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 <bog...@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:13101234...@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:2...@172.16.100.159:21874 > > <http://sip:2...@172.16.100.159:21874>> > > To: "13101234567"<sip:13101234...@ask00-rvn> > > From: "20000"<sip:2...@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 <r...@dimension-virtual.com > > <mailto:r...@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 > > > > _______________________________________________ > > 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 > > > > > _______________________________________________ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >
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