[asterisk-users] PhpAgi call generation

[Nasir Iqbal]

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
> >>
> >>
> >>
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