[asterisk-users] AMI originate and PHP
Bruce Nik
brucevoip at gmail.com
Wed Dec 23 18:15:29 CST 2009
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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