[asterisk-users] DIALSTATUS on CANCEL

[Michael]

Anyone??

Thanks.

On Mon, Dec 20, 2010 at 10:42 AM, VoIP Question <voip.question at gmail.com>wrote:

> 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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