Internal calls not working

asincom · **Joined:** Wed Aug 22, 2007 4:36 am **Posts:** 97

Hi everybody, and thanks Areski for A2Billing.
I recently install trixbox 2.0 with A2Billing V1.3.
Everything is working fine but I can not make calls between two customers or card numbers. I dont know what number I have to dial, crad number or card alias. In my case, I changed username and password to match with existents customer`s accounts, but I tried with two new cards customers and one card can not make the calls to the other one.
I tested on Sipura, Lynjksis, Xlite, etc and no change, also I can see in Asterisk log calls are maded but are passing through my previously default trunk and provider dont handle call because number doesnt exist.
Any clue what changes should I try ?
Thanks in advanced.

Regards, :roll:

Aldo

howard · **Joined:** Wed Jul 25, 2007 4:42 am **Posts:** 21

I think it may be related to the context

if you are using trixbox, all the extension (your customer's account) should be using context = from-internal

when you create sip friend using a2billing, then the context by default is a2billing.

I suggest to double check the context settings.

in order to have the internal calling work properly, you should use context=from-internal

asincom · **Joined:** Wed Aug 22, 2007 4:36 am **Posts:** 97

jroper · **Joined:** Fri Jun 23, 2006 3:56 pm **Posts:** 4065

Hi

There is an option to call SIP and IAX friends for free in the agi-conf section.

By default it looks like this: -
; enable the option to call sip/iax friend for free (values : YES - NO)
sip_iax_friends = NO

; if SIP_IAX_FRIENDS is active, you can define a prefix for the dialed digits to call a pstn number
; values : number
sip_iax_pstn_direct_call_prefix = 555

The number you dial is the cardalias - so as to protect the customer account number, which of course is used for authentication.

Joe

asincom · **Joined:** Wed Aug 22, 2007 4:36 am **Posts:** 97

I set up values in agi conf in yes and I set up a prefix 999 to dont match any country code and the result is extension unavailable. Asterisk cli says rateengine.php could not find any match for the code, and of ocurse if I set up for example 1, that would be a call to USA.
I dont want the IVR either, I just want to dial another card alias and make a free internal user call.
How can I do that ?
I realize, if I change context and create both accounts as extensions in free pbx, both can call between each other, but that call is out of a2billing and all cals maded to pstn will not being charged. So I change context to a2billing again, but can not make internal calls.
Also I tried to set up a diferent string in a2billing context where all my card alias (_7XXXX) supose to call or answer the call as a reguilar extension and not to pass through a2billing.php, but doesnt work either.
the call still is handled by php script, which goes to rateengine.php after validation.
I know this solution might be very simple, but I can not figure out how to handle internal calls in a2billing yet.
Please Areski or anybody ¡¡¡ help ¡¡¡
rgds

gue · **Joined:** Mon Jun 19, 2006 5:14 pm **Posts:** 218

follow the insructions jroper gave you in previous posting

Quote:

; enable the option to call sip/iax friend for free (values : YES - NO)
sip_iax_friends = YES

; if SIP_IAX_FRIENDS is active, you can define a prefix for the dialed digits to call a pstn number
; values : number
sip_iax_pstn_direct_call_prefix = 555

set those parameter in the a2billing.conf agi-conf section

then dial 555+cardalias

asincom · **Joined:** Wed Aug 22, 2007 4:36 am **Posts:** 97

Hi Gue
Sorry, but as I said before, it is not work¡¡¡
I have two extension 71467 and 71468
I set up sip_iax_friends=yes and then I try to dial an extension, 71468
An IVR said "if you want to call to a sip or iax friend, press 9. So I dial 71468 again. Rate engine handle call to Russia¡¡¡
Then I dial 55571468, IVR again, so I press 9 and I dial again 55571468. This time Rate Engine send the call to Brazil.
Everything else is working great, but this issue is driving me crazy.
This feature is absolutely necesary in any voip system, I mean users have to be able to call between each other for free.
Any clue ? Please advice me
Thanks in advanced.
Rgds

gue · **Joined:** Mon Jun 19, 2006 5:14 pm **Posts:** 218

start from the begining:

1. cardalias is what you dial. it is a value you set in the customer record it is not your card number and it must be numeric

2. make sure your you are using the same agi-conf that is specified in the call to a2billing.php

3. in that agi-conf these 3 parameters are to be set exactly the way i have them

; enable the option to call sip/iax friend for free (values : YES - NO)
sip_iax_friends = YES

; if SIP_IAX_FRIENDS is active, you can define a prefix for the dialed digits to call a pstn number
; values : number
sip_iax_pstn_direct_call_prefix = 555

; this will enable a prompt to enter your destination number.
; if number start by sip_iax_pstn_direct_call_prefix we do directly a sip iax call, if not we do a normal call
sip_iax_pstn_direct_call = YES

4. dial 555+card alias

asincom · **Joined:** Wed Aug 22, 2007 4:36 am **Posts:** 97

I solved it.
Well, after set up sip matters in a2billing.conf as Gue and Jroper posted, I still can not make internal calls.
My problem was I changed default account numbers. So I went to database and cc_card table still have the original value so, I changed to the same card alias=username=accountcode=reg extension and voila, it is works.
Certainly everything should work if I dont change default accounts.
Thanks Gue an Jroper for your help.
Regards,

gue · **Joined:** Mon Jun 19, 2006 5:14 pm **Posts:** 218

now that you have it working choose a card-alias that is NOT your card number. You don't wnat those numbers to be the same. making it the same as giving out your number would be the same as giving out the pin number for your account

asincom · **Joined:** Wed Aug 22, 2007 4:36 am **Posts:** 97

Yes, I understand your point.
Now I have a diferent problem.
I´m running a platform based on asterisk and Astbill. I found A2Billing which is more flexible and have much more features so I want to change. The problem is all my customers have predefined account originally created on astbill. Those accounts have 7XXXX format, so I dont want to visit each one to change sip settings, and not all of them can give remote acces, so I tried to match Astbill accounts in my new a2billing.
The problem is: if i changed settings to match account in 7XXXX format, a2billing lost validation and prompt me for my pin number. Many of my customers are call shops, and they dont want to validate pin numbers in each call they make.
So, my question is: there is a way to match this format in order to keep my custiomers sip settings and not to create new a2billing accounts ?
Thanks again.
Rgds

gue · **Joined:** Mon Jun 19, 2006 5:14 pm **Posts:** 218

You can most anything:

in a2b you can austhenticate by PIN and by accountcode
accountcode is the same as PIN but accountcode can be carried in the SIP.conf.

For your callshop you want to use the accountcode feature but you should use a different agi-conf for these clients. creae and use one where you are not authenticating by callerid. this way it will authenticate by accountcode. So when the callshop access your server they will not be prompted for PIN

For you clients who are calling in via access numbers creeate a agi-conf that authenticate with callerid or PIN if no callerid is presented

You users who already have accounts where there "username" was in the 7xxxxx range you should use the import feature for the customer record.
generate csv file with 7xxxx as the customer card number and then import the file into a2b ... it is like moving one set of accounts from one sytem to a2b.

Asterisk is authenticating your sip client on username and secret so you have to make sure you can import all that relevent stuff into the appropriate tables

ia2b is authenticating on PIN, accountcode or callerid

In a2b remember that cardalias is what you dial and also the same as username to log in the customer UI so you have to think thru what features you wnat to use that requires these variable.

asincom · **Joined:** Wed Aug 22, 2007 4:36 am **Posts:** 97

Thanks Gue¡¡¡
I understand better how its works.
I really apreciate your help.
Regards,

provided by Star2Billing S.L.

Internal calls not working

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