[Asterisk-Users] Transferring a call with IAX

Aaron Daniel amdtech at shsu.edu
Fri Mar 24 17:15:35 MST 2006 

Glad it worked out... I learned something new today, and I'm probably 
gonna program that into my dialplan cause I never realized it didn't make 
it past the dial when they answered :-X

No, it doesn't support #include or hinting yet, there's an AEL2, but I'm 
not too thrilled about how it handles apps.  It requires you to maintain a 
list of apps that it compares your dialplan to to make sure that your 
dialplan doesn't include bad options and stuff... too much work for the 
person setting up the system in my opinion.

We're converting back to extensions.conf (actually, I'm doing that right 
now lol) since we need the hinting :)

Aaron

On Fri, 24 Mar 2006, Douglas Garstang wrote:

> Good grief! That did it!
>
> I think I'm gonna poo my pants....
>
> Thanks Aaron... and btw, I see your using AEL... know  if that supports #include yet?
>
> Douglas.
>
>> -----Original Message-----
>> From: Aaron Daniel [mailto:amdtech at shsu.edu]
>> Sent: Friday, March 24, 2006 4:59 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: RE: [Asterisk-Users] Transferring a call with IAX
>>
>>
>> Ok, add g to the option list on the dial:
>>
>> Dial(IAX2/pbxuser@${ARG1}/${ARG2}@${ARG3},25,wWg)
>>
>>      g    - Proceed with dialplan execution at the current
>> extension if the
>>             destination channel hangs up.
>>
>> Aaron
>>
>> On Fri, 24 Mar 2006, Douglas Garstang wrote:
>>
>>> I just changed the macro to:
>>>
>>> exten => s,1,Dial(IAX2/pbxuser@${ARG1}/${ARG2}@${ARG3},25,wW)
>>> exten => s,2,NoOp(${DIALSTATUS})
>>>
>>> and the NoOp doesn't get executed. Bloody hell!
>>> Console has:
>>>    -- Hungup 'IAX2/acdserver1-3'
>>>  == Spawn extension (macro-DialIAX, s, 1) exited non-zero
>> on 'SIP/2944093-9ef2' in macro 'DialIAX'
>>>  == Spawn extension (macro-DialIAX, s, 1) exited non-zero
>> on 'SIP/2944093-9ef2'
>>>
>>>
>>>> -----Original Message-----
>>>> From: Douglas Garstang
>>>> Sent: Friday, March 24, 2006 4:50 PM
>>>> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
>>>> Subject: RE: [Asterisk-Users] Transferring a call with IAX
>>>>
>>>>
>>>> Nope. Still no go.
>>>>
>>>> Caller has this:
>>>>     -- Hungup 'IAX2/acdserver1-2'
>>>>   == Spawn extension (macro-DialIAX, s, 1) exited non-zero on
>>>> 'SIP/2944093-6f31' in macro 'DialIAX'
>>>>   == Spawn extension (macro-DialIAX, s, 1) exited non-zero on
>>>> 'SIP/2944093-6f31'
>>>>
>>>> and the macro has:
>>>> exten => s,1,Dial(IAX2/pbxuser@${ARG1}/${ARG2}@${ARG3},25,wW)
>>>> exten => s,2,Goto(s-${DIALSTATUS},1)
>>>> exten => s-ANSWER,1,NoOp(HERE I AM) #Goto(s-OK,1)
>>>>
>>>> It never gets to s-ANSWER, eventhough the debug shows DIAL
>>>> returns ANSWER. If I shut my ACD server down, I get
>>>> CHANUNAVAIL, and THAT jumps to s-CHANUNAVAIL.
>>>>
>>>> *sigh*
>>>>
>>>>
>>>>> -----Original Message-----
>>>>> From: Aaron Daniel [mailto:amdtech at shsu.edu]
>>>>> Sent: Friday, March 24, 2006 4:41 PM
>>>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>>>> Subject: RE: [Asterisk-Users] Transferring a call with IAX
>>>>>
>>>>>
>>>>> Why do you have s-ANSWER jumping to s-OK?  Try putting a NoOp
>>>>> in s-ANSWER
>>>>> and see if it's making it there... Also, when the call
>>>>> doesn't make it
>>>>> through, does it jump through the s-DIALSTATUS priorities?
