Hi, You don't give the logs and sip traces - its no fun for you or anyone here. I recommend you to change your FreeSwitch/Asterisk settings to stop re-invites (canreinvite=no) and in newer version (directmedia=no) - See if that helps.
I'll again repeat that either you don't engage RTPproxy in the call at all, or you send the Media via the RTPproxy by setting your A-leg & B-leg telephony engines. that'll solve your problem I hope you can handle it from here now. Regards, Sammy On Wed, Jul 4, 2012 at 10:15 PM, Rodrigo Ferreira < [email protected]> wrote: > I’m trying to debug those calls using wireshark and VoIP calls analisys > ... > > And i guess it might be a network problem, maybe some packets being > dropped .. Is there any chance of being that? > Because my Opensips box and my FreePBX (asterisk) they are in the same > network, just passing through a switch, among with other servers (including > another opensips who is giving me the same problem) > > *From:* Rodrigo Ferreira <[email protected]> > *Sent:* Wednesday, July 04, 2012 9:10 AM > *To:* OpenSIPS users mailling list <[email protected]> > *Subject:* Re: [OpenSIPS-Users] Calls being disconnected > > That was my bad, its different IPs, but I wrote it wrong. > > I will get some new traces today, and I will write the IPs correctly > > *From:* SamyGo <[email protected]> > *Sent:* Wednesday, July 04, 2012 1:54 AM > *To:* OpenSIPS users mailling list <[email protected]> > *Subject:* Re: [OpenSIPS-Users] Calls being disconnected > > I will definitely try to read the SIP traces. I love trying to solve the > puzzle but the very first line !! > > U 201.71.XXX.XXX:5060 -> 201.71.XXX.XXX:5060 > > Is it the same server originating SIP INVITE sending to the same server !! > if not please differentiate the two addresses like. > > U 201.71.XXX.XXX:5060 -> 201.71.XXX.YYY:5060 > > Goodness , its FBX,OpenSIPS, gatewaySTS, and even Asterisk 1.8 -- are they > all using the same IP !! > > Caller(FPBX)<--->OpenSIPS+RTPproxy<---->gatewaySTS<---->Asterisk > ||<============================RTP===============>|| > > Please carefully use different IP addresses of all the SIP servers/UACs in > your particular call scenario. What I'm still suspecting with more > intensity is that somehow the RTPproxy is called but Asterisk and FPBX > negotiate media directly in SDPs and bam..RTPproxy gets angry at this > insulting attitude and disconnects the call !! That's my guess. > > OpenSIPS logs along with the new SIP traces will help clarify the real > cause. > > Thanks > Sammy > > On Tue, Jul 3, 2012 at 7:23 PM, Rodrigo Ferreira < > [email protected]> wrote: > >> I’m attaching the log that I got from a call that was disconnected >> >> *From:* Rodrigo Ferreira <[email protected]> >> *Sent:* Tuesday, July 03, 2012 11:09 AM >> *To:* OpenSIPS users mailling list <[email protected]> >> *Subject:* Re: [OpenSIPS-Users] Calls being disconnected >> >> I’m getting the logs, but let me ask you something, force_rport will >> help me? >> >> *From:* SamyGo <[email protected]> >> *Sent:* Tuesday, July 03, 2012 8:18 AM >> *To:* OpenSIPS users mailling list <[email protected]> >> *Subject:* Re: [OpenSIPS-Users] Calls being disconnected >> >> Specifically talking about one where call gets dropped - what is the >> network scenario !? Make one such call and take a sip trace on the >> opensips server using tcpdump or sipgrep. Share the pcap. As well as the >> opensips logs please. >> >> >> On Tue, Jul 3, 2012 at 4:12 PM, Rodrigo Ferreira < >> [email protected]> wrote: >> >>> I’ve a lot of different scenarios through my network, I have clients >>> behind NAT, from other networks, because I work for a telephony company. >>> >>> *From:* SamyGo <[email protected]> >>> *Sent:* Tuesday, July 03, 2012 8:02 AM >>> *To:* OpenSIPS users mailling list <[email protected]> >>> *Subject:* Re: [OpenSIPS-Users] Calls being disconnected >>> >>> RTPproxy - thats what I suspected. What i'm guessing is that RTPproxy >>> is engaged but somehow the end points receive each other's Direct IPs in >>> SDP...hmm...tell me if your both end points are on the same LAN and server >>> is placed somewhere in internet !! >>> >>> On Tue, Jul 3, 2012 at 3:58 PM, Rodrigo Ferreira < >>> [email protected]> wrote: >>> >>>> I will take a look at my rtp .. I’m using rtpproxy on my box, I will >>>> try to get some logs today too >>>> >>>> *From:* SamyGo <[email protected]> >>>> *Sent:* Tuesday, July 03, 2012 4:34 AM >>>> *To:* aamir chougule <[email protected]> ; OpenSIPS users mailling >>>> list <[email protected]> >>>> *Subject:* Re: [OpenSIPS-Users] Calls being disconnected >>>> >>>> Besides logs and any sip traces as Aamir stated it'd be hard to pin >>>> point