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

[Dan]

I'm not sure that I understand you.
Why not to do transcoding if sometimes required?

Thanks,
Dan
----- Original Message ----- 
From: "John Todd" <jtodd at loligo.com>
To: <asterisk-users at lists.digium.com>
Sent: Saturday, May 31, 2003 7:35 PM
Subject: Re: [Asterisk-Users] Passing audio stream through Asterisk or not?


> There is one more note: make sure you don't have any options in your
> Dial statement that require the Asterisk server to do transcoding.
> Such options would be "r", or "m", or "t", which will cause Asterisk
> to need to listen and/or insert sounds in an audio stream if I
> understand previous conversations here to be correct.  I would just
> remove all options from your Dial statments entirely and see what you
> get.
>
> JT
>
>
> >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
> >>  >
> >>  >
> >>
> >>
> >>  _______________________________________________
> >>  Asterisk-Users mailing list
> >>  Asterisk-Users at lists.digium.com
> >>  http://lists.digium.com/mailman/listinfo/asterisk-users
> >--
> >Steven Critchfield <critch at basesys.com>
> >
> >_______________________________________________
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>
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