asterisk users - Feb 2010 - Optimization of call from server 1 to 2 and then back to 1

Hi All,

suppose this call flow:

there are two Asterisk servers, they are connected through a IAX2 trunk.

The users use SIP.

The user A on the Asterisk server 1
calls the user B on the Asterisk server 2.

They talk for a while and then the user B does an attendant transfer to the user
C on the Asterisk server 1.

Question: is it possible to optimize the voice flow or the music on hold flow
so that it is done inside the Asterisk server 1 instead of forward and back:
from server 1 to 2 and then back to 1 ?


Thanks for your attention and for supporting,
have a nice day.
Mike

Difficult to say since you don't say if you are on 1.2, 1.4 or 1.6, but my
WAG would be that the IAX connection takes this out Asterisk 1's hands.  The
attendant transfer never breaks the IAX connection; it actually creates an
extra IAX connection to let A talk to C like this:
Original call
A --> IAX --> B
B --> IAX --> C
A --> IAX --> IAX --> C
You should be able to verify this with a core show channels during the two
legs.  
At any rate, MOH is controlled by the "holding" party, so when A puts
B or C
on hold, Asterisk 1 is controlling; B - Asterisk 2; C - Asterisk 2 via IAX;

Go ahead, shoot me down if I'm wrong; just an educated WAG
--


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
mancyborg at gmail.com
Sent: Wednesday, February 10, 2010 8:47 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Optimization of call from server 1 to 2 and
thenback to 1

Hi All,

suppose this call flow:

there are two Asterisk servers, they are connected through a IAX2 trunk.

The users use SIP.

The user A on the Asterisk server 1
calls the user B on the Asterisk server 2.

They talk for a while and then the user B does an attendant transfer to the
user C on the Asterisk server 1.

Question: is it possible to optimize the voice flow or the music on hold
flow
so that it is done inside the Asterisk server 1 instead of forward and back:
from server 1 to 2 and then back to 1 ?


Thanks for your attention and for supporting,
have a nice day.
Mike

-- 
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users