anything. One of the big reason for automatically dropping an >>>> established call specially after a specific amount of time i.e 30 >>>> seconds..thats because of lack of RTPs flowing through the server and if >>>> media-relaying tools are used they will send hangup to both ends, even >>>> though both ends have their RTP flowing *directly*. >>>> >>>> Regards, >>>> Sammy >>>> >>>> On Tue, Jul 3, 2012 at 12:21 PM, aamir chougule <[email protected]>wrote: >>>> >>>>> Hi Rodrigo, >>>>> >>>>> The only which I know is after the call gets connected i.e. 200OK from >>>>> the other end there should always be an ACK for that 200OK, if there is no >>>>> ACK for the 200OK then the call will get disconnected after sometime. >>>>> >>>>> Can you please give us the logs for the call by tracing it through >>>>> ngrep tool because without any sip traces or opensips log we can't tell >>>>> what is going on within the box. >>>>> >>>>> Regards, >>>>> >>>>> Aamir Chougule >>>>> Cell: 09167989111 >>>>> >>>>> ------------------------------ >>>>> *From:* Rodrigo Ferreira <[email protected]> >>>>> *To:* OpenSIPS users mailling list <[email protected]> >>>>> *Sent:* Monday, 2 July 2012 10:40 PM >>>>> *Subject:* [OpenSIPS-Users] Calls being disconnected >>>>> >>>>> Hey guys, >>>>> >>>>> I’m having problems with calls being “disconnected” and I dont know >>>>> where I can start to look at, because on my CDR all those calls ends with >>>>> a >>>>> 200OK, but that isnt true, because you are talking all the suddenly the >>>>> call stop. >>>>> >>>>> Any ideas where I should start looking at? >>>>> >>>>> Engº Rodrigo Ferreira >>>>> Supervisor de Telefonia >>>>> VIPWay Telecom >>>>> Tel.: +55 13 4010-1000 >>>>> Cel.: +55 13 8136-5839 >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> [email protected] >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> [email protected] >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> >>>> ------------------------------ >>>> _______________________________________________ >>>> Users mailing list >>>> [email protected] >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> ------------------------------ >>>> >>>> Nenhum vírus encontrado nessa mensagem. >>>> Verificado por AVG - www.avgbrasil.com.br >>>> Versão: 2012.0.2179 / Banco de dados de vírus: 2437/5106 - Data de >>>> Lançamento: 07/02/12 >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> [email protected] >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>> ------------------------------ >>> _______________________________________________ >>> Users mailing list >>> [email protected] >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> ------------------------------ >>> >>> Nenhum vírus encontrado nessa mensagem. >>> Verificado por AVG - www.avgbrasil.com.br >>> Versão: 2012.0.2179 / Banco de dados de vírus: 2437/5106 - Data de >>> Lançamento: 07/02/12 >>> >>> _______________________________________________ >>> Users mailing list >>> [email protected] >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >> ------------------------------ >> _______________________________________________ >> Users mailing list >> [email protected] >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> ------------------------------ >> >> Nenhum vírus encontrado nessa mensagem. >> Verificado por AVG - www.avgbrasil.com.br >> Versão: 2012.0.2179 / Banco de dados de vírus: 2437/5108 - Data de >> Lançamento: 07/03/12 >> ------------------------------ >> _______________________________________________ >> Users mailing list >> [email protected] >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> ------------------------------ >> >> Nenhum vírus encontrado nessa mensagem. >> Verificado por AVG - www.avgbrasil.com.br >> Versão: 2012.0.2179 / Banco de dados de vírus: 2437/5108 - Data de >> Lançamento: 07/03/12 >> >> _______________________________________________ >> Users mailing list >> [email protected] >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > ------------------------------ > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > ------------------------------ > > Nenhum vírus encontrado nessa mensagem. > Verificado por AVG - www.avgbrasil.com.br > Versão: 2012.0.2179 / Banco de dados de vírus: 2437/5108 - Data de > Lançamento: 07/03/12 > > ------------------------------ > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > ------------------------------ > > Nenhum vírus encontrado nessa mensagem. > Verificado por AVG - www.avgbrasil.com.br > Versão: 2012.0.2179 / Banco de dados de vírus: 2437/5110 - Data de > Lançamento: 07/04/12 > > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >
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