provided by Star2Billing S.L.

ok thx ill get on that next week,
not quite sure which settings but ill try to get all the settings out ouf A2
(verry little to configure for the trunks in A2)

Ok I went ahead and installed A2 on a trixbox manchine (i found a great script that does the instalation for you)
the installation went whithoud a hitch for a change.

So now i had a clean install of trixbox with A2. After fixing a few initial configuration problems i was able to create a customer and have it connect to the server. now in order to test asterisk i created an extention in freepbx and setup the trunk in free pbx as well and using the extention created in freepbx im able do dail out succesfully.
so now i know the trunk is working correctly.

however i ran into a porblem I though I previusly tackeld,
it seems A2 is unable to find a rate for the destination.
When i try and simulate the call I get a message "No matching rate found !" on the phone when I dail I get a message that the number I have dailed is currently unavailable.
In oder to fix this I attempted matching the settings from my other A2 testing machine with this new one so I set it up the same as the previus machine. now on the first machine these settings produce results in the simulation and it attempts to dail out but in the new machine it fails to find a rate and thus cancels the call.

is there any way i can check why its now not finding a rate? logs i could look into or crusial settings i mist?

I added rate and trunk settings here
mind you the trunk is working when i connect using an extention created in freepbx (running on the same asterisk server of cause)


needles to say this is frustrating being close but not quite there yet

well i been putting some echo's in the code and i found sometning that i hope might help the insiders point me in the tricht direction

I have been comparing the input for the RateEngine between the two A2 systemes
it turns out there are two key difrences between the two

the working A2 uses tarifgrouid 1 while the other uses tarifgroupid 0 now there is no tarif group with id 0
the working A2 remover the two leading 00 from the dailing number the other doesnt

i had the query generated by the working A2 system echoed and i rabn it in the database of the A2 system that doesnt work and id did produce a hit

now from all this i figure the settings in the call plan and rates are probably right since a query agains the database does produce a de desired results. however A2 isnt producing the right query.
it seems somehow the second A2 system is not preparing the info corectly.

how does A2 determine the tarifgroupid to use and whether it should remove the two leading zeros ?

Hi

Things to check, after having checked the trunk works in Asterisk / FreePBX

1. Use the rate simulator to check there is a rate for the call. If not:-
2. Check the rate card is attached to the callplan.
3. Check whether the callplan is removing the international prefix (00 or 011)
4. Check there is a rate in the ratecard matching your dialed digits.
5. Now recheck the rate simulator, if still not working:-
5. Check the system is sending the correct number to your carrier.

Joe

i have done 1 thru 4
but at step 5 things keep failing

im my previus post i described that after cechkking in the code of A2 itself i noticed that it somehow it trying to get the rate from tarifgroup 0 there is no tarig group 0 this isnt even posible since the frist tarifgroup you make is always 1
i have no idea where this came from it was a clean install no previus settings or database.

The order in which you create tarriff system is important.

Create:
Providers
Trunks
Ratecards
Import Ratecards
Call Plan

In other words, you should have in place Provider, trunk, and the ratecard defined before you import rates.

Deleting these in backward order and add again.

ok made some progres now

I been looking into it with a college and it apears to be alomost working.
We get the introduction message and once we put in the number and pres the pound key A2 tels us how manny minutes we have left, then it rings once and we get a message "your outbound call has been disabeld"

in the log i see one message



what does this message mean?
The trunk is reachable.
We have a rate asociated with the prefix.
There is sufficiant credit in the user account.

What else could be stopping it from dailing out?

this message seems to be coused by the user hanging up.

when hanging up while asterisk is still sending a stream of audio this message apears however it seems to be harmless.

"your outbound call has been disabeld"

That message does not appear to be coming from A2B or Asterisk. It would be interesting to see that log. Your provider might be blocking your call for some reason. If your account is up to date then another thing to consider is, some of them will deny the call if they detect bad or bogus caller id information.

You are correct about this. ignore.