[Asterisk-Users] xpro codecs and asterisk

Mailing List MailingList at mobilcom.net
Tue May 3 09:30:43 MST 2005


Please show your dialing context from extensions.conf


_____________
Mobilcom
http://www.mobilcom.net


----- Original Message ----- 
From: Dov Bigio
To: asterisk-users at 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



_______________________________________________
Asterisk-Users mailing list
Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users 




More information about the asterisk-users mailing list