[asterisk-users] Set Call Codec in extension.conf

[Danny Nicholas]

My guess is that you should set the codec either before SIPADDHEADER or
before ANSWER.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

I tried also in asterisk 1.8 setting outbound variable .... but didnt work
also ....
https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables
check the above ... I changed it and tried .... but still I get a video call
________________________________________
From: asterisk-users-bounces at lists.digium.com
[asterisk-users-bounces at lists.digium.com] On Behalf Of Eric Wieling
[EWieling at nyigc.com]
Sent: Wednesday, January 04, 2012 11:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

1.6 does not support setting the outbound codec.    1.8 uses different
variables to set the outbound codec.  See UGRADE.txt in the Asterisk source
for the 1.8 information,.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 12:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

1.6 and 1.8  ... I tried changing stuff on both ....
when I make audio call from my client which supports both audio and video
its sent to the other client as video call .....I tried settings the
SIP_CODED_INBOUND and outbound also ... but no luck
________________________________________
From: asterisk-users-bounces at lists.digium.com
[asterisk-users-bounces at lists.digium.com] On Behalf Of Eric Wieling
[EWieling at nyigc.com]
Sent: Wednesday, January 04, 2012 11:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

Providing which version of Asterisk you are using might be helpful.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 12:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

anyhelp guys?
 I tried a lot of stuff but it doesnt work .... the Codec for audio call
only cannt be set...how I can set the call type video/audio at dail plan?
________________________________________
From: asterisk-users-bounces at lists.digium.com
[asterisk-users-bounces at lists.digium.com] On Behalf Of Faraj Khasib
[fkhasib at iconnecths.com]
Sent: Wednesday, January 04, 2012 5:53 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Set Call Codec in extension.conf

Hi All,
I am trying to set call codec at extension.conf but it doesnt work ... its
like my command doesnt change anything


exten=6500,1,Answer
exten=6500,2,Playback(welcome)
exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)})
exten=6500,4,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:
%M:%S_%A)}_${SIP_HEADER(email)}.wav,b)
exten=6500,5,set(SIP_CODEC=gsm) -- this is not changed ....
exten=6500,6,Queue(${EXTEN})

can any body help me with that?
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users