[Asterisk-Dev] Cisco BTS 10200 G.729 problem

[Bill Hamlin]

I'm getting an INVITE from the Cisco softswitch looking like this (from sip
debug trace):

Sip read:
INVITE sip:9043940358 at 206.165.120.52;user=phone SIP/2.0
Via: SIP/2.0/UDP  sia-ATLCA146.telefyne.com:5060;branch=z9hG4bK_1146_38l5
From:
<sip:7278673170 at sia-ATLCA146.telefyne.com;user=phone>;tag=1_1146_f153592_732
j
To: <sip:9043940358 at 206.165.120.52;user=phone>
Call-ID: 2654525937 at sia-ATLCA146.telefyne.com
CSeq: 1 INVITE
Max-Forwards: 70
Supported: 100rel,precondition,timer
Contact: <sip:7278673170 at sia-ATLCA146.telefyne.com:5060>
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,REFER,UPD
ATE
User-Agent: BTS10200/900-04.02.00.V09(SIA)
Content-Length:    141
Content-Type: application/sdp

v=0
o=BTS 1097198076 1097198076 IN IP4 209.12.137.38
s=CISCO SDP 0
c=IN IP4 209.12.137.38
t=0 0
m=audio 19432/1 RTP/AVP 18
a=ptime:10

13 headers, 7 lines
Using latest request as basis request
Sending to 209.82.173.36 : 5060 (non-NAT)
Oct  7 19:10:47 WARNING[1093708096]: chan_sip.c:2667 process_sdp: Error in
codec string '=audio 19432/1 RTP/AVP 18'
redhat2*CLI>


This error makes the resulting call have no audio.  Does anyone know what
this codec problem is?  It seems like the extra /1 on the m= line is
different but I have no idea what that is supposed to be telling me.

Thanks for any help,
Bill Hamlin