asterisk users - Oct 2005 - call to a particular 800 numbernevershowsanswered on Zap channel

> 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@lists.digium.com
> > [mailto:asterisk-users-bounces@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 na?ve, 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@lists.digium.com
> > [mailto:asterisk-users-bounces@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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> 
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> Asterisk-Users@lists.digium.com
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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@lists.digium.com
>>>[mailto:asterisk-users-bounces@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 na?ve, 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@lists.digium.com
>>>[mailto:asterisk-users-bounces@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}
>>_______________________________________________
>>--Bandwidth and Colocation sponsored by Easynews.com --
>>
>>Asterisk-Users mailing list
>>Asterisk-Users@lists.digium.com
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> _______________________________________________
> --Bandwidth and Colocation sponsored by Easynews.com --
> 
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> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
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