>>>>>
>>>>> Aaron
>>>>>
>>>>> On Fri, 24 Mar 2006, Douglas Garstang wrote:
>>>>>
>>>>>> Nope... that's not the problem here. I put a NoOp right
>>>>> before the MacroExit, and it didn't execute that either.
>>>>>>
>>>>>> Dial returns ANSWER, and so it should execute (2),but it
>>>>> doesn't. This drives me insane. I have lost count of how many
>>>>> days I've wasted trying to get the most basic things to work
>>>>> in Asterisk.
>>>>>>
>>>>>> exten => s,1,Dial(IAX2/pbxuser@${ARG1}/${ARG2}@${ARG3},25,wW)
>>>>>> exten => s,2,Goto(s-${DIALSTATUS},1)
>>>>>> exten => s-ANSWER,1,Goto(s-OK,1)
>>>>>> exten => s-NOANSWER,1,Goto(s-ERROR,1)
>>>>>> exten => s-CONGESTION,1,Goto(s-ERROR,1)
>>>>>> exten => s-CHANUNAVAIL,1,Goto(s-ERROR,1)
>>>>>> exten => s-ERROR,1,Answer()
>>>>>> exten => s-ERROR,2,Wait,1
>>>>>> exten => s-ERROR,3,Set(i=1)
>>>>>> exten => s-ERROR,4,While($[${i} < 4])
>>>>>> exten => s-ERROR,5,Playback(cannot-complete-network-error)
>>>>>> exten => s-ERROR,6,Playback(message-number)
>>>>>> exten => s-ERROR,7,Playback(letters/o)
>>>>>> exten => s-ERROR,8,Playback(letters/e)
>>>>>> exten => s-ERROR,9,Playback(digits/9)
>>>>>> exten => s-ERROR,10,Playback(digits/0)
>>>>>> exten => s-ERROR,11,Playback(digits/0)
>>>>>> exten => s-ERROR,12,Set(i=$[${i} + 1])
>>>>>> exten => s-ERROR,13,EndWhile
>>>>>> exten => s-ERROR,14,Hangup()
>>>>>> exten => s-OK,1,NoOP(CONTROL BACK INSIDE MACRO)
>>>>>> exten => s-OK,2,MacroExit
>>>>>>
>>>>>>> -----Original Message-----
>>>>>>> From: Aaron Daniel [mailto:amdtech at shsu.edu]
>>>>>>> Sent: Friday, March 24, 2006 4:33 PM
>>>>>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>>>>>> Subject: RE: [Asterisk-Users] Transferring a call with IAX
>>>>>>>
>>>>>>>
>>>>>>> Looking at your macro, I don't have any MacroExits in mine.
>>>>>>> I use AEL,
>>>>>>> and it doesn't put that on the macros.  Try changing your
>>>>>>> MacroExit to a
>>>>>>> NoOp(Macro Finished) and see if that drops you back into the
>>>>>>> original call
>>>>>>> structure.
>>>>>>>
>>>>>>> Aaron
>>>>>>>
>>>>>>> On Fri, 24 Mar 2006, Douglas Garstang wrote:
>>>>>>>
>>>>>>>> Aaron, this is what I get, debug turned up and all...
>>>>>>>>
>>>>>>>> Mar 24 16:17:47 DEBUG[19475] chan_iax2.c: Immediately
>>>>>>> destroying 3, having received hangup
>>>>>>>> Mar 24 16:17:47 DEBUG[29506] channel.c: Didn't get a frame
>>>>>>> from channel: IAX2/acdserver1-3
>>>>>>>> Mar 24 16:17:47 DEBUG[29506] channel.c: Bridge stops
>>>>>>> bridging channels SIP/2944093-20ac and IAX2/acdserver1-3
>>>>>>>> Mar 24 16:17:47 DEBUG[29506] chan_iax2.c: We're hanging up
>>>>>>> IAX2/acdserver1-3 now...
>>>>>>>> Mar 24 16:17:47 DEBUG[29506] chan_iax2.c: Really destroying
>>>>>>> IAX2/acdserver1-3 now...
>>>>>>>> Mar 24 16:17:47 VERBOSE[29506] logger.c:     -- Hungup
>>>>>>> 'IAX2/acdserver1-3'
>>>>>>>> Mar 24 16:17:47 DEBUG[29506] app_dial.c: Exiting with
>>>>>>> DIALSTATUS=ANSWER.
