There is no event for Asterisk to recognize. The PROGRESS message just
says that there is an audio message available for the caller to listen
to. Asterisk just passes the indication to the peer channel and opens
the audio path. It is the caller who must recognize any audio message
that their
Richard,
I tried calling the same number outside of Asterisk, by making direct calls
from a landline telephone and a mobile phones. When the user rejected the
call, the call was immediately cancelled.
This implies that for whatever reason, the call reject signal is not
available for asterisk to
Richard,
Thanks for the explanation. You were right about the lack of signalling to
indicate that the call has been rejected, One particular service provider,
instead of signalling rightaway that the call has been rejected, gives a
voice message saying 'The user is busy. Please call later.', and
Hi Richard,
There is no event for Asterisk to recognize. The PROGRESS message just
says that there is an audio message available for the caller to listen
to. Asterisk just passes the indication to the peer channel and opens
the audio path. It is the caller who must recognize any audio
find the inline comment...
On 07/29/2011 12:11 AM, Ishwar Sridharan wrote:
The dialplan is very simple. When the call comes in, we hand the call
over to adhearsion.
This is how the dialplan looks:
;group 0 will be used for incoming calls
EXOIN = DAHDI/g0
;group 11 for outgoing
EXOOUT =
Try to Add h extensions in frompstn context and print ${HANGUPCAUSE} in that
you will receive in that ,
also read this for better implementation.
http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause
regards
Dhaval
On Fri, Jul 29, 2011 at 11:58 AM, Nikhil d.nik...@cem-solutions.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Capturing call Reject/Decline events on a
PRI
line
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
call Reject/Decline events on a PRI
line
-- Called G11/yy
-- DAHDI/i1/y-137 is proceeding passing it to DAHDI/i1/a-136
-- DAHDI/i1/y-137 is ringing
# At this point, y rejected the call. Asterisk doesn't recognise this, and
continues to dial for 30s(the default
: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Ishwar Sridharan
Sent: Friday, July 29, 2011 9:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Capturing call Reject/Decline events on a
PRI
] On Behalf Of Ishwar Sridharan
Sent: Friday, July 29, 2011 9:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Capturing call Reject/Decline events on a
PRI
line
-- Called G11/yy
-- DAHDI/i1/y-137 is proceeding passing it to DAHDI
We enable pri intense debug with the standard asterisk PRI dialplan,
collected the logs and you can find the logs attached to the mail.
After the call was made, the called party cut the call, and asterisk
doesn't seem to recognise the event.
I can't make much sense of the logs given my
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
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
On Thursday 28 Jul 2011, Ishwar Sridharan wrote:
Is there a reason why asterisk doesn't recognise the call decline, and
does it need any configuration changes to enable this?
What are you seeing for ${HANGUPCAUSE} when this happens ? (Put a line such
as
exten = y, n, NoOp(Hangup cause
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Nikhil
Sent: Thursday, July 28, 2011 9:03 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Capturing call Reject/Decline events
The dialplan is very simple. When the call comes in, we hand the call over
to adhearsion.
This is how the dialplan looks:
;group 0 will be used for incoming calls
EXOIN = DAHDI/g0
;group 11 for outgoing
EXOOUT = DAHDI/G11
;This will be used by adhearsion
EXOCID=
[general]
Hi AJS,
Our dialplan doesn't have a Dial() statement as that's taken care of by
adhearsion.
However, I added exten = y, n, NoOp(Hangup cause was ${HANGUPCAUSE})
at the end of our context, restarted asterisk.
The log doesn't have anything new.
--
Thanks,
Ishwar.
On Thu, Jul 28, 2011 at
] On Behalf Of Nikhil
Sent: Thursday, July 28, 2011 9:03 AM
To: asterisk-users@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
...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Ishwar Sridharan
Sent: Thursday, July 28, 2011 2:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Capturing call Reject/Decline events on a PRI
line
Hi Eric,
There weren't any
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