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
