Hello all, I’m running Asterisk 1.4 via PiaF on a dozen small PBXs; there are a couple of gateway boxes handling inbound and outbound calls, all using DUNDi for load balancing/failover.
When I upgraded a test PBX from FreePBX 2.4 to 2.5, it started stealing all the inbound calls from the other PBXs. The big change that jumped out at me in extensions.conf was the addition of a ‘from-pstn-e164-us’ context, not sure if that might be responsible.
I’m not getting into too much detail at this point, as I’m planning to wipe the test PBX and try the upgrade again with some more debugging enabled. Just hoping maybe this happened to someone before. Thanks.
I’m at 2.4 right now, there is no “upgrade to 2.8” option available, only “upgrade to 2.5” is there. I was going to step through 2.5, 2.6, 2.7, then 2.8. I don’t want to get into manual installs, that’s why we got PiaF!
The only trunks set up on the PBXs are the DUNDi connections to the gateway PBXs so none of the PBXs talk to carriers. The problem seems to be that the PBX starts answering every DUNDi request that comes it’s way.
I just inherited all this, so sorry if I’m not being clear on anything. I’m learning as I go!
THe DUNDi situation is going to be the same if they are configured to talk to the same boxes.
As far as upgrading goes, you are best to simply step through all the upgrades to get to the highest level that you are trying to achieve, and then look into any issues.
There are many more bugs and issues addressed on the latest release. The only reason there is only the upgrade module to go to the next release is that most people step up one release at a time as they come out.
well the first question that comes to mind, any particular reason you are “upgrading” to a version that is 2 revisions out of support?
If upgrading, you may want to look at going to at least 2.7 if not 2.8…
As far as your issue, it sounds like you may have SIP trunks that register and when you setup your test box, it registered with your carrier which is what tells your carrier where to send the calls. Disable those trunks and then do a sip reload on your production box and that should correct your issue.