[asterisk-users] Asterisk NOT in the media path

[Kevin P. Fleming]

On 02/23/2012 01:48 PM, Jonas Kellens 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

This indicates that it *is* working. Asterisk has setup a 'native' RTP 
bridge between these two call legs. If they accept the re-INVITES that 
are sent, then the media will flow directly between them.

-- 
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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