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 <jar...@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, <jar...@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 <brucev...@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. > > _______________________________________________ > > -- 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 >
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