Thanks Nasir, That helped alot...
Cheers, Nitesh Nasir Iqbal wrote: > 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