[asterisk-users] Asterisk NOT in the media path

[Jonas Kellens]

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


Regards,
Jonas.