;did=5db.a2833e17 > > 13 external ownProxy Request: BYE sip:ownProxy;did=5db.a2833e17 > > 14 ownProxy external Status: 200 OK > > version: opensips 2.3.0 (x86_64/linux) > > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > MAX_URI_SIZE 1024, BUF_SIZE 65535 > > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. > > git revision: fe0ebcb45 > > main.c compiled on 09:02:02 Sep 14 2017 with gcc 6.3.0 > > rtpengine version 5.4.0.0+0~mr5.4.0.0 git-master-a9d55c6 > > Thanks and best regards!!! > > Marco > > > > _______________________________________________ > 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 xaled at web.de Tue Nov 21 05:30:28 2017 From: xaled at web.de (xaled) Date: Tue, 21 Nov 2017 11:30:28 +0100 Subject: [OpenSIPS-Users] call_center module and loose routing In-Reply-To: References: <010301d3620e$526d8610$f7489230$@web.de> Message-ID: <007e01d362b3$c9b66c00$5d234400$@web.de> Hi Bogdan, I use opensips in an scenario where OPENSIPS is addressed from the main SIPSERVER using Route header. OPENSIPS performs different modifications on the INVITE request and gives the INVITE back to the main SIPSERVER. This works fine with the loose_route() so far. Now I want to use the call_center module for RURI agent addressing, queues, etc while still preserving the Route header based next hop routing between main SIPSERVER and OPENSIPS. Would manually parsing the Route Header at OPENSIPS, deleting first route header, saving the hostport value of the second Route header to some variable and provide it as a kind of $du to the call_center module for the next hop routing back to SIPSERVER work? For my scenario to work the Route header with the second Route header value has not to be deleted or modified by the call_center module. Here is the table showing the pseudo values of the RURI and the Route Header at the ingress of SIPSERVER and OPENSIPS respectively. SIPSERVER -> OPENSIPS -> SIPSERVER -> AGENT RURI: 0800CallCenter 0800CallCenter agent agent Route: opensips, sipserver sipserver Does this make it a bit more clearer? Thanks, xaled From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Tuesday, November 21, 2017 10:55 AM To: users at lists.opensips.org; xaled Subject: Re: [OpenSIPS-Users] call_center module and loose routing Hi, The routing in Call Center module is not compatible with the loose_route(). The loose_route is a mechanism used by a SIP proxy staying in the middle of a SIP session. On the other hand, the Call Center is using a SIP Back-2-Back engine in order to implement the queuing. Better try to explain what are you trying to achieve here. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/20/2017 04:46 PM, xaled wrote: Hi I want to use opensips with call_center module in a loose route scenario where the INVITE has to be routed by opensips to the next hop (sipserver) based on what is left from Route header after loose_route() is performed. The call_center module has to update the RURI with agent location and let the next hop from the route header perform the actual routing to the agent. Here are the expected route and RURI header values at the opensips, sipserver and agent: opensips sipserver agent RURI: 0800CallCenter agent agent Route: opensips, sipserver sipserver Will something like this work as described above: loose_route(); cc_handle_call(test_flow); Thanks, xaled _______________________________________________ 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 21 05:28:03 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 21 Nov 2017 12:28:03 +0200 Subject: [OpenSIPS-Users] Topology Hiding Presence Proxy In-Reply-To: References: <68aeb282-9df4-fcb9-a006-cd9895ed195b@opensips.org> Message-ID: <5c93b408-ac60-2c05-7a6d-c3ab674cfd38@opensips.org> Hi Nate, I missed this update from you - you did what I just asked in my last email :). Could you share (off-line if needed) the pcap for SUBSCRIBE+NOTIFY and the opensips logs in debug level ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/20/2017 08:57 PM, Nathan Baker wrote: > I just wanted to follow up on this one more time with some additional > details. In the topology hiding tutorial > (http://www.opensips.org/Documentation/Tutorials-Topology-Hiding) it > mentions that presence dialogs should be supported on top of the TM > module: > > "When running strictly on top of the TM module, the topology > hiding SIP messages will be bigger when compared to the initial > requests ( since OpenSIPS will encode all the needed information > in a parameter of the Contact header ), but all type of SIP > requests and dialogs will be supported ( INVITE dialogs, Presence > dialogs, SIP MESSAGE, etc )." > > > When I attempt to do this, the SUBSCRIBE works fine, but when the > NOTIFY comes back from the server I get errors when it does the > topology_hiding_match(), usually something like: > > ERROR:core:parse_params: invalid character, ; expected, found o > ERROR:core:do_parse_rr_body: failed to parse params > ERROR:core:do_parse_rr_body: failed to parse RR headers > ERROR:topology_hiding:topo_no_dlg_seq_handling: failed parsing route set > > I can see the th_contact_encode_param value in the request URI, but > there is no route set in the NOTIFY message. Is it expecting that > there is one? I guess I need to turn on debugging or dig into the > topology hiding module code to see what's causing the errors. Any > help would be greatly appreciated! > > Thanks, > Nate > > > On Fri, Nov 17, 2017 at 8:57 AM, Nathan Baker > wrote: > > Hi Bogdan, > > Yes, I think that's correct, although I didn't see any module that > would implement a mid-presence agent. I don't so much need to > hide IPs, but I do need to change the Contact header, so I started > using topology_hiding for that since it has it built in. I > originally tried to use mid_registrar but stopped for now because > it doesn't store locations in the database and doesn't support > presence (yet). Any insight or suggestions would be greatly > appreciated, I would basically be fine with a B2BUA if there was > one that did forward registrations and presence. > > Thanks, > Nate > > > On Nov 17, 2017 4:28 AM, "Bogdan-Andrei Iancu" > > wrote: > > Hi Nathan, > > Based on your description you do not want topology-hiding (to > simply hide IPs from the end-points), but a mid-presence > agent, right ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > > On 11/17/2017 08:13 AM, Nathan Baker wrote: >> Hello, >> >> I have been looking for some examples on how to use the >> topology hiding module and proxying presence (SUBSCRIBE, >> NOTIFY, etc.) together, similar to how the mid_registrar >> module would handle registrations. Does anyone have any >> examples or suggestions on how to do this? >> >> I have the clients subscribing to the server (through >> OpenSIPS proxy to another presence server), but I'm not sure >> how to store the subscriptions and how to route the presence >> messages. For registrations you can just do >> save("location"), but is there an equivalent for >> subscriptions/presence? It seems like the handle_subscribe() >> function from the presence module wouldn't apply because it's >> part of a presence server, or can you just use it for storing >> subscriptions but never publish anything? I don't see a >> function to look up these stored subscriptions. >> >> Would it be better to just assume that the subscriptions are >> valid and relay the messages? If so, what is the best way to >> handle the routing? >> >> Any help or insight would be greatly appreciated! >> >> Thanks, >> Nate >> >> >> _______________________________________________ >> 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 marco.hierl at next-id.de Tue Nov 21 05:49:52 2017 From: marco.hierl at next-id.de (Marco Hierl) Date: Tue, 21 Nov 2017 10:49:52 +0000 Subject: [OpenSIPS-Users] rtp port changed in REPLY's SDP after announcement In-Reply-To: <70fdf3f1-9752-69e4-d5fc-96212d72f4ae@opensips.org> References: <58ac999d138842ec933c290b76ac188a@next-id.de> <70fdf3f1-9752-69e4-d5fc-96212d72f4ae@opensips.org> Message-ID: Hi Bogdan, no, he did not say anything yet, but I expect that it is paragraph 3.2: "A UAS cannot send a new offer in the reliable provisional response, so the UPDATE method is the only method for a UAS to update an early session." (I'm not quite sure if this really fits to our situation, because the UAS itself changed.) According to your suggestion: I used "rtpengine_offer(external internal..." to adjust the SDP the first time. So I get an internal IP in the INVITE to the announcement. And this will be seen in the 2nd INVITE to the callee if I do NOT make "rtpengine_offer(external external..." to the INVITE to the callee. But the callee has public IP and I need to use the external interface. Thanks for your help! Marco Von: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Gesendet: Dienstag, 21. November 2017 11:10 An: OpenSIPS users mailling list; Marco Hierl Betreff: Re: [OpenSIPS-Users] rtp port changed in REPLY's SDP after announcement Hello Marco, Did the interconnection partner mentioned the chapter/paragraph in RFC6337 where the restriction on the port is mentioned (I was not able to find it on a quick scan) ? it is the first time I hear about anything like this. Now, if you use rtpproxy (I do not know about rtpengine) on your side and you do rtpproxy_offer only once (when getting the call from the carrier) , you can do rtpproxy_answer multiple time (during the serial forking). In this case, the RTP IP:port towards the carrier will never change, but the RTP callee leg will be changed. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/21/2017 11:07 AM, Marco Hierl wrote: Dear all, I want to play an early announcement before connecting a call. The announcement can be heard by the caller, but the callee can't hear the caller after the call is connected! To be more precise: The openSIPS proxy receives a call from an external interconnection partner (public IP on external interface) and sends it on to an announcement server (private IP on internal interface). After REPLY 183 the announcement starts and the caller can hear the announcement. It finishes with REPLY603 that comes to the openSIPS failure route. Some stuff is done (e.g. changing R-URI), a new t_relay will be done and an INVITE is send to the callee (public IP on external interface). The callee sends 200ok, it's forwarded to the caller, that sends ACK like usual. But the caller does not send RTP to the IP/PORT indicated in the 200OK. The interconnection partner said that the problem is, that the RTP port in REPLY 183 is different to that in REPLY200 (not allowed, see RFC6337). (just to mention: the to-tag is different too) I'm using openSIPS 2.3 and in order to overcome the issue with internal and external addresses and also to keep the rtp streams I'm using rtpengine 5.4 (on a different machine) Is the change of the RTP Port allowed in this way? If not, what I can do better? Or is it a fault from rtp-engine, should it keep the first port? Any help is very appreciated! I tried to make a picture... maybe it helps a bit to understand what I was not able to explain above... No From To Message 1 external ownProxy Request: INVITE sip:+49xxx at us.de;user=phone 2 ownProxy external Status: 100 Giving a try rtpengine_offer(external internal replace-session-connection ICE=remove trust-address replace-origin) 3 to announcement server Request: INVITE sip:ANNC at mediaserver:5060;user=phone | 4 from announcement server Status: 100 Trying | 5 from announcement server Status: 183 Session Progress | o=root 1135152900 1135152900 IN IP4 mediaserver m=audio 13788 RTP/AVP 8 101 rtpengine_answer(internal external replace-session-connection ICE=remove trust-address replace-origin) 6 ownProxy external Status: 183 Session Progress o=root 1135152900 1135152900 IN IP4 ownProxy m=audio 23024 RTP/AVP 8 101 < ========== ANNOUNCEMENT ========= > 7 from announcement server Status: 603 Declined openSIPS failure route and rtpengine_offer(external external replace-session-connection ICE=remove trust-address replace-origin) and new t_relay 8 to callee Request: INVITE sip:bob at externalcarrier:5060;user=phone 9 from callee Status: 100 Trying 10 from callee Status: 200 OK o=- 239036129 823630844 IN IP4 10.0.203.7 m=audio 1044 RTP/AVP 8 101 rtpengine_answer(external external replace-session-connection ICE=remove trust-address replace-origin) 11 ownProxy external Status: 200 OK o=- 239036129 823630844 IN IP4 ownProxy m=audio 23056 RTP/AVP 8 101 < ========== RTP from B to A is ok, A send NO RTP to B ========= > 12 external ownProxy Request: ACK sip:ownProxy;did=5db.a2833e17 13 external ownProxy Request: BYE sip:ownProxy;did=5db.a2833e17 14 ownProxy external Status: 200 OK version: opensips 2.3.0 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. git revision: fe0ebcb45 main.c compiled on 09:02:02 Sep 14 2017 with gcc 6.3.0 rtpengine version 5.4.0.0+0~mr5.4.0.0 git-master-a9d55c6 Thanks and best regards!!! Marco _______________________________________________ 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 diptesh.patel at ecosmob.com Tue Nov 21 08:08:24 2017 From: diptesh.patel at ecosmob.com (Dipteshkumar Patel) Date: Tue, 21 Nov 2017 18:38:24 +0530 Subject: [OpenSIPS-Users] Degrades the performance of opensips after 100k calls Message-ID: hello all I am using opensips-2.3.2 running on 16 cores of cpu and 32GB of RAM I am facing an isssue. After making 100k of calls, suddenly opensips performance degrades and it takes much time to process a packet. load average of cpu incresing significantly.after restarting opensips it works fine for another 100k calls then again problem appears. It is an SBC server, using 6-7 sql queries with async with database optimisation(using proper indexing), permission module for auth, aaa_radius module for accounting and some string operations. Summary of operations 1. Sanity checks 2. Use IP authentication 3. Send Requet for accounting 4. Some string operations for accounting I am sending global parameters and module parameters of my opensips config file Please find here https://pastebin.com/B7Y5HRgH Please suggest what is wrong and how to improve. *Diptesh Patel* Jr. Software Developer Ecosmob Technologies Ltd Ahmedabad Mo:*+919898962659* -------------- next part -------------- An HTML attachment was scrubbed... URL: From mas_66 at tiscali.co.uk Tue Nov 21 08:46:53 2017 From: mas_66 at tiscali.co.uk (mas_66 at tiscali.co.uk) Date: Tue, 21 Nov 2017 13:46:53 +0000 (GMT) Subject: [OpenSIPS-Users] Opensips 2.3.2 Message-ID: <26890667.2767301511272013887.JavaMail.defaultUser@defaultHost> Hello there I'm having an issue with placing external calls. I can register softphones and place calls between them fine, but have failed in the next step which is to hook up to my SBC(working) to call the outside world. On Opensips my interfaces (with fictional IP's are: 192.168.54.55 ENS160 (159.99.8.111 is my Public IP that points to LAN 192.168.54.55 for my softphones to register via) 172.23.100.55 ENS224(this communicates with an interface on my SBC 172.23.100.250) so a call path will be: 159.99.8.111->192.168.54.55->172.23.100.55->172. 23.100.250->carrier Im sure it has to do with the interface presented to my SBC which currently shows as extn111 at 192.168.54.55. the call hits my SBC but no further, or my routing set up. I want to force the call to be 111 at 172.23.100.55 but not sure how to. I have experimented with domains and routing but as a new Opensipper im guessing a lot of it and trying to read online too but not much out there that relates. If someone can offer some examples of Dynamic routing or anything that would help that would be great. Many Thanks From razvan at opensips.org Tue Nov 21 10:04:11 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Tue, 21 Nov 2017 17:04:11 +0200 Subject: [OpenSIPS-Users] Degrades the performance of opensips after 100k calls In-Reply-To: References: Message-ID: <0b878b98-54bc-f046-55a0-daf232e83746@opensips.org> Hello! I see that you have configured different thresholds in your script. Do you get any alerts from those thresholds? Also, how many concurrent calls are there when you are experiencing this? Best regards, Răzvan Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/21/2017 03:08 PM, Dipteshkumar Patel wrote: > hello all > > I am using opensips-2.3.2 running on 16 cores of cpu and 32GB of RAM > > I am facing an isssue. After making 100k of calls, suddenly opensips > performance degrades and it takes much time to process a packet. load > average of cpu incresing significantly.after restarting opensips it > works fine for another 100k calls then again problem appears. > > It is an SBC server, using 6-7 sql queries with async with database > optimisation(using proper indexing), permission module for auth, > aaa_radius module for accounting and some string operations. > > Summary of operations > 1. Sanity checks > 2. Use IP authentication > 3. Send Requet for accounting > 4. Some string operations for accounting > I am sending global parameters and module parameters of my opensips > config file > > Please find here https://pastebin.com/B7Y5HRgH > > Please suggest what is wrong and how to improve. > > *Diptesh Patel* > Jr. Software Developer > Ecosmob Technologies Ltd > Ahmedabad > Mo:*+919898962659* > > > _______________________________________________ > 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 diptesh.patel at ecosmob.com Tue Nov 21 10:29:32 2017 From: diptesh.patel at ecosmob.com (Dipteshkumar Patel) Date: Tue, 21 Nov 2017 20:59:32 +0530 Subject: [OpenSIPS-Users] Degrades the performance of opensips after 100k calls In-Reply-To: <0b878b98-54bc-f046-55a0-daf232e83746@opensips.org> References: <0b878b98-54bc-f046-55a0-daf232e83746@opensips.org> Message-ID: hello Bogdan Thanks for your reply. But after 100k calls almost all actions take too many time Nov 21 14:00:36 OS /usr/local/sbin/opensips[7235]: WARNING:core:log_expiry: threshold exceeded : msg processing took too long - 61006 us.Source : SIP/2.0 200 OK#015#012Via: SIP/2.0/UDP xxx.xxx.x.226:5060;branch=z9hG4bKf59a.7f50e365.0;rport=5060#015#012From: ;tag=BFH7y5ycBmc4m#015#012To: ;tag=3D4vXraevcj6r#015#012Call-ID: 22b7aa0c-4967-1236-989f-a0369f590b64#015#012CSeq: 115300342 INVITE#015#012Contact: #015#012User-Agent: VOXSBCSWITCH#015#012Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY#015#012Require: timer#015#012Supported: timer, path, replaces#015#012Allow-Events: talk, hold, conference, refer#015#012Session-Expires: 3600;refresher=uac#015#012Content-Type: application/sdp#015#012Content-Disposition: session#015#012Content-Length: 269#015#012X-FS-Support: update_display,send_info#015#012#015#012v=0#015#012o=FreeSWITCH 1511254322 1511254323 IN IP4 xxx.xxx.x.227#015#012s=FreeSWITCH#015#012c=IN IP4 xxx.xxx.x.227#015#012t=0 0#015#012a=sendrecv#015#012m=audio 18498 RTP/AVP 8 101#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-16#015#012a=ptime:20#015#012a=rtcp:18499 IN IP4 xxx.xxx.x.227 Nov 21 14:00:36 OS /usr/local/sbin/opensips[7235]: WARNING:core:log_expiry: threshold exceeded : msg processing took too long - 61006 us.Source : SIP/2.0 200 OK#015#012Via: SIP/2.0/UDP xxx.xxx.x.226:5060;branch=z9hG4bKf59a.7f50e365.0;rport=5060#015#012From: ;tag=BFH7y5ycBmc4m#015#012To: ;tag=3D4vXraevcj6r#015#012Call-ID: 22b7aa0c-4967-1236-989f-a0369f590b64#015#012CSeq: 115300342 INVITE#015#012Contact: #015#012User-Agent: SBCSWITCH#015#012Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY#015#012Require: timer#015#012Supported: timer, path, replaces#015#012Allow-Events: talk, hold, conference, refer#015#012Session-Expires: 3600;refresher=uac#015#012Content-Type: application/sdp#015#012Content-Disposition: session#015#012Content-Length: 269#015#012X-FS-Support: update_display,send_info#015#012#015#012v=0#015#012o=FreeSWITCH 1511254322 1511254323 IN IP4 xxx.xxx.x.227#015#012s=FreeSWITCH#015#012c=IN IP4 xxx.xxx.x.227#015#012t=0 0#015#012a=sendrecv#015#012m=audio 18498 RTP/AVP 8 101#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-16#015#012a=ptime:20#015#012a=rtcp:18499 IN IP4 xxx.xxx.x.227 Nov 21 14:51:33 OS /usr/local/sbin/opensips[7233]: WARNING:core:log_expiry: threshold exceeded : msg processing took too long - 117018 us.Source : SIP/2.0 503 Service Unavailable#015#012Via: SIP/2.0/UDP xxx.xxx.x.226:5060;branch=z9hG4bKa7a3.2fc757f5.0;rport=5060#015#012Max-Forwards: 69#015#012From: ;tag=FQH9vNpjUD89p#015#012To: ;tag=F5m876094y75B#015#012Call-ID: 4e565b83-496e-1236-f5b6-a0369f590b64#015#012CSeq: 115301882 INVITE#015#012User-Agent: SBCSWITCH#015#012Accept: application/sdp#015#012Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY#015#012Supported: timer, path, replaces#015#012Allow-Events: talk, hold, conference, refer#015#012Reason: Q.850;cause=41;text="NORMAL_TEMPORARY_FAILURE"#015#012Content-Length: 0 Nov 21 14:51:33 OS /usr/local/sbin/opensips[7231]: WARNING:core:log_expiry: #1 is a module action : t_reply - 106282us - line 1112 Nov 21 14:51:33 OS /usr/local/sbin/opensips[7238]: WARNING:core:log_expiry: #5 is a core action : 69 - 8us - line 987 Nov 21 14:51:33 OS /usr/local/sbin/opensips[7228]: WARNING:core:log_expiry: #2 is a core action : 82 - 184us - line 696 Nov 21 14:51:33 OS /usr/local/sbin/opensips[7235]: WARNING:core:log_expiry: #1 is a module action : t_reply - 116776us - line 1112 Nov 21 14:51:33 OS /usr/local/sbin/opensips[7236]: WARNING:core:log_expiry: #2 is a module action : search - 26us - line 906 Nov 21 14:51:33 OS /usr/local/sbin/opensips[7221]: WARNING:core:log_expiry: #1 is a module action : t_reply - 68668us - line 1112 Nov 21 14:51:33 OS /usr/local/sbin/opensips[7233]: WARNING:core:log_expiry: #1 is a module action : t_reply - 116606us - line 1112 Nov 21 14:51:33 OS /usr/local/sbin/opensips[7231]: WARNING:core:log_expiry: #2 is a module action : rtpproxy_engage - 2175us - line 811 Nov 21 14:51:33 OS /usr/local/sbin/opensips[7228]: WARNING:core:log_expiry: #3 is a module action : search - 81us - line 906 Nov 21 14:51:33 OS /usr/local/sbin/opensips[7235]: WARNING:core:log_expiry: #2 is a module action : search - 31us - line 906 Nov 21 14:51:33 OS /usr/local/sbin/opensips[7236]: WARNING:core:log_expiry: #3 is a core action : 52 - 17us - line 957 Nov 21 14:51:33 OS /usr/local/sbin/opensips[7221]: WARNING:core:log_expiry: #2 is a module action : t_check_status - 42us - line 879 Nov 21 14:51:33 OS /usr/local/sbin/opensips[7233]: WARNING:core:log_expiry: #2 is a module action : search - 101us - line 906 Nov 21 14:51:33 OS /usr/local/sbin/opensips[7231]: WARNING:core:log_expiry: #3 is a core action : 83 - 340us - line 764 Nov 21 14:51:33 OS /usr/local/sbin/opensips[7228]: WARNING:core:log_expiry: #4 is a module action : t_check_status - 27us - line 956 Nov 21 14:51:33 OS /usr/local/sbin/opensips[7235]: WARNING:core:log_expiry: #3 is a core action : 52 - 17us - line 957 Nov 21 14:51:33 OS /usr/local/sbin/opensips[7236]: WARNING:core:log_expiry: #4 is a module action : t_check_status - 16us - line 879 Nov 21 14:51:33 OS /usr/local/sbin/opensips[7221]: WARNING:core:log_expiry: #3 is a module action : search - 30us - line 906 Nov 21 14:51:33 OS /usr/local/sbin/opensips[7233]: WARNING:core:log_expiry: #3 is a module action : t_check_status - 43us - line 956 Nov 21 14:51:33 OS /usr/local/sbin/opensips[7231]: WARNING:core:log_expiry: #4 is a core action : 82 - 189us - line 696 Nov 21 14:51:33 OS /usr/local/sbin/opensips[7228]: WARNING:core:log_expiry: #5 is a module action : t_check_status - 19us - line 1111 Nov 21 14:51:33 OS /usr/local/sbin/opensips[7235]: WARNING:core:log_expiry: #4 is a module action : t_check_status - 14us - line 879 Nov 21 14:51:33 OS /usr/local/sbin/opensips[7236]: WARNING:core:log_expiry: #5 is a core action : 69 - 8us - line 987 Nov 21 14:51:33 OS /usr/local/sbin/opensips[7221]: WARNING:core:log_expiry: #4 is a module action : t_check_status - 19us - line 1111 Nov 21 14:51:33 OS /usr/local/sbin/opensips[7233]: WARNING:core:log_expiry: #4 is a module action : t_check_status - 41us - line 1111 Nov 21 14:51:33 OS /usr/local/sbin/opensips[7231]: WARNING:core:log_expiry: #5 is a module action : t_relay - 60us - line 843 Nov 21 14:51:33 OS /usr/local/sbin/opensips[7235]: WARNING:core:log_expiry: #5 is a module action : t_check_status - 11us - line 956 Nov 21 14:51:33 OS /usr/local/sbin/opensips[7221]: WARNING:core:log_expiry: #5 is a core action : 52 - 16us - line 957 Nov 21 14:51:33 OS /usr/local/sbin/opensips[7233]: WARNING:core:log_expiry: #5 is a core action : 69 - 18us - line 987 Nov 21 15:03:03 OS /usr/local/sbin/opensips[7216]: WARNING:core:handle_timer_job: utimer job has a 60000 us delay in execution *Diptesh Patel* Jr. Software Developer Ecosmob Technologies Ltd Ahmedabad Mo:*+919898962659* On Tue, Nov 21, 2017 at 8:34 PM, Răzvan Crainea wrote: > Hello! > > I see that you have configured different thresholds in your script. Do you > get any alerts from those thresholds? > > Also, how many concurrent calls are there when you are experiencing this? > > Best regards, > Răzvan > > Răzvan Crainea > OpenSIPS Developerwww.opensips-solutions.com > > On 11/21/2017 03:08 PM, Dipteshkumar Patel wrote: > > hello all > > I am using opensips-2.3.2 running on 16 cores of cpu and 32GB of RAM > > I am facing an isssue. After making 100k of calls, suddenly opensips > performance degrades and it takes much time to process a packet. load > average of cpu incresing significantly.after restarting opensips it works > fine for another 100k calls then again problem appears. > > It is an SBC server, using 6-7 sql queries with async with database > optimisation(using proper indexing), permission module for auth, aaa_radius > module for accounting and some string operations. > > Summary of operations > 1. Sanity checks > 2. Use IP authentication > 3. Send Requet for accounting > 4. Some string operations for accounting > > I am sending global parameters and module parameters of my opensips config > file > > Please find here https://pastebin.com/B7Y5HRgH > > Please suggest what is wrong and how to improve. > > *Diptesh Patel* > Jr. Software Developer > Ecosmob Technologies Ltd > Ahmedabad > Mo:*+919898962659* > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Tue Nov 21 10:43:16 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Tue, 21 Nov 2017 17:43:16 +0200 Subject: [OpenSIPS-Users] Degrades the performance of opensips after 100k calls In-Reply-To: References: <0b878b98-54bc-f046-55a0-daf232e83746@opensips.org> Message-ID: <3444bdd6-b47e-a81a-059c-8222cfdbf624@opensips.org> Are you seeing any other errors, such as out of memory issues? Perhaps you are not configuring enough memory. What values are you using for -m and -M arguments at startup? Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/21/2017 05:29 PM, Dipteshkumar Patel wrote: > hello Bogdan > > Thanks for your reply. > > But after 100k calls almost all actions take too many time > > Nov 21 14:00:36 OS /usr/local/sbin/opensips[7235]: > WARNING:core:log_expiry: threshold exceeded : msg processing took too > long - 61006 us.Source : SIP/2.0 200 OK#015#012Via: SIP/2.0/UDP > xxx.xxx.x.226:5060;branch=z9hG4bKf59a.7f50e365.0;rport=5060#015#012From: > ;tag=BFH7y5ycBmc4m#015#012To: > ;tag=3D4vXraevcj6r#015#012Call-ID: > 22b7aa0c-4967-1236-989f-a0369f590b64#015#012CSeq: 115300342 > INVITE#015#012Contact: > #015#012User-Agent: > VOXSBCSWITCH#015#012Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, > INFO, UPDATE, REFER, NOTIFY#015#012Require: timer#015#012Supported: > timer, path, replaces#015#012Allow-Events: talk, hold, conference, > refer#015#012Session-Expires: 3600;refresher=uac#015#012Content-Type: > application/sdp#015#012Content-Disposition: > session#015#012Content-Length: 269#015#012X-FS-Support: > update_display,send_info#015#012#015#012v=0#015#012o=FreeSWITCH > 1511254322 1511254323 IN IP4 > xxx.xxx.x.227#015#012s=FreeSWITCH#015#012c=IN IP4 > xxx.xxx.x.227#015#012t=0 0#015#012a=sendrecv#015#012m=audio 18498 > RTP/AVP 8 101#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:101 > telephone-event/8000#015#012a=fmtp:101 > 0-16#015#012a=ptime:20#015#012a=rtcp:18499 IN IP4 xxx.xxx.x.227 > Nov 21 14:00:36 OS /usr/local/sbin/opensips[7235]: > WARNING:core:log_expiry: threshold exceeded : msg processing took too > long - 61006 us.Source : SIP/2.0 200 OK#015#012Via: SIP/2.0/UDP > xxx.xxx.x.226:5060;branch=z9hG4bKf59a.7f50e365.0;rport=5060#015#012From: > ;tag=BFH7y5ycBmc4m#015#012To: > ;tag=3D4vXraevcj6r#015#012Call-ID: > 22b7aa0c-4967-1236-989f-a0369f590b64#015#012CSeq: 115300342 > INVITE#015#012Contact: > #015#012User-Agent: > SBCSWITCH#015#012Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, > INFO, UPDATE, REFER, NOTIFY#015#012Require: timer#015#012Supported: > timer, path, replaces#015#012Allow-Events: talk, hold, conference, > refer#015#012Session-Expires: 3600;refresher=uac#015#012Content-Type: > application/sdp#015#012Content-Disposition: > session#015#012Content-Length: 269#015#012X-FS-Support: > update_display,send_info#015#012#015#012v=0#015#012o=FreeSWITCH > 1511254322 1511254323 IN IP4 > xxx.xxx.x.227#015#012s=FreeSWITCH#015#012c=IN IP4 > xxx.xxx.x.227#015#012t=0 0#015#012a=sendrecv#015#012m=audio 18498 > RTP/AVP 8 101#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:101 > telephone-event/8000#015#012a=fmtp:101 > 0-16#015#012a=ptime:20#015#012a=rtcp:18499 IN IP4 xxx.xxx.x.227 > > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7233]: > WARNING:core:log_expiry: threshold exceeded : msg processing took too > long - 117018 us.Source : SIP/2.0 503 Service Unavailable#015#012Via: > SIP/2.0/UDP > xxx.xxx.x.226:5060;branch=z9hG4bKa7a3.2fc757f5.0;rport=5060#015#012Max-Forwards: > 69#015#012From: > ;tag=FQH9vNpjUD89p#015#012To: > ;tag=F5m876094y75B#015#012Call-ID: > 4e565b83-496e-1236-f5b6-a0369f590b64#015#012CSeq: 115301882 > INVITE#015#012User-Agent: SBCSWITCH#015#012Accept: > application/sdp#015#012Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, > MESSAGE, INFO, UPDATE, REFER, NOTIFY#015#012Supported: timer, path, > replaces#015#012Allow-Events: talk, hold, conference, > refer#015#012Reason: > Q.850;cause=41;text="NORMAL_TEMPORARY_FAILURE"#015#012Content-Length: 0 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7231]: > WARNING:core:log_expiry: #1 is a module action : t_reply - 106282us - > line 1112 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7238]: > WARNING:core:log_expiry: #5 is a core action : 69 - 8us - line 987 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7228]: > WARNING:core:log_expiry: #2 is a core action : 82 - 184us - line 696 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7235]: > WARNING:core:log_expiry: #1 is a module action : t_reply - 116776us - > line 1112 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7236]: > WARNING:core:log_expiry: #2 is a module action : search - 26us - line 906 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7221]: > WARNING:core:log_expiry: #1 is a module action : t_reply - 68668us - > line 1112 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7233]: > WARNING:core:log_expiry: #1 is a module action : t_reply - 116606us - > line 1112 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7231]: > WARNING:core:log_expiry: #2 is a module action : rtpproxy_engage - > 2175us - line 811 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7228]: > WARNING:core:log_expiry: #3 is a module action : search - 81us - line 906 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7235]: > WARNING:core:log_expiry: #2 is a module action : search - 31us - line 906 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7236]: > WARNING:core:log_expiry: #3 is a core action : 52 - 17us - line 957 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7221]: > WARNING:core:log_expiry: #2 is a module action : t_check_status - 42us > - line 879 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7233]: > WARNING:core:log_expiry: #2 is a module action : search - 101us - line 906 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7231]: > WARNING:core:log_expiry: #3 is a core action : 83 - 340us - line 764 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7228]: > WARNING:core:log_expiry: #4 is a module action : t_check_status - 27us > - line 956 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7235]: > WARNING:core:log_expiry: #3 is a core action : 52 - 17us - line 957 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7236]: > WARNING:core:log_expiry: #4 is a module action : t_check_status - 16us > - line 879 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7221]: > WARNING:core:log_expiry: #3 is a module action : search - 30us - line 906 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7233]: > WARNING:core:log_expiry: #3 is a module action : t_check_status - 43us > - line 956 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7231]: > WARNING:core:log_expiry: #4 is a core action : 82 - 189us - line 696 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7228]: > WARNING:core:log_expiry: #5 is a module action : t_check_status - 19us > - line 1111 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7235]: > WARNING:core:log_expiry: #4 is a module action : t_check_status - 14us > - line 879 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7236]: > WARNING:core:log_expiry: #5 is a core action : 69 - 8us - line 987 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7221]: > WARNING:core:log_expiry: #4 is a module action : t_check_status - 19us > - line 1111 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7233]: > WARNING:core:log_expiry: #4 is a module action : t_check_status - 41us > - line 1111 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7231]: > WARNING:core:log_expiry: #5 is a module action : t_relay - 60us - line 843 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7235]: > WARNING:core:log_expiry: #5 is a module action : t_check_status - 11us > - line 956 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7221]: > WARNING:core:log_expiry: #5 is a core action : 52 - 16us - line 957 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7233]: > WARNING:core:log_expiry: #5 is a core action : 69 - 18us - line 987 > Nov 21 15:03:03 OS /usr/local/sbin/opensips[7216]: > WARNING:core:handle_timer_job: utimer job has a 60000 us > delay in execution > > > *Diptesh Patel* > Jr. Software Developer > Ecosmob Technologies Ltd > Ahmedabad > Mo:*+919898962659* > > On Tue, Nov 21, 2017 at 8:34 PM, Răzvan Crainea > wrote: > > Hello! > > I see that you have configured different thresholds in your > script. Do you get any alerts from those thresholds? > > Also, how many concurrent calls are there when you are > experiencing this? > > Best regards, > Răzvan > > Răzvan Crainea > OpenSIPS Developer > www.opensips-solutions.com > > On 11/21/2017 03:08 PM, Dipteshkumar Patel wrote: >> hello all >> >> I am using opensips-2.3.2 running on 16 cores of cpu and 32GB of RAM >> >> I am facing an isssue. After making 100k of calls, suddenly >> opensips performance degrades and it takes much time to process a >> packet. load average of cpu incresing significantly.after >> restarting opensips it works fine for another 100k calls then >> again problem appears. >> >> It is an SBC server, using 6-7 sql queries with async with >> database optimisation(using proper indexing), permission module >> for auth, aaa_radius module for accounting and some string >> operations. >> >> Summary of operations >> 1. Sanity checks >> 2. Use IP authentication >> 3. Send Requet for accounting >> 4. Some string operations for accounting >> I am sending global parameters and module parameters of my >> opensips config file >> >> Please find here https://pastebin.com/B7Y5HRgH >> >> >> Please suggest what is wrong and how to improve. >> >> *Diptesh Patel* >> Jr. Software Developer >> Ecosmob Technologies Ltd >> Ahmedabad >> Mo:*+919898962659* >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From diptesh.patel at ecosmob.com Tue Nov 21 11:06:07 2017 From: diptesh.patel at ecosmob.com (Dipteshkumar Patel) Date: Tue, 21 Nov 2017 21:36:07 +0530 Subject: [OpenSIPS-Users] Degrades the performance of opensips after 100k calls In-Reply-To: <3444bdd6-b47e-a81a-059c-8222cfdbf624@opensips.org> References: <0b878b98-54bc-f046-55a0-daf232e83746@opensips.org> <3444bdd6-b47e-a81a-059c-8222cfdbf624@opensips.org> Message-ID: hello Bogdan I am using -m 12000 -M 24 at startup. thanks *Diptesh Patel* Jr. Software Developer Ecosmob Technologies Ltd Ahmedabad Mo:*+919898962659* On Tue, Nov 21, 2017 at 9:13 PM, Răzvan Crainea wrote: > Are you seeing any other errors, such as out of memory issues? Perhaps you > are not configuring enough memory. What values are you using for -m and -M > arguments at startup? > > Best regards, > > Răzvan Crainea > OpenSIPS Developerwww.opensips-solutions.com > > On 11/21/2017 05:29 PM, Dipteshkumar Patel wrote: > > hello Bogdan > > Thanks for your reply. > > But after 100k calls almost all actions take too many time > > Nov 21 14:00:36 OS /usr/local/sbin/opensips[7235]: > WARNING:core:log_expiry: threshold exceeded : msg processing took too long > - 61006 us.Source : SIP/2.0 200 OK#015#012Via: SIP/2.0/UDP > xxx.xxx.x.226:5060;branch=z9hG4bKf59a.7f50e365.0;rport=5060#015#012From: > > ;tag=BFH7y5ycBmc4m#015#012To: > ;tag=3D4vXraevcj6r#015#012Call-ID: > 22b7aa0c-4967-1236-989f-a0369f590b64#015#012CSeq: 115300342 > INVITE#015#012Contact: > #015#012User-Agent: > VOXSBCSWITCH#015#012Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, > INFO, UPDATE, REFER, NOTIFY#015#012Require: timer#015#012Supported: timer, > path, replaces#015#012Allow-Events: talk, hold, conference, > refer#015#012Session-Expires: 3600;refresher=uac#015#012Content-Type: > application/sdp#015#012Content-Disposition: > session#015#012Content-Length: 269#015#012X-FS-Support: > update_display,send_info#015#012#015#012v=0#015#012o=FreeSWITCH > 1511254322 1511254323 IN IP4 xxx.xxx.x.227#015#012s=FreeSWITCH#015#012c=IN > IP4 xxx.xxx.x.227#015#012t=0 0#015#012a=sendrecv#015#012m=audio 18498 > RTP/AVP 8 101#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:101 > telephone-event/8000#015#012a=fmtp:101 0-16#015#012a=ptime:20#015#012a=rtcp:18499 > IN IP4 xxx.xxx.x.227 > Nov 21 14:00:36 OS /usr/local/sbin/opensips[7235]: > WARNING:core:log_expiry: threshold exceeded : msg processing took too long > - 61006 us.Source : SIP/2.0 200 OK#015#012Via: SIP/2.0/UDP > xxx.xxx.x.226:5060;branch=z9hG4bKf59a.7f50e365.0;rport=5060#015#012From: > > ;tag=BFH7y5ycBmc4m#015#012To: > ;tag=3D4vXraevcj6r#015#012Call-ID: > 22b7aa0c-4967-1236-989f-a0369f590b64#015#012CSeq: 115300342 > INVITE#015#012Contact: > #015#012User-Agent: > SBCSWITCH#015#012Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, > UPDATE, REFER, NOTIFY#015#012Require: timer#015#012Supported: timer, path, > replaces#015#012Allow-Events: talk, hold, conference, > refer#015#012Session-Expires: 3600;refresher=uac#015#012Content-Type: > application/sdp#015#012Content-Disposition: > session#015#012Content-Length: 269#015#012X-FS-Support: > update_display,send_info#015#012#015#012v=0#015#012o=FreeSWITCH > 1511254322 1511254323 IN IP4 xxx.xxx.x.227#015#012s=FreeSWITCH#015#012c=IN > IP4 xxx.xxx.x.227#015#012t=0 0#015#012a=sendrecv#015#012m=audio 18498 > RTP/AVP 8 101#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:101 > telephone-event/8000#015#012a=fmtp:101 0-16#015#012a=ptime:20#015#012a=rtcp:18499 > IN IP4 xxx.xxx.x.227 > > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7233]: > WARNING:core:log_expiry: threshold exceeded : msg processing took too long > - 117018 us.Source : SIP/2.0 503 Service Unavailable#015#012Via: > SIP/2.0/UDP xxx.xxx.x.226:5060;branch=z9hG4bKa7a3.2fc757f5.0;rport=5060#015#012Max-Forwards: > 69#015#012From: > ;tag=FQH9vNpjUD89p#015#012To: > > ;tag=F5m876094y75B#015#012Call-ID: 4e565b83-496e-1236-f5b6-a0369f590b64#015#012CSeq: > 115301882 INVITE#015#012User-Agent: SBCSWITCH#015#012Accept: > application/sdp#015#012Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, > INFO, UPDATE, REFER, NOTIFY#015#012Supported: timer, path, > replaces#015#012Allow-Events: talk, hold, conference, refer#015#012Reason: > Q.850;cause=41;text="NORMAL_TEMPORARY_FAILURE"#015#012Content-Length: 0 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7231]: > WARNING:core:log_expiry: #1 is a module action : t_reply - 106282us - line > 1112 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7238]: > WARNING:core:log_expiry: #5 is a core action : 69 - 8us - line 987 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7228]: > WARNING:core:log_expiry: #2 is a core action : 82 - 184us - line 696 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7235]: > WARNING:core:log_expiry: #1 is a module action : t_reply - 116776us - line > 1112 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7236]: > WARNING:core:log_expiry: #2 is a module action : search - 26us - line 906 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7221]: > WARNING:core:log_expiry: #1 is a module action : t_reply - 68668us - line > 1112 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7233]: > WARNING:core:log_expiry: #1 is a module action : t_reply - 116606us - line > 1112 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7231]: > WARNING:core:log_expiry: #2 is a module action : rtpproxy_engage - 2175us - > line 811 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7228]: > WARNING:core:log_expiry: #3 is a module action : search - 81us - line 906 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7235]: > WARNING:core:log_expiry: #2 is a module action : search - 31us - line 906 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7236]: > WARNING:core:log_expiry: #3 is a core action : 52 - 17us - line 957 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7221]: > WARNING:core:log_expiry: #2 is a module action : t_check_status - 42us - > line 879 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7233]: > WARNING:core:log_expiry: #2 is a module action : search - 101us - line 906 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7231]: > WARNING:core:log_expiry: #3 is a core action : 83 - 340us - line 764 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7228]: > WARNING:core:log_expiry: #4 is a module action : t_check_status - 27us - > line 956 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7235]: > WARNING:core:log_expiry: #3 is a core action : 52 - 17us - line 957 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7236]: > WARNING:core:log_expiry: #4 is a module action : t_check_status - 16us - > line 879 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7221]: > WARNING:core:log_expiry: #3 is a module action : search - 30us - line 906 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7233]: > WARNING:core:log_expiry: #3 is a module action : t_check_status - 43us - > line 956 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7231]: > WARNING:core:log_expiry: #4 is a core action : 82 - 189us - line 696 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7228]: > WARNING:core:log_expiry: #5 is a module action : t_check_status - 19us - > line 1111 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7235]: > WARNING:core:log_expiry: #4 is a module action : t_check_status - 14us - > line 879 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7236]: > WARNING:core:log_expiry: #5 is a core action : 69 - 8us - line 987 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7221]: > WARNING:core:log_expiry: #4 is a module action : t_check_status - 19us - > line 1111 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7233]: > WARNING:core:log_expiry: #4 is a module action : t_check_status - 41us - > line 1111 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7231]: > WARNING:core:log_expiry: #5 is a module action : t_relay - 60us - line 843 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7235]: > WARNING:core:log_expiry: #5 is a module action : t_check_status - 11us - > line 956 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7221]: > WARNING:core:log_expiry: #5 is a core action : 52 - 16us - line 957 > Nov 21 14:51:33 OS /usr/local/sbin/opensips[7233]: > WARNING:core:log_expiry: #5 is a core action : 69 - 18us - line 987 > Nov 21 15:03:03 OS /usr/local/sbin/opensips[7216]: > WARNING:core:handle_timer_job: utimer job has a 60000 us delay > in execution > > > *Diptesh Patel* > Jr. Software Developer > Ecosmob Technologies Ltd > Ahmedabad > Mo:*+919898962659* > > On Tue, Nov 21, 2017 at 8:34 PM, Răzvan Crainea > wrote: > >> Hello! >> >> I see that you have configured different thresholds in your script. Do >> you get any alerts from those thresholds? >> >> Also, how many concurrent calls are there when you are experiencing this? >> >> Best regards, >> Răzvan >> >> Răzvan Crainea >> OpenSIPS Developerwww.opensips-solutions.com >> >> On 11/21/2017 03:08 PM, Dipteshkumar Patel wrote: >> >> hello all >> >> I am using opensips-2.3.2 running on 16 cores of cpu and 32GB of RAM >> >> I am facing an isssue. After making 100k of calls, suddenly opensips >> performance degrades and it takes much time to process a packet. load >> average of cpu incresing significantly.after restarting opensips it works >> fine for another 100k calls then again problem appears. >> >> It is an SBC server, using 6-7 sql queries with async with database >> optimisation(using proper indexing), permission module for auth, aaa_radius >> module for accounting and some string operations. >> >> Summary of operations >> 1. Sanity checks >> 2. Use IP authentication >> 3. Send Requet for accounting >> 4. Some string operations for accounting >> >> I am sending global parameters and module parameters of my opensips >> config file >> >> Please find here https://pastebin.com/B7Y5HRgH >> >> Please suggest what is wrong and how to improve. >> >> *Diptesh Patel* >> Jr. Software Developer >> Ecosmob Technologies Ltd >> Ahmedabad >> Mo:*+919898962659* >> >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Tue Nov 21 12:16:04 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Tue, 21 Nov 2017 19:16:04 +0200 Subject: [OpenSIPS-Users] Degrades the performance of opensips after 100k calls In-Reply-To: References: <0b878b98-54bc-f046-55a0-daf232e83746@opensips.org> <3444bdd6-b47e-a81a-059c-8222cfdbf624@opensips.org> Message-ID: So you are not seeing any memory reports, are you? You should also check what opensips is doing when the CPU usage goes up. For example, when the issue occurs, write down the PIDs involved and then do a full gdb snapshot using the 'opensipsctl trap' command. Get this information and send the resulted info here. Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/21/2017 06:06 PM, Dipteshkumar Patel wrote: > hello Bogdan > > I am using -m 12000 -M 24 at startup. > > thanks > > > > *Diptesh Patel* > Jr. Software Developer > Ecosmob Technologies Ltd > Ahmedabad > Mo:*+919898962659* -------------- next part -------------- An HTML attachment was scrubbed... URL: From diptesh.patel at ecosmob.com Wed Nov 22 00:56:31 2017 From: diptesh.patel at ecosmob.com (Dipteshkumar Patel) Date: Wed, 22 Nov 2017 11:26:31 +0530 Subject: [OpenSIPS-Users] Degrades the performance of opensips after 100k calls In-Reply-To: References: <0b878b98-54bc-f046-55a0-daf232e83746@opensips.org> <3444bdd6-b47e-a81a-059c-8222cfdbf624@opensips.org> Message-ID: hello Razvan Sorry for addressing you as Bogdan. We found following statistics after making lots of calls opensipsctl fifo get_statistics shmem: shmem:total_size:: 12582912000 shmem:used_size:: 7009825993193851937 shmem:real_used_size:: 7009825993216059273 shmem:max_used_size:: 18446711925787847093 shmem:free_size:: 11436918093076404343 shmem:fragments:: 497092 After restarting opensips opensipsctl fifo get_statistics shmem: shmem:total_size:: 12582912000 shmem:used_size:: 3369296 shmem:real_used_size:: 3428304 shmem:max_used_size:: 3435928 shmem:free_size:: 12579483696 shmem:fragments:: 1043 Thanks *Diptesh Patel* Jr. Software Developer Ecosmob Technologies Ltd Ahmedabad Mo:*+919898962659* On Tue, Nov 21, 2017 at 10:46 PM, Răzvan Crainea wrote: > So you are not seeing any memory reports, are you? > You should also check what opensips is doing when the CPU usage goes up. > For example, when the issue occurs, write down the PIDs involved and then > do a full gdb snapshot using the 'opensipsctl trap' command. Get this > information and send the resulted info here. > > Best regards, > > Răzvan Crainea > OpenSIPS Developerwww.opensips-solutions.com > > On 11/21/2017 06:06 PM, Dipteshkumar Patel wrote: > > hello Bogdan > > I am using -m 12000 -M 24 at startup. > > thanks > > > > *Diptesh Patel* > Jr. Software Developer > Ecosmob Technologies Ltd > Ahmedabad > Mo:*+919898962659* > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Wed Nov 22 04:06:49 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Wed, 22 Nov 2017 11:06:49 +0200 Subject: [OpenSIPS-Users] Degrades the performance of opensips after 100k calls In-Reply-To: References: <0b878b98-54bc-f046-55a0-daf232e83746@opensips.org> <3444bdd6-b47e-a81a-059c-8222cfdbf624@opensips.org> Message-ID: <300b6419-55e0-97a3-2a3c-c5d73d3fd6cc@opensips.org> So you are seeing ERROR logs regarding memory exhaution, right? If so, most likely that's the problem. Can you tell me how many concurrent calls you are having? You can find out by running 'opensipsctl fifo get_statistics dialog:'. Also, what memory alocator are you using? Can you post the output of 'opensips -V'? Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/22/2017 07:56 AM, Dipteshkumar Patel wrote: > hello Razvan > > Sorry for addressing you as Bogdan. > > We found following statistics after making lots of calls > > opensipsctl fifo get_statistics shmem: > shmem:total_size:: 12582912000 > shmem:used_size:: 7009825993193851937 > shmem:real_used_size:: 7009825993216059273 > shmem:max_used_size:: 18446711925787847093 > shmem:free_size:: 11436918093076404343 > shmem:fragments:: 497092 > > After restarting opensips > > opensipsctl fifo get_statistics shmem: > shmem:total_size:: 12582912000 > shmem:used_size:: 3369296 > shmem:real_used_size:: 3428304 > shmem:max_used_size:: 3435928 > shmem:free_size:: 12579483696 > shmem:fragments:: 1043 > > > Thanks > > > > > > > *Diptesh Patel* > Jr. Software Developer > Ecosmob Technologies Ltd > Ahmedabad > Mo:*+919898962659* > > On Tue, Nov 21, 2017 at 10:46 PM, Răzvan Crainea > wrote: > > So you are not seeing any memory reports, are you? > You should also check what opensips is doing when the CPU usage > goes up. For example, when the issue occurs, write down the PIDs > involved and then do a full gdb snapshot using the 'opensipsctl > trap' command. Get this information and send the resulted info here. > > Best regards, > > Răzvan Crainea > OpenSIPS Developer > www.opensips-solutions.com > > On 11/21/2017 06:06 PM, Dipteshkumar Patel wrote: >> hello Bogdan >> >> I am using -m 12000 -M 24 at startup. >> >> thanks >> >> >> >> *Diptesh Patel* >> Jr. Software Developer >> Ecosmob Technologies Ltd >> Ahmedabad >> Mo:*+919898962659* > > _______________________________________________ > 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 diptesh.patel at ecosmob.com Wed Nov 22 05:25:54 2017 From: diptesh.patel at ecosmob.com (Dipteshkumar Patel) Date: Wed, 22 Nov 2017 15:55:54 +0530 Subject: [OpenSIPS-Users] Degrades the performance of opensips after 100k calls In-Reply-To: <300b6419-55e0-97a3-2a3c-c5d73d3fd6cc@opensips.org> References: <0b878b98-54bc-f046-55a0-daf232e83746@opensips.org> <3444bdd6-b47e-a81a-059c-8222cfdbf624@opensips.org> <300b6419-55e0-97a3-2a3c-c5d73d3fd6cc@opensips.org> Message-ID: hello Razvan This is the output of opensips -V flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. git revision: 823044b main.c compiled on 08:28:39 Nov 22 2017 with gcc 4.8.5 and i am using 100 cps with 1500 concurent calls result of opensipsctl fifo get_statistics dialog: dialog:active_dialogs:: 1288 dialog:early_dialogs:: 120 dialog:processed_dialogs:: 7077 dialog:expired_dialogs:: 0 dialog:failed_dialogs:: 14 dialog:create_sent:: 0 dialog:update_sent:: 0 dialog:delete_sent:: 0 dialog:create_recv:: 0 dialog:update_recv:: 0 dialog:delete_recv:: 0 thanks *Diptesh Patel* Jr. Software Developer Ecosmob Technologies Ltd Ahmedabad Mo:*+919898962659* On Wed, Nov 22, 2017 at 2:36 PM, Răzvan Crainea wrote: > So you are seeing ERROR logs regarding memory exhaution, right? If so, > most likely that's the problem. Can you tell me how many concurrent calls > you are having? You can find out by running 'opensipsctl fifo > get_statistics dialog:'. > Also, what memory alocator are you using? Can you post the output of > 'opensips -V'? > > Best regards, > > Răzvan Crainea > OpenSIPS Developerwww.opensips-solutions.com > > On 11/22/2017 07:56 AM, Dipteshkumar Patel wrote: > > hello Razvan > > Sorry for addressing you as Bogdan. > > We found following statistics after making lots of calls > > opensipsctl fifo get_statistics shmem: > shmem:total_size:: 12582912000 > shmem:used_size:: 7009825993193851937 > shmem:real_used_size:: 7009825993216059273 > shmem:max_used_size:: 18446711925787847093 > shmem:free_size:: 11436918093076404343 > shmem:fragments:: 497092 > > After restarting opensips > > opensipsctl fifo get_statistics shmem: > shmem:total_size:: 12582912000 > shmem:used_size:: 3369296 > shmem:real_used_size:: 3428304 > shmem:max_used_size:: 3435928 > shmem:free_size:: 12579483696 > shmem:fragments:: 1043 > > > Thanks > > > > > > > *Diptesh Patel* > Jr. Software Developer > Ecosmob Technologies Ltd > Ahmedabad > Mo:*+919898962659* > > On Tue, Nov 21, 2017 at 10:46 PM, Răzvan Crainea > wrote: > >> So you are not seeing any memory reports, are you? >> You should also check what opensips is doing when the CPU usage goes up. >> For example, when the issue occurs, write down the PIDs involved and then >> do a full gdb snapshot using the 'opensipsctl trap' command. Get this >> information and send the resulted info here. >> >> Best regards, >> >> Răzvan Crainea >> OpenSIPS Developerwww.opensips-solutions.com >> >> On 11/21/2017 06:06 PM, Dipteshkumar Patel wrote: >> >> hello Bogdan >> >> I am using -m 12000 -M 24 at startup. >> >> thanks >> >> >> >> *Diptesh Patel* >> Jr. Software Developer >> Ecosmob Technologies Ltd >> Ahmedabad >> Mo:*+919898962659* >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Nov 22 10:09:43 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 22 Nov 2017 17:09:43 +0200 Subject: [OpenSIPS-Users] call_center module and loose routing In-Reply-To: <007e01d362b3$c9b66c00$5d234400$@web.de> References: <010301d3620e$526d8610$f7489230$@web.de> <007e01d362b3$c9b66c00$5d234400$@web.de> Message-ID: <91d951c8-ffaf-d035-e4d8-47da0c3e9c9c@opensips.org> Hi, So you use pre-loaded route in order to send initial INVITEs from SIPSERVER to OPENSIPS, right ? And what you want to achieve is to have all the calls generated by OpenSIPS Call Center (to agents, to announcements) to actually go again via the SIPSERVER? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/21/2017 12:30 PM, xaled wrote: > > Hi Bogdan, > > I use opensips in an scenario where OPENSIPS is addressed from the > main SIPSERVER using Route header. OPENSIPS performs different > modifications on the INVITE request and gives the INVITE back to the > main SIPSERVER. This works fine with the loose_route() so far. > > Now I want to use the call_center module for RURI agent addressing, > queues, etc while still preserving the Route header based next hop > routing between main SIPSERVER and OPENSIPS. > > Would manually parsing the Route Header at OPENSIPS, deleting first > route header, saving the hostport value of the second Route header to > some variable and provide it as a kind of $du to the call_center > module for the next hop routing back to SIPSERVER work? > > For my scenario to work the Route header with the second Route header > value has not to be deleted or modified by the call_center module. > > Here is the table showing the pseudo values of the RURI and the Route > Header at the ingress of SIPSERVER and OPENSIPS respectively. > > SIPSERVER -> > OPENSIPS -> SIPSERVER -> AGENT > > RURI: 0800CallCenter > 0800CallCenter agent agent > > Route: opensips, sipserver sipserver > > Does this make it a bit more clearer? > > Thanks, > > xaled > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* Tuesday, November 21, 2017 10:55 AM > *To:* users at lists.opensips.org; xaled > *Subject:* Re: [OpenSIPS-Users] call_center module and loose routing > > Hi, > > The routing in Call Center module is not compatible with the > loose_route(). > > The loose_route is a mechanism used by a SIP proxy staying in the > middle of a SIP session. > > On the other hand, the Call Center is using a SIP Back-2-Back engine > in order to implement the queuing. > > Better try to explain what are you trying to achieve here. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 11/20/2017 04:46 PM, xaled wrote: > > Hi > > I want to use opensips with call_center module in a loose route > scenario where the INVITE has to be routed by opensips to the next > hop (sipserver) based on what is left from Route header after > loose_route() is performed. The call_center module has to update > the RURI with agent location and let the next hop from the route > header perform the actual routing to the agent. > > Here are the expected route and RURI header values at the > opensips, sipserver and agent: > > opensips sipserver agent > > RURI: 0800CallCenter agent agent > > Route: opensips, sipserver sipserver > > Will something like this work as described above: > > loose_route(); > > cc_handle_call(test_flow); > > Thanks, > > xaled > > > > > _______________________________________________ > > 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 22 10:14:38 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 22 Nov 2017 17:14:38 +0200 Subject: [OpenSIPS-Users] rtp port changed in REPLY's SDP after announcement In-Reply-To: References: <58ac999d138842ec933c290b76ac188a@next-id.de> <70fdf3f1-9752-69e4-d5fc-96212d72f4ae@opensips.org> Message-ID: Hi Marco, If I understand correctly, you *NEED* to do get different SDP's in the 2 INVITE branches as the first branch going to the media server needs to have a private IP, while the second branch going to the calle needs to have a public IP, right ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/21/2017 12:49 PM, Marco Hierl wrote: > > Hi Bogdan, > > no, he did not say anything yet, but I expect that it is paragraph 3.2: > > “A UAS cannot send a new offer in the reliable > > provisional response, so the UPDATE method is the only method for a > > UAS to update an early session.” > > (I’m not quite sure if this really fits to our situation, because the > UAS itself changed.) > > According to your suggestion: > > I used “rtpengine_offer(external *internal*…” to adjust the SDP the > first time. So I get an internal IP in the INVITE to the announcement. > And this will be seen in the 2^nd INVITE to the callee if I do NOT > make “rtpengine_offer(external *external*…” to the INVITE to the > callee. But the callee has public IP and I need to use the external > interface. > > Thanks for your help! > > Marco > > *Von:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Gesendet:* Dienstag, 21. November 2017 11:10 > *An:* OpenSIPS users mailling list; Marco Hierl > *Betreff:* Re: [OpenSIPS-Users] rtp port changed in REPLY's SDP after > announcement > > Hello Marco, > > Did the interconnection partner mentioned the chapter/paragraph in > RFC6337 where the restriction on the port is mentioned (I was not able > to find it on a quick scan) ? it is the first time I hear about > anything like this. > > Now, if you use rtpproxy (I do not know about rtpengine) on your side > and you do rtpproxy_offer only once (when getting the call from the > carrier) , you can do rtpproxy_answer multiple time (during the serial > forking). In this case, the RTP IP:port towards the carrier will never > change, but the RTP callee leg will be changed. > > Best regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 11/21/2017 11:07 AM, Marco Hierl wrote: > > Dear all, > > I want to play an early announcement before connecting a call. The > announcement can be heard by the caller, but the callee can't hear > the caller after the call is connected! > > To be more precise: > > The openSIPS proxy receives a call from an external > interconnection partner (public IP on external interface) and > sends it on to an announcement server (private IP on internal > interface). > > After REPLY 183 the announcement starts and the caller can hear > the announcement. It finishes with REPLY603 that comes to the > openSIPS failure route. Some stuff is done (e.g. changing R-URI), > > a new t_relay will be done and an INVITE is send to the callee > (public IP on external interface). The callee sends 200ok, it's > forwarded to the caller, that sends ACK like usual. > > But the caller does not send RTP to the IP/PORT indicated in the > 200OK. > > The interconnection partner said that the problem is, that the RTP > port in REPLY 183 is different to that in REPLY200 (not allowed, > see RFC6337). (just to mention: the to-tag is different too) > > I'm using openSIPS 2.3 and in order to overcome the issue with > internal and external addresses and also to keep the rtp streams > I'm using rtpengine 5.4 (on a different machine) > > Is the change of the RTP Port allowed in this way? If not, what I > can do better? Or is it a fault from rtp-engine, should it keep > the first port? > > Any help is very appreciated! > > I tried to make a picture... maybe it helps a bit to understand > what I was not able to explain above... > > No From To Message > > 1 external ownProxy Request: INVITE sip:+49xxx at us.de;user=phone > > 2 ownProxy external Status: 100 Giving a try > > rtpengine_offer(external > internal replace-session-connection ICE=remove trust-address > replace-origin) > > 3 to announcement > server Request: INVITE sip:ANNC at mediaserver:5060;user=phone | > > 4 from announcement server Status: 100 Trying | > > 5 from announcement server Status: 183 > Session Progress | > > o=root > 1135152900 1135152900 IN IP4 mediaserver > > m=audio 13788 RTP/AVP 8 101 > > rtpengine_answer(internal external replace-session-connection > ICE=remove trust-address replace-origin) > > 6 ownProxy external Status: 183 Session Progress > > o=root 1135152900 > 1135152900 IN IP4 ownProxy > > m=audio 23024 RTP/AVP 8 101 > > < ========== ANNOUNCEMENT ========= > > > 7 from announcement server Status: 603 Declined > > openSIPS failure route > > and > rtpengine_offer(external external replace-session-connection > ICE=remove trust-address replace-origin) > > and new t_relay > > 8 to callee Request: INVITE > sip:bob at externalcarrier:5060;user=phone > > 9 from callee Status: 100 Trying > > 10 from callee Status: 200 OK > > o=- > 239036129 823630844 IN IP4 10.0.203.7 > > m=audio 1044 RTP/AVP 8 101 > > rtpengine_answer(external external > replace-session-connection ICE=remove trust-address replace-origin) > > 11 ownProxy external Status: 200 OK > > o=- 239036129 823630844 IN IP4 ownProxy > > m=audio 23056 RTP/AVP 8 101 > > < ========== RTP from B to A is ok, A send NO RTP to B > ========= > > > 12 external ownProxy Request: ACK sip:ownProxy;did=5db.a2833e17 > > 13 external ownProxy Request: BYE sip:ownProxy;did=5db.a2833e17 > > 14 ownProxy external Status: 200 OK > > version: opensips 2.3.0 (x86_64/linux) > > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN > 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 > > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. > > git revision: fe0ebcb45 > > main.c compiled on 09:02:02 Sep 14 2017 with gcc 6.3.0 > > rtpengine version 5.4.0.0+0~mr5.4.0.0 git-master-a9d55c6 > > Thanks and best regards!!! > > Marco > > > > > _______________________________________________ > > 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 xaled at web.de Wed Nov 22 10:44:20 2017 From: xaled at web.de (Xaled) Date: Wed, 22 Nov 2017 16:44:20 +0100 Subject: [OpenSIPS-Users] call_center module and loose routing In-Reply-To: <91d951c8-ffaf-d035-e4d8-47da0c3e9c9c@opensips.org> References: <010301d3620e$526d8610$f7489230$@web.de> <007e01d362b3$c9b66c00$5d234400$@web.de> <91d951c8-ffaf-d035-e4d8-47da0c3e9c9c@opensips.org> Message-ID: Hi Correct. Thanks, Xaled > On 22 Nov 2017, at 16:09, Bogdan-Andrei Iancu wrote: > > Hi, > > So you use pre-loaded route in order to send initial INVITEs from SIPSERVER to OPENSIPS, right ? > > And what you want to achieve is to have all the calls generated by OpenSIPS Call Center (to agents, to announcements) to actually go again via the SIPSERVER ? > > Regards, > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com >> On 11/21/2017 12:30 PM, xaled wrote: >> Hi Bogdan, >> >> I use opensips in an scenario where OPENSIPS is addressed from the main SIPSERVER using Route header. OPENSIPS performs different modifications on the INVITE request and gives the INVITE back to the main SIPSERVER. This works fine with the loose_route() so far. >> >> Now I want to use the call_center module for RURI agent addressing, queues, etc while still preserving the Route header based next hop routing between main SIPSERVER and OPENSIPS. >> Would manually parsing the Route Header at OPENSIPS, deleting first route header, saving the hostport value of the second Route header to some variable and provide it as a kind of $du to the call_center module for the next hop routing back to SIPSERVER work? >> For my scenario to work the Route header with the second Route header value has not to be deleted or modified by the call_center module. >> >> Here is the table showing the pseudo values of the RURI and the Route Header at the ingress of SIPSERVER and OPENSIPS respectively. >> >> SIPSERVER -> OPENSIPS -> SIPSERVER -> AGENT >> RURI: 0800CallCenter 0800CallCenter agent agent >> Route: opensips, sipserver sipserver >> >> Does this make it a bit more clearer? >> >> Thanks, >> xaled >> >> >> From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] >> Sent: Tuesday, November 21, 2017 10:55 AM >> To: users at lists.opensips.org; xaled >> Subject: Re: [OpenSIPS-Users] call_center module and loose routing >> >> Hi, >> >> The routing in