[Asterisk-Users] Passing audio stream through Asterisk or not?

[Dan]

Hi,
> if you turn off the reinvite in the asterisk configs for those ata186s
> then it will pass through asterisk even if asterisk doesn't understand
> the codec.
So I must have:
canreinvite = no
in sip.conf file for the specific phone?

Then the call is passed through Asterisk without any conversion?

How can I do to pass all the calls through Asterisk, even if a codec
conversion is required or not?

Thanks,
Dan

----- Original Message ----- 
From: "Steven Critchfield" <critch at basesys.com>
To: <asterisk-users at lists.digium.com>
Sent: Saturday, May 31, 2003 5:27 PM
Subject: Re: [Asterisk-Users] Passing audio stream through Asterisk or not?


> On Sat, 2003-05-31 at 08:06, Dan wrote:
> > Hi all,
> >
> > One short question.
> > When one extension (let's say ATA-186, SIP image, G.723 codec
> > selected) try to call an external SIP address like:
> > SIP/user at domain.com, where another identical ATA-186 is available with
> > G.723 codec selectrd,
> > after the signaling phase, the call is established through Asterisk or
> > directly between the two ATAs?
> > There is no G.723 codec available on Asterisk
> > I need to know this because of the firewall.
>
> if you turn off the reinvite in the asterisk configs for those ata186s
> then it will pass through asterisk even if asterisk doesn't understand
> the codec.
>
> -- 
> Steven Critchfield <critch at basesys.com>
>
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