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.
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