[Asterisk-Users] sip rfc bye violated?

[Matt Hess]

I have this in sip show history for a particular channel marked as dead 
(should be removed) in sip show channels:

1. TxReqRel        INVITE / 102 INVITE
2. Rx              SIP/2.0 / 102 INVITE
3. CancelDestroy
4. Rx              SIP/2.0 / 102 INVITE
5. CancelDestroy
6. Unhold          SIP/2.0
7. Rx              SIP/2.0 / 102 INVITE
8. CancelDestroy
9. Unhold          SIP/2.0
10. Rx              SIP/2.0 / 102 INVITE
11. CancelDestroy
12. Unhold          SIP/2.0
13. TxReq           ACK / 102 ACK
14. TxReqRel        INVITE / 103 INVITE
15. Rx              SIP/2.0 / 103 INVITE
16. CancelDestroy
17. Rx              SIP/2.0 / 103 INVITE
18. CancelDestroy
19. Unhold          SIP/2.0
20. TxReq           ACK / 103 ACK
21. TxReqRel        INVITE / 104 INVITE
22. Rx              BYE / 302 BYE
23. TxResp          SIP/2.0 / 302 BYE
24. Rx              SIP/2.0 / 104 INVITE
25. CancelDestroy

Why is asterisk allowing an invite after receiving a bye on a particular 
session/channel? From what I've read.. a bye should be the termination 
of the session/channel and therefore it should be hungup and removed.. 
yet it is not.

I am using cvs head from 2005-10-08 00:00 .. I can't use the latest cvs 
head as it's rather ugly with sip right now.. especially on 
refer/redirect/reinvites.. but that will be left for a different topic.

I believe from looking at things that the sip gateway involved with the 
sip session is re-using a particular call identifier immediately after 
it believes that call from before is gone.. (possibly a bug on the 
vendor side as far as that goes) but regardless of whether the vendor is 
immediately re-using a session id or not should not matter as the fact 
seems to be that asterisk allows this situation to happen when (from 
what I've been reading) it should not. Does anyone have any comments or 
thoughts on this?

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