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

[Dan]

Many thanks,
Dan

----- Original Message ----- 
From: "Steven Critchfield" <critch at basesys.com>
To: <asterisk-users at lists.digium.com>
Sent: Saturday, May 31, 2003 7:29 PM
Subject: Re: [Asterisk-Users] Passing audio stream through Asterisk or not?


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