[asterisk-dev] Questions regarding g.729 and g.711 in Asterisk

[Chan Kwang Mien]

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