Sure... The call comes up as PCMU:
INVITE sip:5888@10.70.0.99 SIP/2.0
Call-ID: 80ea31a017f6de1d53e4a9c52f00
CSeq: 1 INVITE
From: sip:9413122830@smh.sip.local;tag=80ea31a017f6de1d43e4a9c52f00
Record-Route: <sip:10.70.0.65:5060;lr>,<sip:10.70.0.69;lr;transport=tcp>
To: "5888" <sip:5888@10.70.0.99>
Via: SIP/2.0/UDP
10.70.0.65:5060;branch=z9hG4bK838383030303565656105e9.0,SIP/2.0/TCP
10.70.0.69;psrrposn=2;received=10.70.0.69;branch=z9hG4bK80ea31a017f6de1d63e4a9c52f00
Content-Length: 206
Content-Type: application/sdp
Contact: <sip:9413122830@10.70.0.69;transport=tcp>
Max-Forwards: 70
User-Agent: Avaya CM/R015x.02.0.947.3
Allow: INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO,PUBLISH
Supported: timer,replaces,join,histinfo,100rel
Alert-Info: <cid:external@10.70.0.99>;avaya-cm-alert-type=external
Min-SE: 1200
Session-Expires: 1200;refresher=uac
P-Asserted-Identity: sip:9413122830@smh.sip.local
P-Charging-Vector: icid-value="AAS:13283-a031ea801def6179c4a3ed3f52"
History-Info: <sip:5888@10.70.0.99>;index=1,"5888"
<sip:5888@10.70.0.99>;index=1.1
v=0
o=- 1 1 IN IP4 10.70.0.69
s=-
c=IN IP4 10.70.0.22
b=AS:64
t=0 0
m=audio 2176 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
We don't support G729 so this call comes up as PCMU when we answer
and then that codec is first in the codec list...
--
On Wed, Dec 16, 2009 at 2:26 PM, Anthony Minessale
<anthony.minessale@gmail.com> wrote:
> can you do another trace to show the inbound invite too?
>
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com
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