[asterisk-users] Does Early Media have to be Ulaw?

[Kristian Kielhofner]

On 6/22/07, Douglas Garstang <DGarstang at interainc.com> wrote:
>
>
>
>
> 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:19256002182 at xxx.yyy.34.195>;tag=d80222d2-27
>
> To: <sip:13033372500 at xxx.yyy.34.195>;tag=as4fe079a5
>
> Call-ID: call-71EEDB19-F102-2A10-0B0C-1B at xxx.yyy.34.210
>
> CSeq: 2 INVITE
>
> User-Agent: Asterisk PBX
>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>
> Supported: replaces
>
> Contact: <sip:13033372500 at xxx.yyy.34.195>
>
> ontent-Type: application/sdp
>
> Content-Length: 261
>
>
>
> v=0
>
> o=root 2235 2235 IN IP4 xxx.yyy.34.195
>
> s=session
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> c=IN IP4 xxx.yyy.34.195
>
> t=0 0
>
> m=audio 10484 RTP/AVP 18 101
>
> a=rtpmap:18 G729/8000
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> a=fmtp:18 annexb=no
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> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-16
>
> a=silenceSupp:off - - - -
>
> a=ptime:20
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> 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.
>

Doug,

  The SDP says it is G729 (no surprise there).  It looks like Asterisk
is trying to transcode that to slin from g729.  What dialplan logic is
this going into?  Can you post the section from extensions.conf?

  My guess is that some application in Asterisk (Dial, Queue,
something) is trying to generate ringing (playtones_alloc from
indications.c is a dead giveaway) but the call fails because you don't
have a g729 codec installed and can't transcode from slin (ringing).
Just a guess...


-- 
Kristian Kielhofner