Oh, you need Dial application instead of origination.
so no need to AGI Script simply add the dialplan called for ".call" should look like this exten => yourexten,1,BackGround(your_menu_ivr) exten => yourexten,n,WaitExten() exten => 1,1,Dial(SIP/xo-out/$supervisor_num) ;for Supervisor exten => 2,1,Dial(SIP/xo-out/$manager_num) ;for Manager exten => 3,1,Voicemail(your_voice_mail_box) Regards Nasir Iqbal On Tue, 2007-07-31 at 12:21 -0400, Nitesh Divecha wrote: > Thanks Nasir, > > By putting "'Exten'=> your_extensions_here" it will create a new channel > to that extension, correct? > > What I want to do is to join two channels... Join the User A channel > which is active with supervisor. > > Cheers, > Nitesh > > > > Nasir Iqbal wrote: > > 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 > > > > > _______________________________________________ --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