[asterisk-users] Asterisk NOT in the media path

[Jonas Kellens]

On 02/24/2012 10:51 PM, Jared Geiger wrote:
>
>
> On Thu, Feb 23, 2012 at 2:48 PM, Jonas Kellens 
> <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote:
>
>     On 01/20/2012 03:42 PM, Kevin P. Fleming wrote:
>
>         On 01/20/2012 08:07 AM, Jonas Kellens wrote:
>
>             Hello,
>
>             I want to place an Asterisk-server A in front of 2 other
>             Asterisk-servers (B1 & B2).
>
>             This first Asterisk-server A needs to send incoming calls
>             to one of the
>             2 available Asterisk-servers (B1 or B2) behind it.
>
>             So I want the first Asterisk-server A to accept the call,
>             and based upon
>             some checks in the dialplan send the call through to one
>             of the other
>             Asterisk-servers (B1 or B2) which further handle the call.
>
>             The first Asterisk-server A then needs to pull itself from the
>             media-path. There's no further need for this Asterisk to
>             stay within the
>             audio-path.
>
>             1. Is this possible ?
>             2. Using Asterisk 1.6.2.22, do I just use canreinvite=yes
>             in the peer
>             definition of Asterisk B1 and Asterisk B2 ?
>
>             So I have :
>
>             Provider >>> Asterisk A1 >>> Asterisk B1 & Asterisk B2
>
>             I want the audio to go directly from Provider to server B1
>             when the call
>             has been set up.
>
>
>         As long as there are no NATs involved, yes, this should work.
>         You will also need 'canreinvite' ('directmedia' in Asterisk
>         1.8 and later) in the peer definition for the provider.
>
>
>     Hello again,
>
>     this is currently not really working.
>
>     I see on the Asterisk CLI that the call streams through my
>     Asterisk A1 (which should stay out of the media path) :
>
>     [Feb 23 22:24:47]     -- Called Mast/980419
>     [Feb 23 22:24:47]     -- SIP/Mast-0000000e answered
>     SIP/VOXBONEin-0000000d
>     [Feb 23 22:24:47]     -- Native bridging SIP/VOXBONEin-0000000d
>     and SIP/Mast-0000000e
>     *CLI>
>     *CLI> core show channels
>     Channel              Location             State   Application(Data)
>     SIP/Mast-000000 (None)               Up      AppDial((Outgoing Line))
>     SIP/VOXBONEin-000000 980419 at VOXBONEin Up      Dial(SIP/Mast/980419)
>     2 active channels
>     1 active call
>
>     Peer VoxBone and peer Mast should re-invite and leave this
>     Asterisk out of the media path on call answer.
>
>     These are my SIP peer definitions :
>
>     [VOXBONEin]
>     type=peer
>     host=XX.XX.XX.XX
>     context=VOXBONEin
>     disallow=all
>     allow=alaw
>     allow=gsm
>     canreinvite=yes
>     qualify=yes
>     dtmfmode=rfc2833
>
>     [Mast]
>     type=peer
>     host=XX.XX.XX.XX
>     defaultuser=Mast
>     secret=guessme
>     disallow=all
>     allow=alaw
>     allow=gsm
>     canreinvite=yes
>     qualify=yes
>     dtmfmode=rfc2833
>
>
>     Am I missing a setting ? Using Asterisk 1.6.2.22
>
>
> The Asterisk server still stays in the SIP Signaling path of the call, 
> just media does not flow through the server. You can verify this by 
> running a SIP debug and looking at the media endpoints.

What is it that I should be looking for in the SIP debug information ? 
Is it in the SDP-body ?


Kind regards,
Jonas.
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