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

[Dome C.]

Please try this config
sip.conf

[sip1]
type=friend
host=dynamic
secret=pass
disallow=all
allow=g729

[sip2]
type=friend
host=dynamic
secret=pass
disallow=all
allow=g729

Chan Kwang Mien wrote:
> 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
>
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