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

[Dan]

Many thanks for clarification,
Dan

----- Original Message ----- 
From: "John Todd" <jtodd at loligo.com>
To: <asterisk-users at lists.digium.com>
Sent: Sunday, June 01, 2003 5:35 AM
Subject: Re: [Asterisk-Users] Passing audio stream through Asterisk or not?


>
> Because your original question asked about G.723, which Asterisk
> cannot transcode because there is no codec support for it within
> Asterisk.
>
> Summary: If all you want to do is have Asterisk "relay" the RTP
> (voice) data between two endpoints, pretty much any codec can be
> used, since Asterisk doesn't have to interpret the data stream for
> any reason - it's just moving the data around, and not "listening" or
> "talking" in the stream.  However, if you want to do something clever
> with that data/sound stream, such as listening for the "#" key (the
> "t" option in a Dial statement) then you'll need to be using a codec
> that Asterisk understands (G711, gsm, iLIBC, etc.)
>
> JT
>
>
> >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
> >  > >>  >
> >  > >>  >
> >  > >>
> >  > >>
> >  > >>  _______________________________________________
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> >  > >>  Asterisk-Users at lists.digium.com
> >  > >>  http://lists.digium.com/mailman/listinfo/asterisk-users
> >>  >--
> >>  >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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> >>
> >
> >
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