provided by Star2Billing S.L.

Hello,

I was wondering how could I use FASTAGI to connect to a2billing.php|2 via our server to handle a2billing issues without the need of running a local copy of FastAGI.

eg:
- Server A has A2Billing on it and is the master server.
- Server B is a local PBX in a condominium to handle all local calls.
- Server B needs to call Server A's A2Billing DeadAGI php2billing.php|2
- Server A processes the information and dials out without any problems.

Has anyone ever done this solution yet? I am trying to eliminate running A2Billing on the Condo's Servers so that I can have everything centralized.

from what I deduced with FastAGI is that I would have to enter
exten => s,n,AGI(agi://Server A/a2billing.php|2)

Is there anything else I need to enter in Server A and/or Server B?

Thanks in advance for any assistance.

you can do this a totally different way.

Trunk the servers and use the CallerID to authenticate the caller and place the call

This way all the a2b lives on the billing server and the other server does not even know that a billing agent exist

Could you supply an example in trunking the servers?
would that be like AMP:SIP/$OUTNUM$@SERVERA

hello cyberglobe,

Do you able to get this one working at all ? I am also trying to do the same but get no lead to accomplish this task.

Not yet since I have not built my secondary server to test it out.

But I think to do this you would need to do the following:
Server A (Billing Server)
1: Setup Incoming Route ANY:ANY to goto the A2Billing Module
2: ** Anything else? **

Server B (Condo Server)
1: Setup a Trunk to Server A
2: Setup an outgoing route for 1NXXNXXXXXX and 011XXXXXXX. and map it to the Trunk to Server A

But then, the problem here would be that Server A would be handling the call control and not Server B. I would like for Server B to handle the Call Control and have Server A just reference everything to Server B. But would Server A still get the accounting records right?

If I did the FastAGI remotely, who will be processing the Call Control? Server A or Server B? Anyone knows?

I have 2 servers and I did as follow

Server A (Billing Server)
1: Setup Incoming Route ANY:ANY to goto the A2Billing Module

Server B (Condo Server)
1: Setup a Trunk to Server A
2: Setup an outgoing route for 1NXXNXXXXXX and 011XXXXXXX. and map it to the Trunk to Server A

HI Richard,

Can you check the following for me:

- Ensure that you have Reinvite=yes in your trunk that you built on Server B.
- Call via extension from Server B to dial your provider.
- on Server A, goto your SSH Shell and run asterisk -r
- type rtp debug
- make your long distance call and make sure the call picks up.
- after the call has been answered, check to see if you still have RTP traffic going to Server A.
- Check also at Server B for rtp traffic while the call is answered.

I am trying to avoid extension -> server B -> server A -> VoIP Provider RTP call path. I would like a Extension -> VoIP RTP Call path.

Call control can go through Servers A/B/VoIP Provider.

Hi Cyberglobe,

I don't know where to put Reinvite=yes in my trunk of server B. Any way, using extension from server B, I called a 1800 number and the call answered. setting rtp debug on both server A & server B, I observed the RTP traffic is going like this while the call is going

extension -> server B -> server A -> VoIP Provider RTP ( calling ....)


extension -> server A -> VoIP Provider RTP ( while the call is still on )


Server B is stop transmitting RTP packets .

Ok, now for the Extension, IF your extension is not behind NAT, try this:

- Edit the extension that is not behind NAT
- Set the Reinvite to YES and NAT to NO
Save and reload and then see the RTP call path.

It should look like Extension -> LD Provider
If you are not behind NAT though.

I think my extension is behind NAT. Do you know of a way to determine if an extension is behind NAT or not ? what is the advantage of Extension --> Voip provider ?

If your IP Address is 192.168.x.x or 10.0.x.x You are a NATed Client.

The advantage is to reduce the amount of machines that your call must go through.

If you go Extension -> VoIP Provider is the Fastest/Shortest route. This is like flying a plane from point A to B.

If you go Extension -> Server A -> Server B -> VoIP Provider, this is like driving to the destination where you can get congested in the servers you visit on the way.

I prefer to have the call RTP path to always be direct instead of running via the Servers. This way Asterisk can handle hundreds of calls on a small box without the need to process audio streams between Extension to the VoIP provider.

If you have the ability to get a Class C, I highly recommend trying to use it for VoIP. The more ways to reduce servers from the audio stream the better. I have had problems with providers with SIP and H323 on the same box. Their servers have to convert SIP Streams to H323 and this has caused problems (Server outage, degraded sound quality, bad echoing due to conversion delays).

Thank you for a valuable lesson. My server A is behind NAT and server B is not and that's explains why the last test the rtp packets flow like this Extension --> Server A --> Voip Provider ( while calling still going on ..)
So if the Server A is not behind NAT then for sure the rtp ---> Voip Provider as you would have expected.

Hi Cyberglobe,

I noticed SIP-friend extension in A2B is not passing RTP audio stream to Voip provider even though canreinvite=yes was set in the extension. Unlike, extension in Trixbox, I am able to shift all RTP audio stream to Voip provider.
Have any idea on this ?

make sure your VoIP Provider also has Reinvite=yes turned on or it will fail too.