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