Here is the INVITE:<br><br>INVITE sip:13101234567@ask00-rvn SIP/2.0<br>Record-Route: <sip:10.1.3.130;lr;ftag=c020195b;did=d08.3a8259b2><br>Via: SIP/2.0/UDP 10.1.3.130;branch=z9hG4bK88c5.4ae45bf5.0<br>Via: SIP/2.0/UDP 172.16.100.159:21874;received=172.16.100.159;branch=z9hG4bK-d87543-2376e4785757b07b-1--d87543-;rport=21874<br>
Max-Forwards: 69<br>Contact: <<a href="http://sip:2000@172.16.100.159:21874">sip:2000@172.16.100.159:21874</a>><br>To: "13101234567"<sip:13101234567@ask00-rvn><br>From: "20000"<sip:2000@ask00-rvn>;tag=c020195b<br>
Call-ID: NDg4Y2Y0ZWU5MGM4NjhiNWVlZGNiZTc1ZGQxMjlhYzc.<br>CSeq: 1 INVITE<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO<br>Content-Type: application/sdp<br>User-Agent: X-Lite release 1011s stamp 41150<br>
Content-Length: 529<br><br>v=0<br>o=- 8 2 IN IP4 172.16.100.159<br>s=CounterPath X-Lite 3.0<br>c=IN IP4 172.16.100.159<br>t=0 0<br>m=audio 39148 RTP/AVP 107 119 100 106 0 105 98 8 101<br>a=alt:1 3 : 5AkMoAfO yRnFlRIn 172.16.100.159 39148<br>
a=alt:2 2 : 7PbWVKqn VccqHBD1 192.168.2.59 39148<br>a=alt:3 1 : TXSbExav /8BXXCL+ 192.168.176.152 39148<br>a=fmtp:101 0-15<br>a=rtpmap:107 BV32/16000<br>a=rtpmap:119 BV32-FEC/16000<br>a=rtpmap:100 SPEEX/16000<br>a=rtpmap:106 SPEEX-FEC/16000<br>
a=rtpmap:105 SPEEX-FEC/8000<br>a=rtpmap:98 iLBC/8000<br>a=rtpmap:101 telephone-event/8000<br>a=sendrecv<br><br><br><div class="gmail_quote">2009/10/21 Raúl Alexis Betancor Santana <span dir="ltr"><<a href="mailto:rabs@dimension-virtual.com">rabs@dimension-virtual.com</a>></span><br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div><div></div><div class="h5">On Wednesday 21 October 2009 23:13:36 Justin L wrote:<br>
> Hi,<br>
><br>
> I have a question related to my load balancing configuration of opensips.<br>
><br>
> I have an X-Lite softphone that connects to Opensips server, which<br>
> transfers the INVITE request to one of the asterisk boxes.<br>
> All of them are behind firewall on the same network. Then asterisk calls to<br>
> my cell phone through the voip provider.<br>
><br>
> The SIP balancing works fine and I get the call, but there is no audio. The<br>
> firewall should be configured correctly to transfer the SIP and RTP ports.<br>
><br>
> Since I just started to use opensips it sounds to me like a very basic<br>
> problem, that many people probably have faced.<br>
> Could you please recommend me a way to troubleshoot this issue?<br>
><br>
> Thanks a lot,<br>
><br>
> Justin.<br>
<br>
</div></div>Some SIP trace would be nice to begin ...<br>
<font color="#888888"><br>
--<br>
Raúl Alexis Betancor Santana<br>
Dimensión Virtual<br>
<br>
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