We are currently running Asterisk 1.6.2.11 and FreePBX 2.9.0.12 on our 2 servers. We have extensions on both, a SIP trunk between the 2 servers where users can call extensions between the servers, and conference number on each server that work fine among the extensions for that particular server.
The question is how are the conferences configured so that users on one side can call into and access the conference numbers on the other side, and vice a versa? I tried adding the conference numbers to the trunk, that didn’t work. Somewhat confused how a conference number is accessed as a common resource within the global configuration of FreePBX servers.
I’ve looked at various sites, but can’t seem to find an example or an explanation. Help or tip would be great!
Thanks
Most users map a DID to each conference room.
If you don’t have enough DID’s you can create an IVR.
If you are doing heavy conferencing the WebMeetme package might be a big benefit for you. You can find it over at Sourceforge
Swimming deeper in the FreePBX pond!
I’ll research DIDs and IVRs, still a rookie, doing this research as our network radios will soon be SIP capable. This is research, so there is no performance issues just concepts and use cases, but thanks for the tip On Web MeetMe. I’ll read about these features and hopefully be able to ask some questions, or heck get it working!
Thanks for the “instant” help; folks on this forum are really great!
Did much research over the week-end to no avail today. Some questions…
- Can the DID be a 4 digit extension? This is just lab testing.
- Once you create the DID number and you set the destination to the confernce is that all that is needed?
I’m still dead in the water on this one, it seems like lots of folks would do this scenario but can’t find any examples.
Andvice?