>>>>>>>> Mar 24 16:17:47 VERBOSE[29506] logger.c:   == Spawn
>>>>>>> extension (macro-DialIAX, s, 1) exited non-zero on
>>>>>>> 'SIP/2944093-20ac' in macro 'DialIAX'
>>>>>>>> Mar 24 16:17:47 VERBOSE[29506] logger.c:   == Spawn
>>>>>>> extension (macro-DialIAX, s, 1) exited non-zero on
>>>>> 'SIP/2944093-20ac'
>>>>>>>> Mar 24 16:17:47 DEBUG[29506] cdr_addon_mysql.c: cdr_mysql:
>>>>>>> inserting a CDR record.
>>>>>>>>
>>>>>>>> It's all greek to me... actually you can see it exits with
>>>>>>> DIALSTATUS=Answer. My macro calls MacroExit() on ANSWER,
>>>>>>> which should return control back to where the Macro was
>>>>>>> called from! How weird.. it looks like I _AM_ getting control
>>>>>>> back, sort of...
>>>>>>>>
>>>>>>>> Doug.
>>>>>>>>
>>>>>>>>> -----Original Message-----
>>>>>>>>> From: Aaron Daniel [mailto:amdtech at shsu.edu]
>>>>>>>>> Sent: Friday, March 24, 2006 4:07 PM
>>>>>>>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>>>>>>>> Subject: RE: [Asterisk-Users] Transferring a call with IAX
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> Hmm... and nothing in the macro after the dial command is
>>>>>>>>> being executed?
>>>>>>>>> What does the CLI say on the caller server when the
>>>> ACD server is
>>>>>>>>> finished?
>>>>>>>>>
>>>>>>>>> Aaron
>>>>>>>>>
>>>>>>>>> On Fri, 24 Mar 2006, Douglas Garstang wrote:
>>>>>>>>>
>>>>>>>>>> Aaron... I don't think that's it.
>>>>>>>>>>
>>>>>>>>>> When I comment out the Macro call on the ACD server, the
>>>>>>>>> NoOP(QUEUE DONE) is called, and that's where it stops.
>>>>>>>>> Without the macro being called on the ACD server, control
>>>>>>>>> should return to the PBX server and it does not.
>>>>>>>>>>
>>>>>>>>>> Here's what the caller has:
>>>>>>>>>> exten => 2944000,1,Dial(SIP/2944030,15,tr)
>>>>>>>>>> exten => 2944000,2,Answer
>>>>>>>>>> exten => 2944000,3,Wait,1
>>>>>>>>>> exten => 2944000,4,Playback(thank-you-for-calling)
>>>>>>>>>> exten => 2944000,5,Playback(customer-service)
>>>>>>>>>> exten =>
>>>>>>>>> 2944000,6,Macro(DialIAX,acdserver1,oe_custcare,oneeighty_acd)
>>>>>>>>>> exten => 2944000,7,NoOp(CONTROL RETURNED) <-- this does
>>>>>>> NOT execute
>>>>>>>>>>
>>>>>>>>>> and here's what the callee has:
>>>>>>>>>> exten => oe_custcare,1,Answer
>>>>>>>>>> exten => oe_custcare,2,Queue(oe_custcare||||120)
>>>>>>>>>> exten => oe_custcare,3,NoOP(QUEUE DONE) <-- this executes
>>>>>>>>>> exten => oe_custcare,4,Hangup <-- this also executes
>>>>>>>>>>
>>>>>>>>>> and here's the caller's macro:
>>>>>>>>>> exten => s,1,Dial(IAX2/pbxuser@${ARG1}/${ARG2}@${ARG3},25,wW)
>>>>>>>>>> exten => s,2,Goto(s-${DIALSTATUS},1)
>>>>>>>>>> exten => s-ANSWER,1,Goto(s-OK,1)
>>>>>>>>>> exten => s-NOANSWER,1,Goto(s-ERROR,1)
>>>>>>>>>> exten => s-CONGESTION,1,Goto(s-ERROR,1)
>>>>>>>>>> exten => s-CHANUNAVAIL,1,Goto(s-ERROR,1)
>>>>>>>>>> exten => s-ERROR,1,Answer()
>>>>>>>>>> exten => s-ERROR,2,Wait,1
>>>>>>>>>> exten => s-ERROR,3,Set(i=1)
>>>>>>>>>> exten => s-ERROR,4,While($[${i} < 4])
>>>>>>>>>> exten => s-ERROR,5,Playback(cannot-complete-network-error)
>>>>>>>>>> exten => s-ERROR,6,Playback(message-number)
