[asterisk-users] Asterisk NOT in the media path

[Jared Geiger]

On Thu, Feb 23, 2012 at 2:48 PM, Jonas Kellens <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.

If you don't want the call to maintain in the server at all, you need to
look into doing 302 redirects.

Regards,
Jared
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