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I am working on a web dialer interface in order to make phone calls through the browser. I am using sipml5 on the frontend and I have an asterisk/freepbx server for the backend. Everything has been going smoothly so far, but there is still one piece of functionality that I am missing. I cannot seem to bridge together two currently existing calls.

I have been using this library to send AMI commands to asterisk from the webserver.

When I try to use the AMI bridge command, the asterisk logs show my channels as being zombies and drops the call on both lines.

Following is a pastebin of the asterisk verbose/debug logs when i attempt to execute the command.

http://pastebin.com/ErmnT31w

I am an asterisk newb so I may not be thinking of this the right way.

Any suggestions on how to bridge calls via a web dialer using webrtc?

Thanks for your time

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1 Answers

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You have use single channel, one leg call out, other leg to operator

You HAVE NOT use TWO channel.

So there are no any need in bridge if it developed correct way.