Hi Nitesh, you are missing Extension try with
$call = $asm->send_request('Originate', array('Channel'=>"SIP/xo-out/$supervisor_num", 'Context'=>'default', 'Exten'=> your_extensions_here, 'Priority'=>1, 'Callerid'=>$cid)); or you must put an "s" extensions in your desired context in this case it is "default". Regards Nasir Iqbal On Tue, 2007-07-31 at 10:08 -0400, Nitesh Divecha wrote: > Hello All, > > Can anyone help me with this... This is what my program does: - > > 1) At certain time the system generates a ".call" and make a call to User A. > > 2) When User A picks up the phone call, system will play a menu select > option. > a) Press 1 to call your supervisor. > b) Press 2 to call your manager. > c) Press 3 to leave a voice message. > > 3) When the User A press 1 to call his supervisor... The system has to > put the User A on hold and place a call to the supervisor. > > 4) Once the supervisor picks up the call, User A has to be in session > with his supervisor. > > Now I have already got part 1 and 2 done... but I am stuck with part 3 > and 4. > > This is how I generate my call to the supervisor: - > =================================== > if($asm->connect()) > { > $call = $asm->send_request('Originate', > array('Channel'=>"SIP/xo-out/$supervisor_num", > 'Context'=>'default', > 'Priority'=>1, > 'Callerid'=>$cid)); > $asm->disconnect(); > } > > One the *CLI I do see the call, but its failing: - > > AGI Rx << STREAM FILE > /var/spool/asterisk//tmp//text2wav_e08db16aede0af38ebb90a1c69ee19e3 "" 0 > AGI Tx >> 200 result=0 endpos=26224 > == Parsing '/etc/asterisk/manager.conf': Found > == Manager 'phpagi' logged on from 127.0.0.1 > > Channel SIP/xo-out-08f8ae10 was answered. > == Starting SIP/xo-out-08f8ae10 at default,,1 failed so falling back > to exten 's' > == Manager 'phpagi' logged off from 127.0.0.1 > AGI Rx << STREAM FILE goodbye "" 0 > > Can anyone put some light what I am missing here... Why the call is > dropped on both end...? > > Cheers, > Nitesh > > > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users