Call Center module is not compatible with the loose_route(). >> >> The loose_route is a mechanism used by a SIP proxy staying in the middle of a SIP session. >> >> On the other hand, the Call Center is using a SIP Back-2-Back engine in order to implement the queuing. >> >> Better try to explain what are you trying to achieve here. >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> On 11/20/2017 04:46 PM, xaled wrote: >> Hi >> >> I want to use opensips with call_center module in a loose route scenario where the INVITE has to be routed by opensips to the next hop (sipserver) based on what is left from Route header after loose_route() is performed. The call_center module has to update the RURI with agent location and let the next hop from the route header perform the actual routing to the agent. >> >> Here are the expected route and RURI header values at the opensips, sipserver and agent: >> >> opensips sipserver agent >> RURI: 0800CallCenter agent agent >> Route: opensips, sipserver sipserver >> >> Will something like this work as described above: >> >> loose_route(); >> cc_handle_call(test_flow); >> >> Thanks, >> xaled >> >> >> >> _______________________________________________ >> 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 marco.hierl at next-id.de Wed Nov 22 11:37:52 2017 From: marco.hierl at next-id.de (Marco Hierl) Date: Wed, 22 Nov 2017 16:37:52 +0000 Subject: [OpenSIPS-Users] rtp port changed in REPLY's SDP after announcement In-Reply-To: References: <58ac999d138842ec933c290b76ac188a@next-id.de> <70fdf3f1-9752-69e4-d5fc-96212d72f4ae@opensips.org> Message-ID: <08c1e9f75c2f449d9bc1273182f9b526@next-id.de> Hi Bogdan, yes, that's correct! Best regards Marco Von: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Gesendet: Mittwoch, 22. November 2017 16:15 An: Marco Hierl; OpenSIPS users mailling list Betreff: Re: AW: [OpenSIPS-Users] rtp port changed in REPLY's SDP after announcement Hi Marco, If I understand correctly, you *NEED* to do get different SDP's in the 2 INVITE branches as the first branch going to the media server needs to have a private IP, while the second branch going to the calle needs to have a public IP, right ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/21/2017 12:49 PM, Marco Hierl wrote: Hi Bogdan, no, he did not say anything yet, but I expect that it is paragraph 3.2: "A UAS cannot send a new offer in the reliable provisional response, so the UPDATE method is the only method for a UAS to update an early session." (I'm not quite sure if this really fits to our situation, because the UAS itself changed.) According to your suggestion: I used "rtpengine_offer(external internal..." to adjust the SDP the first time. So I get an internal IP in the INVITE to the announcement. And this will be seen in the 2nd INVITE to the callee if I do NOT make "rtpengine_offer(external external..." to the INVITE to the callee. But the callee has public IP and I need to use the external interface. Thanks for your help! Marco Von: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Gesendet: Dienstag, 21. November 2017 11:10 An: OpenSIPS users mailling list; Marco Hierl Betreff: Re: [OpenSIPS-Users] rtp port changed in REPLY's SDP after announcement Hello Marco, Did the interconnection partner mentioned the chapter/paragraph in RFC6337 where the restriction on the port is mentioned (I was not able to find it on a quick scan) ? it is the first time I hear about anything like this. Now, if you use rtpproxy (I do not know about rtpengine) on your side and you do rtpproxy_offer only once (when getting the call from the carrier) , you can do rtpproxy_answer multiple time (during the serial forking). In this case, the RTP IP:port towards the carrier will never change, but the RTP callee leg will be changed. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/21/2017 11:07 AM, Marco Hierl wrote: Dear all, I want to play an early announcement before connecting a call. The announcement can be heard by the caller, but the callee can't hear the caller after the call is connected! To be more precise: The openSIPS proxy receives a call from an external interconnection partner (public IP on external interface) and sends it on to an announcement server (private IP on internal interface). After REPLY 183 the announcement starts and the caller can hear the announcement. It finishes with REPLY603 that comes to the openSIPS failure route. Some stuff is done (e.g. changing R-URI), a new t_relay will be done and an INVITE is send to the callee (public IP on external interface). The callee sends 200ok, it's forwarded to the caller, that sends ACK like usual. But the caller does not send RTP to the IP/PORT indicated in the 200OK. The interconnection partner said that the problem is, that the RTP port in REPLY 183 is different to that in REPLY200 (not allowed, see RFC6337). (just to mention: the to-tag is different too) I'm using openSIPS 2.3 and in order to overcome the issue with internal and external addresses and also to keep the rtp streams I'm using rtpengine 5.4 (on a different machine) Is the change of the RTP Port allowed in this way? If not, what I can do better? Or is it a fault from rtp-engine, should it keep the first port? Any help is very appreciated! I tried to make a picture... maybe it helps a bit to understand what I was not able to explain above... No From To Message 1 external ownProxy Request: INVITE sip:+49xxx at us.de;user=phone 2 ownProxy external Status: 100 Giving a try rtpengine_offer(external internal replace-session-connection ICE=remove trust-address replace-origin) 3 to announcement server Request: INVITE sip:ANNC at mediaserver:5060;user=phone | 4 from announcement server Status: 100 Trying | 5 from announcement server Status: 183 Session Progress | o=root 1135152900 1135152900 IN IP4 mediaserver m=audio 13788 RTP/AVP 8 101 rtpengine_answer(internal external replace-session-connection ICE=remove trust-address replace-origin) 6 ownProxy external Status: 183 Session Progress o=root 1135152900 1135152900 IN IP4 ownProxy m=audio 23024 RTP/AVP 8 101 < ========== ANNOUNCEMENT ========= > 7 from announcement server Status: 603 Declined openSIPS failure route and rtpengine_offer(external external replace-session-connection ICE=remove trust-address replace-origin) and new t_relay 8 to callee Request: INVITE sip:bob at externalcarrier:5060;user=phone 9 from callee Status: 100 Trying 10 from callee Status: 200 OK o=- 239036129 823630844 IN IP4 10.0.203.7 m=audio 1044 RTP/AVP 8 101 rtpengine_answer(external external replace-session-connection ICE=remove trust-address replace-origin) 11 ownProxy external Status: 200 OK o=- 239036129 823630844 IN IP4 ownProxy m=audio 23056 RTP/AVP 8 101 < ========== RTP from B to A is ok, A send NO RTP to B ========= > 12 external ownProxy Request: ACK sip:ownProxy;did=5db.a2833e17 13 external ownProxy Request: BYE sip:ownProxy;did=5db.a2833e17 14 ownProxy external Status: 200 OK version: opensips 2.3.0 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. git revision: fe0ebcb45 main.c compiled on 09:02:02 Sep 14 2017 with gcc 6.3.0 rtpengine version 5.4.0.0+0~mr5.4.0.0 git-master-a9d55c6 Thanks and best regards!!! Marco _______________________________________________ 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 22 12:17:17 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 22 Nov 2017 19:17:17 +0200 Subject: [OpenSIPS-Users] call_center module and loose routing In-Reply-To: References: <010301d3620e$526d8610$f7489230$@web.de> <007e01d362b3$c9b66c00$5d234400$@web.de> <91d951c8-ffaf-d035-e4d8-47da0c3e9c9c@opensips.org> Message-ID: <54607825-e6e6-f0dd-17fd-5656ad42deb0@opensips.org> Have you tried to use the local route to identify the call to the agents (based on RURI ?) and force the SIPSERVER address in the $du ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/22/2017 05:44 PM, Xaled wrote: > Hi > > Correct. > > Thanks, > Xaled > > > On 22 Nov 2017, at 16:09, Bogdan-Andrei Iancu > wrote: > >> Hi, >> >> So you use pre-loaded route in order to send initial INVITEs from >> SIPSERVER to OPENSIPS, right ? >> >> And what you want to achieve is to have all the calls generated by >> OpenSIPS Call Center (to agents, to announcements) to actually go >> again via the SIPSERVER? >> >> Regards, >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> On 11/21/2017 12:30 PM, xaled wrote: >>> >>> Hi Bogdan, >>> >>> I use opensips in an scenario where OPENSIPS is addressed from the >>> main SIPSERVER using Route header. OPENSIPS performs different >>> modifications on the INVITE request and gives the INVITE back to the >>> main SIPSERVER. This works fine with the loose_route() so far. >>> >>> Now I want to use the call_center module for RURI agent addressing, >>> queues, etc while still preserving the Route header based next hop >>> routing between main SIPSERVER and OPENSIPS. >>> >>> Would manually parsing the Route Header at OPENSIPS, deleting first >>> route header, saving the hostport value of the second Route header >>> to some variable and provide it as a kind of $du to the call_center >>> module for the next hop routing back to SIPSERVER work? >>> >>> For my scenario to work the Route header with the second Route >>> header value has not to be deleted or modified by the call_center >>> module. >>> >>> Here is the table showing the pseudo values of the RURI and the >>> Route Header at the ingress of SIPSERVER and OPENSIPS respectively. >>> >>> SIPSERVER -> >>> OPENSIPS -> SIPSERVER -> AGENT >>> >>> RURI: 0800CallCenter >>> 0800CallCenter agent agent >>> >>> Route: opensips, sipserver sipserver >>> >>> Does this make it a bit more clearer? >>> >>> Thanks, >>> >>> xaled >>> >>> *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] >>> *Sent:* Tuesday, November 21, 2017 10:55 AM >>> *To:* users at lists.opensips.org; xaled >>> *Subject:* Re: [OpenSIPS-Users] call_center module and loose routing >>> >>> Hi, >>> >>> The routing in Call Center module is not compatible with the >>> loose_route(). >>> >>> The loose_route is a mechanism used by a SIP proxy staying in the >>> middle of a SIP session. >>> >>> On the other hand, the Call Center is using a SIP Back-2-Back engine >>> in order to implement the queuing. >>> >>> Better try to explain what are you trying to achieve here. >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> OpenSIPS Founder and Developer >>> http://www.opensips-solutions.com >>> >>> On 11/20/2017 04:46 PM, xaled wrote: >>> >>> Hi >>> >>> I want to use opensips with call_center module in a loose route >>> scenario where the INVITE has to be routed by opensips to the >>> next hop (sipserver) based on what is left from Route header >>> after loose_route() is performed. The call_center module has to >>> update the RURI with agent location and let the next hop from >>> the route header perform the actual routing to the agent. >>> >>> Here are the expected route and RURI header values at the >>> opensips, sipserver and agent: >>> >>> opensips sipserver agent >>> >>> RURI: 0800CallCenter agent agent >>> >>> Route: opensips, sipserver sipserver >>> >>> Will something like this work as described above: >>> >>> loose_route(); >>> >>> cc_handle_call(test_flow); >>> >>> Thanks, >>> >>> xaled >>> >>> >>> >>> >>> _______________________________________________ >>> >>> 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 22 12:16:03 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 22 Nov 2017 19:16:03 +0200 Subject: [OpenSIPS-Users] rtp port changed in REPLY's SDP after announcement In-Reply-To: <08c1e9f75c2f449d9bc1273182f9b526@next-id.de> References: <58ac999d138842ec933c290b76ac188a@next-id.de> <70fdf3f1-9752-69e4-d5fc-96212d72f4ae@opensips.org> <08c1e9f75c2f449d9bc1273182f9b526@next-id.de> Message-ID: Marco, In this is case it exclusivelydepends on the rtpengine if it updates or creates a new session when you call the "offer" for the second time. If the existing session (created for the first branch) will be updated on the second "offer", the port in the replies (towards caller) will not change, right ? But, based on your tests, I assume it does change the port send to caller after the second offer :( . Again, this depends on the media relay implementation, it is not something to be controlled by OpenSIPS. As asimple alternative, move your media server on the public network, so you will not need a second offer. For the second branch, you will simply do asecond answer. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/22/2017 06:37 PM, Marco Hierl wrote: > > Hi Bogdan, > > yes, that’s correct! > > Best regards > > Marco > > *Von:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Gesendet:* Mittwoch, 22. November 2017 16:15 > *An:* Marco Hierl; OpenSIPS users mailling list > *Betreff:* Re: AW: [OpenSIPS-Users] rtp port changed in REPLY's SDP > after announcement > > Hi Marco, > > If I understand correctly, you *NEED* to do get different SDP's in the > 2 INVITE branches as the first branch going to the media server needs > to have a private IP, while the second branch going to the calle needs > to have a public IP, right ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 11/21/2017 12:49 PM, Marco Hierl wrote: > > Hi Bogdan, > > no, he did not say anything yet, but I expect that it is paragraph > 3.2: > > “A UAS cannot send a new offer in the reliable > > provisional response, so the UPDATE method is the only method for a > > UAS to update an early session.” > > (I’m not quite sure if this really fits to our situation, because > the UAS itself changed.) > > According to your suggestion: > > I used “rtpengine_offer(external *internal*…” to adjust the SDP > the first time. So I get an internal IP in the INVITE to the > announcement. And this will be seen in the 2^nd INVITE to the > callee if I do NOT make “rtpengine_offer(external *external*…” to > the INVITE to the callee. But the callee has public IP and I need > to use the external interface. > > Thanks for your help! > > Marco > > *Von:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Gesendet:* Dienstag, 21. November 2017 11:10 > *An:* OpenSIPS users mailling list; Marco Hierl > *Betreff:* Re: [OpenSIPS-Users] rtp port changed in REPLY's SDP > after announcement > > Hello Marco, > > Did the interconnection partner mentioned the chapter/paragraph in > RFC6337 where the restriction on the port is mentioned (I was not > able to find it on a quick scan) ? it is the first time I hear > about anything like this. > > Now, if you use rtpproxy (I do not know about rtpengine) on your > side and you do rtpproxy_offer only once (when getting the call > from the carrier) , you can do rtpproxy_answer multiple time > (during the serial forking). In this case, the RTP IP:port towards > the carrier will never change, but the RTP callee leg will be changed. > > Best regards, > > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > On 11/21/2017 11:07 AM, Marco Hierl wrote: > > Dear all, > > I want to play an early announcement before connecting a call. > The announcement can be heard by the caller, but the callee > can't hear the caller after the call is connected! > > To be more precise: > > The openSIPS proxy receives a call from an external > interconnection partner (public IP on external interface) and > sends it on to an announcement server (private IP on internal > interface). > > After REPLY 183 the announcement starts and the caller can > hear the announcement. It finishes with REPLY603 that comes to > the openSIPS failure route. Some stuff is done (e.g. changing > R-URI), > > a new t_relay will be done and an INVITE is send to the callee > (public IP on external interface). The callee sends 200ok, > it's forwarded to the caller, that sends ACK like usual. > > But the caller does not send RTP to the IP/PORT indicated in > the 200OK. > > The interconnection partner said that the problem is, that the > RTP port in REPLY 183 is different to that in REPLY200 (not > allowed, see RFC6337). (just to mention: the to-tag is > different too) > > I'm using openSIPS 2.3 and in order to overcome the issue with > internal and external addresses and also to keep the rtp > streams I'm using rtpengine 5.4 (on a different machine) > > Is the change of the RTP Port allowed in this way? If not, > what I can do better? Or is it a fault from rtp-engine, should > it keep the first port? > > Any help is very appreciated! > > I tried to make a picture... maybe it helps a bit to > understand what I was not able to explain above... > > No From To Message > > 1 external ownProxy Request: INVITE > sip:+49xxx at us.de;user=phone > > 2 ownProxy external Status: 100 Giving a try > > rtpengine_offer(external > internal replace-session-connection ICE=remove trust-address > replace-origin) > > 3 to > announcement server Request: INVITE > sip:ANNC at mediaserver:5060;user=phone | > > 4 from announcement server Status: 100 Trying | > > 5 from announcement server Status: > 183 Session Progress | > > o=root > 1135152900 1135152900 IN IP4 mediaserver > > m=audio 13788 RTP/AVP 8 101 > > rtpengine_answer(internal external replace-session-connection > ICE=remove trust-address replace-origin) > > 6 ownProxy external Status: 183 Session Progress > > o=root 1135152900 > 1135152900 IN IP4 ownProxy > > m=audio 23024 RTP/AVP 8 101 > > < ========== ANNOUNCEMENT ========= > > > 7 from announcement server Status: 603 Declined > > openSIPS failure route > > and > rtpengine_offer(external external replace-session-connection > ICE=remove trust-address replace-origin) > > and new t_relay > > 8 to callee Request: INVITE > sip:bob at externalcarrier:5060;user=phone > > 9 from callee Status: 100 Trying > > 10 from callee Status: 200 OK > > o=- > 239036129 823630844 IN IP4 10.0.203.7 > > m=audio 1044 RTP/AVP 8 101 > > rtpengine_answer(external external > replace-session-connection ICE=remove trust-address > replace-origin) > > 11 ownProxy external Status: 200 OK > > o=- 239036129 823630844 IN IP4 ownProxy > > m=audio 23056 RTP/AVP 8 101 > > < ========== RTP from B to A is ok, A send NO RTP to B > ========= > > > 12 external ownProxy Request: ACK > sip:ownProxy;did=5db.a2833e17 > > 13 external ownProxy Request: BYE > sip:ownProxy;did=5db.a2833e17 > > 14 ownProxy external Status: 200 OK > > version: opensips 2.3.0 (x86_64/linux) > > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, > PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, > MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 > > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. > > git revision: fe0ebcb45 > > main.c compiled on 09:02:02 Sep 14 2017 with gcc 6.3.0 > > rtpengine version 5.4.0.0+0~mr5.4.0.0 git-master-a9d55c6 > > Thanks and best regards!!! > > Marco > > > > > > _______________________________________________ > > 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 22 12:35:15 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 22 Nov 2017 19:35:15 +0200 Subject: [OpenSIPS-Users] [OpenSIPS-News] Introducing OpenSIPS 2.4 In-Reply-To: <972d9f71-89d4-89f1-9183-f7fc0f1c506a@opensips.org> References: <972d9f71-89d4-89f1-9183-f7fc0f1c506a@opensips.org> Message-ID: Hello all, Thank you all the participants to the conf call. For the people who were not able to join us, please find the audio recording http://opensips.org/html/media/Introducing_OpenSIPS_2-4_2018-11-21.mp3 And keep in mind that whatever comment, objection, idea or generic feedback you may have on the topic we are working on for OpenSIPS 2.4 is more than welcome ;) Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/01/2017 07:16 PM, Bogdan-Andrei Iancu wrote: > One more year, one more evolution cycle, one more OpenSIPS major > release. So let me introduce you the upcoming OpenSIPS 2.4 . > > For the OpenSIPS 2.4 release we decided to focus on the */clustering > abilities/*. Today’s VoIP world is getting more and more dynamic, > services are moving into Clouds and more and more flexibility is > needed for the application to fully exploit such environments. But > let’s pin point the main reasons for going for a clustered approach : > > * scaling up with the processing/traffic load > * geographical distribution > * redundancy and High-Availability > > For the OpenSIPS 2.4 we laid down a roadmap that addresses the > clustering both from the clustering engine itself (the underlayer) and > from the functionalities that will perform on top of the clustering > layer, to share data and state. > > With OpenSIPS 2.4, it will never be easier to built a consistent and > powerful clustered solution: > > * *clustering engine* – enhances the capabilities of controlling the > cluster topology, like re-routing for bypassing broken links, > dynamic joining of new nodes, support for multiple capabilities > per node, data syncing between nodes and many more; > * *distributed user location* – this is a very complex topic as it > exceeds the simple concept of data sharing. By the nature of the > data (the user registrations), you may have different constraints > on how data is roaming in a cluster – registrations may be tied to > a node due NAT or TCP constraints. Even more, a sharding aspect > must be addressed when looking at distributing the pinging effort > across the cluster. So, multiple solutions are viable here, > depending on what is to be achieved (scaling, redundancy) and what > are the network constraints – see a detailed presentation > > of the available solutions; > * *distributed presence server* – quite similar (but less complex) > as the distributed user location, a distributed presence server > provides a consistent, but distributed way of sharing presence > information – SIP entities may publish data via different nodes in > the SIP cluster, while the subscribers may fetch presence data via > multiple various nodes. Two approaches are under work : (a) a > cluster built around a noSQL DB based as primary data storage and > (b) a cluster exclusively relying on OpenSIPS for data sharing; > * *anycast support* – to be able to build a fully-flavored anycast > support (addressing both redundancy and balancing) requires > OpenSIPS to