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

Chan Kwang Mien kwangmien at asgent-tech.com
Mon Aug 7 01:58:57 MST 2006


Thanks. By setting allow=g729, sip1 was able to connect to sip2. 

Does that mean that Asterisk chooses the codec to be used between the
Caller and Callee ? In this case, since sip1 informs Asterisk that it
supports g.711 and g.729, Asterisk chooses g.729 since the Callee also
supports g.729. From the SIP Messages exchange, it doesn't seem that
Asterisk chooses the codec.

My previous setting was "allow=all". I was expecting "allow=all" to work
since that would also imply "allow=g729", isn't it ?


> 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
> >
> > _______________________________________________
> > --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
> >   
> 
> _______________________________________________
> --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
> 




More information about the asterisk-dev mailing list