yah so the codec chosen by the inbound leg is always offered in the outbound sdp to try and prevent transcoding.<br>if you set {absolute_codec_string=G722} in the bridge string you will bypass this feature.<br><br><br><div class="gmail_quote">
On Wed, Dec 16, 2009 at 1:41 PM, Kristian Kielhofner <span dir="ltr"><<a href="mailto:kristian.kielhofner@gmail.com">kristian.kielhofner@gmail.com</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;">
Sure... The call comes up as PCMU:<br>
<br>
INVITE <a href="mailto:sip%3A5888@10.70.0.99">sip:5888@10.70.0.99</a> SIP/2.0<br>
Call-ID: 80ea31a017f6de1d53e4a9c52f00<br>
CSeq: 1 INVITE<br>
From: sip:9413122830@smh.sip.local;tag=80ea31a017f6de1d43e4a9c52f00<br>
Record-Route: <sip:10.70.0.65:5060;lr>,<sip:10.70.0.69;lr;transport=tcp><br>
To: "5888" <<a href="mailto:sip%3A5888@10.70.0.99">sip:5888@10.70.0.99</a>><br>
Via: SIP/2.0/UDP<br>
10.70.0.65:5060;branch=z9hG4bK838383030303565656105e9.0,SIP/2.0/TCP<br>
10.70.0.69;psrrposn=2;received=10.70.0.69;branch=z9hG4bK80ea31a017f6de1d63e4a9c52f00<br>
Content-Length: 206<br>
Content-Type: application/sdp<br>
Contact: <<a href="mailto:sip%3A9413122830@10.70.0.69">sip:9413122830@10.70.0.69</a>;transport=tcp><br>
Max-Forwards: 70<br>
User-Agent: Avaya CM/R015x.02.0.947.3<br>
Allow: INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO,PUBLISH<br>
Supported: timer,replaces,join,histinfo,100rel<br>
Alert-Info: <<a href="mailto:cid%3Aexternal@10.70.0.99">cid:external@10.70.0.99</a>>;avaya-cm-alert-type=external<br>
Min-SE: 1200<br>
Session-Expires: 1200;refresher=uac<br>
P-Asserted-Identity: sip:9413122830@smh.sip.local<br>
P-Charging-Vector: icid-value="AAS:13283-a031ea801def6179c4a3ed3f52"<br>
History-Info: <<a href="mailto:sip%3A5888@10.70.0.99">sip:5888@10.70.0.99</a>>;index=1,"5888"<br>
<<a href="mailto:sip%3A5888@10.70.0.99">sip:5888@10.70.0.99</a>>;index=1.1<br>
<br>
v=0<br>
o=- 1 1 IN IP4 10.70.0.69<br>
s=-<br>
c=IN IP4 10.70.0.22<br>
b=AS:64<br>
t=0 0<br>
m=audio 2176 RTP/AVP 18 0 101<br>
a=rtpmap:18 G729/8000<br>
a=fmtp:18 annexb=no<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:101 telephone-event/8000<br>
<br>
We don't support G729 so this call comes up as PCMU when we answer<br>
and then that codec is first in the codec list...<br>
<div class="im"><br>
On Wed, Dec 16, 2009 at 2:26 PM, Anthony Minessale<br>
<<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>> wrote:<br>
> can you do another trace to show the inbound invite too?<br>
><br>
<br>
</div>--<br>
<div><div></div><div class="h5">Kristian Kielhofner<br>
<a href="http://www.astlinux.org" target="_blank">http://www.astlinux.org</a><br>
<a href="http://blog.krisk.org" target="_blank">http://blog.krisk.org</a><br>
<a href="http://www.star2star.com" target="_blank">http://www.star2star.com</a><br>
<a href="http://www.submityoursip.com" target="_blank">http://www.submityoursip.com</a><br>
<a href="http://www.voalte.com" target="_blank">http://www.voalte.com</a><br>
<br>
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</div></div></blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
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