replicate/share transaction state across the nodes in > the cluster (nodes sharing the same anycast IP). Depending on the > nature of the replication (full transaction versus transaction > meta-data) a full anycast and light anycast support will be > available – here is a detailed description on the anycast > > support; > * *clustered media relays* – as OpenSIPS has the ability to work > together with several flavors of media relays (such as RTPproxy, > RTPEngine, MediaProxy), the clustering support will help OpenSIPS > do distributed load-balancing over the relays – even if a relay is > used by multiple nodes in the cluster, all the nodes will share > information on the load on the relay, to avoid overloading or idle > time; > * *distributed call center* – an agent is able to register with > multiple queues on different nodes on a cluster. Still, all the > queues do share the status / availability of the agent and its > statistics for call distribution; > * *custom clustering* – the OpenSIPS clustering underlayer provides > at script level the ability to broadcast (in the cloud) or send to > a given node a custom message/action (with replying possibility) – > this is a very flexible and powerful way to build your custom > distributed functionality directly at script level. > > And because we started on the integration path with OpenSIPS 2.3, and > because we did it well, we decided to push forward on this path with > the 2.4 version as well: > > * more *Homer integration *to be able to report TCP statistics, DB > events and media relay events via HEP; > * *SIPREC integration* for standard call recording. The new SIPREC > module > > provides a standard and transparent (for the call parties) way to > do call recording against an external recorder like Oreka > provided by Orecx > ; > * more *FreeSWITCH integration* in order to capture the call-events > (DTMFs, call status) from FreeSWITCH into OpenSIPS script or for > being able to control a FreeSWITCH call from OpenSIPS script via ESL > * *Asterisk flavored* Load-Balancing for a more realistic and > accurate traffic balancing over Asterisk clusters (as the load > information is fetched in realtime from Asterisk); > > ------------------------------------------------------------------------ > > The timeline for OpenSIPS 2.4 is: > > * Beta Release – 12-16 March 2018 > * Stable Release – 23-27 April 2018 > * General Availability – 1st of May 2018, during OpenSIPS Summit > 2018 > > To talk more about the features of this new release, a public audio > conference will be > available on 21st of November 2017, 4 pm GMT > , thanks to > the kind sponsorship of UberConference > . Anyone is welcome to join to find > out more details or to ask questions about OpenSIPS > 2.4 . > > This is a public and open conference, so no registration is needed, > but if you want to announce your intention to participate, please let > us know here: > > http://blog.opensips.org/2017/11/01/introducing-opensips-2-4/ > > > Best regards, > > -- > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > > _______________________________________________ > News mailing list > News at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/news -------------- next part -------------- An HTML attachment was scrubbed... URL: From xaled at web.de Wed Nov 22 15:46:07 2017 From: xaled at web.de (xaled) Date: Wed, 22 Nov 2017 21:46:07 +0100 Subject: [OpenSIPS-Users] call_center module and loose routing In-Reply-To: <54607825-e6e6-f0dd-17fd-5656ad42deb0@opensips.org> References: <010301d3620e$526d8610$f7489230$@web.de> <007e01d362b3$c9b66c00$5d234400$@web.de> <91d951c8-ffaf-d035-e4d8-47da0c3e9c9c@opensips.org> <54607825-e6e6-f0dd-17fd-5656ad42deb0@opensips.org> Message-ID: <027901d363d2$ebff3790$c3fda6b0$@web.de> Hi Bogdan, thanks for an idea. I will try. Greetings, Xaled From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Wednesday, November 22, 2017 6:17 PM To: Xaled Cc: users at lists.opensips.org Subject: Re: [OpenSIPS-Users] call_center module and loose routing Have you tried to use the local route to identify the call to the agents (based on RURI ?) and force the SIPSERVER address in the $du ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/22/2017 05:44 PM, Xaled wrote: Hi Correct. Thanks, Xaled On 22 Nov 2017, at 16:09, Bogdan-Andrei Iancu > wrote: Hi, So you use pre-loaded route in order to send initial INVITEs from SIPSERVER to OPENSIPS, right ? And what you want to achieve is to have all the calls generated by OpenSIPS Call Center (to agents, to announcements) to actually go again via the SIPSERVER ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/21/2017 12:30 PM, xaled wrote: Hi Bogdan, I use opensips in an scenario where OPENSIPS is addressed from the main SIPSERVER using Route header. OPENSIPS performs different modifications on the INVITE request and gives the INVITE back to the main SIPSERVER. This works fine with the loose_route() so far. Now I want to use the call_center module for RURI agent addressing, queues, etc while still preserving the Route header based next hop routing between main SIPSERVER and OPENSIPS. Would manually parsing the Route Header at OPENSIPS, deleting first route header, saving the hostport value of the second Route header to some variable and provide it as a kind of $du to the call_center module for the next hop routing back to SIPSERVER work? For my scenario to work the Route header with the second Route header value has not to be deleted or modified by the call_center module. Here is the table showing the pseudo values of the RURI and the Route Header at the ingress of SIPSERVER and OPENSIPS respectively. SIPSERVER -> OPENSIPS -> SIPSERVER -> AGENT RURI: 0800CallCenter 0800CallCenter agent agent Route: opensips, sipserver sipserver Does this make it a bit more clearer? Thanks, xaled From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Tuesday, November 21, 2017 10:55 AM To: users at lists.opensips.org ; xaled Subject: Re: [OpenSIPS-Users] call_center module and loose routing Hi, The routing in Call Center module is not compatible with the loose_route(). The loose_route is a mechanism used by a SIP proxy staying in the middle of a SIP session. On the other hand, the Call Center is using a SIP Back-2-Back engine in order to implement the queuing. Better try to explain what are you trying to achieve here. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/20/2017 04:46 PM, xaled wrote: Hi I want to use opensips with call_center module in a loose route scenario where the INVITE has to be routed by opensips to the next hop (sipserver) based on what is left from Route header after loose_route() is performed. The call_center module has to update the RURI with agent location and let the next hop from the route header perform the actual routing to the agent. Here are the expected route and RURI header values at the opensips, sipserver and agent: opensips sipserver agent RURI: 0800CallCenter agent agent Route: opensips, sipserver sipserver Will something like this work as described above: loose_route(); cc_handle_call(test_flow); Thanks, xaled _______________________________________________ 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 23 04:59:43 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 23 Nov 2017 11:59:43 +0200 Subject: [OpenSIPS-Users] fix_nated_sdp and rtpproxy In-Reply-To: References: <268801e7-54c7-2693-c756-efd01dcd0717@opensips.org> Message-ID: <73143ac8-8cb8-68bb-f6ef-bc6c758f573d@opensips.org> Hi Yuri, No, in the answer() you need to use a separate set of flags, reflecting your intentions in regards to the SDP you received from the callee side. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/21/2017 02:15 PM, Yury Kirsanov wrote: > Hi Bogdan, > Yes, you're correct, I should try to analyze if initial INVITE packet > coming from outside (Internet) contains a private IP in SDP and in > that case just not use "r" flag in rtpproxy_offer(). Thanks a lot for > clarifying this! Just one more question - do I need to specify same > flags in rtpproxy_answer() or is it sufficient just to call it without > parameters? Thanks! > > Best regards, > Yury. > > 2017-11-21 20:34 GMT+11:00 Bogdan-Andrei Iancu >: > > Hi Yury, > > OK, so SIP goes like 192.168.1.1 / 1.1.1.1 -> 2.2.2.2 -> 3.3.3.3 , > while RTP 192.168.1.1 / 1.1.1.1 -> 3.3.3.4/5 . > or ? what is the full path for SIP and RTP ? > > Without the "r" flag, rtpproxy will stay and listen for incoming > traffic in order to learn the end points, it will not use the SIP > signaling IP for anything (rtpproxy makes no assumptions based on > the SIP ips) > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 11/16/2017 10:25 PM, Yury Kirsanov wrote: >> Bogdan, >> But in SIP packets from upstream carriers there's a public IP >> address in SDP header, pointing to different IP than SIP packet >> is coming from. Both of these IP addresses are public. But >> customers who connect via public Internet to OpenSIPS server >> sitting on public interface are actually behind NAT and that's >> why packets from them contain private IP address in SDP header. >> Example, to clarify things up: >> >> Client has a PBX behind a router that's doing NAT. Internal IP of >> PBX is 192.168.1.1, public IP of client is 1.1.1.1, router is >> doing just NAT without SIP ALG. Customer connects over the >> Internet to OpenSIPS at 2.2.2.2, so packets are reaching >> OpenSIPS. But in SDP of customer's SIP packets there's >> c=192.168.1.1 and all the rest of descriptors are with the same IP. >> >> Now, provider is at IP of 3.3.3.3 having RTP proxies at 3.3.3.4 >> and 3.3.3.5. So, packets from provider are reaching OpenSIPS at >> 2.2.2.2 via the Internet and contain different IP address in SDP >> from SIP packet (3.3.3.4 or 3.3.3.5 in SDP, SIP has 3.3.3.3) >> >> RTP Proxy at my side is operating in bridge mode, one side is >> connected to public interface with IP of 4.4.4.4 and 4.4.4.5 (two >> proxies), and another side is connected to DMZ internal LAN, IPs >> 10.0.0.4 and 10.0.0.5. >> >> So, in order to use fix_nated_sdp() AND engage RTP proxy when the >> call is coming from customer - I should NOT use flag "r". But if >> the call is coming from provider - I must be using that flag or >> otherwise RTP Proxy would be expecting packets to be coming from >> 3.3.3.3 rather than 3.3.3.4 or 3.3.3.5. That's why my idea was >> first to update customer's part of SDP from 192.168.1.1 to >> 1.1.1.1 (where packets from customer are coming from) and then >> pass this modified SIP packet with new SDP to RTP proxy, so it >> would have flag "r" enabled and start sending packets to IP of >> 1.1.1.1 straight away. >> >> Thanks and best regards, >> Yury. >> >> >> 2017-11-17 3:26 GMT+11:00 Bogdan-Andrei Iancu >> >: >> >> Using the "i" and "e" flags to control the RTP interfaces >> does not affect the "learning" mode. You should you "r" flag >> only if the received IP in SDP is public. Otherwise do not >> use it. >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> >> On 11/16/2017 03:07 PM, Yury Kirsanov wrote: >>> Hi Bogdan, >>> Thanks for clarification, I didn't know that RTP Proxy >>> learns from source packets, it's quite hard to find >>> documentation on it. The problem here is that I'm using RTP >>> Proxy in bridge mode, connecting public interface with >>> internal LAN and my upstream providers are using sets of >>> their own proxies too, besides customers connecting to this >>> OpenSIPS server. So I have to use flag "r" (and, of course, >>> flags of direction - "i" and "e") or else there would be no >>> audio on calls when SDP in provider's answer is different >>> from SIP packet source IP. Of course, I can analyze where >>> the initial INVITE packet is coming from and enable >>> different options but that would be more complex rather than >>> just have SDP modified before passing it to RTP Proxy. >>> Regards, >>> Yury. >>> >>> 2017-11-17 0:01 GMT+11:00 Bogdan-Andrei Iancu >>> >: >>> >>> Hi Yury, >>> >>> RTPproxy learns not only from SDP, but based on the >>> incoming RTP traffic. Your Client will send RTP to >>> RTPproxy (as the Client will receive back a valid IP in >>> the SDP) and RTPproxy will see where the RTP traffic >>> comes from and starts sending back RTP to that IP:port. >>> >>> Just be sure you do not use the "a" and "r" flags when >>> calling rtpproxy engage. >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> OpenSIPS Founder and Developer >>> http://www.opensips-solutions.com >>> >>> >>> On 11/16/2017 02:48 PM, Yury Kirsanov wrote: >>>> Hi Bogdan, >>>> Thanks for your answer and support! But the issue is >>>> that there is a private IP address in SDP coming from >>>> the client so if I just pass it via RTP proxy - it >>>> would learn that incorrect IP and would try to forward >>>> packets to it causing one-way audio. That's why I first >>>> wanted to overstamp that incorrect IP with received IP >>>> and then pass it to the RTP Proxy. Hope this makes my >>>> idea clear. >>>> >>>> Regards, >>>> Yury. >>>> >>>> 2017-11-16 23:32 GMT+11:00 Bogdan-Andrei Iancu >>>> >: >>>> >>>> Hi Yury, >>>> >>>> Why do you need to update the incoming SDP before >>>> engaging RTPproxy ? Just pass to it whatever you >>>> get into SDP (even if private) and let rtpproxy to >>>> "learn" the correct IP:port of the media end points >>>> based on the incoming traffic. >>>> >>>> Regards, >>>> >>>> Bogdan-Andrei Iancu >>>> OpenSIPS Founder and Developer >>>> http://www.opensips-solutions.com >>>> >>>> >>>> On 11/14/2017 04:05 AM, Yury Kirsanov wrote: >>>>> Hi, >>>>> I have a problem with remote client who can't >>>>> properly configure NAT and is sending private IP >>>>> in all SDP and SIP packets. I've resolved issues >>>>> with registration but can't resolve the issue with >>>>> SDP. My topology is as following: >>>>> >>>>> ----- ------- >>>>> ---------- -------- ----------- >>>>> >>>>> So, I need to simultaneously engage RTP Proxy to >>>>> bridge external Internet and our local LAN and >>>>> need to rewrite SDP body of original Client's >>>>> packet with received IP address. If I'm doing >>>>> fix_nated_sdp and then rtpproxy_offer - I'm >>>>> getting doubled IPs in SDP message, like this: >>>>> >>>>> 100.11.100.200192.168.20.200 >>>>> >>>>> >>>>> Is there any way to update SDP of client's packet >>>>> and then command RTP Proxy to use updated IP? 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 netaskd at gmail.com Thu Nov 23 06:10:27 2017 From: netaskd at gmail.com (Dmitry) Date: Thu, 23 Nov 2017 14:10:27 +0300 Subject: [OpenSIPS-Users] Question about run exec from route script periodically. Message-ID: <003601d3644b$abaccdf0$030669d0$@gmail.com> Hello All, I don’t found the info in google, so I’ll try to ask here. Data: I have some custom SIP client for iIOS with logic wake up via push notification and most of the time it working in a background. The client send REGISTER in first time and go to the background. Now I have a special _useragent_ for next steps for detect it as my app. When a caller try to call to special client, opensips detect _useragent_ and run php script for send push notification (before send INVITE forth) to iOS server and will wait 2 sec in async mode. In this time, iOS client (callee) receive push, “wake up” and send REGISTER to opensips. So, registration completed, we run LOOKUP route with search updated “Contact” data for callee. ==opensips router snippet == async(exec("/etc/opensips/ios/wakeup.php '$var(ua)' '$var(key)'", "$var(out)", "$var(err)"), LOOKUP); === My question is: How can I run an external script (via exec/etc app inside the routing script) _periodically_ for keep alive _useragent_ record in location table? Something like a cron inside opensips maybe. I’ll glad to any ideas. PS. For keep alive “normal” users I use nathelper directives: sipping_bflag remove_on_timeout_bflag natping_interval ping_threshold max_pings_lost BR, Dmitry -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Nov 23 06:15:59 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 23 Nov 2017 13:15:59 +0200 Subject: [OpenSIPS-Users] Opensips 2.3.2 In-Reply-To: <26890667.2767301511272013887.JavaMail.defaultUser@defaultHost> References: <26890667.2767301511272013887.JavaMail.defaultUser@defaultHost> Message-ID: <7a94e5fc-bc4c-e322-9a0a-915cfe3dd8e8@opensips.org> Hi, I'm 100% sure I understood your scenario, but what you need to do is an interface exchange for the calls from users to the SBC. You can do this using the force_send_socket() function (http://www.opensips.org/Documentation/Script-CoreFunctions-2-3#toc16) before doing the relay. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/21/2017 03:46 PM, mas_66--- via Users wrote: > Hello there > > I'm having an issue with placing external calls. > I can register softphones and place calls between them fine, but have failed in > the next step which is to hook up to my SBC(working) to call the outside world. > On Opensips my interfaces (with fictional IP's are: > > 192.168.54.55 ENS160 > (159.99.8.111 is my Public IP that points to LAN 192.168.54.55 for my > softphones to register via) > 172.23.100.55 ENS224(this communicates with an interface on my SBC > 172.23.100.250) > > so a call path will be: 159.99.8.111->192.168.54.55->172.23.100.55->172. > 23.100.250->carrier > > Im sure it has to do with the interface presented to my SBC which currently > shows as extn111 at 192.168.54.55. > the call hits my SBC but no further, or my routing set up. > I want to force the call to be 111 at 172.23.100.55 but not sure how to. > I have experimented with domains and routing but as a new Opensipper im > guessing a lot of it and trying to read online too but not much out there that > relates. > If someone can offer some examples of Dynamic routing or anything that would > help that would be great. > > Many Thanks > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From denys.pozniak at crazycall.com Thu Nov 23 07:13:48 2017 From: denys.pozniak at crazycall.com (Denys Pozniak) Date: Thu, 23 Nov 2017 13:13:48 +0100 Subject: [OpenSIPS-Users] Question about script variables Message-ID: Hello! I am newbie in OpenSIPS and probably my question is stupid, but I could not sleep without correct answer :) In the book *Building Telephony Systems with OpenSIPS Second Edition* I found next part (p. 97): *The script variables* *As the name indicates, these variables are strictly bound to the scripting routes. The* *variables are visible only in the routing blocks and they have no persistency—when* *the route (with all its subroutes) ends, the variables are discarded.* Is it correct explanation?According to this, variable lives on SIP message level when script are executing, but I think something is not correct as variable lives on proccess level. Please help me to sort out this! BR Denys -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Thu Nov 23 08:34:51 2017 From: liviu at opensips.org (Liviu Chircu) Date: Thu, 23 Nov 2017 15:34:51 +0200 Subject: [OpenSIPS-Users] Question about script variables In-Reply-To: References: Message-ID: <3783ee34-f9bf-5ec3-26c7-29419202878e@opensips.org> Hi Denys, The book is doing the correct thing - educating the script writer that he must not rely on any $var() holders once an instance of script execution ends for a given message. You, on the other hand, have the additional knowledge of an underlying optimization: the $var() values are not reset (actually, they are rarely touched at all beyond the script manipulations). The fact that you occasionally find the value that you left in the previous message is just a coincidence, and you cannot reliably use their initial values. Since this is just a "nice to know" developer quirk, I think it would cause confusion if the authors were to include it in the book. If you talk C, you can think of $vars as "uninitialized stack variables". Regards, Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 23.11.2017 14:13, Denys Pozniak wrote: > Hello! > > I am newbie in OpenSIPS and probably my question is stupid, but I > could not sleep without correct answer :) > In the book *Building Telephony Systems with OpenSIPS Second Edition* > I found next part (p. 97): > > /The script variables/ > /As the name indicates, these variables are strictly bound to the > scripting routes. The/ > /variables are visible only in the routing blocks and they have no > persistency—when/ > /the route (with all its subroutes) ends, the variables are discarded./ > > Is it correct explanation?According to this, variable lives on SIP > message level when script are executing, but I think something is not > correct as variable lives on proccess level. > > Please help me to sort out this! > > BR > Denys > > > _______________________________________________ > 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 denys.pozniak at crazycall.com Thu Nov 23 08:57:06 2017 From: denys.pozniak at crazycall.com (Denys Pozniak) Date: Thu, 23 Nov 2017 14:57:06 +0100 Subject: [OpenSIPS-Users] Question about script variables In-Reply-To: <3783ee34-f9bf-5ec3-26c7-29419202878e@opensips.org> References: <3783ee34-f9bf-5ec3-26c7-29419202878e@opensips.org> Message-ID: Thank you for the explanation! On 23 November 2017 at 14:34, Liviu Chircu wrote: > Hi Denys, > > The book is doing the correct thing - educating the script writer that he > must not rely on any $var() holders once an instance of script execution > ends for a given message. > > You, on the other hand, have the additional knowledge of an underlying > optimization: the $var() values are not reset (actually, they are rarely > touched at all beyond the script manipulations). The fact that you > occasionally find the value that you left in the previous message is just a > coincidence, and you cannot reliably use their initial values. Since this > is just a "nice to know" developer quirk, I think it would cause confusion > if the authors were to include it in the book. > > If you talk C, you can think of $vars as "uninitialized stack variables". > > Regards, > > Liviu Chircu > OpenSIPS Developerhttp://www.opensips-solutions.com > > On 23.11.2017 14:13, Denys Pozniak wrote: > > Hello! > > I am newbie in OpenSIPS and probably my question is stupid, but I could > not sleep without correct answer :) > In the book *Building Telephony Systems with OpenSIPS Second Edition* I > found next part (p. 97): > > *The script variables* > *As the name indicates, these variables are strictly bound to the > scripting routes. The* > *variables are visible only in the routing blocks and they have no > persistency—when* > *the route (with all its subroutes) ends, the variables are discarded.