[asterisk-dev] Failing SIP sip_hangup()

[Steve Davies]

Hi,

(Hope this is a suitably dev-list topic)

I have a scenario where a SIP Invite/Replaces is generated through the
press of a directed-pickup button on a phone, but the SIP pickup code
do_magic_pickup() cannot find the call in question (it is in the wrong
context), In this case I would expect the Invite/Replaces call to be
hung-up, and that it what the code "tries" to do, as per line 13 of
the history:

pabx*CLI> sip show history 3c2671cbaa85-4c7i5s86qb7v
  * SIP Call
1. Rx              INVITE / 1 INVITE / sip:201 at 10.0.0.1
2. AuthChal        Auth challenge sent for  - nc 0
3. TxRespRel       SIP/2.0 / 1 INVITE - 401 Unauthorized
4. SchedDestroy    6400 ms
5. Rx              ACK / 1 ACK / sip:201 at 10.0.0.1
6. Rx              INVITE / 2 INVITE / sip:201 at 10.0.0.1
7. CancelDestroy
8. Invite          New call: 3c2671cbaa85-4c7i5s86qb7v
9. AuthOK          Auth challenge succesful for snom360
10. NewChan         Channel SIP/snom360-0000000f - from
3c2671cbaa85-4c7i5s86qb7v
11. Xfer            INVITE/Replace received
12. TxResp          SIP/2.0 / 2 INVITE - 100 Trying
13. Hangup          Cause Unknown
14. SchedDestroy    6400 ms
15. CancelDestroy

I would expect to see a "Cancel" being sent between lines 13 and 14 -
Any ideas why it is missing? This results in the Pickup call not being
cleaned up correctly.

Regards,
Steve