Greetings
I've noticed something odd while messing around with a test system and I'm not sure if this is a bug or not. I have three phones connected to an asterisk system in a remote office over a point to point T1 (no nat) all set up with canreinvite=yes. The phones are a Polycom 601, 501 and a budgetone 102. Here's what I am seeing:
Phone A calls phone B, phone B answers. RTP traffic travels between A and B directly like it is supposed to. Phone B does an attended transfer (using the phone's transfer features) to the parking extension, waits for the parked number and then completes the transfer. RTP travels from phone A to asterisk and hold music is played like it is supposed to. Now, phone B calls the parked extension and the call is reconnected, except the RTP traffic is now traveling A <-> Asterisk <-> B. I would have thought traffic should resume going between A and B directly. Is this an incorrect assumption or is it a bug?
I've tested this on 1.2.11 and SVN-branch-1.2-r41989. Thanks -Dave _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
