[Asterisk-Users] call to a particular 800 numbernevershowsanswered on Zap channel

Tue Oct 11 12:25:00 MST 2005

Yes, but you don't need the ${EXTEN}:
exten => 18004267378,1,Answer
exten => 18004267378,2,Dial(Zap/g1/18004267378)
exten => 18004267378,3,Congestion

Andy Goss wrote:
>>Watch the output of 'pri debug span 1' on the Asterisk server while
>>placing the call - bug #4468 (http://bugs.digium.com/view.php?id=4468)
>>might be relevant.
> 
> 
> Yes, this is exactly what is happening.  Thanks a lot.  I am thinking about adding a special case for the IBM 800 number since it is the only one my company is complaining about.  Currently I have this in my dialplan:
> 
> ; outgoing calls
> ;
> ; 7 digit
> exten => _NXXXXXX,1,Dial(Zap/g1/${EXTEN})
> exten => _NXXXXXX,2,Congestion
> ; 10 digit
> exten => _NXXNXXXXXX,1,Dial(Zap/g1/1${EXTEN})
> exten => _NXXNXXXXXX,2,Congestion
> ; 11 digit
> exten => _1NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN})
> exten => _1NXXNXXXXXX,2,Congestion
> 
> If I wanted to add a special case for IBM, with the Answer before the dial as suggested for an ugly fix, would I need to add this before the 10 and 11 digit patterns or does it matter?  Would adding this work:
> 
> exten => 18004267378,1,Answer()
> exten => 18004267378,2,Dial(Zap/g1/${EXTEN})
> exten => 18004267378,3,Congestion
> 
> Thanks,
> Andy  
> 
> 
> 
> 
>>>-----Original Message-----
>>>From: asterisk-users-bounces at lists.digium.com
>>>[mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of Andy Goss
>>>Sent: Monday, October 10, 2005 5:58 AM
>>>To: Asterisk Users Mailing List - Non-Commercial Discussion
>>>Subject: RE: [Asterisk-Users] call to a particular 800 number
>>>nevershowsanswered on Zap channel
>>>
>>>
>>>I am still looking to solve this problem, does anyone have any ideas?
>>>
>>>Thanks,
>>>Andy
>>>
>>>-----Original Message-----
>>>From: Andy Goss
>>>Sent: Friday, October 07, 2005 5:37 PM
>>>To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
>>>Subject: RE: [Asterisk-Users] call to a particular 800 number
>>>never showsanswered on Zap channel
>>>
>>>Thanks for the reply.  Forgive me for being naïve, however
>>>have jumped in to this asterisk project at work due to some
>>>circumstances beyond my control and I don't know a lot about
>>>carriers and how this all works.  I am figuring it out, but
>>>it's a lot of trial by fire.
>>>
>>>As far as I know, we only use 1 carrier for our system.  We
>>>have a PRI from NuVox and we use 7 channels for our asterisk
>>>server.  So, I have a few questions:
>>>
>>>Is asterisk or the carrier causing the disconnect?
>>>
>>>Is IBM (the 800 number I am dialing) not passing the answer
>>>supervision or is that a function of the carrier?
>>>
>>>Is there a way to make asterisk not drop the call or to force
>>>the answer on this number?  Seems like a hard-PBX would have
>>>to be able to handle this type of situation.
>>>
>>>Thanks,
>>>Andy
>>>
>>>-----Original Message-----
>>>From: asterisk-users-bounces at lists.digium.com
>>>[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
>>>Garth Summey
>>>Sent: Friday, October 07, 2005 5:18 PM
>>>To: Asterisk Users Mailing List - Non-Commercial Discussion
>>>Subject: Re: [Asterisk-Users] call to a particular 800 number
>>>never showsanswered on Zap channel
>>>
>>>This one drove me crazy for a while too.  I found out that some
>>>companies don't exactly play fair and don't pass answer
>>>supervision on a
>>>call until you are actually speaking with a live person.  The
>>>person I
>>>spoke to about this wasn't sure if that was even legal, but
>>>he said it
>>>happens quite a bit.  I was lucky in that I use multiple carriers
>>>(voipjet and broadvoice), voipjet disconnected the call after 60
>>>seconds, but broadvoice did not, so when I find one of those
>>>800 numbers
>>>I route it through broadvoice.
>>>
>>>Hope that helps,
>>>
>>>G
>>>
>>>Andy Goss wrote:
>>>
>>>>Whenever we call IBM, the call counter on the phone never
>>>
>>>starts and in
>>>
>>>>the CLI the zap channel never gets the answered signal from the PRI.
>>>>See below.
>>>>
>>>>    -- Executing Dial("SIP/5933-645d", "Zap/g1/18004267378") in new
>>>>stack
>>>>    -- Requested transfer capability: 0x00 - SPEECH
>>>>    -- Called g1/18004267378
>>>>
>>>>At this point, I am in IBM's menu system.  However the call never
>>>>indicates that it is answered either on the phone or in the
>>>
>>>CLI.  After
>>>
>>>>60 seconds, the call disconnects.
>>>>
>>>>    -- Hungup 'Zap/1-1'
>>>>  == Spawn extension (main, 18004267378, 1) exited non-zero on
>>>>'SIP/5933-7bff'
>>>>    -- Executing Hangup("SIP/5933-7bff", "") in new stack
>>>>  == Spawn extension (main, h, 1) exited non-zero on 'SIP/5933-7bff'
>>>>
>>
>>{clip}
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