Variable passing between servers

Hello all,

Let’s say I have two asterisk servers, A and B.
[ul]

  • The user enters the dialplan in server A.

  • Server A calls server B using Dial() application.

  • Once inside server B, the user is put inside a conference using the MeetMe() application.

  • To exit the conference, the user presses any key corresponding to a valid extension inside the dialplan of server B.

  • Server B hangs up, ending the Dial() command issued from server A.

  • The user then returns to server A and continues dialplan over there.
    [/ul]
    In this situation, the dialplan in server B knows which key was pressed to end the conference.

Here is the question: [color=blue]How is it possible to let server A know about this key? How can I send this information over from A to B in a way that server A knows which key the user pressed to exit the conference?[/color]

We would be glag to hear any suggestions. Thanks in advance for the help.

maybe you can get ideas from this:

http://lists.digium.com/pipermail/asterisk-dev/2006-July/021896.html

Thanks for the reply.

I had bumped into this thread a couple of days back, but as the question is posted, well… has it ever been implemented?

We were looking around to see how values could be passed using SIP headers, but we would like to do that without having to hack asterisk’s code in fear of introducing any unknown bugs into the system. We are keeping this option as a last resource.

We could think of ways to get around, but we can’t seem to find any clues of how to do this elegantly. We thought, as an example, of using one of our FastAGI servers to work as a bridge for passing values between the two servers, but that sounds like constructing a cannon to shoot small fly.

Thanks anyway for the sugestion, we really appreciate it.
Regards.

actually that just might work but both sides will have to know what they are looking for…

on the sending side use SipAddHeader() to insert headers with names that won’t conflict with anything

on the other side use SipGetHeader() or ${SIP_HEADER(headername)}…

let us know how it works!