Fw: [Asterisk-Users] Possible Sip logic bug?

[Rich Adamson]

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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