Hi all, I'm experiencing a problem with SIP channels going ZOmBIE after the following sequence of events:
- IAX2 client calls SIP client - SIP client consultive transfers (using sip REFER) the call to a MeetMe extension, and hangs up. At this point, the IAX2 client will indeed be in the meetme room, but a 'show channels' at the * CLI reveals that the SIP channels that were involved in the consultive transfer are still bridged and one is ZOMBIE. This will persist until issuing a 'soft hangup' to them. Oddly, I can only duplicate this problem when it's an IAX2 call being transfered (by a SIP client) to a meetme room. The phone's method of transfer (REFER) also seems to be a variable, as server side (#) transfers don't exhibit the problem. I've tested with both the Sayson 480i and the Uniden uip200. I'm using Asterisk v1.0.2 (CVS-v1-0-11/01/04) What could possibly be causing this, and can anyone reproduce it? Thanks Ryan _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
