Hi Eric,<br><br>There weren't any lines with "PRI channel =>" in the chan_dahdi.conf<br><br>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.<br>
<br>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?<br><br>--<br>Thanks,<br>Ishwar.<br><br><br><div class="gmail_quote">
On Thu, Jul 28, 2011 at 6:37 PM, Eric Wieling <span dir="ltr"><<a href="mailto:EWieling@nyigc.com">EWieling@nyigc.com</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;">
<br>
<br>
> -----Original Message-----<br>
> From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-">asterisk-users-</a><br>
> <a href="mailto:bounces@lists.digium.com">bounces@lists.digium.com</a>] On Behalf Of Nikhil<br>
> Sent: Thursday, July 28, 2011 9:03 AM<br>
> To: <a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br>
> Subject: Re: [asterisk-users] Capturing call Reject/Decline events on a PRI<br>
> line<br>
<div class="im">><br>
> Can you share the dialplan ,where SIP call is dialing...<br>
> Thanks<br>
> Nikhil<br>
><br>
> On 07/28/2011 06:15 PM, Ishwar Sridharan wrote:<br>
><br>
> 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<br>
> the event when the called party declines/cuts the call. This happens<br>
> specifically over calls on a PRI line. For calls over SIP, call decline event is<br>
> captured properly.<br>
><br>
> I wasn't able to find a solution on the asterisk-users mailing list<br>
> archive. Any suggestions/help would be much appreiciated :) I can share the<br>
> 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<br>
> 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<br>
> 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>
</div>> <<a href="http://127.0.0.1:5063/" target="_blank">http://127.0.0.1:5063/</a>><br>
<div class="im">> -- 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<br>
> 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<br>
> in asterisk is the following:<br>
> -- DAHDI/i1/xxxxx-18f8 is making progress passing it to<br>
> 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",<br>
> and does it need any configuration changes to enable this?<br>
><br>
> Thanks for your help.<br>
<br>
<br>
</div>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.<br>
<br>
facilityenable=yes<br>
priindication=outofband<br>
<div><div></div><div class="h5"><br>
<br>
<br>
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</div></div></blockquote></div><br>