SIP Channels Getting Stuck

Asterisk Asterisk Support
1

I am having a weird problem where I have some SIP channels which aren’t cleaning up properly.

The thing I find strange is that nothing shows in “core show channels”, as in there are 0 active channels, but when I do “sip show channels” i get a couple of ‘stuck’ channels. The output shows that the last message received was BYE, which would indicate to me that they should have cleaned up.

atl-asterisk5*CLI> core show channels
Channel              Location             State   Application(Data)
0 active channels
0 active calls
atl-asterisk5*CLI> sip show channels
Peer             User/ANR    Call ID      Seq (Tx/Rx)  Format           Hold     Last Message
216.82.XXX.XXX   +17066XXX  6b25419936b  00102/00001  0x0 (nothing)    No  (d)  Rx: BYE
216.82.XXX.XXX   +14042XXX  3955ba5d7c2  00102/00001  0x0 (nothing)    No  (d)  Rx: BYE
2 active SIP channels

When I look at the detail of one of the channels, it appears they have their “need destroy” flag set, but they weren’t ever destroyed.

atl-asterisk5*CLI> sip show channel 6b25419936b
atl-asterisk5*CLI>
  * SIP Call5*CLI>
  Curr. trans. direction:  Outgoing
  Call-ID:                6b25419936bd5cac4b4dbe6568042787@blah.com
  Owner channel ID:       <none>
  Our Codec Capability:   260
  Non-Codec Capability (DTMF):   1
  Their Codec Capability:   256
  Joint Codec Capability:   256
  Format:                 0x0 (nothing)
  MaxCallBR:              384 kbps
  Theoretical Address:    216.82.XXX.XXX:5060
  Received Address:       216.82.XXX.XXX:5060
  SIP Transfer mode:      open
  NAT Support:            Always
  Audio IP:               67.220.XXX.XXX (local)
  Our Tag:                as1f608769
  Their Tag:              VPST506071629460
  SIP User agent:
  Username:               +1706687XXXX
  Peername:               bw_g729
  Original uri:           sip:+170668XXXX@216.82.XXX.XXX
  Need Destroy:           2
  Last Message:           Rx: BYE
  Promiscuous Redir:      No
  Route:                  N/A
  DTMF Mode:              rfc2833
  SIP Options:            (none)

I am running what should be a stable 1.4 version.

Asterisk 1.4.21.2 built by root @ atl-asterisk5.blah.com on a x86_64 running Linux on 2008-09-26 20:23:10 UTC

Thanks,
G