asterisk users - Oct 2005 - 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'
> 
> Does anyone have any ideas?
> 
> Thanks,
> Andy
> 
> --
> H. Andy Goss
> Network Engineer
> Network Advocates Inc.
> Main: 502.412.1050
> DID: 502.992.5933
> Mobile: 502.387.8216
> agoss@ntad.com
> _______________________________________________
> --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 --

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

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'
> 
> Does anyone have any ideas?
> 
> Thanks,
> Andy
> 
> --
> H. Andy Goss
> Network Engineer
> Network Advocates Inc.
> Main: 502.412.1050
> DID: 502.992.5933
> Mobile: 502.387.8216
> agoss@ntad.com
> _______________________________________________
> --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 --

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

> First, there is nothing "unfair" or "illegal" going on.
Large
toll-free
> users have enough clout that they can negotiate contracts, where they
> are not billed during the service selection phase of a call.  For
example,
> when you call American Airlines, billing doesn't start until an agent
> answers, or the caller selects "automated flight information" or
a
similar
> IVR service.  Answer supervision is used to tell the carrier when to
start
> billing.  This system is quite common and used by hundreds of
companies.

This makes good sense, thanks for clearing it up.
> 
> With Asterisk, three things might go wrong:
> 
> You may have two-way communication with the IVR, but the call gets
> disconnected before answer supervision is received.   Find out if it's
> your carrier or Asterisk that is timing out.  If the latter, just put
> a longer timeout in your Dial statement; 180 seconds should be enough.
This is the situation I am in.  If I am really fast, I can navigate the
menu system in enough time to be transferred to a real person, or at
least the real-person queue and I get the answer supervision message.

Is there a way I can tell if it is asterisk or the carrier that is
timing out from the CLI?  I thought if the timeout was not specified in
the Dial statement it was unlimited, but perhaps I am looking in the
wrong place.

Also, is there a way to force the phone to start the call counter or
force the answer on the asterisk-side.  

Thanks,
Andy

> Is there a way I can tell if it is asterisk or the carrier that is
> timing out from the CLI?
Sorry, I don't have PRI and don't know the details.
However, I'm sure that if you set a high enough verbose or debug
level, you'll see the ISDN messages between * and the carrier's
switch.  I don't know which terminology will be used, but you
should see * send an IAM (perhaps called Initial Address Message
or Setup) and the switch reply with ACM (perhaps Address Complete
Message or Call Proceeding).  Then, about 60 seconds later,
you'll see REL (Release).  Who sends it?  If it's your carrier,
ask them why it comes so soon.  If it's *, perhaps your SIP
phone is the culprit.  If it's not obvious from its config,
set up a local extension that doesn't time out to voice mail,
call it from your SIP phone and see if it will ring for more
than a minute.  If neither your carrier nor your phone is
timing out, then I guess it must be * but I don't know where
that might be.  Perhaps some * guru can help.
> Also, is there a way to force the phone to start the call
> counter or force the answer on the asterisk-side.
I would guess that if you called Answer() before Dial(),
then the call counter would start.  However, it would
also start on busy signals, rejected calls, etc.  Sorry,
I don't know if there is a way to have it start only
when call progress is received.

--Stewart