Hello, I am working on a click-2-call type system. connecting a local sip extension and a 3rd party call seems relatively straight-forward. I am having some difficulty connecting two 3rd party PSTN numbers together. I am currently using a meetme conference to pull this off.
I am working with a PHP script that connects to the asterick manager with fsockopen, etc.
Here is an example of how I am able to connect a local extension to a 3rd party number:
fputs($socket, “Action: Originate\r\n” );
fputs($socket, “CallerID: Calling… <{$_GET[‘f’]}>\r\n”);
fputs($socket, “Channel: Local/101@from-internal\r\n” );
fputs($socket, “Context: from-internal\r\n” );
fputs($socket, “Variable: _SIPADDHEADER55=Call-Info: answer-after=0\r\n” );
fputs($socket, “Exten: {$_GET[‘f’]}\r\n” );
fputs($socket, “Priority: 1\r\n” );
fputs($socket, “Async: yes\r\n\r\n” );
And I am able to adapt this to connect two third parties buy making two calls to two different numbers with the Channel being a meetme conference extension.
I am interested in either of the following:
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is there a way to connect two third parties directly without using the meetme conference.
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if not, is there a way to boot users out of the meetme conference (built in, by altering some functionality of it, or maybe through agi or asterisk manager, etc). The problem is that If I connect two third parties and one of the third parties does not hangup, then they are sitting around in the conference indefinitely. It could be a real person that is doing that intentionally or more likely it could be a voicemail that does not detect silence and continues to listen indefinitely. Either way, I would like to be able to boot all users from the conference after I am done connecting the two third parties (one of which will always be my cellphone)
Maybe there is a command in the asterisk manager that will reboot certain aspects of asterisk which will in effect close down all of the connections to the conference. I am open to anything. Thanks, Jason