[Asterisk-Users] select codec based on extension

[pdhales at optusnet.com.au]

Off hand, I agree that it's probably doable...even if you have to put
another sip server inbetween.
(or pay the $10 per channel for the g729 licence if it's only a few
channels)

PaulH

----- Original Message ----- 
From: "Simone Cittadini" <mymailforlists at gmail.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Sent: Thursday, December 29, 2005 7:52 PM
Subject: Re: [Asterisk-Users] select codec based on extension


> Leandro Rzezak ha scritto:
>
> > I'm having same problem. Were you able to solve it?
>
> No, codecs became a secondary problem later in our project so we ended
> up with 711 on all servers and more bandwidth,  anyway the post refers
> to asterisk 1.0.something and I never investigated the problem in more
> detail... I think it's possible, usually when you receive no answers (as
> the case of that post) you have made a really silly question :)
>
> >
> > On 10/18/05, *Simone Cittadini* <mymailforlists at gmail.com
> > <mailto:mymailforlists at gmail.com>> wrote:
> >
> >     I've the following installation :
> >
> >     |asterisk client| --- > |asterisk server| --- > |other asterisk
> >     server|
> >
> >     all the connections are made in IAX, the client and first server
> >     allows
> >     711 and 729
> >     the other server only allows 729 since it has low bandwidth at
> >     disposal
> >
> >     all the numbers but a few are routed to a digium card in the first
> >     server, the others are routed to the other server, this way :
> >
> >     [default]
> >
> >     exten => _123X.,1,Dial(IAX2/otherserver/${EXTEN})
> >     exten => _123X.,2,Hangup
> >
> >     exten => _X.,1,Dial(Zap/g1/${EXTEN})
> >     exten => _X.,2,Hangup
> >
> >     when I call 123456 from the client box ...
> >
> >     on the client :
> >     Call accepted by asterisk server (format alaw)
> >
> >     on the server :
> >     Call accepted by other asterisk server (format g729)
> >
> >     on the other server :
> >     Called 123456 at something
> >
> >     and then on the server in the middle :
> >     Oct 18 18:00:37 NOTICE[2846]: channel.c:1724 ast_set_write_format:
> >     Unable to find a path from alaw to g729
> >     Oct 18 18:00:37 NOTICE[2846]: channel.c:1757 ast_set_read_format:
> >     Unable
> >     to find a path from g729 to alaw
> >
> >     since that "something" at the end of the call and the paps which
sits
> >     before the first asterisk server both have g729, I don't like too
> >     much
> >     having to pay to translate something which need not translation.
> >     Is there a clever combination of sip.conf, iax.conf and
> >     extensions.conf
> >     I'm missing to solve my problem ?
> >     _______________________________________________
> >
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