[Asterisk-Users] Possible Sip logic bug?

[Rich Adamson]

Anyone recognize this as a sip logic bug?

Example Case:
 C7960 -> * -> sip gateway -> pstn
 (sip gateway config'ed with canreinvite=no, but shouldn't have an
  impact on this.)

Outgoing call initiated from C7960. Call is completed and conversation
is very much normal. All equipment on the same wire; no nat.

The C7960 user hangs up the phone. Pkt flows (as observed by sniffer)
are:

C7960 sends sip BYE packet to *
  * returns 200 OK
* sends INVITE to sip gateway    <<================ where is BYE?
  sip gateway responds with 100 Trying
  sip gateway responds with 200 OK
  sip gateway responds with 200 OK
  sip gateway responds with 200 OK

The end result, the sip gateway does not drop the pstn line until the
"called" number hangs up.

It would appear that asterisk has an issue dropping the call. When the
C7960 issues the BYE, I would expect * to send a BYE to the sip g/w.
Is this a * logic problem (or my logic problem)?

(I'm actually running CVS-12/04/03-14:24:40 and has been very stable
in this production environment. Is it time to update this one even
though it is 99% sip hardphone based?)

Rich