My system IS working - I just have a problem adding additional Dial Patterns in my outbound route because I already have like 2400 lines listed in there (~!)
My [Stuipid] voice circuit provider INSISTS that I send 10 digits for local calls, and 1+10digits for long distance calls.
When I set up my Freepbx distro server (lastest version with latest updates), I decided to configure so that our users should ALWAYS dial a 1+10, and I imported all 2400 dial patterns which we can dial (MUST) dial without 1+10. and this worked just fine.
The problem that NOW I need to add another Dial Pattern - and in FreePBX gui, only about 1/2 of my Dial Patterns display. I am concerned that if I add the one new dial pattern and hit save - that I MIGHT LOSE the other dial patterns which ARE currently working right but not displayed in freepbx web gui.
Note that when I did the import, the wizard said something like “you have so many, let me switch you to the other view” which is great - and now instead of forms-view - it’s just a white box with many lines of dial patterns - one per line. but I don’t see any way to view more than about the first 1/2 (or maybe it’s even the first 1/4 not quite sure.)
Should I just try and edit the asterisk extension file which actually stores this data? or should I re-import all of the rules again and this time use wild-cards 4232[2-6]XXXXX to make less entries? or other ideas?
– I see some of the other posts where their voice provider switched their accounts so that they can ALWAYS dial 1+10 - but alas - my voice T1 provider won’t do it…
To minimize, I would look at the first few pages of your local white page directory, add the exceptions to long distance office-codes inside your area-code. I am sure they would be less than 2400. Then make the various prioritized outbound routes to suit.
I live in a “local long distance” area with three local area codes that requires 10-digit dialing, and I don’t have nearly that many numbers in my dial pattern list.
There is precedent for your provider requiring “local calls” not use 1 and long distance using 1, so finding a solution is important, .
Is there any way to simplify by area code and NPX? Your example of 4232[2-6]XXXXX is going to become unwieldy quickly. Remember that there is an “advanced setting” for the old-style text-based rule list.
If you still want to do that, I’d recommend writing a program that took the dial patterns in your current data and build a text file that you can use to limit your calling. You should NOT try to directly edit the extensions_additional.conf file - the program will just over-write it. You need to stick with the GUI to avoid weird problems with numbers that are in supposed to be there (and aren’t) or vice versa.
It might be time to think outside the box, though. Depending on your costs, a T1 might NOT be the most cost effective solution for your needs. For example, SIPStation, Alcazar Network and VOIP Innovations both provide SIP connections at very reasonable rates. The long-distance termination and origination rates were about 1/10th the cost of the calls we were placing on our PRIs. The amount of “free calling” we were getting from the PRI (think 23-channel T1) didn’t offset the amount of money we “lost” when we started having to pay for every call.
Thanks for the kind responses.
Earthlink thinks they’re being helpful by extending the free calling area far beyond what AT&T for example would give us - but that’s what is causing the problem really…
I have primarily relied on SIP trunks in the past - and with the SIP trunk providers I’ve used - you just use 1+10digits for all calls - which makes this very simple…
I’m meeting with another provider next week, I’ll talk with them about it - we may end up switching providers over this issue - of course it’ll be a PAIN because we have like 80 DID numbers and 2 - 800 numbers, etc.
Also, the advantage of using Asterisk is that the in-processing of calls is split between the trunk and the inbound routes.
Adding a new (or multiple) trunks (as @lgaetz pointed out) means that the trunks can pick and choose which inbound route (or routes) the incoming calls will use.
So, having 80 DIDs is really not an issue for Asterisk. You set up your trunk to the provider and let them decide which calls come in and which don’t. Once they arrive at the trunk, you can route the calls to your inbound routes for DID processing, so once your system is set up, you can add/move/delete numbers or trunks at will and the system should properly route all of your calls correctly.