Try it again after executing: "sip debug" and give us the actual SIP messages. The devil
is usually in the details.
Rich Adamson wrote:
Anyone have comments on this? Really could use some suggestions or ideas why this is happening. Thanks. Rich
------------------------
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
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