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

[John Todd]

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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>--
>Steven Critchfield <critch at basesys.com>
>
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