>>>>>>>>>> exten => s-ERROR,7,Playback(letters/o)
>>>>>>>>>> exten => s-ERROR,8,Playback(letters/e)
>>>>>>>>>> exten => s-ERROR,9,Playback(digits/9)
>>>>>>>>>> exten => s-ERROR,10,Playback(digits/0)
>>>>>>>>>> exten => s-ERROR,11,Playback(digits/0)
>>>>>>>>>> exten => s-ERROR,12,Set(i=$[${i} + 1])
>>>>>>>>>> exten => s-ERROR,13,EndWhile
>>>>>>>>>> exten => s-ERROR,14,Hangup()
>>>>>>>>>> exten => s-OK,1,MacroExit
>>>>>>>>>>
>>>>>>>>>> ... on callee:
>>>>>>>>>>    -- Executing NoOp("IAX2/216.187.142.203:4569-5", "QUEUE
>>>>>>>>> DONE") in new stack
>>>>>>>>>>    -- Executing Hangup("IAX2/216.187.142.203:4569-5", "")
>>>>>>>>> in new stack
>>>>>>>>>>
>>>>>>>>>> ... on the caller:
>>>>>>>>>>    -- Hungup 'IAX2/acdserver1-3'
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>> -----Original Message-----
>>>>>>>>>>> From: Aaron Daniel [mailto:amdtech at shsu.edu]
>>>>>>>>>>> Sent: Friday, March 24, 2006 3:52 PM
>>>>>>>>>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>>>>>>>>>> Subject: RE: [Asterisk-Users] Transferring a call with IAX
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> My bad, sorry, one of those days.
>>>>>>>>>>>
>>>>>>>>>>> Change priority 4 on the ACD server to a Hangup and ignore
>>>>>>>>>>> what I said
>>>>>>>>>>> before about putting in priority 5.  Put the macro call you
>>>>>>>>>>> had on the ACD
>>>>>>>>>>> server on the PBX server, and that should fix your problem.
>>>>>>>>>>> Since you're
>>>>>>>>>>> having the ACD server do a macro of it's own, it's not
>>>>>>>>>>> getting sent back
>>>>>>>>>>> directly to the PBX server.
>>>>>>>>>>>
>>>>>>>>>>> Let me know how that works.
>>>>>>>>>>>
>>>>>>>>>>> Aaron
>>>>>>>>>>>
>>>>>>>>>>> On Fri, 24 Mar 2006, Douglas Garstang wrote:
>>>>>>>>>>>
>>>>>>>>>>>> Aaron.
>>>>>>>>>>>>
>>>>>>>>>>>> Uhm... yes. I thought you picked up on that.
>>>>>>>>>>>> It's like this:
>>>>>>>>>>>>
>>>>>>>>>>>> PBX Server -> ACD Server(queue times out) -> VM Server
>>>>>>>>>>>>
>>>>>>>>>>>> I'd like it to go like this:
>>>>>>>>>>>>
>>>>>>>>>>>> PBX Server -> ACD Server(queue times out) -> PBX Server
>>>>>>>>> -> VM Server
>>>>>>>>>>>>
>>>>>>>>>>>> So, after the pbx server dials the acd server, and the
>>>>>>>>>>> queue times out, I wanted to have control returned
>> to the pbx
>>>>>>>>>>> server where _it_ could dial the VM server, instead of the
>>>>>>>>>>> ACD server doing it. I thought you where doing
>>>>> something similar?
>>>>>>>>>>>>
>>>>>>>>>>>> Douglas.
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>> -----Original Message-----
>>>>>>>>>>>>> From: Aaron Daniel [mailto:amdtech at shsu.edu]
>>>>>>>>>>>>> Sent: Friday, March 24, 2006 2:51 PM
>>>>>>>>>>>>> To: Asterisk Users Mailing List - Non-Commercial
>> Discussion
>>>>>>>>>>>>> Subject: RE: [Asterisk-Users] Transferring a call with IAX
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> Hhhmmm... I missed something... You're jumping from one
>>>>>>>>>>>>> calling server
>>>>>>>>>>>>> through a "callee" server, and then from there to another
>>>>>>>>>>> server for
>>>>>>>>>>>>> voicemail?
