[asterisk-users] SIREN14 call setup and record/playback

[Tom Browning]

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?

In sip.conf I have:

disallow=all                   ; First disallow all codecs
allow=siren14                ;  Is this the right name?


And the INVITE comes from the Polycom softphone with an SDP of:

...
User-Agent: Polycom VV 8.0.4.4035.
...
m=audio 12386 RTP/AVP 99 98 97 102 101 103 9 15 18 0 8.
a=rtpmap:99 SIREN14/16000.
a=fmtp:99 bitrate=48000.
a=rtpmap:98 SIREN14/16000.
a=fmtp:98 bitrate=32000.
a=rtpmap:97 SIREN14/16000.
a=fmtp:97 bitrate=24000.
a=rtpmap:102 G7221/16000.
a=fmtp:102 bitrate=32000.
a=rtpmap:101 G7221/16000.
a=fmtp:101 bitrate=24000.
a=rtpmap:103 G7221/16000.
a=fmtp:103 bitrate=16000.
a=rtpmap:9 G722/8000.
a=rtpmap:15 G728/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=sendrecv.
m=video 12388 RTP/AVP 109 34 96 31.
b=TIAS:384000.
a=rtpmap:109 H264/90000.
a=fmtp:109 profile-level-id=42800d; max-mbps=40000; max-fs=1792;
max-br=1025.
a=rtpmap:34 H263/90000.
a=fmtp:34 CIF4=1;CIF=1;


Thanks in advance for any tips,

Tom
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