when they only return 1, that means that is what they want you to use.<br>The call should establish in that case with g726-32<br><br><br><div class="gmail_quote">On Wed, Jan 6, 2010 at 6:57 PM, Brian West <span dir="ltr"><<a href="mailto:brian@freeswitch.org">brian@freeswitch.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">NO the codec map includes 2 aka g726-32 which you don't have to list in the map if its defined in the standard... what I need is a pcap of the whole process.<br>
<font color="#888888"><br>
/b<br>
</font><div><div></div><div class="h5"><br>
On Jan 6, 2010, at 6:43 PM, Mark Campbell-Smith wrote:<br>
<br>
> Thanks guys for your response. I'll have a read through the links sent to me.<br>
><br>
> On the INVITE my FS box sends to Phonzo (the SIP provider) the codecs<br>
> it supports are G726-32, PCMU, PCMA.<br>
><br>
> v=0<br>
> o=FreeSWITCH 1262742568 1262742569 IN IP4 xxx.xxx.xxx.xx<br>
> s=FreeSWITCH<br>
> c=IN IP4 xxx.xxx.xxx.xxx<br>
> t=0 0<br>
> m=audio 31566 RTP/AVP 2 0 8 101 13<br>
> a=rtpmap:2 G726-32/8000<br>
> a=rtpmap:0 PCMU/8000<br>
> a=rtpmap:8 PCMA/8000<br>
> a=rtpmap:101 telephone-event/8000<br>
> a=fmtp:101 0-16<br>
> a=rtpmap:13 CN/8000<br>
> a=ptime:20<br>
><br>
> However, Phonzo responds with the following in the Session Progress message:<br>
><br>
> v=0<br>
> o=Sippy 158698636 1 IN IP4 80.232.37.178<br>
> s=-<br>
> t=0 0<br>
> m=audio 61812 RTP/AVP 2 101 13<br>
> c=IN IP4 213.50.91.3<br>
> a=rtpmap:101 telephone-event/8000<br>
> a=fmtp:101 0-15<br>
><br>
> Does that make sense? The codec rtpmap only included DTMF ...<br>
<br>
<br>
</div></div><div><div></div><div class="h5">_______________________________________________<br>
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