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 Sent: Wednesday, July 04, 2012 1:54 AM To: OpenSIPS users mailling list 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 Sent: Tuesday, July 03, 2012 11:09 AM To: OpenSIPS users mailling list 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 Sent: Tuesday, July 03, 2012 8:18 AM To: OpenSIPS users mailling list 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 Sent: Tuesday, July 03, 2012 8:02 AM To: OpenSIPS users mailling list 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 Sent: Tuesday, July 03, 2012 4:34 AM To: aamir chougule ; OpenSIPS users mailling list 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. 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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
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