[asterisk-users] AMI originate and PHP

[DHAVAL INDRODIYA]

Hi, Bruce ,

would you remove Async from your php script,
and give it a try

regards
Dhaval
On Thu, Dec 24, 2009 at 5:45 AM, Bruce Nik <brucevoip at gmail.com> wrote:

> Jarrod,
>
> Thanks for the input. Can you please include a sample of your work? It will
> really save me days of headache and tests if I can start with something that
> is tested to work.
>
> I really appreciate your response.
>
> In the meantime, I will go check meetme creation rules.
>
> Regards,
> Bruce
>
>
> On Wed, Dec 23, 2009 at 7:03 PM, Jarrod Lash <jarrod at fed-com.com> wrote:
>
>> Bruce,
>>
>> What I have done for apps like this is call the first guy and at the
>> end of your dialplan put him in a meetme room.  In your script watch
>> for the meetme room to be created in the AMI output.
>>
>> Once the room is created originate a call to the other guy and dump
>> him into that meetme room when he answers.
>>
>>
>> --
>> Jarrod Lash, <jarrod at fed-com.com>
>> Federated Communications, LLC.
>> www.fed-com.com
>> Office: +1-412-357-2127
>> Mobile: +1-412-999-0049
>> Fax: +1-412-545-8368
>>
>>
>> On Wed, Dec 23, 2009 at 6:19 PM, Bruce Nik <brucevoip at gmail.com> wrote:
>> > Hi Guys,
>> > I am trying to make a web form where a person is allowed to put in
>> > $phoneNumber, $dialNumber, and $spoofNumber to make a call with spoof
>> caller
>> > ID. There are a few problems that I am facing with Asterisk AMI
>> Originate
>> > command. The reason why I want to use the darn AMI Originate is because
>> I am
>> > sending calls to mobile phones and I want to have some accountability
>> and to
>> > know if a call was connected for billing purposes or not. Calls go to
>> PSTN
>> > through SIP provider so all signaling is available.
>> > First, if i use AMI Originate to dial both parties with the set CallerID
>> > then, one party may pick up than the other and channel is not bridged at
>> > ringing. So, this can confuse the callee. So, I thought I should send
>> calls
>> > to a context first and then ask customer enter $spoofNumber and then
>> place
>> > call but then I am facing another problem. Using that, the internal
>> context
>> > is called first and all announcements are made and then the
>> > SIP/sipProvider/$phoneNumb is dialed. Or at least it's dialed at the
>> same
>> > time but since it takes time to pick ones phone context already goes
>> over
>> > it's announcement for putting in spoof number and dialnumber. Please
>> guide
>> > me how to do this properly. Following is the code and the context:
>> > $sys_ip = "127.0.0.1";
>> > $User_str = "test";
>> > $Secret_str = "test";
>> > $phoneNumb = "14167777777";
>> > $dialNumb = "14168888888";
>> > $spoofNumb = "1416999999";
>> > $oSocket = fsockopen($sys_ip, 5038, $errnum, $errdesc) or
>> die("Connection to
>> > host failed");
>> > fputs($oSocket, "Action: login\r\n");
>> > fputs($oSocket, "Username: $User_str\r\n");
>> > fputs($oSocket, "Secret: $Secret_str\r\n\r\n");
>> > fputs($oSocket, "Events: off\r\n\r\n");
>> > fputs($oSocket, "Action: originate\r\n");
>> > fputs($oSocket, "Channel: SIP/testTrunk/$phoneNumb\r\n");
>> > fputs($oSocket, "Exten: $dialNumb\r\n");
>> > fputs($oSocket, "Context: testphp\r\n");
>> > fputs($oSocket, "Priority: 1\r\n\r\n");
>> > fputs($oSocket, "Timeout: 10000\r\n");
>> > fputs($oSocket, "CallerId: $spoofNumb\r\n");
>> > fputs($oSocket, "Async: true\r\n");
>> > fputs($oSocket, "Action: Logoff\r\n\r\n");
>> > fclose($oSocket);
>> >
>> > /etc/asterisk/extensions.conf
>> > [testphp]
>> > exten => _X.,1,Answer()
>> > exten =>
>> >
>> _X.,n,Playback(/var/lib/asterisk/sounds/please_enter_dialnumber_and_spoof_callerid)
>> > exten => _X.,n,Read(dialnumber,,10)
>> > exten => _X.,n,Read(spoofnumber,,10)
>> > exten => _X.,n,Playback(connecting_now)
>> > exten => _X.,n,Dial(SIP/testTrunk/$dialNumb)
>> > exten => _X.,n,Hangup()
>> > Thanks a lot.
>> > _______________________________________________
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>
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