[Asterisk-Users] Transferring a call with IAX

Douglas Garstang dgarstang at oneeighty.com
Fri Mar 24 16:49:37 MST 2006


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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