asterisk users - May 2003 - G723 - Has anyone gotten SIP_CODEC= to work?

FYI, asterisk DOES now support g723, but you have to pay for it:

http://store.yahoo.com/asteriskpbx/asteriskg729.html

-----Original Message-----
From: Dan Fernandez <danfernandez00@hotmail.com>
Date: Mon, 5 May 2003 17:33:05 -0300
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Has anyone gotten SIP_CODEC=   to work?

Basically, since I?d like to use g723 for sip communication between
endpoints and * does not support it, I need to change codecs when a user
wants to check voicemail, use a zap channel, etc.

I have configured sip.conf and extensions.conf as below but when I try it I
keep getting the following:

chan_sip.c ...(sip_answer):Changing codec to GSM for this call because of
${SIP_CODEC} variable
channel.c ..(ast_set_write_format): Unable to find a path from 2 to 1.

Any ideas?



sip.conf
disallow=all
allow=g723.1
allow=gsm


extensions.conf

exten => 1000,1,SetVar,SIP_CODEC=gsm
exten => 1000,2,VoiceMailMain

_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users

Not quite.  G.723 is not G.729, and thus G.723 for Asterisk is not 
currently sold on the Asterisk store, nor anywhere else that I've 
been able to find.  Anyone with alternate information is welcome to 
forward it to me...

JT

>FYI, asterisk DOES now support g723, but you have to pay for it:
>
>http://store.yahoo.com/asteriskpbx/asteriskg729.html
>
[snip]

Joe,

Thanks. I know * doesn?t support g723 (actually you can still have a SIP to
SIP call with g723) and that?s exactly why I need to change codecs on the
fly.

Dan

----- Original Message -----
From: "Joe Antkowiak" <joe@jsci.net>
To: <asterisk-users@lists.digium.com>; "Dan Fernandez"
<danfernandez00@hotmail.com>
Sent: Monday, May 05, 2003 7:19 PM
Subject: RE: [Asterisk-Users] G723 - Has anyone gotten SIP_CODEC= to work?

> FYI, asterisk DOES now support g723, but you have to pay for it:
>
> http://store.yahoo.com/asteriskpbx/asteriskg729.html
>
> -----Original Message-----
> From: Dan Fernandez <danfernandez00@hotmail.com>
> Date: Mon, 5 May 2003 17:33:05 -0300
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Has anyone gotten SIP_CODEC=   to work?
>
> Basically, since I?d like to use g723 for sip communication between
> endpoints and * does not support it, I need to change codecs when a user
> wants to check voicemail, use a zap channel, etc.
>
> I have configured sip.conf and extensions.conf as below but when I try it
I
> keep getting the following:
>
> chan_sip.c ...(sip_answer):Changing codec to GSM for this call because of
> ${SIP_CODEC} variable
> channel.c ..(ast_set_write_format): Unable to find a path from 2 to 1.
>
> Any ideas?
>
>
>
> sip.conf
> disallow=all
> allow=g723.1
> allow=gsm
>
>
> extensions.conf
>
> exten => 1000,1,SetVar,SIP_CODEC=gsm
> exten => 1000,2,VoiceMailMain
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>

Yes, my mistake, misread the e-mail...

Sorry.

-----Original Message-----
From: John Todd <jtodd@loligo.com>
Date: Mon, 5 May 2003 16:31:46 -0700
To: asterisk-users@lists.digium.com <asterisk-users@lists.digium.com>
Subject: RE: [Asterisk-Users] G723 - Has anyone gotten SIP_CODEC=   to work?

Not quite.  G.723 is not G.729, and thus G.723 for Asterisk is not 
currently sold on the Asterisk store, nor anywhere else that I've 
been able to find.  Anyone with alternate information is welcome to 
forward it to me...

JT

>FYI, asterisk DOES now support g723, but you have to pay for it:
>
>http://store.yahoo.com/asteriskpbx/asteriskg729.html
>
[snip]
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users