[asterisk-users] Optimization of call from server 1 to 2 and then back to 1

[mancyborg at gmail.com]

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