Incredible PBX 4.11.4 for RasPBX on a Beagle Bone Black
Asterisk: 11.13.
FreePBX: 2.11.0.34
I am setting up a SPA3102 ATA to connect to my PSTN line. I am replicating the process I used on two other SPA3102. Both several years ago.
I am having problems with the GUI in setting up the trunk and out bound routes. For example I create a route, use the dial pattern wizard (Lookup local prefixes) for my exchange, add trunks and click submit. This deletes all the trunks. I noticed that there were more patterns than my previous setups. Deleting patterns eventually allow me to save the trunks. I think using 50586 generates ~300 patterns. Deleting half seems to fix it.
I looked in extensions_additional.conf and see the dial patterns.
I tried using the wizard in the trunk deletes the trunk name and details.
Jens,
I also have a SPA3102 in production connecting PSTN line to raspbx on raspberry pi
Why not keep it simple. I have the end user just enter 10digits
Local, long distance, and 800 = 10digits.
Dial plan like
Nxxnxxxxxx
911
411
611
I’ve had pstn providers that required 1 for long distance. Instead of struggling with dial plan creation to accommodate local vs long distance dialing, i called the provider and had them change to allow 10digit for everything
The issue does not really impact me. I can receive calls from the ATA. I use Google Voice for the out-bound calls. I wanted to use the PSTN for a secondary link. Your solution would almost work if it were not for my wife factor. I have the GV trunk handle all calls like the PSTN does. This is easy since all I do is add the 1505 to the NXXXXXXX. My solution was to add a CallWithUs trunk and use it as a secondary trunk.
In the past it was that hard to do what I wanted.to do. Looking at the asterisk configuration files, it appears that the trunk information is lost. I was hoping that someone would say they see the same thing.
I use the PSTN for 911 and information using two separate routes.
Update
When I searched the FreePBX site, I apparently did not check the bug reports. This is an issue that is scheduled to be address in version 12.
My long term plans are to port the number to a VoIP Provider. In the interim, I may look into figuring out how add a custom route that does not get overwritten.