[asterisk-users] How to pick a codec on the fly

[Danny Nicholas]

----------------------------------------
> From: danny at debsinc.com
> To: daniel at tryba.nl; asterisk-users at lists.digium.com
> Date: Mon, 27 Sep 2010 13:30:08 -0500
> Subject: Re: [asterisk-users] How to pick a codec on the fly
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Daniel Tryba
> Sent: Monday, September 27, 2010 1:17 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] How to pick a codec on the fly
>
> On Mon, Sep 27, 2010 at 01:02:04PM -0500, Danny Nicholas wrote:
> > I'm trying to test an IVR system with recorded prompts and would
> > like to be able to call 1234 and have the codec be gsm, 2234 slin, 3234
> > ulaw, etc. I know I can set up 3 users where #1 is gsm, #2 is ulaw and
#3
> > is slin; Need it the other way so I can do DAHDI--> IAX testing.
>
> exten => 1234,1,Set(_SIP_CODEC=alaw)
> exten => 1234,n,Goto(0234,1)
> exten => 2234,1,Set(_SIP_CODEC=slin)
> exten => 2234,n,Goto(0234,1)
>
> Should do the trick.
>
> --
>
> Daniel Tryba
>
> Nice try, Daniel, but apparently _SIP_CODEC is no longer useful in 1.4X.
> -- Executing [s at from-pstn:7] Goto("DAHDI/1-1", "default|s|1") in new stack
> -- Goto (default,s,1)
> -- Executing [s at default:1] Answer("DAHDI/1-1", "") in new stack
> -- Executing [s at default:2] Goto("DAHDI/1-1", "select-func|s|1") in new
> stack
> -- Goto (select-func,s,1)
> -- Executing [s at select-func:1] WaitExten("DAHDI/1-1", "5|m") in new
> stack
> -- Started music on hold, class 'default', on DAHDI/1-1
> -- Stopped music on hold on DAHDI/1-1
> == CDR updated on DAHDI/1-1
> -- Executing [2 at select-func:1] Set("DAHDI/1-1", "_SIP_CODEC=ulaw") in
> new stack
> -- Executing [2 at select-func:2] Dial("DAHDI/1-1", "IAX2/xxx/332|30|m") in
> new stack
> -- Called xxx/332
> -- Started music on hold, class 'default', on DAHDI/1-1
> -- Call accepted by XXX.XXX.XX.XX (format gsm)
> -- Format for call is gsm
> -- IAX2/ffb-18075 answered DAHDI/1-1
> -- Stopped music on hold on DAHDI/1-1
> -- Hungup 'IAX2/xxx-18075'
> == Spawn extension (select-func, 2, 2) exited non-zero on 'DAHDI/1-1'
> -- Hungup 'DAHDI/1-1'
>
>
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Tarek Sawah
Sent: Monday, September 27, 2010 1:40 PM
To: Asterisk Users
Subject: Re: [asterisk-users] How to pick a codec on the fly


I think it's SIP_CODEC now .. and not _SIP_CODEC?





Tarek Sawah

Information Technology Adviser

Integrated Digital Systems

CCNP, MCSE, RHCE, TELECOM

USA: +1 386 492 9993


FWIW, SIP_CODEC is value for use in Asterisk 1, _SIP_CODEC passes the value
on to Asterisk 2.