[asterisk-users] DIALSTATUS on CANCEL

[VoIP Question]

Hello,

We have a strange situation (asterisk 1.6.2.14), where we get a result for
DIALSTATUS for BUSY and No-ANSWER, but nothing for CANCEL.

This is the (relevant) test dialplan:
--------------------------------
[incoming-private]
exten => _X., n, Dial(SIP/1001,30)
exten => _X., n, NoOp(${DIALSTATUS})
exten => _X., n, Gosub(incoming-status,s-${DIALSTATUS},1)

[incoming-status]
exten => s-CANCEL,1, NoOp()
exten => s-CANCEL,n, Return()
exten => s-NOANSWER,1, NoOp()
exten => s-NOANSWER,n, Return()
exten => s-BUSY,1, NoOp()
exten => s-BUSY,n,  Return()


This is what we get on a BUSY call:
-----------------------------------
    -- Executing [11111111 at incoming-private:3] Dial("SIP/Proxy-0000002b",
"SIP/1001,50") in new stack
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP CoS mark 6
  == Using UDPTL CoS mark 5
    -- Called 1001
    -- Got SIP response 486 "Busy Here" back from 10.0.0.1
    -- SIP/1001-0000002c is busy
  == Everyone is busy/congested at this time (1:1/0/0)
    -- Executing [11111111 at incoming-private:4] NoOp("SIP/Proxy-0000002b",
"BUSY") in new stack
    -- Executing [11111111 at incoming-private:5] Gosub("SIP/Proxy-0000002b",
"incoming-status,s-BUSY,1") in new stack

This is what we get on a NO ANSWER call:
---------------------------------------
    -- Executing [11111111 at incoming-private:3] Dial("SIP/Proxy-0000002f",
"SIP/1001,30") in new stack
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP CoS mark 6
  == Using UDPTL CoS mark 5
    -- Called 1001
    -- SIP/1001-00000030 is ringing
    -- Nobody picked up in 30000 ms
    -- Executing [11111111 at incoming-private:4] NoOp("SIP/Proxy-0000002f",
"NOANSWER") in new stack
    -- Executing [11111111 at incoming-private:5] Gosub("SIP/Proxy-0000002f",
"incoming-status,s-NOANSWER,1") in new stack

This is what we get on a CANCEL call:
-------------------------------------
    -- Executing [11111111 at incoming-private:3] Dial("SIP/Proxy-00000031",
"SIP/1001,30") in new stack
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP CoS mark 6
  == Using UDPTL CoS mark 5
    -- Called 1001
    -- SIP/1001-00000032 is ringing
  == Spawn extension (incoming-private, 11111111, 3) exited non-zero on
'SIP/Proxy-00000031'

There's no event indicating that a DIALSTATUS is generated and the call
simply doesn't go to the next step in the dialplan. Unless I'm missing
something, it seems to me that it might be a bug.

I would be happy to get feedback from other users of the DIALSTATUS value
(or Digium), especially in the CANCEL scenario.

Thank you,

Michael
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