Running FreePBX Distro 2.10. I have one at each of 2 offices connected via VPN. I’ve been able to create an IAX2 trunk to connect the two offices. I also created a ring group at office1 that includes extensions at office2 but they won’t ring. Only the extensions at office1 ring.
I’m perplexed because I can call from one to the other using extension numbers.
What you probably want is not a bunch of external extensions in a local ring group, because when the external extension goes to voice mail it will grab the call from the ring group. What you probably want is a remote ring group containing the remote extensions and then ring the remote ring group from the local ring group, again using the # suffix.
I’m not sure I understand the problem or the solution.
I need all incoming calls to ring all extensions. If no one gets it, I need it to roll to voicemail.
Ring Groups have a destination on no answer. I have voice mail on my extension and my extension is in a ring group. When a call comes in I route it to the ring group if there is no answer I send it to a second ring group. If that ring group does not answer the call is sent to a cell phone. All of the extensions in both ring groups have voicemail. If the call is not answered the call does not go to voice mail at all.
Now I’m even more confused. When creating a ring group there is an option at the bottom to change the destination on no answer. I set it to a specific voicemail box and it goes there.
The only weird thing about this is, if I add a remote extension to the ring group, that extension is the voicemail box it lands in on a no answer. Even if I have a different mailbox specified for that ring group.
This is what I was hinting at above. When you add an external number to a ring group (lets say it is your cell number suffixed #) the PBX has zero control of that number (assuming confirm calls is not enabled). If your cell VM answers the call, FreePBX has no way of knowing if it is a human or not, and so transfers the ring group call to the cell number and the ring group is done. The destination if no answer never happens because the call is always answered by your cell vm. Your external PBX extensions work the same way, if an external extension VM answers the ring group, the local PBX will hand off the call and be done with it.
The solution in your case is to have a ring group on the remote PBX with all the remote extensions added, then ring the remote group from the local PBX by adding the remote ring group number with the # to the local group.
For the remote ring group, do I simply create a ring group on my local PBX that includes each remote extension with a # at the end or is there a way to create a ring group that can be dialed by simply dialing the ring group number with a # at the end? I tried that but cannot make it work.
local PBX has 1 sip trunk & 2 extensions 301 & 302. I created a ring group 300 with both of them in it. All incoming calls on that sip trunk go to that ring group.
remote PBX has 2 extensions 601 & 602. I created ring group 600 with both of them in it. It’s simply a mirror image of the local PBX with different extension and ring group numbers. It also has 1 sip trunk with all incoming calls ringing that ring group.
My initial goal was to make all extensions ring simulaneously for all incoming calls and have unanswered calls go to local voicemail. Now I understand the local PBX cannot control any call forwarded to the remote PBX.
So am I missing something?..If not, I have 2 questions:
First. Is there a way to get these two PBX’s to act as one? That may be a lofty goal but I thought since they were already connected via VPN it wouldn’t be that tough.
If the context of your “tie-line” trunk is from-internal, you will have better access to the endpoints on the remote machine by routing the other PBX’ enddpoint numbers to the other side.
This is just how contexts work in asterisk, and from-pstn does not include such things as Ring-Groups or IVR’s . . .
You will never get them to “act as one” because they are “Back-to-Back User Agents”, so at best they will always be two. There are SIP “proxies” out there that will perhaps make it easier to realize what you want but I don’t think Asterisk will do it for you.