Re: [OpenSIPS-Users] Dialog destroys when answering call!!
Hello Bogdan, Here is the output from opensips log with debug =4. May 12 03:50:39 devserver /usr/local/sbin/opensips[11590]: WARNING:dialog:dlg_onroute: unable to find dialog for ACK with route param '30f.e3e8b527' -- Neo From: Bogdan-Andrei Iancu To: OpenSIPS users mailling list Sent: Mon, May 10, 2010 10:00:59 PM Subject: Re: [OpenSIPS-Users] Dialog destroys when answering call!! Could you set debug = 4 in cfg and run a single call through your opensips and send me the output ? Regards, Bogdan Neo Anderson wrote: > Hello Bogdan, > > Yes when opensips gets 200 OK & then send ACK back to the Callee, that > time dialog destroys. > When making call, I am executing opensipsctl fifo dlg_list . > Till receiving 200 OK I can see the dialog but when sending ACK dialog > destroys. Also I found Status 5 in dialog list. > Please let me know if anything I did wrong in configurations. > > dialog module configurations: > modparam("dialog", "dlg_flag", 10) > modparam("dialog", "dlg_match_mode", 1) > modparam("dialog", "profiles_with_value", "caller") > modparam("dialog", "default_timeout", > 43200) > modparam("dialog", "timeout_avp", "$avp(i:100)") > > Thanks. > > -- > Neo > > > > > > *From:* Bogdan-Andrei Iancu > *To:* OpenSIPS users mailling list > *Sent:* Mon, May 10, 2010 6:20:55 PM > *Subject:* Re: [OpenSIPS-Users] Dialog destroys when answering call!! > > Hi Neo, > > you are saying the dialogs (in dialog module) are destroyed when they > are answered (200 ok ) ? what makes you say that? I mean what do you see > to confirm this? > > Regards, > Bogdan > > Neo Anderson wrote: > > Hi, > > > > I am using OpenSIPS 1.5.3 . > > I have implemented call-limit based on the tutorial. > > > > http://www.opensips.org/Resources/DocsTutConcurrentCalls > > <http://www.opensips.org/Resources/DocsTutConcurrentCalls> > > > > But when call gets answered, dialog destroys. That's why call limit is > > not working. > > Would you please let me know what I am doing wrong? > > I have followed the same instructions given in the tutorial. > > I am using carrier-route module to route outbound calls. > > > > Thanks in advance!!! > > > > -- > > Neo > > > > > > > > ___ > > Users mailing list > > Users@lists.opensips.org <mailto:Users@lists.opensips.org> > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > -- > Bogdan-Andrei Iancu > www.voice-system.ro<http://www.voice-system.ro> > > > ___ > Users mailing list > Users@lists.opensips.org <mailto:Users@lists.opensips.org> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Dialog destroys when answering call!!
Hello Bogdan, Yes when opensips gets 200 OK & then send ACK back to the Callee, that time dialog destroys. When making call, I am executing opensipsctl fifo dlg_list . Till receiving 200 OK I can see the dialog but when sending ACK dialog destroys. Also I found Status 5 in dialog list. Please let me know if anything I did wrong in configurations. dialog module configurations: modparam("dialog", "dlg_flag", 10) modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "profiles_with_value", "caller") modparam("dialog", "default_timeout", 43200) modparam("dialog", "timeout_avp", "$avp(i:100)") Thanks. -- Neo From: Bogdan-Andrei Iancu To: OpenSIPS users mailling list Sent: Mon, May 10, 2010 6:20:55 PM Subject: Re: [OpenSIPS-Users] Dialog destroys when answering call!! Hi Neo, you are saying the dialogs (in dialog module) are destroyed when they are answered (200 ok ) ? what makes you say that? I mean what do you see to confirm this? Regards, Bogdan Neo Anderson wrote: > Hi, > > I am using OpenSIPS 1.5.3 . > I have implemented call-limit based on the tutorial. > > http://www.opensips.org/Resources/DocsTutConcurrentCalls > <http://www.opensips.org/Resources/DocsTutConcurrentCalls> > > But when call gets answered, dialog destroys. That's why call limit is > not working. > Would you please let me know what I am doing wrong? > I have followed the same instructions given in the tutorial. > I am using carrier-route module to route outbound calls. > > Thanks in advance!!! > > -- > Neo > > > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Dialog destroys when answering call!!
Hi, I am using OpenSIPS 1.5.3 . I have implemented call-limit based on the tutorial. http://www.opensips.org/Resources/DocsTutConcurrentCalls But when call gets answered, dialog destroys. That's why call limit is not working. Would you please let me know what I am doing wrong? I have followed the same instructions given in the tutorial. I am using carrier-route module to route outbound calls. Thanks in advance!!! -- Neo ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] opensips performance
Hi, 1. CPU : Quad Core 64 bit CPU 2. 8 GB RAM 3. Centos 5.4 64 bit OS 4. OpenSIPS will have db lookup with radius integration for accounting. Please let me know how many simultaneous calls OpenSIPS can process for above mentioned situations? -- VoipExpert ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenSIPS Crashed!!
Hi, Right now I am using OpenSIPS 1.5.3 no-tls version in production. Suddenly OpenSIPS got crashed. I did check coredump but can not understand it. Please help me out to interpret it. Here is the pastebin link which contains output of bt. http://pastebin.com/m49520853 Thanks in advance!!! -- VoipExpert ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Call-control stops to work
Hi, I am using OpenSIPS 1.5.3 no-tls with call-control support for prepaid billing. There can be many parallel sessions using one balance per subscriber. But sometimes call-control gets stuck. Getting no credit found evenif there's enough credit to make call. I did increase socket timeout in callcontrol config file. That has decreased the frequency of failing. Another thing is, it's a cluster setup. There are 2 opensips servers , handles 50%-50% calls. Both servers sends request to one CDRTool server to fetch maxsessiontime & debit balance. But still sometimes facing issue with call-control. Can you please let me know how to resolve this? -- VoipExpert ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] CRITICAL:tm:t_should_relay_response: pick_branch failed (lowest==-1) for code 487
Hi I am using OpenSIPS 1.5.3 notls version. I have integrated opensips with sems to play early announcements from SEMS for failed calls. For example when opensips gets 404 error code...it sends command to SEMS to play early message for 404 error code. now When SEMS is playing message & in between listening the message, caller hangsup the call. I am getting CRITICAL:tm:t_should_relay_response: pick_branch failed (lowest==-1) for code 487 , in opensips logs. Basically Caller sends CANCEL to opensips, & opensips can not process the CANCEL. Would you please let me know how to resolve this? Thanks in advance!!! -- voipexpert ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users