Hi Folks,
I am running a rather old version of Asterisk (see below), and all I need to do is to somehow ignore the multiple m= coming into from a Cisco 7961G phone. Below is how the SIP OK looks like coming from the Cisco 7961G Phone. If you look at the OK below, you will see that there are multiple m= in the SDP. m=audio and m=video. The problem is that the Cisco phone is sending all 0’s in the m=video portion (don’t know why), and the Asterisk is using that for audio as well, which translates to one way audio.
Is there a quick way to ignore the second m= in the SDP in Asterisk ?
Thanks
Atif…
Asterisk 1.4.22-4 RPM
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.99:5060;branch=z9hG4bK5fe0a3e9;rport
From: sip:114@192.168.100.99;tag=as1ddfa911
To: sip:103@192.168.100.123:5060;transport=udp;tag=001759e96ce045fe7e739029-4cf9f199
Call-ID: 00baf1ad650361113e276d0257859a71@192.168.100.99
Date: Mon, 28 Oct 2013 20:03:27 GMT
CSeq: 102 INVITE
Server: Cisco-CP7961G/9.3.1
Contact: sip:103@192.168.100.123:5060;transport=udp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Remote-Party-ID: sip:103@192.168.100.99;party=called;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.0,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Content-Length: 251
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 21707 0 IN IP4 192.168.100.123
s=SIP Call
t=0 0
m=audio 32636 RTP/AVP 0 101
c=IN IP4 192.168.100.123
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 0 RTP/AVP 34
c=IN IP4 0.0.0.0