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 href="http://127.0.0.1">127.0.0.1</a>)<br><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 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;"><u></u>
<div text="#000000" bgcolor="#ffffff">
Can you share the dialplan ,where SIP call is dialing...<br>
Thanks<br>
Nikhil<div><div></div><div class="h5"><br>
<br>
On 07/28/2011 06:15 PM, Ishwar Sridharan wrote:
</div></div><blockquote type="cite"><div><div></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 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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