[asterisk-dev] Failing SIP sip_hangup()

[Olle E. Johansson]

12 maj 2010 kl. 12.55 skrev 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.
Cancel is sent by the caller to interrupt. We send error codes in response
to the hangup, which happened in line 13 I hope.

/O