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>