Hello all, I am allowing a reinvite between a snom 320 phone and a SIP gateway to take load off my Asterisk server. When I put the caller on hold, for example, Asterisk successfully reinserts itself into the rtp stream to play music on hold to the caller, but when I do a chanspy Asterisk does not seem to pull the call back. If I am spying on a channel when the call build up happens the reinvite never occurs and it works, but I cannot jump in and spy on a call in progress once the reinvite has happened.
Has anyone run into this issue any maybe have a solution, or does anyone know of a good way to get that call back onto the Asterisk switch from another extension prior to calling chanspy? Thanks much, Franklin Webb -- Franklin Webb Asst Project Manager Inter Medi@ Marketing Solutions 610-701-9670 [EMAIL PROTECTED] _______________________________________________ -- 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
