[OpenSIPS-Users] Audio problem

[Justin L]

Thanks for reply.

Actually I already solved the problem. It was an asterisk configuration
issue.
Thanks anyway.

Justin.

On Wed, Oct 21, 2009 at 9:52 PM, Bogdan-Andrei Iancu <bogdan at voice-system.ro
> wrote:

> Hi Justin,
>
> a trace means all SIP messages from that call (not only the INVITE) :).
>
> Also, "audio problem" means there is not audio at all or means you have
> one way audio ?
>
> Regards,
> Bogdan
>
> Justin L wrote:
> > Here is the INVITE:
> >
> > INVITE sip:13101234567 at ask00-rvn SIP/2.0
> > Record-Route: <sip:10.1.3.130;lr;ftag=c020195b;did=d08.3a8259b2>
> > Via: SIP/2.0/UDP 10.1.3.130;branch=z9hG4bK88c5.4ae45bf5.0
> > Via: SIP/2.0/UDP
> > 172.16.100.159:21874
> ;received=172.16.100.159;branch=z9hG4bK-d87543-2376e4785757b07b-1--d87543-;rport=21874
> > Max-Forwards: 69
> > Contact: <sip:2000 at 172.16.100.159:21874
> > <http://sip:2000@172.16.100.159:21874>>
> > To: "13101234567"<sip:13101234567 at ask00-rvn>
> > From: "20000"<sip:2000 at ask00-rvn>;tag=c020195b
> > Call-ID: NDg4Y2Y0ZWU5MGM4NjhiNWVlZGNiZTc1ZGQxMjlhYzc.
> > CSeq: 1 INVITE
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
> > SUBSCRIBE, INFO
> > Content-Type: application/sdp
> > User-Agent: X-Lite release 1011s stamp 41150
> > Content-Length: 529
> >
> > v=0
> > o=- 8 2 IN IP4 172.16.100.159
> > s=CounterPath X-Lite 3.0
> > c=IN IP4 172.16.100.159
> > t=0 0
> > m=audio 39148 RTP/AVP 107 119 100 106 0 105 98 8 101
> > a=alt:1 3 : 5AkMoAfO yRnFlRIn 172.16.100.159 39148
> > a=alt:2 2 : 7PbWVKqn VccqHBD1 192.168.2.59 39148
> > a=alt:3 1 : TXSbExav /8BXXCL+ 192.168.176.152 39148
> > a=fmtp:101 0-15
> > a=rtpmap:107 BV32/16000
> > a=rtpmap:119 BV32-FEC/16000
> > a=rtpmap:100 SPEEX/16000
> > a=rtpmap:106 SPEEX-FEC/16000
> > a=rtpmap:105 SPEEX-FEC/8000
> > a=rtpmap:98 iLBC/8000
> > a=rtpmap:101 telephone-event/8000
> > a=sendrecv
> >
> >
> > 2009/10/21 Raúl Alexis Betancor Santana <rabs at dimension-virtual.com
> > <mailto:rabs at dimension-virtual.com>>
> >
> >     On Wednesday 21 October 2009 23:13:36 Justin L wrote:
> >     > Hi,
> >     >
> >     > I have a question related to my load balancing configuration of
> >     opensips.
> >     >
> >     > I have an X-Lite softphone that connects to Opensips server, which
> >     > transfers the INVITE request to one of the asterisk boxes.
> >     > All of them are behind firewall on the same network. Then
> >     asterisk calls to
> >     > my cell phone through the voip provider.
> >     >
> >     > The SIP balancing works fine and I get the call, but there is no
> >     audio. The
> >     > firewall should be configured correctly to transfer the SIP and
> >     RTP ports.
> >     >
> >     > Since I just started to use opensips it sounds to me like a very
> >     basic
> >     > problem, that many people probably have faced.
> >     > Could you please recommend me a  way to troubleshoot this issue?
> >     >
> >     > Thanks a lot,
> >     >
> >     > Justin.
> >
> >     Some SIP trace would be nice to begin ...
> >
> >     --
> >     Raúl Alexis Betancor Santana
> >     Dimensión Virtual
> >
> >     _______________________________________________
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> >     Users at lists.opensips.org <mailto:Users at lists.opensips.org>
> >     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> >
> > ------------------------------------------------------------------------
> >
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> >
>
>
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