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 Post subject: How to send all incomming calls to a2billing ?
PostPosted: Mon Oct 03, 2011 5:42 am 
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Joined: Mon Oct 03, 2011 5:27 am
Posts: 9
Hello Everybody,

I have installed Elastix 2.2.0 RC3.
Asterisk Version: 1.8.7.0
Freepbx Version: 2.8.1.4
a2billing Version: 1.8.1

What I want to do: I want all incoming calls to my asterisk server should go to a2billing and then a2billing should route that calls to the trunks associated with a2billing and freepbx.

What I have already tested: I have already tested incoming calls to my asterisk server to trunks gateways. And I have done this is the following ways....

1. I have set "Allow Anonymous Inbound SIP Calls?= yes" from general settings in freepbx menu.(myserverip/admin).
2. Then I have added a SIP trunk for my voip gsm gateway(DINSTAR VOIP GSM GW)
3. After that, an inbound route, named "DINSTAR2008" has been created for the DID "_8801XXXXXXXXX."
4. Now if I call to any number that is like 880173XXXXXXX(cell phone number), successful call connection is made between my PC dialer(x-lite) and my cellphone via asterisk and DINSTAR VOIP GSM GW. I just dial from x-lite without registering to astering server.


Now I want to do the above using a2billing so that every billing should be possible for every incoming calls. Please tell me the steps to follow.......

Please share your knowledge with us.

Regards,
Zahid


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 Post subject: Re: How to send all incomming calls to a2billing ?
PostPosted: Mon Oct 03, 2011 7:06 am 
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Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4065
Hi

Create another inbound route with _. as the DID, much the same as _8801XXXXXXXXX.

Joe


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 Post subject: Re: How to send all incomming calls to a2billing ?
PostPosted: Mon Oct 03, 2011 7:18 am 
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Joined: Mon Oct 03, 2011 5:27 am
Posts: 9
Hi Joe,

Thank you very much for your reply. Let me try and update you.

Regards,
Zahid


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 Post subject: Re: How to send all incomming calls to a2billing ?
PostPosted: Mon Oct 03, 2011 7:35 am 
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Joined: Mon Oct 03, 2011 5:27 am
Posts: 9
Hello Joe,

Yes I have created another inbound route with _. as DID. I can call from xlite to my cell phone.
xlite >>> asterisk Server>>> voip gsm GW >>>GSM network.

Now I want to do the same call should pass through a2billling as like as bellow:
xlite >>> a2billing>>>asteriskserver>>> voip gsm GW >>>GSM network.

Please advise me what have to do in a2billing configuration and freepbx configuration.

Regards,
Zahid Hasan


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 Post subject: Re: How to send all incomming calls to a2billing ?
PostPosted: Mon Oct 03, 2011 7:37 am 
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Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4065
Hi

Simply send the call into the appropriate A2Billing context, in FreePBX this will be done with a custom destination of CONTEXT,EXTENSION,PRIORITY, e.g. a2billing,${EXTEN},1

Joe


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 Post subject: Re: How to send all incomming calls to a2billing ?
PostPosted: Tue Oct 04, 2011 6:13 am 
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Joined: Mon Oct 03, 2011 5:27 am
Posts: 9
Hello Joe,

I have done the following so far...

1. I have created a trunk named "DX157" in free-PBX. This is a trunk to voip-gsm GW.

2. Then I have added custom destination named "Let me try" from tools in free-PBX having custom destination "a2billing,${EXTEN},1".

3. After that I setup an inbound route named "GO_DX157" having DID number "_." which points to custom destination "Let me try"

4. After doing all this in free-PBX, I have added the following lines in /etc/extensions_a2billing.conf
[a2billing]
exten => _X.,1,Answer
exten => _X.,n,Wait(1)
exten => _X.,n,DeadAGI(a2billing.php,1)
exten => _X.,n,Hangup

5. I also set up provider/trunk/prefix/callplan/ratecard/rates etc in a2billing.

6. But now when I am trying to call 8801730075290 from x-lite , I am hearing "Please enter your complete pin number". I did not set up any pin number for any incomming calls. here is the cli output......

