[Asterisk-Users] codecs

Martin Pycko martinp at digium.com
Mon Feb 24 08:14:26 MST 2003


Yes, it's there.

regards
Martin

On 24 Feb 2003, Marian Danisek wrote:

> great !!!
> it is in cvs ?
>
> best regards ?
>
>
> > Actually now you can use SIP_CODEC variable
> >
> > eg:
> >
> > [sip-context]
> > exten => 8500,1,SetVar,SIP_CODEC=alaw
> > exten => 8500,2,VoiceMailMain
> > .....
> >
> > now when you normally have
> > dissallow=all
> > allow=g729
> > in sip.conf configuration file ... then when you place
> > a call with your SIP phone to 8500 asterisk will
> > force your phone to talk with alaw codec.
> >
> > regards
> > Martin
> >
> > On Wed, 19 Feb 2003, Martin Pycko wrote:
> >
> > > Yes, you can,
> > >
> > > you have to modify a little bit chan_sip
> > > to set up a codec that you need for certain extensions.
> > > It's going to be hardcoded.
> > >
> > > ps. maybe this will be a feature soon in the asterisk
> > >
> > > regards
> > > Martin
> > >
> > > On 19 Feb 2003, Marian Danisek wrote:
> > >
> > > > Hello,
> > > >
> > > > can i use different audio codecs when i calling between sip devices (
> > > > snom phones ) and different when i making call from isdn to sip or from
> > > > sip to isdn ?
> > > >
> > > > best regards
> > > >
> > > > Marian
> > > >
> > > > --
> > > > SUNTEQ s. r. o.
> > > > Bojnicka cesta 35 # Prievidza # 971 04 # Slovak republic
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> > > > http://www.sunteq.sk/
> > > > ------------------------------------------------------------
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> > > >
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> --
> SUNTEQ s. r. o.
> Bojnicka cesta 35 # Prievidza # 971 04 # Slovak republic
> Tel: +421-46-5430 754 # Fax: +421-46-5439 144
> http://www.sunteq.sk/
> ------------------------------------------------------------
> A mind is like a parachute... it only works when it's open.
>
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