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

[Joe Pukepail]

I am using T1 Signaling and seeing the same problems (I think), so I don't
think its just E1.

On 12/29/05, Javier Ergas <jergas at gmx.net> wrote:
>
>  I have tried both inband and outofband too unsuccessfully. I think the
> priindication parameter says how Asterisk reports Busy and Congestion to the
> PSTN, not the other way around.
>
> In the Asterisk config sirrix.conf (
> http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sirrix.conf)
> there is a providetones parameter, witch I think handles the way that
> interface receives the signalization from the PSTN, but I think it won't
> work for zaptel/Zapata.
>
>
>
> Today I tried Asterisk 1.2 in another Telco and I experienced the same
> behavior. I'm starting to think this is a bug in the Asterisk E1
> signalization.
>  ------------------------------
>
> *De:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *En nombre de *Joe Pukepail
> *Enviado el:* Jueves, 29 de Diciembre de 2005 15:22
> *Para:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Asunto:* Re: [Asterisk-Users] PRI: This number has been disconnected
>
>
>
> I have tried both inband and outofband, doesn't seem to make a
> difference.  I added the congension and playtones(congestion) to the dial
> plan after the dial, but the users just get a busy instead of "Do-De-Dah The
> number of have reached is not in service <fastbusy>". PRI Debug below.
>
>  …
>
> …
>
>  On 12/29/05, *Adam Goryachev* <mailinglists at websitemanagers.com.au>
> wrote:
>
> On Wed, 2005-12-28 at 14:00 -0300, Javier Ergas 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?
> >
>
> Sounds like the difference between doing inband signalling or out of
> band signalling. I think by default, a PRI uses out of band signalling,
> ie, it just sends a message saying "this number if un reachable" so
> asterisk just hangs up and plays the local congestion dialplan.
>
> What you need to do is use inband signalling, so that asterisk won't
> hangup, and instead will pass the audio from the telco through.
>
> See /etc/asterisk/zapata.conf:
> ; PRI Out of band indications.
> ; Enable this to report Busy and Congestion on a PRI using out-of-band
> ; notification. Inband indication, as used by Asterisk doesn't seem to
> work
> ; outofband:      Signal Busy/Congestion out of band with
> RELEASE/DISCONNECT
> ; inband:         Signal Busy/Congestion using in-band tones
> priindication = outofband
>
>
> Regards,
> Adam
>
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