Connected to Asterisk 1.8.7.0 currently running on SW1 (pid = 11692)
Verbosity is at least 17
-- Remote UNIX connection
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [8801730075290@from-sip-external:1] NoOp("SIP/114.141.208.30-0000000d", "Received incoming SIP connection from unknown peer to 8801730075290") in new stack
-- Executing [8801730075290@from-sip-external:2] Set("SIP/114.141.208.30-0000000d", "DID=8801730075290") in new stack
-- Executing [8801730075290@from-sip-external:3] Goto("SIP/114.141.208.30-0000000d", "s,1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("SIP/114.141.208.30-0000000d", "1?checklang:noanonymous") in new stack
-- Goto (from-sip-external,s,2)
-- Executing [s@from-sip-external:2] GotoIf("SIP/114.141.208.30-0000000d", "0?setlanguage:from-trunk,8801730075290,1") in new stack
-- Goto (from-trunk,8801730075290,1)
-- Executing [8801730075290@from-trunk:1] Set("SIP/114.141.208.30-0000000d", "__FROM_DID=8801730075290") in new stack
-- Executing [8801730075290@from-trunk:2] Gosub("SIP/114.141.208.30-0000000d", "app-blacklist-check,s,1") in new stack
-- Executing [s@app-blacklist-check:1] GotoIf("SIP/114.141.208.30-0000000d", "0?blacklisted") in new stack
-- Executing [s@app-blacklist-check:2] Set("SIP/114.141.208.30-0000000d", "CALLED_BLACKLIST=1") in new stack
-- Executing [s@app-blacklist-check:3] Return("SIP/114.141.208.30-0000000d", "") in new stack
-- Executing [8801730075290@from-trunk:3] ExecIf("SIP/114.141.208.30-0000000d", "0 ?Set(CALLERID(name)=206550)") in new stack
-- Executing [8801730075290@from-trunk:4] Set("SIP/114.141.208.30-0000000d", "__CALLINGPRES_SV=allowed_not_screened") in new stack
-- Executing [8801730075290@from-trunk:5] Set("SIP/114.141.208.30-0000000d", "CALLERPRES()=allowed_not_screened") in new stack
-- Executing [8801730075290@from-trunk:6] Goto("SIP/114.141.208.30-0000000d", "a2billing,8801730075290,1") in new stack
-- Goto (a2billing,8801730075290,1)
-- Executing [8801730075290@a2billing:1] Answer("SIP/114.141.208.30-0000000d", "") in new stack
-- Executing [8801730075290@a2billing:2] Wait("SIP/114.141.208.30-0000000d", "1") in new stack
-- Executing [8801730075290@a2billing:3] DeadAGI("SIP/114.141.208.30-0000000d", "a2billing.php,1") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
-- <SIP/114.141.208.30-0000000d> Playing 'prepaid-enter-pin-number.gsm' (language 'en')
-- <SIP/114.141.208.30-0000000d> Playing 'prepaid-enter-pin-number.gsm' (language 'en')
-- <SIP/114.141.208.30-0000000d>AGI Script a2billing.php completed, returning 4
== Spawn extension (a2billing, 8801730075290, 3) exited non-zero on 'SIP/114.141.208.30-0000000d'


Please help me out!

Regards,
Zahid Hasan


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 Post subject: Re: How to send all incomming calls to a2billing ?
PostPosted: Tue Oct 04, 2011 6:20 am 
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Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4065
Hi

Can you explain again the desired call-flow.

Joe


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 Post subject: Re: How to send all incomming calls to a2billing ?
PostPosted: Tue Oct 04, 2011 6:37 am 
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Joined: Mon Oct 03, 2011 5:27 am
Posts: 9
Hi,

Let me describe the desired call flow...

all incomming calls to asterisk server(Maybe from voip softswitch or from anywhere, voip termination call to gsm networks) >>>>> a2billing>>>>>voip gsm gateway >>>>> GSM network
Or simply speaking, I want to make a2billing and asterisk as like as a voip softswitch.

Please ask me any question if i am failed to make you understand what i am saying.

Regards,
Zahid


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 Post subject: Re: How to send all incomming calls to a2billing ?
PostPosted: Tue Oct 04, 2011 6:42 am 
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Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4065
Hi

So where does the 8801730075290 come into the equation.

Is this an access number to access the platform, or is it the number you are trying to route outbound?

Joe


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 Post subject: Re: How to send all incomming calls to a2billing ?
PostPosted: Tue Oct 04, 2011 6:49 am 
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Joined: Mon Oct 03, 2011 5:27 am
Posts: 9
Joe,

I want to call this number "8801730075290" from x-lite. This is just an example. Actually need to terminate all voip termination call(whole-sale termination call) to gsm network(8801 prefix) via my asterisk and a2billing server.

x-lite= calling party
8801730075290 or something like that = called party

Regards,
Zahid


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 Post subject: Re: How to send all incomming calls to a2billing ?
PostPosted: Tue Oct 04, 2011 7:01 am 
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Joined: Mon Oct 03, 2011 5:27 am
Posts: 9
Hello Joe,

I want to make call between x-lite(or voip wholesale customer) and gsm networks where voip-gsm GW is connected to asterisk server.

Regards,
Zahid


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 Post subject: Re: How to send all incomming calls to a2billing ?
PostPosted: Tue Oct 04, 2011 4:45 pm 
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Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4065
Set up a SIP account in A2Billing, plumb in the settings into xlite, create trunks, callplans, ratecards, and rates, then call the number, making sure that you have adjusted your agi-conf to suit voip calling.

Joe


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