provided by Star2Billing S.L.

I have gotten a2billing working, using the current version 1.7.1, and I can make calls out through our Asterisk server. But right now my sticking point is Caller ID. Before we can sell this product to our users, we need to be able to set the outgoing caller id. Having it default to the account id# is useless.

I have found that by manually editing the file /etc/asterisk/additional_a2billing_sip.conf and putting in a line for the user "cid_number=##########" with the right number does the trick.

However, having to manually edit that file FOR EVERY CUSTOMER is unreasonable. I am pretty sure I somehow managed to do this once for one test user through the web interface, but I cannot seem to replicate that with a second user.

How exactly do I go about setting that field through the web interface without having to bring up a shell account?

You can set this under customers | Voip Settings.

OK, I click "Customers" then "Voip Settings". I can then for each customer choose either "Username" (their ID#) or the pencil to edit it. If I select the Username then I can find an option to set "Caller ID" farther down; however it is already set to the Caller ID I want showing, and therefore that can't possibly be the option to edit /etc/asterisk/additional_a2billing_sip.conf or else it would already have BEEN in the .conf file, yes?

So moving on, I try clicking the pencil. But guess what? the Caller ID is ALSO listed there already, and it's STILL not set in the .conf file. I see nothing else on that page which might possibly affect the additional_a2billing_sip.conf file whatsoever.

And so I ask again. What do I click on in order to get it to rewrite the user's data in the additional_a2billing_sip.conf file so that it will add in the "cid_number=" line? I imagine that my only current workaround will be to delete the field from some webform or other, and then resave it again. But which page, which field?

Hi

The /etc/asterisk/additional_a2billing_sip.conf file is generated from this screen.

You change it here - regenerate the /etc/asterisk/additional_a2billing_sip.conf file reload, and it will appear, providing of course you have realtime set to no in the system settings.

joe

That appears to have been the problem -- the DEFAULT setting for "use_realtime" is "Yes". When I switched that to "No" instead, then the Customers->Voip Settings screen helpfully includes an option for "Generate additional_a2billing_sip.conf", which when I click rewrites the whole .conf file from scratch.

I am still confused though. Somehow my first user's "cid_number" setting got written into that file without "use_realtime" disabled, and I am wondering how to get it to write that into the users' settings, by default, when we create the account in the first place!

Are there any other side effects of changing "use_realtime"?

OK, apparently now it's completely broken again. I made that one system-wide change for "realtime", and it regenerated the ..._sip.conf file with the callerid information. However, NOW the outgoing caller id is NOT set for either user. I have confirmed that the previously working data for the originally working user has not changed in the slightest, and the call display information is no longer being passed along.

So this "fix" appears to have made things worse.

This was a simple vanilla install of a2billing 1.7.1 on a working Trixbox system that has been otherwise working fine for months. So I can only figure that everything I'm seeing as going wrong basically goes wrong by default?

How hard can it be to set up so that, when a sip device calls out through this system (that much worked) that the receiving end gets the caller id on their screen?