Hi, My test-bed is
SIP Phone 1 <--> Asterisk IP PBX<--> SIP Phone 2 Asterisk IP PBX does not have the g.729 codec licence installed. SIP Phone 1 supports g.729 and g.711 while SIP Phone 2 supports only g.729. When SIP Phone 1 calls SIP Phone 2, SIP Phone 2 rings but it hangs up after it was answered. The Logs are as follows: Aug 7 03:52:59 WARNING[3331]: channel.c:2357 set_format: Unable to find a codec translation path from g729 to ulaw -- SIP/2003-33d5 answered SIP/2006-3753 Aug 7 03:52:59 WARNING[3336]: channel.c:2725 ast_channel_make_compatible: No path to translate from SIP/2006-3753(4) to SIP/2003-33d5(256) Aug 7 03:52:59 WARNING[3336]: app_dial.c:1608 dial_exec_full: Had to drop call because I couldn't make SIP/2006-3753 compatible with SIP/2003-33d5 My questions are : 1) Shouldn't the 2 SIP Phones able to connect to each other since both of them support the g.729 codecs ? 2) It seems that SIP Phone 1 uses g.711 codec to transmit its data. Would it be possible to configure Asterisk such that it forces the SIP Phone 1 to use g.729 since SIP Phone 2 is using g.729 ? Thank you. Best Regards, Kwang Mien _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev