<br>I've got a fresh (1 day old) svn trunk release SVN-trunk-r225360 of Asterisk and I'm trying to get it to accept a SIREN14 call from Polycom's softphone. Having trouble with SDP negotiation, I want to only allow SIREN14 and nothing else. I also want to record and playback files, any tips on what the Record function parameters should be?<br>
<br>In sip.conf I have:<br><br>disallow=all ; First disallow all codecs<br>allow=siren14 ; Is this the right name?<br><br><br>And the INVITE comes from the Polycom softphone with an SDP of:<br>
<br>...<br>User-Agent: Polycom VV 8.0.4.4035.<br>...<br>m=audio 12386 RTP/AVP 99 98 97 102 101 103 9 15 18 0 8.<br>a=rtpmap:99 SIREN14/16000.<br>a=fmtp:99 bitrate=48000.<br>a=rtpmap:98 SIREN14/16000.<br>a=fmtp:98 bitrate=32000.<br>
a=rtpmap:97 SIREN14/16000.<br>a=fmtp:97 bitrate=24000.<br>a=rtpmap:102 G7221/16000.<br>a=fmtp:102 bitrate=32000.<br>a=rtpmap:101 G7221/16000.<br>a=fmtp:101 bitrate=24000.<br>a=rtpmap:103 G7221/16000.<br>a=fmtp:103 bitrate=16000.<br>
a=rtpmap:9 G722/8000.<br>a=rtpmap:15 G728/8000.<br>a=rtpmap:18 G729/8000.<br>a=fmtp:18 annexb=no.<br>a=rtpmap:0 PCMU/8000.<br>a=rtpmap:8 PCMA/8000.<br>a=sendrecv.<br>m=video 12388 RTP/AVP 109 34 96 31.<br>b=TIAS:384000.<br>
a=rtpmap:109 H264/90000.<br>a=fmtp:109 profile-level-id=42800d; max-mbps=40000; max-fs=1792; max-br=1025.<br>a=rtpmap:34 H263/90000.<br>a=fmtp:34 CIF4=1;CIF=1;<br><br><br>Thanks in advance for any tips,<br><br>Tom<br>