[asterisk-users] PhpAgi call generation

[Nitesh Divecha]

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
>>>>
>>>>
>>>>
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>>>>         
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>
>
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