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

[Steven Critchfield]

On Sat, 2003-05-31 at 10:51, Dan wrote:
> 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?

yes

> Then the call is passed through Asterisk without any conversion?

yes

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

canreinvite=no
The whole point is you don't reinvite the phones to talk to each other
instead of passing through asterisk.

> ----- 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>
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> 
> 
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-- 
Steven Critchfield <critch at basesys.com>