[Asterisk-Users] PRI: This number has been disconnected

[Javier Ergas]

I believe this behavior has nothing to do with the A at H Scripts. I think the
problem is in the PRI signalization.
I can see the zap hangup messages when trying to call a disconnected number.
	.....
    -- Executing Dial("SIP/9349-1787", "ZAP/g0/2514990") in new stack
    -- Called g0/2514990
    -- Channel 0/2, span 1 got hangup
    -- Hungup 'Zap/2-1'
  == No one is available to answer at this time
    -- Executing Goto("SIP/9349-1787", "s-NOANSWER|1") in new stack
    -- Goto (macro-dialout-trunk,s-NOANSWER,1)
	....
The telco says they are sending inband information with the status of the
call, but Asterisk is hanging up the channel instead of connecting it to let
hear the audio message.

There is a post with a similar issue here:
http://mailgate.supereva.com/comp/comp.dcom.isdn.capi/msg04138.html

Is anyone experiencing the same behavior?


-----Mensaje original-----
De: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] En nombre de Francesco
Peeters (Asterisk)
Enviado el: Martes, 27 de Diciembre de 2005 20:09
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [Asterisk-Users] PRI: This number has been disconnected

On Tue, December 27, 2005 23:37, Javier Ergas said:
> Hi,
>
>
>
> I'm running Asterisk at home 1.5 with TE110P E1 PRI in Chile.
>
> When calling an invalid number using, I expect to hear:
>
> "We're sorry you have reached a number which has been disconnected ..."
>
> And that is indeed what I hear when I dial out from [*] using analog FXO,
> or
> VoicePulse or NuPhone.  When I dial that same number trough the T1 / PRI
> interface however, I only hear the allison7/all-circuits-busy-now message.
>
>
>
> There was another issue like this in an old post
> (http://lists.digium.com/pipermail/asterisk-users/2004-April/043597.html)
> but I think it isn't the same.
>
>
<SNIP>

I believe this has to do with the AMP macro's being used in A at H. I am
seeing similar things.

For instance: One issue I have is that when a route has multiple trunks,
and the first trunk after a while returns with 'NOANSWER', it merrily
continues to the next trunk, which is not quite the behavior I'd expect.
Especially as the primary trunk (IAX/VoipBuster) is *much* cheaper (ie
free) as compared to the second trunk (Zap/g1), but the switch is made
without any message. This could mean that you might be talking to someone
on a different trunk, and instead of a free call, be paying normal fees.

This could become expensive if you're calling the USA from Europe!...

I am currently looking in to ways to enhance those macro's to respond more
reliably, as well as return more useful information (busy tone on busy and
no-answer, number disconnected info, etc.) when needed.

If I do get to a satifactory set of macro's, I will put them up on the
Wiki and let the list know... (I'm just starting on doing manual
configuring, so it will be a tough job to crack, but also a learning
experience...)

-- 
F Peeters
  PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch
  2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0
    Cologne HFC-S pins #52, #54, #55 connected in parallel for synching.
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