HI Eric, Nikhil,<br><br>Thanks a lot for the responses. Bear with me a little as I'm very new to asterisk.<br><br>I reproduced the problem using standard dialplan. The following are the configuration files:<br><b>chan_dahdi.conf</b><br>
<i>[trunkgroups]<br>[channels]<br>language=en<br>nationalprefix=+91<br>pridialplan=national ; or national or local?<br>
usecallerid=yes<br>hidecallerid=no<br>callwaiting=yes<br>allow_call_waiting_calls=no<br>usecallingpres=yes<br>callwaitingcallerid=yes<br>threewaycalling=yes<br>transfer=yes<br>canpark=yes<br>cancallforward=yes<br>callreturn=yes<br>
echocancel=yes<br>echocancelwhenbridged=no ;Might have to play with this.<br>callerid=asreceived<br>facilityenable=yes<br>priindication=outofband<br><br>cidsignalling=dtmf ; most likely dtmf based on the India link below<br>
cidstart=polarity_IN<br><br>#include dahdi-channels.conf</i><br><br><b>extensions.conf:</b><br><i>[frompstn]<br>exten => xxxxx,1,Ringing<br>exten => xxxxx,n,Dial(Dahdi/G11/yyyyy)<br>exten => xxxxx,n,Hangup()</i><br>
<br>When aaaaa calls xxxxx, we dial yyyyy<br><br>This is what I find in the logs:<br> -- Accepting call from 'aaaaa' to 'xxxxx on channel 0/24, span 1<br> -- Executing [xxxxx@frompstn:1] Ringing("DAHDI/i1/aaaaa-136", "") in new stack<br>
-- Executing [xxxxx@frompstn:2] Dial("DAHDI/i1/aaaaaa-136", "Dahdi/G11/yyyyyy") in new stack<br> -- Requested transfer capability: 0x10 - 3K1AUDIO<br> -- Called G11/yyyyyy<br> -- DAHDI/i1/yyyyy-137 is proceeding passing it to DAHDI/i1/aaaaa-136<br>
-- DAHDI/i1/yyyyy-137 is ringing<br># At this point, yyyyy rejected the call. Asterisk doesn't recognise this, and continues to dial for 30s(the default) before hanging up.<br> -- DAHDI/i1/yyyyy-137 is making progress passing it to DAHDI/i1/aaaaa-136<br>
-- Nobody picked up in 30000 ms<br><br><br>I'll try out "pri intense debug" during night time when the traffic on our servers is low, and update the list with the logs.<br><br>In the mean time, is there anything missing in the configuration that rejected calls aren't detected?<br>
<br>--<br>Thanks,<br>Ishwar.<br><br><br><div class="gmail_quote">On Fri, Jul 29, 2011 at 1:10 AM, Eric Wieling <span dir="ltr"><<a href="mailto:EWieling@nyigc.com" target="_blank">EWieling@nyigc.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
1) You have to have channels configured for your PRI SOMEWHERE in the Asterisk DAHDI configs.<br>
2) Can't troubleshoot when everything important is masked by an AGI script. Reproduce the problem using standard dialplan stuff.<br>
<div><br>
> -----Original Message-----<br>
> From: <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-" target="_blank">asterisk-users-</a><br>
</div>> <a href="mailto:bounces@lists.digium.com" target="_blank">bounces@lists.digium.com</a>] On Behalf Of Ishwar Sridharan<br>
> Sent: Thursday, July 28, 2011 2:52 PM<br>
> To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
<div><div></div><div>> Subject: Re: [asterisk-users] Capturing call Reject/Decline events on a PRI<br>
> line<br>
><br>
> 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,<br>
> 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<br>
> over to adhearsion. Does that affect how events are handled in asterisk?<br>
><br>
> --<br>
> Thanks,<br>
> Ishwar.<br>
><br>
><br>
><br>
> On Thu, Jul 28, 2011 at 6:37 PM, Eric Wieling <<a href="mailto:EWieling@nyigc.com" target="_blank">EWieling@nyigc.com</a>> wrote:<br>
><br>
><br>
><br>
><br>
> > -----Original Message-----<br>
> > From: <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-" target="_blank">asterisk-</a><br>
> users-<br>
> > <a href="mailto:bounces@lists.digium.com" target="_blank">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" target="_blank">asterisk-users@lists.digium.com</a><br>
> > Subject: Re: [asterisk-users] Capturing call Reject/Decline events<br>
> on a PRI<br>
> > line<br>
><br>
> ><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<br>
> recognise<br>
> > the event when the called party declines/cuts the call. This<br>
> happens<br>
> > specifically over calls on a PRI line. For calls over SIP, call decline<br>
> 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<br>
> 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<br>
> to<br>
> > SIP/xxxxx-00000001<br>
> > -- Started music on hold, class 'default', on SIP/xxxxx-<br>
> 00000001<br>
> > -- SIP/xxxxx-00000000 requested special control 20, passing it<br>
> 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>
><br>
> > <<a href="http://127.0.0.1:5063/" target="_blank">http://127.0.0.1:5063/</a>><br>
><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<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<br>
> DAHDI/i1/yyyyy<br>
> > -- DAHDI/i1/xxxxx-18f8 is ringing<br>
> > # At this point in time, xxxxx rejects the call. The event that's<br>
> 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<br>
> decline",<br>
> > and does it need any configuration changes to enable this?<br>
> ><br>
> > Thanks for your help.<br>
><br>
><br>
><br>
> Try adding the following before your PRI channel => lines in your<br>
> chan_dahdi.conf. If you are using a GUI like FreePBX, you will have place<br>
> the info where you need to for FreePBX.<br>
><br>
> facilityenable=yes<br>
> priindication=outofband<br>
><br>
><br>
><br>
><br>
> --<br>
> ____________________________________________________________<br>
> _________<br>
> -- Bandwidth and Colocation Provided by <a href="http://www.api-" target="_blank">http://www.api-</a><br>
> <a href="http://digital.com" target="_blank">digital.com</a> --<br>
> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
><br>
> asterisk-users mailing list<br>
> To UNSUBSCRIBE or update options visit:<br>
> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
><br>
><br>
<br>
<br>
--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</div></div></blockquote></div><br>