[Asterisk-Users] codecs
Martin Pycko
martinp at digium.com
Sat Feb 22 18:18:35 MST 2003
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
> > 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.
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
More information about the asterisk-users
mailing list