Force SIP REJECT from bridged Asterisk


We have two Asterisk installations bridged via SIP (Locally v13, remotely v1.8 ). Basic calling between the two works great, but what we’re trying to do is send queue calls to the other server but to have the remote server reject the SIP INVITE if no one is available to take a call so our server can just try the next agent in our queue. Is there any way in the Asterisk dialplan to do this? From what I can see the remote system seems to accept the SIP call, sending the OK message, before running through the dialplan. So a Hangup or Busy call doesn’t do what we need because our system already thinks the call was successfully handed off and removes it from the queue. Any help or suggestions would be appreciated.


There is no need to answer the call to run the dialplan, although some things will not work well unless you do. Queue will certainly work without answering.

The above is true from at least version 1.4. I have no experience of anything earlier.