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

[Javier Ergas]

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 < <mailto:mailinglists at websitemanagers.com.au>
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>
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

_______________________________________________ 
--Bandwidth and Colocation provided by  <http://Easynews.com> Easynews.com
--

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   <http://lists.digium.com/mailman/listinfo/asterisk-users>
http://lists.digium.com/mailman/listinfo/asterisk-users

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051229/2ddd82fc/attachment.htm