I have this in sip.conf:
[general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes progressinband=yes [19256002182] type=friend username=19256002182 callerid="Test hone 1" <+19256002182> host=dynamic canreinvite=no secret=password context=test disallow=all allow=g729 [level3] type=peer host=xxx.yyy.16.99 context=default insecure=port dtmfmode=rfc2833 canreinvite=yes qualify=yes disallow=all ;allow=ulaw allow=g729 Level 3 sends early media... <--- Transmitting (no NAT) to xxx.yyy.34.210:5061 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP xxx.yyy.34.210:5061;branch=z9hG4bK-tenor-d802-22d2-004d;received=xxx.yyy .34.210 From: <sip:[EMAIL PROTECTED]>;tag=d80222d2-27 To: <sip:[EMAIL PROTECTED]>;tag=as4fe079a5 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[EMAIL PROTECTED]> ontent-Type: application/sdp Content-Length: 261 v=0 o=root 2235 2235 IN IP4 xxx.yyy.34.195 s=session c=IN IP4 xxx.yyy.34.195 t=0 0 m=audio 10484 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv and Asterisk responds on the console with: [Jun 22 10:06:03] WARNING[32573]: channel.c:2882 set_format: Unable to find a codec translation path from g729 to slin [Jun 22 10:06:03] WARNING[32573]: indications.c:121 playtones_alloc: Unable to set 'SIP/19256002182-096ac918' to signed linear format (write) This doesn't happen when progressinband=no. It almost seems like Asterisk has to do early media as G711 only. Is that the case??? Doug.
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