asterisk users - Jan 2012 - Asterisk NOT in the media path

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.


Thank you for your input !


Kind regards,
Jonas.
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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.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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