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

[Joe Pukepail]

I am having a similar problem, I thought it was because the PRI card is in
another server that I connect to via IAX from my server, but we are seeing
the same problem, ie getting a hangup instead of unavailable when calling a
number that is not in service.  I'm using T1 and Asterisk 1.21

On 12/28/05, Javier Ergas <jergas at gmx.net> wrote:
>
> 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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