[asterisk-users] Capturing call Reject/Decline events on a PRI line

[Ishwar Sridharan]

Hi Eric,

There weren't any lines with "PRI channel =>" in the chan_dahdi.conf

However, I added the lines you'd mentioned, near the top of the file. Still,
no difference in either the behaviour or the asterisk output.

Please note that as soon as the call lands on asterisk, we pass the control
over to adhearsion. Does that affect how events are handled in asterisk?

--
Thanks,
Ishwar.


On Thu, Jul 28, 2011 at 6:37 PM, Eric Wieling <EWieling at nyigc.com> wrote:

>
>
> > -----Original Message-----
> > From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> > bounces at lists.digium.com] On Behalf Of Nikhil
> > Sent: Thursday, July 28, 2011 9:03 AM
> > To: asterisk-users at lists.digium.com
> > Subject: Re: [asterisk-users] Capturing call Reject/Decline events on a
> PRI
> > line
> >
> > Can you share the dialplan ,where SIP call is dialing...
> > Thanks
> > Nikhil
> >
> > On 07/28/2011 06:15 PM, Ishwar Sridharan wrote:
> >
> >       Hello everybody,
> >
> >       We have an asterisk 1.8.4.1 setup, connected to a PRI line.
> >
> >       We're currently facing an issue where asterisk does not recognise
> > the event when the called party declines/cuts the call. This happens
> > specifically over calls on a PRI line. For calls over SIP, call decline
> event is
> > captured properly.
> >
> >       I wasn't able to find a solution on the asterisk-users mailing list
> > archive. Any suggestions/help would be much appreiciated :) I can share
> the
> > relevant parts of the configuration files, if needed.
> >
> >       Here's an excerpt from asterisk logs for a SIP call.
> >           -- SIP/xxxxx-00000000 requested special control 16, passing it
> to
> > SIP/xxxxx-00000001
> >           -- Started music on hold, class 'default', on
> SIP/xxxxx-00000001
> >           -- SIP/xxxxx-00000000 requested special control 20, passing it
> to
> > SIP/xxxxx-00000001
> >           -- Got SIP response 603 "Decline" back from 127.0.0.1:5063
> > <http://127.0.0.1:5063/>
> >           -- SIP/xxxxx-00000001 is busy
> >           -- Stopped music on hold on SIP/xxxxx-00000001
> >
> >       As you can see, on a SIP call, a call reject event is identified.
> >
> >       For a call over the PRI, on the other hand, this event is not
> > recognised. Here's an excerpt from asterisk log for a call over PRI.
> >       Call from yyyy to xxxx.
> >           -- Requested transfer capability: 0x10 - 3K1AUDIO
> >           -- Called G11/xxxxx
> >           -- Started music on hold, class 'default', on DAHDI/i1/yyyyy
> >           -- DAHDI/i1/xxxxx-18f8 is proceeding passing it to
> DAHDI/i1/yyyyy
> >           -- DAHDI/i1/xxxxx-18f8 is ringing
> >       # At this point in time, xxxxx rejects the call. The event that's
> logged
> > in asterisk is the following:
> >           -- DAHDI/i1/xxxxx-18f8 is making progress passing it to
> > DAHDI/i1/yyyyy
> >       # And the call times out after the default 30s.
> >           -- Nobody picked up in 30000 ms
> >
> >       Is there a reason why asterisk doesn't recognise the "call
> decline",
> > and does it need any configuration changes to enable this?
> >
> >       Thanks for your help.
>
>
> Try adding the following before your PRI channel => lines in your
> chan_dahdi.conf.  If you are using a GUI like FreePBX, you will have place
> the info where you need to for FreePBX.
>
> facilityenable=yes
> priindication=outofband
>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110729/572e3bf5/attachment.htm>