Anyone?? Thanks.
On Mon, Dec 20, 2010 at 10:42 AM, VoIP Question <voip.quest...@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 [11111...@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 [11111...@incoming-private:4] NoOp("SIP/Proxy-0000002b", > "BUSY") in new stack > -- Executing [11111...@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 [11111...@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 [11111...@incoming-private:4] NoOp("SIP/Proxy-0000002f", > "NOANSWER") in new stack > -- Executing [11111...@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 [11111...@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 >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users