>>>>>>>>>>>>>
>>>>>>>>>>>>> Aaron
>>>>>>>>>>>>>
>>>>>>>>>>>>> On Fri, 24 Mar 2006, Douglas Garstang wrote:
>>>>>>>>>>>>>
>>>>>>>>>>>>>> Thanks Aaron, but nope... that didn't do it. I put an
>>>>>>>>>>>>> explicit hangup right after the Queue app on the
>>>> ACD server,
>>>>>>>>>>>>> and I see this when it times out:
>>>>>>>>>>>>>> Executing Hangup("IAX2/216.187.142.203:4569-2", "")
>>>>>>> in new stack
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> However, the calling server never regained control. Ahhh
>>>>>>>>>>>>> Asterisk a marvelous thing... I can see myself
>>>> spending days
>>>>>>>>>>>>> on trying to get this to work.
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Doug
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> -----Original Message-----
>>>>>>>>>>>>>>> From: Aaron Daniel [mailto:amdtech at shsu.edu]
>>>>>>>>>>>>>>> Sent: Friday, March 24, 2006 1:43 PM
>>>>>>>>>>>>>>> To: Asterisk Users Mailing List - Non-Commercial
>>>> Discussion
>>>>>>>>>>>>>>> Subject: RE: [Asterisk-Users] Transferring a
>>>> call with IAX
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Heh, lots of voodoo... I've got a drawer full of
>>>>> dolls shaped
>>>>>>>>>>>>>>> like servers
>>>>>>>>>>>>>>> that we stick pins into when something's not working :)
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Anyway... um, let's see if I can piece this together,
>>>>>>>>> it's kinda
>>>>>>>>>>>>>>> scattered...
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> A call comes from SCM2 (the secondary call
>> server) and it
>>>>>>>>>>>>>>> starts looking
>>>>>>>>>>>>>>> for the phone with this:
>>>>>>>>>>>>>>>                  Dial(SIP/${info_forwardto},25);
>>>>>>>>>>>>>>> then using the DIALSTATUS, if it finds that it's in
>>>>>>>>>>>>>>> CHANUNAVAIL, it sends
>>>>>>>>>>>>>>> it to the primary server:
>>>>>>>>>>>>>>>                  case "CHANUNAVAIL":
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>
>>>> Dial(IAX2/asterisk:password at scm1.shsu.edu/${info_forwardto},25,wW);
>>>>>>>>>>>>>>>                          &uvm(${ext});
>>>>>>>>>>>>>>>                          Hangup;
>>>>>>>>>>>>>>>                          break;
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> In order to keep the call compartmentalized, on SCM1,
>>>>>>>>> we've got:
>>>>>>>>>>>>>>> context from-scm2 {
>>>>>>>>>>>>>>>          _4XXXX => {
>>>>>>>>>>>>>>>                  NoOp(DIALING SIP EXTENSION
>>>> ${EXTEN} - FROM
>>>>>>>>>>>>>>> ${CALLERIDNUM});
>>>>>>>>>>>>>>>                  Dial(SIP/${EXTEN},20,wW);
>>>>>>>>>>>>>>>                  Hangup;
>>>>>>>>>>>>>>>          };
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>          _6XXXX => {
>>>>>>>>>>>>>>>                  NoOp(DIALING SIP EXTENSION
>>>> ${EXTEN} - FROM
>>>>>>>>>>>>>>> ${CALLERIDNUM});
>>>>>>>>>>>>>>>                  Dial(SIP/${EXTEN},20,wW);
>>>>>>>>>>>>>>>                  Hangup;
>>>>>>>>>>>>>>>          };
>>>>>>>>>>>>>>> };
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> I think your problem is that the other server
>>>> isn't hanging
>>>>>>>>>>>>>>> up the line
>>>>>>>>>>>>>>> when it runs out of the queue.  Add this, and it should
>>>>>>>>>>>>> work for you:
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> exten => oe_custcare,5,Hangup
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Let me know if that works :)
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Aaron
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> P.S. It's the same on both servers, just the
>>>>> server names are
>>>>>>>>>>>>>>> switched.
>>>>>>>>>>>>>>> Either server can be the primary.  If you want it in
>>>>>>>>>>>>> extensions.conf
>>>>>>>>>>>>>>> language, let me know.
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> On Fri, 24 Mar 2006, Douglas Garstang wrote:
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Aaron,
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> That's not what I'm seeing. I'd like to know how your
>>>>>>>>> doing it.
