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, <[email protected]> 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 <[email protected]> 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. > _______________________________________________ > -- 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
