asterisk users - May 2005 - xpro codecs and asterisk

Hi all,

I am trying to make a call from an X-Pro with only the G.729 codec enabled to
another with both G.711 and g.729. The Asterisk version is 1.0.3 and
canreeinvite is set to Yes. What happens is that I got an 403 - Forbidden
response and setting verbose 10 in Asterisk I can see the message:
May  2 15:47:36 WARNING[6690]: channel.c:2115 ast_channel_make_compatible: No
path to translate from SIP/victor-a02d(4) to SIP/dediana-1fd9(256)
    -- SIP/victor-a02d is ringing
    -- SIP/victor-a02d answered SIP/dediana-1fd9
May  2 15:47:38 WARNING[6690]: channel.c:2115 ast_channel_make_compatible: No
path to translate from SIP/dediana-1fd9(256) to SIP/victor-a02d(4)
May  2 15:47:38 WARNING[6690]: app_dial.c:1006 dial_exec: Had to drop call
because I couldn't make SIP/dediana-1fd9 compatible with SIP/victor-a02d

If I do the same when both softphones have only G.711 set, everything works
fine. It seems that Asterisk tries to use the first codec in SDP and ignores
others. Does it make sense?

Thanks in advance.
Dov
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----- Original Message ----- 
From: Dov Bigio
To: asterisk-users@lists.digium.com
Sent: Tuesday, May 03, 2005 11:01 AM
Subject: [Asterisk-Users] xpro codecs and asterisk


Hi all,

I am trying to make a call from an X-Pro with only the G.729 codec enabled to
another with both G.711 and g.729. The Asterisk
version is 1.0.3 and canreeinvite is set to Yes. What happens is that I got an
403 - Forbidden response and setting verbose 10 in
Asterisk I can see the message:
May  2 15:47:36 WARNING[6690]: channel.c:2115 ast_channel_make_compatible: No
path to translate from SIP/victor-a02d(4) to
SIP/dediana-1fd9(256)
    -- SIP/victor-a02d is ringing
    -- SIP/victor-a02d answered SIP/dediana-1fd9
May  2 15:47:38 WARNING[6690]: channel.c:2115 ast_channel_make_compatible: No
path to translate from SIP/dediana-1fd9(256) to
SIP/victor-a02d(4)
May  2 15:47:38 WARNING[6690]: app_dial.c:1006 dial_exec: Had to drop call
because I couldn't make SIP/dediana-1fd9 compatible with
SIP/victor-a02d

If I do the same when both softphones have only G.711 set, everything works
fine. It seems that Asterisk tries to use the first
codec in SDP and ignores others. Does it make sense?

Thanks in advance.
Dov



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