If someone has both IAX and SIP clients, would you please attempt to duplicate the below problem? I don't want to submit a bug unless the problem can be verified.
The SIP client must support attended transfers (ie: sayson, uniden): 1) Make a call from an IAX extension to a SIP extension 2) On the SIP phone, use attended transfer (not #) to transfer the call to a meetme room 3) Execute 'show channels' at the * CLI Do you see any 'zombie channels'? Thanks in advance, Ryan On Tue, 2004-23-11 at 14:18 -0700, Ryan Courtnage wrote: > 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 _______________________________________________ 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
