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find the inline comment...<br>
<br>
On 07/29/2011 12:11 AM, Ishwar Sridharan wrote:
<blockquote
cite="mid:CA+fQKYwYW+8fSfu8xwHsc4o1HN2sLA=cHr7ztZ1P-ggvZfjU-A@mail.gmail.com"
type="cite">The dialplan is very simple. When the call comes in,
we hand the call over to adhearsion.<br>
This is how the dialplan looks:<br>
<br>
;group 0 will be used for incoming calls<br>
EXOIN = DAHDI/g0<br>
<br>
;group 11 for outgoing<br>
EXOOUT = DAHDI/G11<br>
<br>
;This will be used by adhearsion<br>
EXOCID=xxxxxxxx<br>
<br>
[general]<br>
autofallthrough = yes ;really?<br>
clearglobalvars = no<br>
<br>
[frompstn]<br>
;Send everything to adhearsion<br>
exten => _X.,1,Ringing<br>
exten => _X.,n,AGI(agi://<a moz-do-not-send="true"
href="http://127.0.0.1">127.0.0.1</a>)<br>
</blockquote>
<font color="#ff6666"> exten => _X.,n,Hangup() ; Please try
this.</font><br>
<blockquote
cite="mid:CA+fQKYwYW+8fSfu8xwHsc4o1HN2sLA=cHr7ztZ1P-ggvZfjU-A@mail.gmail.com"
type="cite"><br>
; End dialplan<br>
<br>
The rest of the logic happens in adhearsion.<br>
<br>
--<br>
Thanks,<br>
Ishwar.<br>
<br>
<br>
<div class="gmail_quote">On Thu, Jul 28, 2011 at 6:33 PM, Nikhil <span
dir="ltr"><<a moz-do-not-send="true"
href="mailto:d.nikhil@cem-solutions.net">d.nikhil@cem-solutions.net</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt
0.8ex; border-left: 1px solid rgb(204, 204, 204);
padding-left: 1ex;">
<div text="#000000" bgcolor="#ffffff"> Can you share the
dialplan ,where SIP call is dialing...<br>
Thanks<br>
Nikhil
<div>
<div class="h5"><br>
<br>
On 07/28/2011 06:15 PM, Ishwar Sridharan wrote: </div>
</div>
<blockquote type="cite">
<div>
<div class="h5">Hello everybody,<br>
<br>
We have an asterisk 1.8.4.1 setup, connected to a PRI
line.<br>
<br>
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.<br>
<br>
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.<br>
<br>
Here's an excerpt from asterisk logs for a SIP call.<br>
-- SIP/xxxxx-00000000 requested special control
16, passing it to SIP/xxxxx-00000001<br>
-- Started music on hold, class 'default', on
SIP/xxxxx-00000001<br>
-- SIP/xxxxx-00000000 requested special control
20, passing it to SIP/xxxxx-00000001<br>
-- Got SIP response 603 "Decline" back from <a
moz-do-not-send="true" href="http://127.0.0.1:5063/"
target="_blank">127.0.0.1:5063</a><br>
-- SIP/xxxxx-00000001 is busy<br>
-- Stopped music on hold on SIP/xxxxx-00000001<br>
<br>
As you can see, on a SIP call, a call reject event is
identified.<br>
<br>
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.<br>
Call from yyyy to xxxx.<br>
-- Requested transfer capability: 0x10 - 3K1AUDIO<br>
-- Called G11/xxxxx<br>
-- Started music on hold, class 'default', on
DAHDI/i1/yyyyy<br>
-- DAHDI/i1/xxxxx-18f8 is proceeding passing it to
DAHDI/i1/yyyyy<br>
-- DAHDI/i1/xxxxx-18f8 is ringing<br>
# At this point in time, xxxxx rejects the call. The
event that's logged in asterisk is the following:<br>
-- DAHDI/i1/xxxxx-18f8 is making progress passing
it to DAHDI/i1/yyyyy<br>
# And the call times out after the default 30s.<br>
-- Nobody picked up in 30000 ms<br>
<br>
Is there a reason why asterisk doesn't recognise the
"call decline", and does it need any configuration
changes to enable this?<br>
<br>
Thanks for your help.<br>
<br>
--<br>
Cheers,<br>
<font color="#888888">Ishwar.</font><br>
</div>
</div>
<pre><fieldset></fieldset>
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<pre wrap="">
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