* > > Is it correct explanation?According to this, variable lives on SIP message > level when script are executing, but I think something is not correct as > variable lives on proccess level. > > Please help me to sort out this! > > BR > Denys > > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- BR Denys Pozniak -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Nov 23 09:11:07 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 23 Nov 2017 16:11:07 +0200 Subject: [OpenSIPS-Users] Question about run exec from route script periodically. In-Reply-To: <003601d3644b$abaccdf0$030669d0$@gmail.com> References: <003601d3644b$abaccdf0$030669d0$@gmail.com> Message-ID: Hello Dmitry, Maybe the timer_route[] will help with this ? http://www.opensips.org/Documentation/Script-Routes-2-3#toc8 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/23/2017 01:10 PM, Dmitry wrote: > > Hello All, > > I don’t found the info in google, so I’ll try to ask here. > > Data: > > I have some custom SIP client for iIOS with logic wake up via push > notification and most of the time it working in a background. The > client send REGISTER in first time and go to the background. Now I > have a special _/useragent/_ for next steps for detect it as my app. > When a caller try to call to special client, opensips detect > _useragent_ and run php script for send push notification (before send > INVITE forth) to iOS server and will wait 2 sec in async mode. In this > time, iOS client (callee) receive push, “wake up” and send REGISTER to > opensips. So, registration completed, we run LOOKUP route with search > updated “Contact” data for callee. > > ==opensips router snippet == > > async(exec("/etc/opensips/ios/wakeup.php '$var(ua)' '$var(key)'", > "$var(out)", "$var(err)"), LOOKUP); > > === > > My question is: > > How can I run an external script (via exec/etc app inside the routing > script) _periodically_ for keep alive _/useragent/_ record in location > table? Something like a cron inside opensips maybe. I’ll glad to any > ideas. > > PS. For keep alive “normal” users I use nathelper directives: > > sipping_bflag > > remove_on_timeout_bflag > > natping_interval > > ping_threshold > > max_pings_lost > > BR, > > Dmitry > > > > _______________________________________________ > 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 netaskd at gmail.com Thu Nov 23 09:56:22 2017 From: netaskd at gmail.com (Dmitry) Date: Thu, 23 Nov 2017 17:56:22 +0300 Subject: [OpenSIPS-Users] Question about run exec from route script periodically. In-Reply-To: References: <003601d3644b$abaccdf0$030669d0$@gmail.com> Message-ID: <005901d3646b$3b111590$b13340b0$@gmail.com> Hello Bogdan-Andrei, Thank you for reply. It seems like that’s what I need. BR, Dmitry From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Thursday, November 23, 2017 5:11 PM To: OpenSIPS users mailling list ; Dmitry Subject: Re: [OpenSIPS-Users] Question about run exec from route script periodically. Hello Dmitry, Maybe the timer_route[] will help with this ? http://www.opensips.org/Documentation/Script-Routes-2-3#toc8 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/23/2017 01:10 PM, Dmitry wrote: Hello All, I don’t found the info in google, so I’ll try to ask here. Data: I have some custom SIP client for iIOS with logic wake up via push notification and most of the time it working in a background. The client send REGISTER in first time and go to the background. Now I have a special _useragent_ for next steps for detect it as my app. When a caller try to call to special client, opensips detect _useragent_ and run php script for send push notification (before send INVITE forth) to iOS server and will wait 2 sec in async mode. In this time, iOS client (callee) receive push, “wake up” and send REGISTER to opensips. So, registration completed, we run LOOKUP route with search updated “Contact” data for callee. ==opensips router snippet == async(exec("/etc/opensips/ios/wakeup.php '$var(ua)' '$var(key)'", "$var(out)", "$var(err)"), LOOKUP); === My question is: How can I run an external script (via exec/etc app inside the routing script) _periodically_ for keep alive _useragent_ record in location table? Something like a cron inside opensips maybe. I’ll glad to any ideas. PS. For keep alive “normal” users I use nathelper directives: sipping_bflag remove_on_timeout_bflag natping_interval ping_threshold max_pings_lost BR, Dmitry _______________________________________________ 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 netaskd at gmail.com Thu Nov 23 17:19:18 2017 From: netaskd at gmail.com (netaskd at gmail.com) Date: Fri, 24 Nov 2017 01:19:18 +0300 Subject: [OpenSIPS-Users] Question about run exec from route script periodically. In-Reply-To: <005901d3646b$3b111590$b13340b0$@gmail.com> Message-ID: <0e122c9e-7a4b-4d45-8cd6-5b3e7e2c48ec@gmail.com> Hi Bogdan-Andrei, I've tested the timer_route, it works well. But ,as it turned out, my issue is broader. === So, if I use the timer_route, it runs some logic at specific time. I'm imagine, that users will be more then 100, so 100+ requests at one time will send keepalive to iOS server... Thereby, I'll try to rephrase my question: How can I run an external script _periodically_ for keep alive _useragent_ record in location table, and do not do it simultaneously. I guess, I need to bind the logic to expires time of the acc. I can try to get it time from the location table, but maybe exists some graceful way for it? BR, Dmitry ________________________________ From: Dmitry Sent: Nov 23, 2017 5:56 PM To: 'Bogdan-Andrei Iancu'; 'OpenSIPS users mailling list' Subject: RE: [OpenSIPS-Users] Question about run exec from route script periodically. Hello Bogdan-Andrei, Thank you for reply. It seems like that’s what I need. BR, Dmitry   From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Thursday, November 23, 2017 5:11 PM To: OpenSIPS users mailling list ; Dmitry Subject: Re: [OpenSIPS-Users] Question about run exec from route script periodically.   Hello Dmitry, Maybe the timer_route[] will help with this ?     http://www.opensips.org/Documentation/Script-Routes-2-3#toc8 Regards, Bogdan-Andrei Iancu   OpenSIPS Founder and Developer   http://www.opensips-solutions.com On 11/23/2017 01:10 PM, Dmitry wrote: > > Hello All, > > I don’t found the info in google, so I’ll try to ask here. > > Data: > > I have some custom SIP client for iIOS with logic wake up via push notification and most of the time it working in a background. The client send REGISTER in first time and go to the background. Now I have a special _useragent_ for next steps for detect it as my app. When a caller try to call to special client, opensips detect _useragent_ and run php script for send push notification (before send INVITE forth) to iOS server and will wait 2 sec in async mode. In this time, iOS client (callee) receive push, “wake up” and send REGISTER to opensips. So, registration completed, we run LOOKUP route with search updated “Contact” data for callee. > > ==opensips router snippet == > > async(exec("/etc/opensips/ios/wakeup.php '$var(ua)' '$var(key)'", "$var(out)", "$var(err)"), LOOKUP); > > === > > My question is: > > How can I run an external script (via exec/etc app inside the routing script) _periodically_ for keep alive _useragent_ record in location table? Something like a cron inside opensips maybe. I’ll glad to any ideas. > > PS. For keep alive “normal” users I use nathelper directives: > > sipping_bflag > > remove_on_timeout_bflag > > natping_interval > > ping_threshold > > max_pings_lost > >   > > BR, > > Dmitry > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users   -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Nov 24 06:35:14 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 24 Nov 2017 13:35:14 +0200 Subject: [OpenSIPS-Users] Question about run exec from route script periodically. In-Reply-To: <0e122c9e-7a4b-4d45-8cd6-5b3e7e2c48ec@gmail.com> References: <0e122c9e-7a4b-4d45-8cd6-5b3e7e2c48ec@gmail.com> Message-ID: Hi Dmitry, So, basically you want to do a kind of user pinging, but not via SIP (as nathelper is doing), but via PN, using your external script ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/24/2017 12:19 AM, netaskd at gmail.com wrote: > Hi Bogdan-Andrei, > I've tested the timer_route, it works well. But ,as it turned out, my > issue is broader. > === > So, if I use the timer_route, it runs some logic at specific time. I'm > imagine, that users will be more then 100, so 100+ requests at one > time will send keepalive to iOS server... > Thereby, I'll try to rephrase my question: > How can I run an external script _periodically_ for keep alive > _/useragent/_ record in location table, and do not do it > simultaneously. I guess, I need to bind the logic to expires time of > the acc. I can try to get it time from the location table, but maybe > exists some graceful way for it? > BR, > Dmitry > > ------------------------------------------------------------------------ > *From:* Dmitry > *Sent:* Nov 23, 2017 5:56 PM > *To:* 'Bogdan-Andrei Iancu'; 'OpenSIPS users mailling list' > *Subject:* RE: [OpenSIPS-Users] Question about run exec from route > script periodically. > > Hello Bogdan-Andrei, > > Thank you for reply. It seems like that’s what I need. > > BR, > > Dmitry > > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* Thursday, November 23, 2017 5:11 PM > *To:* OpenSIPS users mailling list ; Dmitry > > *Subject:* Re: [OpenSIPS-Users] Question about run exec from route > script periodically. > > > Hello Dmitry, > > Maybe the timer_route[] will help with this ? > http://www.opensips.org/Documentation/Script-Routes-2-3#toc8 > > Regards, > > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 11/23/2017 01:10 PM, Dmitry wrote: > > Hello All, > > I don’t found the info in google, so I’ll try to ask here. > > Data: > > I have some custom SIP client for iIOS with logic wake up via push > notification and most of the time it working in a background. The > client send REGISTER in first time and go to the background. Now I > have a special _/useragent/_ for next steps for detect it as my > app. When a caller try to call to special client, opensips detect > _useragent_ and run php script for send push notification (before > send INVITE forth) to iOS server and will wait 2 sec in async > mode. In this time, iOS client (callee) receive push, “wake up” > and send REGISTER to opensips. So, registration completed, we run > LOOKUP route with search updated “Contact” data for callee. > > ==opensips router snippet == > > async(exec("/etc/opensips/ios/wakeup.php '$var(ua)' '$var(key)'", > "$var(out)", "$var(err)"), LOOKUP); > > === > > My question is: > > How can I run an external script (via exec/etc app inside the > routing script) _periodically_ for keep alive _/useragent/_ record > in location table? Something like a cron inside opensips maybe. > I’ll glad to any ideas. > > PS. For keep alive “normal” users I use nathelper directives: > > sipping_bflag > > remove_on_timeout_bflag > > natping_interval > > ping_threshold > > max_pings_lost > > > BR, > > Dmitry > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From netaskd at gmail.com Fri Nov 24 06:52:05 2017 From: netaskd at gmail.com (Dmitry) Date: Fri, 24 Nov 2017 14:52:05 +0300 Subject: [OpenSIPS-Users] Question about run exec from route script periodically. In-Reply-To: References: <0e122c9e-7a4b-4d45-8cd6-5b3e7e2c48ec@gmail.com> Message-ID: <008801d3651a$a6b9b5e0$f42d21a0$@gmail.com> Hi Bogdan-Andrei, Exactly. Script send a push via iOS server, the server send the push notification for wake up a client, it wakes up and send REGISTER to my proxy, and authenticated. Then the client go to background again. That’s all. I don't use nathelper module, because it is not make sense (client will not respond, because it frozen) So, I need to concern about keep alive _useragent_ in DB by himself. From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Friday, November 24, 2017 2:35 PM To: netaskd at gmail.com; 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Question about run exec from route script periodically. Hi Dmitry, So, basically you want to do a kind of user pinging, but not via SIP (as nathelper is doing), but via PN, using your external script ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/24/2017 12:19 AM, netaskd at gmail.com wrote: Hi Bogdan-Andrei, I've tested the timer_route, it works well. But ,as it turned out, my issue is broader. === So, if I use the timer_route, it runs some logic at specific time. I'm imagine, that users will be more then 100, so 100+ requests at one time will send keepalive to iOS server... Thereby, I'll try to rephrase my question: How can I run an external script _periodically_ for keep alive _useragent_ record in location table, and do not do it simultaneously. I guess, I need to bind the logic to expires time of the acc. I can try to get it time from the location table, but maybe exists some graceful way for it? BR, Dmitry _____ From: Dmitry Sent: Nov 23, 2017 5:56 PM To: 'Bogdan-Andrei Iancu'; 'OpenSIPS users mailling list' Subject: RE: [OpenSIPS-Users] Question about run exec from route script periodically. Hello Bogdan-Andrei, Thank you for reply. It seems like that’s what I need. BR, Dmitry From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Thursday, November 23, 2017 5:11 PM To: OpenSIPS users mailling list ; Dmitry Subject: Re: [OpenSIPS-Users] Question about run exec from route script periodically. Hello Dmitry, Maybe the timer_route[] will help with this ? http://www.opensips.org/Documentation/Script-Routes-2-3#toc8 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/23/2017 01:10 PM, Dmitry wrote: Hello All, I don’t found the info in google, so I’ll try to ask here. Data: I have some custom SIP client for iIOS with logic wake up via push notification and most of the time it working in a background. The client send REGISTER in first time and go to the background. Now I have a special _useragent_ for next steps for detect it as my app. When a caller try to call to special client, opensips detect _useragent_ and run php script for send push notification (before send INVITE forth) to iOS server and will wait 2 sec in async mode. In this time, iOS client (callee) receive push, “wake up” and send REGISTER to opensips. So, registration completed, we run LOOKUP route with search updated “Contact” data for callee. ==opensips router snippet == async(exec("/etc/opensips/ios/wakeup.php '$var(ua)' '$var(key)'", "$var(out)", "$var(err)"), LOOKUP); === My question is: How can I run an external script (via exec/etc app inside the routing script) _periodically_ for keep alive _useragent_ record in location table? Something like a cron inside opensips maybe. I’ll glad to any ideas. PS. For keep alive “normal” users I use nathelper directives: sipping_bflag remove_on_timeout_bflag natping_interval ping_threshold max_pings_lost BR, Dmitry _______________________________________________ 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 social at bohboh.info Fri Nov 24 10:22:23 2017 From: social at bohboh.info (Social Boh) Date: Fri, 24 Nov 2017 10:22:23 -0500 Subject: [OpenSIPS-Users] OpenSIPs Control Panel Homer Menu In-Reply-To: <32fe0423-10e7-6f23-2d01-3ebeb4f86b98@bohboh.info> References: <31a910a6-10e7-6aa8-cb53-31d35a6e64ba@bohboh.info> <667cccd5-ae01-9780-9c0d-eb34c39355aa@bohboh.info> <01993f73-4c11-de06-c8fe-3e5b93c62580@opensips.org> <32fe0423-10e7-6f23-2d01-3ebeb4f86b98@bohboh.info> Message-ID: <5d6a3c38-1c73-9534-128f-eab976d3ffc0@bohboh.info> Hello, now, after working fine for a while, I have this error in the OpenSIPs Control Panel, Homer menu: { "sid": "ic8jmj4cmv8a9f1t04mrhs8if4", "auth": "false", "status": 403, "message": "wrong session", "data": [ ] } Access to OCP is over IP. /var/www/html/opensips-cp/config/tools/system/homer/local.inc.php $homer_URL = "http://HomerIP"; define('EXTERNAL_AUTH_URI', "http://OCP_IP/cp/tools/system/homer/auth.php"); Regards --- I'm SoCIaL, MayBe El 03/11/2017 a las 10:22, Social Boh escribió: > > The error was related with this semicolon (before the last parenthesis): > > define('EXTERNAL_AUTH_URI', > "http://5.6.7.8/cp/tools/system/homer/auth.php"*;*); > > Thank you. > > Regards > > --- > I'm SoCIaL, MayBe > El 03/11/2017 a las 07:15, Răzvan Crainea escribió: >> This time there seems to be an issue on the homer side. Can you check >> the logs to see if homer is saying anything useful? >> >> Best regards, >> >> Răzvan Crainea >> OpenSIPS Developer >> www.opensips-solutions.com >> >> On 11/03/2017 02:06 PM, Social Boh wrote: >>> Hello, >>> >>> Now I have this error (homer side): >>> >>> GET >>> /api/v1/redirect?externalid=rx88kxQUs1DNT6es2ybwBvhevecJwjnY&url=http%3A%2F%2F1.2.3.4.0%2F >>> HTTP/1.1. >>> Host: 104.236.105.0. >>> User-Agent: Mozilla/5.0 (Windows NT 6.1; Win64; x64; rv:56.0) >>> Gecko/20100101 Firefox/56.0. >>> Accept: >>> text/html,application/xhtml+xml,application/xml;q=0.9,*/*;q=0.8. >>> Accept-Language: es-ES,es;q=0.8,en-US;q=0.5,en;q=0.3. >>> Accept-Encoding: gzip, deflate. >>> Referer: http://5.6.7.8/cp/tools/system/homer/homer.php. >>> Connection: keep-alive. >>> Upgrade-Insecure-Requests: 1. >>> >>> HTTP/1.0 500 Internal Server Error. >>> Date: Fri, 03 Nov 2017 11:47:36 GMT. >>> Server: Apache/2.4.6 (CentOS) OpenSSL/1.0.2k-fips PHP/5.4.16. >>> X-Powered-By: PHP/5.4.16. >>> Content-Length: 0. >>> Connection: close. >>> Content-Type: text/html; charset=UTF-8. >>> >>> My homer configuration is: >>> >>> define('AUTHENTICATION',"External"); >>> define('EXTERNAL_AUTH_URI', >>> "http://5.6.7.8/cp/tools/system/homer/auth.php";); >>> define('EXTERNAL_AUTH_METHOD', "GET"); >>> define('EXTERNAL_AUTH_PARAM', "param=KEY"); >>> define('EXTERNAL_AUTH_REQUEST_TYPE', "get"); >>> define('EXTERNAL_AUTH_REQUEST_KEY', "externalid"); >>> define('EXTERNAL_AUTH_POSITIVE_REPLY', "200"); >>> define('EXTERNAL_AUTH_REPLY_USER_INFO', "json"); >>> >>> 1.2.3.4 is the Homer IP and 5.6.7.8 is the OpenSIPs Control Panel IP >>> >>> Thank you >>> >>> Regards >>> >>> --- >>> I'm SoCIaL, MayBe >>> >>> El 03/11/2017 a las 04:47, Răzvan Crainea escribió: >>>> Hello! >>>> >>>> You should install the php apc library. On Ubuntu this is php-apcu. >>>> >>>> Best regards, >>>> >>>> Răzvan Crainea >>>> OpenSIPS Developer >>>> www.opensips-solutions.com >>>> >>>> On 11/02/2017 10:09 PM, Social Boh wrote: >>>>> Hello, >>>>> >>>>> each time I try to access Homer Menu on OpenSIPs Control Panel >>>>> 7.3.2, on the apache log: >>>>> >>>>> Call to undefined function apc_store() in >>>>> /var/www/html/opensips-cp/web/tools/system/homer/homer.php on line 5 >>>>> >>>>> Blank page >>>>> >>>>> OpenSIPs Homer Configuration: >>>>> >>>>> $homer_URL = "http://homer.mydomain.org"; >>>>> >>>>> $homer_auth_method = "get"; >>>>> >>>>> Homer OpenSIPs configuration (preference.php) >>>>> >>>>> define('AUTHENTICATION',"External"); >>>>> define('EXTERNAL_AUTH_URI', >>>>> "http://os.mydomain.org/cp/tools/system/homer/auth.php";); >>>>> define('EXTERNAL_AUTH_METHOD', "GET"); >>>>> define('EXTERNAL_AUTH_PARAM', "param=KEY"); >>>>> define('EXTERNAL_AUTH_REQUEST_TYPE', "get"); >>>>> define('EXTERNAL_AUTH_REQUEST_KEY', "externalid"); >>>>> define('EXTERNAL_AUTH_POSITIVE_REPLY', "200"); >>>>> define('EXTERNAL_AUTH_REPLY_USER_INFO', "json"); >>>>> >>>>> Any hint >>>>> >>>>> Thank you in advance. >>>>> >>>>> Regards >>>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From vishalmpai at gmail.com Fri Nov 24 13:47:19 2017 From: vishalmpai at gmail.com (Vishal Pai) Date: Fri, 24 Nov 2017 18:47:19 +0000 Subject: [OpenSIPS-Users] Opensips basic Message-ID: Hello! I am newbie in OpenSIPS. I just want to know what is command to reload opensips after updating .cfg file. Or I just need to restart it. Thanks Vishal Pai -------------- next part -------------- An HTML attachment was scrubbed... URL: From vishalmpai at gmail.com Fri Nov 24 13:56:14 2017 From: vishalmpai at gmail.com (Vishal Pai) Date: Fri, 24 Nov 2017 18:56:14 +0000 Subject: [OpenSIPS-Users] Regarding avp_db_query Message-ID: Hi Everyone I just want to know what is correct to fetch all the row from database using avp_db_query. Currently I am getting only first row in result. One more think I want to know is it possible to get array of for variable in Opensips configuration file. Thanks Vishal Pai -------------- next part -------------- An HTML attachment was scrubbed... URL: From aqsyounas at gmail.com Fri Nov 24 14:55:37 2017 From: aqsyounas at gmail.com (Aqs Younas) Date: Sat, 25 Nov 2017 00:55:37 +0500 Subject: [OpenSIPS-Users] Opensips basic In-Reply-To: References: Message-ID: There is no such command to reload configuration. You had to restart it. On 24 Nov 2017 11:51 pm, "Vishal Pai" wrote: > Hello! > > I am newbie in OpenSIPS. I just want to know what is command to reload > opensips after updating .cfg file. Or I just need to restart it. > > > > > Thanks > Vishal Pai > > > > > > _______________________________________________ > 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 Sat Nov 25 01:28:19 2017 From: vishalmpai at gmail.com (Vishal Pai) Date: Sat, 25 Nov 2017 11:58:19 +0530 Subject: [OpenSIPS-Users] Opensips basic In-Reply-To: References: Message-ID: Thanks Younas. On Sat, Nov 25, 2017 at 1:25 AM, Aqs Younas wrote: > There is no such command to reload configuration. You had to restart it. > > On 24 Nov 2017 11:51 pm, "Vishal Pai" wrote: > >> Hello! >> >> I am newbie in OpenSIPS. I just want to know what is command to reload >> opensips after updating .cfg file. Or I just need to restart