>>>>>>>>>>>>>>>> Here's what the calling system has:
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> exten => 2944000,1,Dial(SIP/2944030,15,tr)
>>>>>>>>>>>>>>>> exten => 2944000,2,Answer
>>>>>>>>>>>>>>>> exten => 2944000,3,Wait,1
>>>>>>>>>>>>>>>> exten => 2944000,4,Playback(thank-you-for-calling)
>>>>>>>>>>>>>>>> exten => 2944000,5,Playback(customer-service)
>>>>>>>>>>>>>>>> exten =>
>>>>>>>>>>>>>>>
>>>>> 2944000,6,Macro(DialIAX,acdserver1,oe_custcare,oneeighty_acd)
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> and on the callee system(acd box) I have:
>>>>>>>>>>>>>>>> exten => oe_custcare,1,Answer
>>>>>>>>>>>>>>>> exten => oe_custcare,2,Queue(oe_custcare||||120)
>>>>>>>>>>>>>>>> exten => oe_custcare,3,NoOP(QUEUE DONE)
>>>>>>>>>>>>>>>> exten =>
>>>>>>>>>>> oe_custcare,4,Macro(DialIAX,vmserver1,2944002,vmdeposit)
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> and here's the Macro on the calling system:
>>>>>>>>>>>>>>>> exten => s,1,Dial(IAX2/pbxuser@${ARG1}/${ARG2}@${ARG3})
>>>>>>>>>>>>>>>> exten => s,2,Goto(s-${DIALSTATUS},1)
>>>>>>>>>>>>>>>> exten => s-ANSWER,1,Goto(s-OK,1)
>>>>>>>>>>>>>>>> exten => s-NOANSWER,1,Goto(s-ERROR,1)
>>>>>>>>>>>>>>>> exten => s-CONGESTION,1,Goto(s-ERROR,1)
>>>>>>>>>>>>>>>> exten => s-CHANUNAVAIL,1,Goto(s-ERROR,1)
>>>>>>>>>>>>>>>> exten => s-ERROR,1,Answer()
>>>>>>>>>>>>>>>> exten => s-ERROR,2,Wait,1
>>>>>>>>>>>>>>>> exten => s-ERROR,3,Set(i=1)
>>>>>>>>>>>>>>>> exten => s-ERROR,4,While($[${i} < 4])
>>>>>>>>>>>>>>>> exten =>
>>>> s-ERROR,5,Playback(cannot-complete-network-error)
>>>>>>>>>>>>>>>> exten => s-ERROR,6,Playback(message-number)
>>>>>>>>>>>>>>>> exten => s-ERROR,7,Playback(letters/o)
>>>>>>>>>>>>>>>> exten => s-ERROR,8,Playback(letters/e)
>>>>>>>>>>>>>>>> exten => s-ERROR,9,Playback(digits/9)
>>>>>>>>>>>>>>>> exten => s-ERROR,10,Playback(digits/0)
>>>>>>>>>>>>>>>> exten => s-ERROR,11,Playback(digits/0)
>>>>>>>>>>>>>>>> exten => s-ERROR,12,Set(i=$[${i} + 1])
>>>>>>>>>>>>>>>> exten => s-ERROR,13,EndWhile
>>>>>>>>>>>>>>>> exten => s-ERROR,14,Hangup()
>>>>>>>>>>>>>>>> exten => s-OK,1,MacroExit
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> The callee system executes the NoOP(QUEUE
>> DONE) when the
>>>>>>>>>>>>>>> queue times out, but does not return control to
>>>> the calling
>>>>>>>>>>>>>>> system. I have to dial the VM server from the ACD box. I
>>>>>>>>>>>>>>> don't understand how that could work anyways.
>> Once you've
>>>>>>>>>>>>>>> transferred the call, you've transferred it.
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> What voodoo are you using?
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Doug.