it. >> >> >> >> >> Thanks >> Vishal Pai >> >> >> >> >> >> _______________________________________________ >> 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 michele.pinassi at unisi.it Sat Nov 25 01:54:45 2017 From: michele.pinassi at unisi.it (michele.pinassi) Date: Sat, 25 Nov 2017 07:54:45 +0100 Subject: [OpenSIPS-Users] R: Re: Opensips basic Message-ID: Using opensipsctl reload can help you minimizing OpenSips downtime. Michele Inviato da smartphone Samsung Galaxy. -------- Messaggio originale --------Da: Aqs Younas Data: 24/11/17 20:55 (GMT+01:00) A: OpenSIPS users mailling list Oggetto: Re: [OpenSIPS-Users] Opensips basic There is no such command to reload configuration. You had to restart it. On 24 Nov 2017 11:51 pm, "Vishal Pai" wrote: Hello! I am newbie in OpenSIPS. I just want to know what is command to reload opensips after updating .cfg file. Or I just need to restart it. ThanksVishal Pai _______________________________________________ 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 Mon Nov 27 03:33:08 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Mon, 27 Nov 2017 10:33:08 +0200 Subject: [OpenSIPS-Users] Regarding avp_db_query In-Reply-To: References: Message-ID: <0e449337-c52e-371a-b18e-468c8272b13e@opensips.org> Hi, Vishal! You can find the answer to your question in the function's documentation[1]: If the result gives many rows, then multiple AVPs with corresponding name will be added. Consider the following query: avp_db_query("select group where username='$tu'", "$avp(group)"); The first line of the result can be accessed using $avp(group) (or $(avp(group)[0])), the second line $(avp(group)[1]), and so on. [1] http://www.opensips.org/html/docs/modules/2.4.x/avpops.html#idp3469424 Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/24/2017 08:56 PM, Vishal Pai wrote: > Hi Everyone > > I just want to know what is correct to fetch all the row from database > using avp_db_query. Currently I am getting only first row in result. > > One more think I want to know is it possible to get array of for > variable in Opensips configuration file. > > > Thanks > > Vishal Pai > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From vishalmpai at gmail.com Mon Nov 27 04:01:35 2017 From: vishalmpai at gmail.com (Vishal Pai) Date: Mon, 27 Nov 2017 14:31:35 +0530 Subject: [OpenSIPS-Users] Regarding avp_db_query In-Reply-To: <0e449337-c52e-371a-b18e-468c8272b13e@opensips.org> References: <0e449337-c52e-371a-b18e-468c8272b13e@opensips.org> Message-ID: Thanks i got that.... On Mon, Nov 27, 2017 at 2:03 PM, Răzvan Crainea wrote: > Hi, Vishal! > > You can find the answer to your question in the function's documentation[1]: > If the result gives many rows, then multiple AVPs with corresponding name > will be added. > > Consider the following query: > avp_db_query("select group where username='$tu'", "$avp(group)"); > > The first line of the result can be accessed using $avp(group) (or > $(avp(group)[0])), the second line $(avp(group)[1]), and so on. > > [1] http://www.opensips.org/html/docs/modules/2.4.x/avpops.html#idp3469424 > > Best regards, > > Răzvan Crainea > OpenSIPS Developerwww.opensips-solutions.com > > On 11/24/2017 08:56 PM, Vishal Pai wrote: > > Hi Everyone > > I just want to know what is correct to fetch all the row from database > using avp_db_query. Currently I am getting only first row in result. > > One more think I want to know is it possible to get array of for variable > in Opensips configuration file. > > > Thanks > > Vishal Pai > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From hunterj91 at hotmail.com Mon Nov 27 06:40:10 2017 From: hunterj91 at hotmail.com (Jonathan Hunter) Date: Mon, 27 Nov 2017 11:40:10 +0000 Subject: [OpenSIPS-Users] Pn-uri - Push Notification with SIP Message-ID: Hi Guys, We use opensips in the core of our network and are looking to implement Push Notification with the Session Initiation Protocol (SIP), as per this spec; https://tools.ietf.org/id/draft-holmberg-sipcore-sip-push-02.html Can anyone recommend a push server software that can be used in the scenario, that opensips can integrate with? I can see there are plenty of open source options out there, just wondered if anyone else had implemented? Thanks Jon -------------- next part -------------- An HTML attachment was scrubbed... URL: From goup2010 at gmail.com Mon Nov 27 06:46:47 2017 From: goup2010 at gmail.com (Dragomir Haralambiev) Date: Mon, 27 Nov 2017 13:46:47 +0200 Subject: [OpenSIPS-Users] get expires from REGISTER Message-ID: Hello, I try to get expires from REGISTER. xlog("Expires=$hdr(expires)\n"); I receive this: Expires= Here is REGISTER Session Initiation Protocol (SIP as raw text) REGISTER sip:sip-server.com:5060 SIP/2.0\r\n Accept: application/reginfo+xml, application/sdp, application/simple-message-summary, message/sipfrag, multipart/mixed, multipart/related\r\n Via: SIP/2.0/UDP 10.152.220.38:5062 ;branch=z9hG4bKde569c7a0f7678fe1;rport\r\n Max-Forwards: 70\r\n From: "55555555" ;tag=026b9b8a13\r\n To: "55555555" \r\n Call-ID: de52071eacc30295\r\n CSeq: 448743846 REGISTER\r\n Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, REFER, SUBSCRIBE, UPDATE\r\n Contact: ;audio;expires=120\r\n User-Agent: Media5-fone/4.25.4.13060 Android/8.0.0\r\n Content-Length: 0\r\n \r\n -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Mon Nov 27 07:14:34 2017 From: liviu at opensips.org (Liviu Chircu) Date: Mon, 27 Nov 2017 14:14:34 +0200 Subject: [OpenSIPS-Users] get expires from REGISTER In-Reply-To: References: Message-ID: <78aa335c-7212-79cb-dca3-980e7ff6ff76@opensips.org> Hi Dragomir, There is no "Expires" in REGISTER, so $hdr(expires) prints "". Try $(hdr(Contact){uri.param,expires}). Regards, Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 27.11.2017 13:46, Dragomir Haralambiev wrote: > Hello, > > I try to get expires from REGISTER. > xlog("Expires=$hdr(expires)\n"); > > I receive this: > Expires= > > Here is REGISTER > > Session Initiation Protocol (SIP as raw text) >     REGISTER sip:sip-server.com:5060 > SIP/2.0\r\n >     Accept: application/reginfo+xml, application/sdp, > application/simple-message-summary, message/sipfrag, multipart/mixed, > multipart/related\r\n >     Via: SIP/2.0/UDP > 10.152.220.38:5062;branch=z9hG4bKde569c7a0f7678fe1;rport\r\n >     Max-Forwards: 70\r\n >     From: "55555555" >;tag=026b9b8a13\r\n >     To: "55555555" >\r\n >     Call-ID: de52071eacc30295\r\n >     CSeq: 448743846 REGISTER\r\n >     Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, REFER, > SUBSCRIBE, UPDATE\r\n >     Contact: >;audio;expires=120\r\n >     User-Agent: Media5-fone/4.25.4.13060 Android/8.0.0\r\n >     Content-Length: 0\r\n >     \r\n > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From goup2010 at gmail.com Mon Nov 27 07:56:56 2017 From: goup2010 at gmail.com (Dragomir Haralambiev) Date: Mon, 27 Nov 2017 14:56:56 +0200 Subject: [OpenSIPS-Users] get expires from REGISTER In-Reply-To: <78aa335c-7212-79cb-dca3-980e7ff6ff76@opensips.org> References: <78aa335c-7212-79cb-dca3-980e7ff6ff76@opensips.org> Message-ID: I try this: xlog("Expires= $(hdr(Contact){uri.param,expires}) \n"); Here is ERRORS: Nov 27 14:53:07 /usr/local/sbin/opensips[28413]: ERROR:core:parse_uri: bad uri, state 0 parsed: < (4) / <;audio;expires=120> (57) Nov 27 14:53:07 /usr/local/sbin/opensips[28413]: ERROR:core:tr_eval_uri: invalid uri [;audio;expires=120] 2017-11-27 14:14 GMT+02:00 Liviu Chircu : > Hi Dragomir, > > There is no "Expires" in REGISTER, so $hdr(expires) prints "". Try > $(hdr(Contact){uri.param,expires}). > > Regards, > > Liviu Chircu > OpenSIPS Developerhttp://www.opensips-solutions.com > > On 27.11.2017 13:46, Dragomir Haralambiev wrote: > > Hello, > > I try to get expires from REGISTER. > xlog("Expires=$hdr(expires)\n"); > > I receive this: > Expires= > > Here is REGISTER > > Session Initiation Protocol (SIP as raw text) > REGISTER sip:sip-server.com:5060 SIP/2.0\r\n > Accept: application/reginfo+xml, application/sdp, > application/simple-message-summary, message/sipfrag, multipart/mixed, > multipart/related\r\n > Via: SIP/2.0/UDP 10.152.220.38:5062;branch=z9hG4bKde569c7a0f7678fe1; > rport\r\n > Max-Forwards: 70\r\n > From: "55555555" ;tag=026b9b8a13\r\n > To: "55555555" \r\n > Call-ID: de52071eacc30295\r\n > CSeq: 448743846 REGISTER\r\n > Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, REFER, > SUBSCRIBE, UPDATE\r\n > Contact: ;audio;expires=120\r\n > User-Agent: Media5-fone/4.25.4.13060 Android/8.0.0\r\n > Content-Length: 0\r\n > \r\n > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From callum.guy at x-on.co.uk Mon Nov 27 08:34:23 2017 From: callum.guy at x-on.co.uk (Callum Guy) Date: Mon, 27 Nov 2017 13:34:23 +0000 Subject: [OpenSIPS-Users] Debug Assistance (All the gateways are disabled) Message-ID: Hi All, This morning one of our OpenSIPs instances (2.1.4 (x86_64/linux)) had an issue that lasted around a minute, I'm trying to establish cause and wondered if anyone would be able to offer suggestions based on our log output. At the time of the issue the server was receiving around genuine 80 calls per second however some did not get timely responses (took ~2.5 seconds) which caused the requests to fail. The following log messages represent the start and end of the failure period: 2017-11-27 08:00:21 ERROR:drouting:route2_carrier: All the gateways are disabled 2017-11-27 08:00:21 ERROR:drouting:route2_carrier: All the gateways are disabled 2017-11-27 08:00:21 ERROR:drouting:route2_carrier: All the gateways are disabled 2017-11-27 08:00:21 CRITICAL:core:mk_proxy: could not resolve hostname: "internal.address" 2017-11-27 08:00:21 ERROR:tm:uri2proxy: bad host name in URI 2017-11-27 08:00:21 ERROR:tm:t_forward_nonack: failure to add branches 2017-11-27 08:00:21 ERROR:rtpproxy:force_rtp_proxy_body: incorrect port 0 in reply from rtp proxy 2017-11-27 08:00:21 CRITICAL:core:mk_proxy: could not resolve hostname: "internal.address" 2017-11-27 08:00:21 ERROR:tm:uri2proxy: bad host name in URI ... 2017-11-27 08:01:17 CRITICAL:core:mk_proxy: could not resolve hostname: "internal.address" 2017-11-27 08:01:17 ERROR:tm:uri2proxy: bad host name in URI 2017-11-27 08:01:17 ERROR:tm:t_forward_nonack: failure to add branches Where the logs state internal.address these are normal and correct domains which have been redacted. Is anyone able to offer any advice on how to interpret those messages? In particular the initial messages "All the gateways are disabled" seem like a good lead however I am unclear how to interpret these. Any advice would be greatly appreciated. Many thanks, Callum -- Callum Guy Head of Information Security X-on -- *0333 332 0000 | www.x-on.co.uk | ** * X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales. Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD. Company Registration No. 2578478. The information in this e-mail is confidential and for use by the addressee(s) only. If you are not the intended recipient, please notify X-on immediately on +44(0)333 332 0000 and delete the message from your computer. If you are not a named addressee you must not use, disclose, disseminate, distribute, copy, print or reply to this email. Views or opinions expressed by an individual within this email may not necessarily reflect the views of X-on or its associated companies. Although X-on routinely screens for viruses, addressees should scan this email and any attachments for viruses. X-on makes no representation or warranty as to the absence of viruses in this email or any attachments. -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Nov 27 10:37:43 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 27 Nov 2017 17:37:43 +0200 Subject: [OpenSIPS-Users] get expires from REGISTER In-Reply-To: References: <78aa335c-7212-79cb-dca3-980e7ff6ff76@opensips.org> Message-ID: <1262b846-42cd-326e-29b1-90f7c2f8258e@opensips.org> $hdr(Contact) return the whole Contact header which is a name-addr spec (according to the RFC3261 grammar). You should try: $(hdr(Contact){nameaddr.param,expires}) See : http://www.opensips.org/Documentation/Script-Tran-2-3#toc68 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/27/2017 02:56 PM, Dragomir Haralambiev wrote: > I try this: > xlog("Expires= $(hdr(Contact){uri.param,expires}) \n"); > > Here is ERRORS: > Nov 27 14:53:07 /usr/local/sbin/opensips[28413]: ERROR:core:parse_uri: > bad uri, state 0 parsed: < (4) / > < >;audio;expires=120> (57) > Nov 27 14:53:07 /usr/local/sbin/opensips[28413]: > ERROR:core:tr_eval_uri: invalid uri > [ >;audio;expires=120] > > > 2017-11-27 14:14 GMT+02:00 Liviu Chircu >: > > Hi Dragomir, > > There is no "Expires" in REGISTER, so $hdr(expires) prints > "". Try $(hdr(Contact){uri.param,expires}). > > Regards, > > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > > On 27.11.2017 13:46, Dragomir Haralambiev wrote: >> Hello, >> >> I try to get expires from REGISTER. >> xlog("Expires=$hdr(expires)\n"); >> >> I receive this: >> Expires= >> >> Here is REGISTER >> >> Session Initiation Protocol (SIP as raw text) >> REGISTER sip:sip-server.com:5060 >> SIP/2.0\r\n >> Accept: application/reginfo+xml, application/sdp, >> application/simple-message-summary, message/sipfrag, >> multipart/mixed, multipart/related\r\n >> Via: SIP/2.0/UDP >> 10.152.220.38:5062;branch=z9hG4bKde569c7a0f7678fe1;rport\r\n >> Max-Forwards: 70\r\n >> From: "55555555" > >;tag=026b9b8a13\r\n >> To: "55555555" > >\r\n >> Call-ID: de52071eacc30295\r\n >> CSeq: 448743846 REGISTER\r\n >> Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, >> REFER, SUBSCRIBE, UPDATE\r\n >> Contact: > >;audio;expires=120\r\n >> User-Agent: Media5-fone/4.25.4.13060 Android/8.0.0\r\n >> Content-Length: 0\r\n >> \r\n >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Nov 27 10:40:14 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 27 Nov 2017 17:40:14 +0200 Subject: [OpenSIPS-Users] Debug Assistance (All the gateways are disabled) In-Reply-To: References: Message-ID: <324279e3-7a95-8450-05fc-c105d966b3d3@opensips.org> Hi Callum, That message means that all the gateways provisioned under the selected carrier are marked as disabled. So, how do you disable the gws ? do you do dr_disable() from script ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/27/2017 03:34 PM, Callum Guy wrote: > Hi All, > > This morning one of our OpenSIPs instances (2.1.4 (x86_64/linux)) had > an issue that lasted around a minute, I'm trying to establish cause > and wondered if anyone would be able to offer suggestions based on our > log output. At the time of the issue the server was receiving around > genuine 80 calls per second however some did not get timely responses > (took ~2.5 seconds) which caused the requests to fail. > > The following log messages represent the start and end of the failure > period: > > 2017-11-27 08:00:21 ERROR:drouting:route2_carrier: All the gateways > are disabled > 2017-11-27 08:00:21 ERROR:drouting:route2_carrier: All the gateways > are disabled > 2017-11-27 08:00:21 ERROR:drouting:route2_carrier: All the gateways > are disabled > 2017-11-27 08:00:21 CRITICAL:core:mk_proxy: could not resolve > hostname: "internal.address" > 2017-11-27 08:00:21 ERROR:tm:uri2proxy: bad host name in URI > > 2017-11-27 08:00:21 ERROR:tm:t_forward_nonack: failure to add branches > 2017-11-27 08:00:21 ERROR:rtpproxy:force_rtp_proxy_body: incorrect > port 0 in reply from rtp proxy > 2017-11-27 08:00:21 CRITICAL:core:mk_proxy: could not resolve > hostname: "internal.address" > 2017-11-27 08:00:21 ERROR:tm:uri2proxy: bad host name in URI > > ... > 2017-11-27 08:01:17 CRITICAL:core:mk_proxy: could not resolve > hostname: "internal.address" > 2017-11-27 08:01:17 ERROR:tm:uri2proxy: bad host name in URI > > 2017-11-27 08:01:17 ERROR:tm:t_forward_nonack: failure to add branches > > Where the logs state internal.address these are normal and correct > domains which have been redacted. > > Is anyone able to offer any advice on how to interpret those messages? > In particular the initial messages "All the gateways are disabled" > seem like a good lead however I am unclear how to interpret these. > > Any advice would be greatly appreciated. > > Many thanks, > > Callum > -- > Callum Guy > Head of Information Security > X-on > > > *^0333 332 0000 | www.x-on.co.uk | > _**_^ > * > X-on is a trading name of Storacall Technology Ltd a limited company > registered in England and Wales. > Registered Office : Avaland House, 110 London Road, Apsley, Hemel > Hempstead, Herts, HP3 9SD. Company Registration No. 2578478. > The information in this e-mail is confidential and for use by the > addressee(s) only. If you are not the intended recipient, please > notify X-on immediately on +44(0)333 332 0000 and delete the > message from your computer. If you are not a named addressee you must > not use, disclose, disseminate, distribute, copy, print or reply to > this email. Views or opinions expressed by an individual > within this email may not necessarily reflect the views of X-on or its > associated companies. Although X-on routinely screens for viruses, > addressees should scan this email and any attachments > for viruses. X-on makes no representation or warranty as to the > absence of viruses in this email or any attachments. > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From k.galinurov at gmail.com Mon Nov 27 12:08:19 2017 From: k.galinurov at gmail.com (Kirill Galinurov) Date: Mon, 27 Nov 2017 20:08:19 +0300 Subject: [OpenSIPS-Users] opensips 2.3 and rtpengine module Message-ID: Hi all. I use latest stable opensips 2.3.2 and rtpengine module. In 2.4 devel branch rtpengine module i can use *in-iface=..., out-iface=. *parmeters. But what about rtpengine module in 2.3.2 ? -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Tue Nov 28 05:49:22 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Tue, 28 Nov 2017 12:49:22 +0200 Subject: [OpenSIPS-Users] opensips 2.3 and rtpengine module In-Reply-To: References: Message-ID: <10a317a8-c3a9-dfa9-3481-df20ec237670@opensips.org> Hi, Kirill! These parameters are not supported by 2.3.2, but you can still use the internal and external options. Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/27/2017 07:08 PM, Kirill Galinurov wrote: > Hi all. I use latest stable opensips 2.3.2 and rtpengine module. In > 2.4 devel branch rtpengine module i can use /in-iface=..., out-iface=. > /parmeters. But what about rtpengine  module in 2.3.2 ? > > > _______________________________________________ > 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 28 05:56:16 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Tue, 28 Nov 2017 12:56:16 +0200 Subject: [OpenSIPS-Users] Pn-uri - Push Notification with SIP In-Reply-To: References: Message-ID: <11258ca1-eaa3-e49c-261f-1656df9e6745@opensips.org> Hi, Jonathan! To be honest, I haven't fully implemented a push notification solution myself, but I've seen some deployments doing it. Basically, when a client registers, you need to store the PRID in a database, and when a call comes to that client, you need to do a REST query to the associated PNS provider, with that ID. The implementation of this service goes hand in hand with Event Based Routing[1] we've been talking about for a while - using this mechanism you can "delay" a call until the client re-registers, so you can forward the call to it. Hope this is helpful. If anyone has a working example that he has successfully deployed, I am also curious about his story :). [1] https://blog.opensips.org/2017/03/10/advanced-sip-scenarios-with-event-based-routing/ Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/27/2017 01:40 PM, Jonathan Hunter wrote: > Hi Guys, > > We use opensips in the core of our network and are looking to > implement Push Notification with the Session Initiation Protocol > (SIP), as per this spec; > > https://tools.ietf.org/id/draft-holmberg-sipcore-sip-push-02.html > > Can anyone recommend a push server software that can be used in the > scenario, that opensips can integrate with? > > I can see there are plenty of open source options out there, just > wondered if anyone else had implemented? > > Thanks > > Jon > > > > _______________________________________________ > 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 hunterj91 at hotmail.com Tue Nov 28 12:07:08 2017 From: hunterj91 at hotmail.com (Jonathan Hunter) Date: Tue, 28 Nov 2017 17:07:08 +0000 Subject: [OpenSIPS-Users] Pn-uri - Push Notification with SIP In-Reply-To: <11258ca1-eaa3-e49c-261f-1656df9e6745@opensips.org> References: , <11258ca1-eaa3-e49c-261f-1656df9e6745@opensips.org> Message-ID: Hi Răzvan, Cool thanks for the response, appreciate it! Its certainly an interesting scenario, will look at both the event based solution and push. Many thanks Jon ________________________________ From: Users on behalf of Răzvan Crainea Sent: 28 November 2017 10:56 To: users at lists.opensips.org Subject: Re: [OpenSIPS-Users] Pn-uri - Push Notification with SIP Hi, Jonathan! To be honest, I haven't fully implemented a push notification solution myself, but I've seen some deployments doing it. Basically, when a client registers, you need to store the PRID in a database, and when a call comes to that client, you need to do a REST query to the associated PNS provider, with that ID. The implementation of this service goes hand in hand with Event Based Routing[1] we've been talking about for a while - using this mechanism you can "delay" a call until the client re-registers, so you can forward the call to it. Hope this is helpful. If anyone has a working example that he has successfully deployed, I am also curious about his story :). [1] https://blog.opensips.org/2017/03/10/advanced-sip-scenarios-with-event-based-routing/ [https://blogopensips.files.wordpress.com/2017/03/ebr_model_wait.png] Advanced SIP scenarios with Event-based-Routing – Drops of ... blog.opensips.org There is an increasing need for more complex SIP scenarios, even for the Class 4 Switches. Such scenarios exceed the capabilities of a liner processing - something ... Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com Home — OpenSIPS Solutions www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 11/27/2017 01:40 PM, Jonathan Hunter wrote: Hi Guys, We use opensips in the core of our network and are looking to implement Push Notification with the Session Initiation Protocol (SIP), as per this spec; https://tools.ietf.org/id/draft-holmberg-sipcore-sip-push-02.html Push Notification with the Session Initiation Protocol (SIP) tools.ietf.org This document describes how push notification mechanisms can be used to wake up suspended Session Initiation Protocol (SIP) User Agents (UAs), in order to be able to receive and generate SIP requests. The document defines new SIP URI parameters, that can be used in a SIP REGISTER request to provide push notification information from the SIP User Agent (UA) to the SIP entity (realized as a SIP proxy in this document) that will send a push request to the push server in order to trigger a push notification towards the SIP UA. Can anyone recommend a push server software that can be used in the scenario, that opensips can integrate with? I can see there are plenty of open source options out there, just wondered if anyone else had implemented? Thanks Jon _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Tue Nov 28 12:42:00 2017 From: liviu at opensips.org (Liviu Chircu) Date: Tue, 28 Nov 2017 19:42:00 +0200 Subject: [OpenSIPS-Users] Pn-uri - Push Notification with SIP In-Reply-To: References: <11258ca1-eaa3-e49c-261f-1656df9e6745@opensips.org> Message-ID: <0460111c-233c-1d18-204d-fdd1b4ec7dcb@opensips.org> Nice to see another draft attempting to solve the same problem. Previously, I only knew about this one [1], though now it looks to be expired. I cannot help wondering: just how many different ones are out there? [1]: https://tools.ietf.org/html/draft-ivanov-sipcore-pnsip-01 Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 28.11.2017 19:07, Jonathan Hunter wrote: > > Hi Răzvan, > > > Cool thanks for the response, appreciate it! > > > Its certainly an interesting scenario, will look at both the event > based solution and push. > > > Many thanks > > > Jon > > > > ------------------------------------------------------------------------ > *From:* Users on behalf of Răzvan > Crainea > *Sent:* 28 November 2017 10:56 > *To:* users at lists.opensips.org > *Subject:* Re: [OpenSIPS-Users] Pn-uri - Push Notification with SIP > Hi, Jonathan! > > To be honest, I haven't fully implemented a push notification solution > myself, but I've seen some deployments doing it. Basically, when a > client registers, you need to store the PRID in a database, and when a > call comes to that client, you need to do a REST query to the > associated PNS provider, with that ID. > The implementation of this service goes hand in hand with Event Based > Routing[1] we've been talking about for a while - using this mechanism > you can "delay" a call until the client re-registers, so you can > forward the call to it. > Hope this is helpful. If anyone has a working example that he has > successfully deployed, I am also curious about his story :). > > [1] > https://blog.opensips.org/2017/03/10/advanced-sip-scenarios-with-event-based-routing/ > > > > Advanced SIP scenarios with Event-based-Routing – Drops of ... > > blog.opensips.org > There is an increasing need for more complex SIP scenarios, even for > the Class 4 Switches. Such scenarios exceed the capabilities of a > liner processing - something ... > > > > > Best regards, > Răzvan Crainea > OpenSIPS Developer > www.opensips-solutions.com > Home — OpenSIPS Solutions > www.opensips-solutions.com > OpenSIPS is a mature Open Source implementation of a SIP server. > OpenSIPS is more than a SIP proxy/router as it includes > application-level functionalities. > > > On 11/27/2017 01:40 PM, Jonathan Hunter wrote: >> Hi Guys, >> >> We use opensips in the core of our network and are looking to >> implement Push Notification with the Session Initiation Protocol >> (SIP), as per this spec; >> >> https://tools.ietf.org/id/draft-holmberg-sipcore-sip-push-02.html >> Push Notification with the Session Initiation Protocol (SIP) >> >> tools.ietf.org >> This document describes how push notification mechanisms can be used >> to wake up suspended Session Initiation Protocol (SIP) User Agents >> (UAs), in order to be able to receive and generate SIP requests. The >> document defines new SIP URI parameters, that can be used in a SIP >> REGISTER request to provide push notification information from the >> SIP User Agent (UA) to the SIP entity (realized as a SIP proxy in >> this document) that will send a push request to the push server in >> order to trigger a push notification towards the SIP UA. >> >> >> >> Can anyone recommend a push server software that can be used in the >> scenario, that opensips can integrate with? >> >> I can see there are plenty of open source options out there, just >> wondered if anyone else had implemented? >> >> Thanks >> >> Jon >> >> >> >> _______________________________________________ >> 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 pat at voxtelesys.com Tue Nov 28 13:10:56 2017 From: pat at voxtelesys.com (Pat Burke) Date: Tue, 28 Nov 2017 12:10:56 -0600 Subject: [OpenSIPS-Users] Pre-load mongodb cache Message-ID: <72779441d693f311451c236c6ff9a2a3@voxtelesys.com> Hello, We are looking at using the cachedb_mongodb module to access a large cache of data (100+ million records) that we wanted to do the following - on a separate server, create a "pre-loaded" cache mongodb file, and then copy the file to each opensips server so that the data is accessed locally - access the mongodb "cache" using the cache_raw_query on the calls as needed - on a nightly basis, replace the build an updated (on a separate server) and replace the cache mongodb file on the local server Can I just replace the file while they system is running?  Do I have to a replicaSet to have auto failover?  Any recommendation? Regards, Pat Burke ______________________________________________________________________________________ Direct: (402) 403-5121   |   Cell: (402) 443-8929  |   Email: pat at voxtelesys.com 1801 23rd Avenue North   |  Suite 217    |  Fargo, North Dakota 58102   -------------- next part -------------- An HTML attachment was scrubbed... URL: From mas_66 at tiscali.co.uk Wed Nov 29 06:55:21 2017 From: mas_66 at tiscali.co.uk (mas_66 at tiscali.co.uk) Date: Wed, 29 Nov 2017 11:55:21 +0000 (GMT) Subject: [OpenSIPS-Users] Opensips 2.3.2 In-Reply-To: <7a94e5fc-bc4c-e322-9a0a-915cfe3dd8e8@opensips.org> References: <26890667.2767301511272013887.JavaMail.defaultUser@defaultHost> <7a94e5fc-bc4c-e322-9a0a-915cfe3dd8e8@opensips.org> Message-ID: <3847644.3127601511956521356.JavaMail.defaultUser@defaultHost> Hi Bogdan I tried your suggestion but unfortunately i am still presenting the incorrect domain when i try to dial 02088253053. So i am still trying to dial out From: "6691" instead of "6691". below is the udp stream if it helps. Thanks Mas U 172.23.100.250:5060 -> 172.23.100.55:5060 OPTIONS sip:172.23.100.55:5060 SIP/2.0. Via: SIP/2.0/UDP 172.23.100.250:5060;branch=z9hG4bK00Bc649aa94943d13d3. From: ;tag=gK007fa50f. To: . Call-ID: 35425181_134052342 at 172.23.100.250. CSeq: 500409 OPTIONS. Max-Forwards: 1. Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,UPDATE, OPTIONS,MESSAGE,PUBLISH. Accept: application/sdp, application/isup, application/dtmf, application/dtmf- relay, multipart/mixed. Contact: . Content-Length: 0. . U 172.23.100.55:5060 -> 172.23.100.250:5060 SIP/2.0 484 Address Incomplete. Via: SIP/2.0/UDP 172.23.100.250:5060;branch=z9hG4bK00Bc649aa94943d13d3. From: ;tag=gK007fa50f. To: ;tag=5588ad0e2fdda81c4b33ac6770683a01.1b10. Call-ID: 35425181_134052342 at 172.23.100.250. CSeq: 500409 OPTIONS. Server: OpenSIPS (2.3.2 (x86_64/linux)). Content-Length: 0. . U 172.23.100.55:5060 -> 172.23.100.250:5060 INVITE sip:02088253053 at 172.23.100.250 SIP/2.0. Record-Route: . Record-Route: . Via: SIP/2.0/UDP 172.23.100.55:5060;branch=z9hG4bK2f3e.20349925.0. Via: SIP/2.0/UDP 192.168.200.162:56511;received=55.59.210.100;branch=z9hG4bK- 524287-1---3b2989732e03e469;rport=12328. Max-Forwards: 69. Contact: . To: . From: "6691";tag=1a24473b. Call-ID: 88254ZjZhZjYwYmI1N2MyYzNjODkzYTNmYzRhMzg2MWFiYTI. CSeq: 1 INVITE. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite release 5.0.3 stamp 88254. Content-Length: 340. P-hint: outbound. . v=0. o=- 13156428154656607 1 IN IP4 192.168.200.162. s=X-Lite release 5.0.3 stamp 88254. c=IN IP4 192.168.200.162. t=0 0. m=audio 59698 RTP/AVP 9 8 120 0 84 101. a=rtpmap:120 opus/48000/2. a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000. a=rtpmap:84 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 172.23.100.250:5060 -> 172.23.100.55:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 172.23.100.55:5060;branch=z9hG4bK2f3e.20349925.0. Via: SIP/2.0/UDP 192.168.200.162:56511;received=55.59.210.100;branch=z9hG4bK- 524287-1---3b2989732e03e469;rport=12328. From: "6691";tag=1a24473b. To: ;tag=gK04ead159. Call-ID: 88254ZjZhZjYwYmI1N2MyYzNjODkzYTNmYzRhMzg2MWFiYTI. CSeq: 1 INVITE. Content-Length: 0. . U 172.23.100.250:5060 -> 172.23.100.55:5060 SIP/2.0 488 Not Acceptable Here. Via: SIP/2.0/UDP 172.23.100.55:5060;branch=z9hG4bK2f3e.20349925.0. Via: SIP/2.0/UDP 192.168.200.162:56511;received=55.59.210.100;branch=z9hG4bK- 524287-1---3b2989732e03e469;rport=12328. From: "6691";tag=1a24473b. To: ;tag=gK04ead159. Call-ID: 88254ZjZhZjYwYmI1N2MyYzNjODkzYTNmYzRhMzg2MWFiYTI. CSeq: 1 INVITE. Content-Length: 0. . U 172.23.100.55:5060 -> 172.23.100.250:5060 ACK sip:02088253053 at 172.23.100.250 SIP/2.0. Via: SIP/2.0/UDP 172.23.100.55:5060;branch=z9hG4bK2f3e.20349925.0. From: "6691";tag=1a24473b. Call-ID: 88254ZjZhZjYwYmI1N2MyYzNjODkzYTNmYzRhMzg2MWFiYTI. To: ;tag=gK04ead159. CSeq: 1 ACK. Max-Forwards: 70. User-Agent: OpenSIPS (2.3.2 (x86_64/linux)). Content-Length: 0. From f4fahmed at gmail.com Wed Nov 29 09:30:40 2017 From: f4fahmed at gmail.com (Farrukh Ahmed) Date: Wed, 29 Nov 2017 17:30:40 +0300 Subject: [OpenSIPS-Users] OpenSIPS 2.2.x - 477/TM Error - TCP/TLS Client Message-ID: Hello, I am using OpenSIPs version 2.2 and when we register we have no issue but when i try to make call i get error message 477/TM. I have searched about it but i could not got any solution to this problem. The user is behind TCP and Error Logs mentioned this. I am getting following error in my syslog. -------------- Syslog Error -------------- Nov 20 10:49:34 s93334 /usr/sbin/opensips[16318]: ERROR:core:tcp_connect_blocking: timeout 99211 ms elapsed from 100000 s Nov 20 10:49:34 s93334 /usr/sbin/opensips[16318]: ERRORroto_tls:tls_sync_connect: tcp_blocking_connect failed Nov 20 10:49:34 s93334 /usr/sbin/opensips[16318]: ERRORroto_tlsroto_tls_send: connect failed Nov 20 10:49:34 s93334 /usr/sbin/opensips[16318]: ERROR:tm:msg_send: send() to 192.168.1.6:56716 for proto tls/3 failed Nov 20 10:49:34 s93334 /usr/sbin/opensips[16318]: ERROR:tm:t_forward_nonack: sending request failed Nov 20 10:49:34 s93334 /usr/sbin/opensips[16318]: ERROR:acc:acc_db_request: database not loaded! Probably database url not defined! -------------------------------- OpenSIPS Configuration File -------------------------------- https://dpaste.de/FBAO​ Looking forward for your help. Thank you.​ -- *Farrukh Ahmed* -------------- next part -------------- An HTML attachment was scrubbed... URL: From brians at iptel.co Wed Nov 29 18:40:20 2017 From: brians at iptel.co (Brian :) Date: Wed, 29 Nov 2017 23:40:20 +0000 Subject: [OpenSIPS-Users] Debug Assistance (All the gateways are disabled) In-Reply-To: References: Message-ID: Dns resolution issue? Do you use IPs or hostnames for your drouting gateways? On Monday, November 27, 2017, Callum Guy wrote: > Hi All, > > This morning one of our OpenSIPs instances (2.1.4 (x86_64/linux)) had an > issue that lasted around a minute, I'm trying to establish cause and > wondered if anyone would be able to offer suggestions based on our log > output. At the time of the issue the server was receiving around genuine 80 > calls per second however some did not get timely responses (took ~2.5 > seconds) which caused the requests to fail. > > The following log messages represent the start and end of the failure > period: > > 2017-11-27 08:00:21 ERROR:drouting:route2_carrier: All the gateways are > disabled > 2017-11-27 08:00:21 ERROR:drouting:route2_carrier: All the gateways are > disabled > 2017-11-27 08:00:21 ERROR:drouting:route2_carrier: All the gateways are > disabled > 2017-11-27 08:00:21 CRITICAL:core:mk_proxy: could not resolve hostname: > "internal.address" > 2017-11-27 08:00:21 ERROR:tm:uri2proxy: bad host name in URI > > 2017-11-27 08:00:21 ERROR:tm:t_forward_nonack: failure to add branches > 2017-11-27 08:00:21 ERROR:rtpproxy:force_rtp_proxy_body: incorrect port > 0 in reply from rtp proxy > 2017-11-27 08:00:21 CRITICAL:core:mk_proxy: could not resolve hostname: > "internal.address" > 2017-11-27 08:00:21 ERROR:tm:uri2proxy: bad host name in URI > > ... > 2017-11-27 08:01:17 CRITICAL:core:mk_proxy: could not resolve hostname: > "internal.address" > 2017-11-27 08:01:17 ERROR:tm:uri2proxy: bad host name in URI > > 2017-11-27 08:01:17 ERROR:tm:t_forward_nonack: failure to add branches > > Where the logs state internal.address these are normal and correct domains > which have been redacted. > > Is anyone able to offer any advice on how to interpret those messages? In > particular the initial messages "All the gateways are disabled" seem like > a good lead however I am unclear how to interpret these. > > Any advice would be greatly appreciated. > > Many thanks, > > Callum > -- > Callum Guy > Head of Information Security > X-on > > > *0333 332 0000 | www.x-on.co.uk | ** > > * > X-on is a trading name of Storacall Technology Ltd a limited company > registered in England and Wales. > Registered Office : Avaland House, 110 London Road, Apsley, Hemel > Hempstead, Herts, HP3 9SD. Company Registration No. 2578478. > The information in this e-mail is confidential and for use by the > addressee(s) only. If you are not the intended recipient, please notify > X-on immediately on +44(0)333 332 0000 and delete the > message from your computer. If you are not a named addressee you must not > use, disclose, disseminate, distribute, copy, print or reply to this email. Views > or opinions expressed by an individual > within this email may not necessarily reflect the views of X-on or its > associated companies. Although X-on routinely screens for viruses, > addressees should scan this email and any attachments > for viruses. X-on makes no representation or warranty as to the absence of > viruses in this email or any attachments. > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From callum.guy at x-on.co.uk Thu Nov 30 04:23:41 2017 From: callum.guy at x-on.co.uk (Callum Guy) Date: Thu, 30 Nov 2017 09:23:41 +0000 Subject: [OpenSIPS-Users] Debug Assistance (All the gateways are disabled) In-Reply-To: References: Message-ID: Hi Bogdan/Brian, It was a false alarm I am afraid and the fault was external to OpenSIPs however the error message did throw me off the scent of the fault for a while! The configuration file used on the affected instance of OpenSIPs relies on an external web service to handle some routing scenarios. This component was running slow on the day in question which caused occupation of all OpenSIPs threads and a backlog of messages was created. We plan on designing the reliance on this service out of the system in our next release. Apologies for the delayed response - I missed your initial reply. Best Regards, Callum On Wed, Nov 29, 2017 at 11:42 PM Brian : wrote: > Dns resolution issue? Do you use IPs or hostnames for your drouting > gateways? > > On Monday, November 27, 2017, Callum Guy wrote: > >> Hi All, >> >> This morning one of our OpenSIPs instances (2.1.4 (x86_64/linux)) had an >> issue that lasted around a minute, I'm trying to establish cause and >> wondered if anyone would be able to offer suggestions based on our log >> output. At the time of the issue the server was receiving around genuine 80 >> calls per second however some did not get timely responses (took ~2.5 >> seconds) which caused the requests to fail. >> >> The following log messages represent the start and end of the failure >> period: >> >> 2017-11-27 08:00:21 ERROR:drouting:route2_carrier: All the gateways are >> disabled >> 2017-11-27 08:00:21 ERROR:drouting:route2_carrier: All the gateways are >> disabled >> 2017-11-27 08:00:21 ERROR:drouting:route2_carrier: All the gateways are >> disabled >> 2017-11-27 08:00:21 CRITICAL:core:mk_proxy: could not resolve hostname: >> "internal.address" >> 2017-11-27 08:00:21 ERROR:tm:uri2proxy: bad host name in URI >> >> 2017-11-27 08:00:21 ERROR:tm:t_forward_nonack: failure to add branches >> 2017-11-27 08:00:21 ERROR:rtpproxy:force_rtp_proxy_body: incorrect port >> 0 in reply from rtp proxy >> 2017-11-27 08:00:21 CRITICAL:core:mk_proxy: could not resolve hostname: >> "internal.address" >> 2017-11-27 08:00:21 ERROR:tm:uri2proxy: bad host name in URI >> >> ... >> 2017-11-27 08:01:17 CRITICAL:core:mk_proxy: could not resolve hostname: >> "internal.address" >> 2017-11-27 08:01:17 ERROR:tm:uri2proxy: bad host name in URI >> >> 2017-11-27 08:01:17 ERROR:tm:t_forward_nonack: failure to add branches >> >> Where the logs state internal.address these are normal and correct >> domains which have been redacted. >> >> Is anyone able to offer any advice on how to interpret those messages? In >> particular the initial messages "All the gateways are disabled" seem >> like a good lead however I am unclear how to interpret these. >> >> Any advice would be greatly appreciated. >> >> Many thanks, >> >> Callum >> -- >> Callum Guy >> Head of Information Security >> X-on >> >> >> *0333 332 0000 | www.x-on.co.uk | ** >> >> * >> X-on is a trading name of Storacall Technology Ltd a limited company >> registered in England and Wales. >> Registered Office : Avaland House, 110 London Road, Apsley, Hemel >> Hempstead, Herts, HP3 9SD. Company Registration No. 2578478. >> The information in this e-mail is confidential and for use by the >> addressee(s) only. If you are not the intended recipient, please notify >> X-on immediately on +44(0)333 332 0000 <+44%20333%20332%200000> and >> delete the >> message from your computer. If you are not a named addressee you must not >> use, disclose, disseminate, distribute, copy, print or reply to this email. Views >> or opinions expressed by an individual >> within this email may not necessarily reflect the views of X-on or its >> associated companies. Although X-on routinely screens for viruses, >> addressees should scan this email and any attachments >> for viruses. X-on makes no representation or warranty as to the absence >> of viruses in this email or any attachments. >> >> _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Callum Guy Head of Information Security X-on -- *0333 332 0000 | www.x-on.co.uk | ** * X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales. Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD. Company Registration No. 2578478. The information in this e-mail is confidential and for use by the addressee(s) only. If you are not the intended recipient, please notify X-on immediately on +44(0)333 332 0000 and delete the message from your computer. If you are not a named addressee you must not use, disclose, disseminate, distribute, copy, print or reply to this email. Views or opinions expressed by an individual within this email may not necessarily reflect the views of X-on or its associated companies. Although X-on routinely screens for viruses, addressees should scan this email and any attachments for viruses. X-on makes no representation or warranty as to the absence of viruses in this email or any attachments. -------------- next part -------------- An HTML attachment was scrubbed... URL: From alain.bieuzent at free.fr Thu Nov 30 10:08:34 2017 From: alain.bieuzent at free.fr (Alain Bieuzent) Date: Thu, 30 Nov 2017 16:08:34 +0100 Subject: [OpenSIPS-Users] Voice Transcoding in OpenSIPS using external hardware Message-ID: <34664F2B-2DF2-44B8-8FCB-886A5600077B@free.fr> Hi All, Is there a way to use external hardware transcoding ? I find the documentation about integration of Sangoma Cards, but is there other implementation already done? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From tali.gillon at vonage.com Tue Nov 28 04:43:44 2017 From: tali.gillon at vonage.com (Gillon, Tali) Date: Tue, 28 Nov 2017 11:43:44 +0200 Subject: [OpenSIPS-Users] avp variables Message-ID: Hi, Is there a limitation for saving dynamic variables? I'm asking this since I would like to collect data about the errors we send. below is an example: send_reply("503", "Service Unavailable 0"); $avp(errorvar) = "ERR_503__Service_Unavailable_0"; update_stat("$avp(errorvar)", "+1"); I'm afraid that we are going to have performance issues if we have too many dynamic variables. Is there a way to delete variables from statistics and not just reset them? Thanks, Tali -------------- next part -------------- An HTML attachment was scrubbed... URL: From vudn5 at fpt.com.vn Tue Nov 28 14:41:36 2017 From: vudn5 at fpt.com.vn (vudn5) Date: Tue, 28 Nov 2017 12:41:36 -0700 (MST) Subject: [OpenSIPS-Users] acc extra_fields In-Reply-To: References: <1496757382356-7607614.post@n2.nabble.com> <1496772246640-7607626.post@n2.nabble.com> <1496839662830-7607641.post@n2.nabble.com> <1496841494547-7607644.post@n2.nabble.com> <1496918544521-7607662.post@n2.nabble.com> Message-ID: <1511898096329-0.post@n2.nabble.com> i have tini trouble with opensips 2.3.2 . In old version 2.2.2 i have a config modparam("acc", "db_extra", "from_uri=$fU; to_uri=$tU; src_uri=$si") to save $fU ,$tU , $si variable to columns from_uri , to_uri , src_uri in table acc ( i create 3 fields) but i don't now how can i do that in opensips 2.3.2? i read a EXTRA_FIELD and it's my config: modparam("acc", "extra_fields", "db: $fU->from_uri; $tU-> to_uri; $si->src_uri ") but it's not work, and i want to khow why it's not work? Thanks -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html