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> --
>>>>>>>>>>>>>>> Aaron Daniel
>>>>>>>>>>>>>>> Computer Systems Technician
>>>>>>>>>>>>>>> Sam Houston State University
>>>>>>>>>>>>>>> amdtech at shsu.edu
>>>>>>>>>>>>>>> (936) 294-4198
>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>> --Bandwidth and Colocation provided by Easynews.com --
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Asterisk-Users mailing list
>>>>>>>>>>>>>>> To UNSUBSCRIBE or update options visit:
>>>>>>>>>>>>>>>
>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>> --Bandwidth and Colocation provided by Easynews.com --
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Asterisk-Users mailing list
>>>>>>>>>>>>>> To UNSUBSCRIBE or update options visit:
>>>>>>>>>>>>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> --
>>>>>>>>>>>>> Aaron Daniel
>>>>>>>>>>>>> Computer Systems Technician
>>>>>>>>>>>>> Sam Houston State University
>>>>>>>>>>>>> amdtech at shsu.edu
>>>>>>>>>>>>> (936) 294-4198
>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>> --Bandwidth and Colocation provided by Easynews.com --
>>>>>>>>>>>>>
>>>>>>>>>>>>> Asterisk-Users mailing list
>>>>>>>>>>>>> To UNSUBSCRIBE or update options visit:
>>>>>>>>>>>>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>>>>>>>>>
>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>> --Bandwidth and Colocation provided by Easynews.com --
>>>>>>>>>>>>
>>>>>>>>>>>> Asterisk-Users mailing list
>>>>>>>>>>>> To UNSUBSCRIBE or update options visit:
>>>>>>>>>>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> --
>>>>>>>>>>> Aaron Daniel
>>>>>>>>>>> Computer Systems Technician
>>>>>>>>>>> Sam Houston State University
>>>>>>>>>>> amdtech at shsu.edu
>>>>>>>>>>> (936) 294-4198
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> --Bandwidth and Colocation provided by Easynews.com --
>>>>>>>>>>>
>>>>>>>>>>> Asterisk-Users mailing list
>>>>>>>>>>> To UNSUBSCRIBE or update options visit:
>>>>>>>>>>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> --Bandwidth and Colocation provided by Easynews.com --
>>>>>>>>>>
>>>>>>>>>> Asterisk-Users mailing list
>>>>>>>>>> To UNSUBSCRIBE or update options visit:
>>>>>>>>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>> --
>>>>>>>>> Aaron Daniel
>>>>>>>>> Computer Systems Technician
>>>>>>>>> Sam Houston State University
>>>>>>>>> amdtech at shsu.edu
>>>>>>>>> (936) 294-4198
>>>>>>>>> _______________________________________________
>>>>>>>>> --Bandwidth and Colocation provided by Easynews.com --
>>>>>>>>>
>>>>>>>>> Asterisk-Users mailing list
>>>>>>>>> To UNSUBSCRIBE or update options visit:
>>>>>>>>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> --Bandwidth and Colocation provided by Easynews.com --
>>>>>>>>
>>>>>>>> Asterisk-Users mailing list
>>>>>>>> To UNSUBSCRIBE or update options visit:
>>>>>>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Aaron Daniel
>>>>>>> Computer Systems Technician
>>>>>>> Sam Houston State University
>>>>>>> amdtech at shsu.edu
>>>>>>> (936) 294-4198
>>>>>>> _______________________________________________
>>>>>>> --Bandwidth and Colocation provided by Easynews.com --
>>>>>>>
>>>>>>> Asterisk-Users mailing list
>>>>>>> To UNSUBSCRIBE or update options visit:
>>>>>>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>>>
>>>>>> _______________________________________________
>>>>>> --Bandwidth and Colocation provided by Easynews.com --
>>>>>>
>>>>>> Asterisk-Users mailing list
>>>>>> To UNSUBSCRIBE or update options visit:
>>>>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>>
>>>>>
>>>>> --
>>>>> Aaron Daniel
>>>>> Computer Systems Technician
>>>>> Sam Houston State University
>>>>> amdtech at shsu.edu
>>>>> (936) 294-4198
>>>>> _______________________________________________
>>>>> --Bandwidth and Colocation provided by Easynews.com --
>>>>>
>>>>> Asterisk-Users mailing list
>>>>> To UNSUBSCRIBE or update options visit:
>>>>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>
>>>>
>>> _______________________________________________
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>>
>> --
>> Aaron Daniel
>> Computer Systems Technician
>> Sam Houston State University
>> amdtech at shsu.edu
>> (936) 294-4198
>> _______________________________________________
>> --Bandwidth and Colocation provided by Easynews.com --
>>
>> Asterisk-Users mailing list
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

-- 
Aaron Daniel
Computer Systems Technician
Sam Houston State University
amdtech at shsu.edu
(936) 294-4198

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