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 Post subject: Using A2Billing for Origination Billing(termination working)
PostPosted: Sun May 07, 2006 9:29 am 
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Joined: Wed Mar 29, 2006 5:46 am
Posts: 15
Location: South Florida
I'm already using a2billing to do my SIP termination billing. A customer will push the call to me SIP, I set the accountcode to their PIN and then push it to A2Billing, which will then do the LCR and push it to one of my 3 upstream providers via SIP. This is all working great, have over 200,000 minutes a month running through the system right now, even have a secondary server for load balancing, accessing the same database off a dedicated database server, doesn't matter which server a customer pushes the call to, it gets charged to the same account. It's running good.

I've gone through some credit checks, background checks, contracts, commitments, 45 day waits, inerconnection testing, etc. and now have awesome rates for Termination and Origination through XO Communications, Global Crossing and Level 3. Thanks to A2Billing, I'm already competing (beating actually) against the likes of NuFone and VoIPJet and new customers sign up daily. I'm going to be offering my rate table in a format ready to be imported into A2Billing for anybody interested.

I want to start offering Origination, DID's to my customers. Trying to route traffic in the opposite direction, and bill for it, will be a fun task. I'm wondering about A2Billing's "DID" system. Can a DID come in from one of my carriers via SIP, hit A2Billing and then A2Billing routes it to my customers SIP gateway? I'm charging a base cost a month for the DID plus usage, so A2Billing would have to know which PIN number/account to bill for usage on that DID and then know where to route it.

I've
A) Created a DID Group called Test
B) Created a DID "1234567890"
Set Billing to "Fixed per month + Dial out rate"
Put it in the DID Group Test, Activated it, and set the fixed rate cost
C) Created a Destination for this DID, to SIP/[email protected] and specified the card ID of the account I want charged

Now what? I've tried calling "1234567890" through A2Billing, thinking it would route it to SIP/[email protected] but instead it just tried to dial it out through one of my carriers, like a regular long distance number. It didn't detect that it was a DID and didn't route it differently like I'd hoped.

Can A2Billing work for what I need? I'm going to be charging something like $2 a month per DID and $0.011 per minute for usage. A2Billing has to take $2 a month from the customers credit, and any usage they do has to come from that balance as well.


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 Post subject:
PostPosted: Sun May 07, 2006 12:47 pm 
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Joined: Tue Mar 21, 2006 7:17 pm
Posts: 141
Location: Washington DC
There is a little bit of info in A2Billing_AGI/INSTALL_AGI.TXT. Basically you have to route the DID to A2Billing in extensions.conf. Add the following section:

[a2billing-did]
exten => _X.,1,DeadAGI(a2billing.php|1|did)
exten => _X.,2,Hangup

And send the incoming DIDs to it.

And about the charges, A2Billing will charge for calls but you'll have to do the monthly charges manually for now. You can specify them but A2Billing will not charge them monthly.

PS: Your website?


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 Post subject:
PostPosted: Sun May 07, 2006 12:58 pm 
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Joined: Tue Mar 21, 2006 7:17 pm
Posts: 141
Location: Washington DC
We'd love to hear some statistics. For example how many concurrent calls can you route using A2Billing (please give some details of the hardware, consumed memory, CPU load) with and without transcoding, and what are the bottlenecks you found.


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 Post subject: Re: Using A2Billing for Origination Billing(termination work
PostPosted: Wed May 10, 2006 3:07 pm 
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Joined: Tue Apr 18, 2006 2:51 am
Posts: 12
Location: Malaysia
You mind sharing how to do the load balancing for a2billing ...??

Thanks.


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 Post subject:
PostPosted: Thu May 11, 2006 4:02 am 
If he does, you would then compete with him...:-)

People in this forum don't want to share any info how to do anything/tips/success stories etc. This will be giving away all of your hard work otherwise.

:-(


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 Post subject:
PostPosted: Thu May 11, 2006 4:27 am 
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Joined: Tue Mar 21, 2006 7:17 pm
Posts: 141
Location: Washington DC
Anonymous wrote:
If he does, you would then compete with him...:-)

People in this forum don't want to share any info how to do anything/tips/success stories etc. This will be giving away all of your hard work otherwise.

:-(


Let me contradict you and give a few hints about load balancing.

One of the most CPU intensive parts is the codec transcoding so you want more Asterisk servers. You can have more Asterisk servers, all of them using A2billing connected to the same database server (as a matter of fact you can have a balanced SQL cluster behind). That's the easy part. A little more complicated would be to balance load on the Asterisk servers. An easy approach is to have some customers using one server, other using another server and so on. But to do it right you need a Session Border Controller which will receive all the incoming SIP signaling and distribute the calls over the available Asterisk servers based on the existing loads. The SBC can use SER or OpenSER.

Good luck... :)


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 Post subject:
PostPosted: Thu May 11, 2006 4:41 am 
I guess my post helped. :-)


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 Post subject:
PostPosted: Thu May 11, 2006 4:48 am 
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Joined: Tue Mar 21, 2006 7:17 pm
Posts: 141
Location: Washington DC
Anonymous wrote:
I guess my post helped. :-)


Nobody asked me before... ;) Anyway, the existing of A2Billing itself proves you wrong.


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 Post subject:
PostPosted: Sat Jan 13, 2007 12:09 pm 
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Joined: Sat Jan 13, 2007 11:30 am
Posts: 3
I did set up the same SIP system in which I use openser for registrar and call routing server. All calls are loadbalaced via 2 asterisk servers which have a2billing AGI inside. The two backend asterisks will do all the a2billing stuffs and terminating calls to upper stream SIP providers.


[SIP Phone, Xlite] ----> [OpenSer] ----> [Asterisk/A2B] ----> [UPPER SIP Providers]


The system works well except some problems:
- When my customers press cancel first to hangup the call, it doesn't hang up unless the remote site (from UPPER Sip provider) presses hangup first.
- When the remote site presses hangup, Asterisk (so a2billing) can receive the signal and update billing records, but our customer's softphone or ip phone can't see anything.

I had tried to reconfigure (openser, asterisk, a2billing), place the whole system in the same network to avoid NAT problem, but no luck.
I recall that Asterisk does not have SIP proxy capabilities, so it can't forward the BYE message back !??! Is that make sense ? If yes, is there any workaround ?

Please help.


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 Post subject:
PostPosted: Sat Jan 13, 2007 10:12 pm 
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Joined: Tue Jun 06, 2006 12:14 pm
Posts: 685
Location: florida
So you have originator with SIP phone that calls into openser which hands the call off to asterisk running A2B (this is one box - Asterisk / A2B) ? And then it is registered with provider that passes it along?

The OpenSer does reinvite so its out of the picture if you are getting to the Asterisk & A2B.

If so, then I wouldn't think you should have issues, cause then it would look like most of our systems. I am routing enduser --> Trixbox/A2b --> Tribox Outbound Providers box --> termination. So it sounds to me like the same except you are initially registering and passing the call on with OpenSer.

Have you tried to shorten your path to locate what is causing the issue? I'd doubt it is your SER, and think it would be probably your provider.


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 Post subject:
PostPosted: Tue Jan 23, 2007 11:48 am 
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Joined: Sat Jan 13, 2007 11:30 am
Posts: 3
Yes, I did try many things, even capturing the signalling packets. I doubt SER too, but don't have any luck. Can u just share me your openser.cfg ?


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 Post subject:
PostPosted: Tue Jan 23, 2007 3:06 pm 
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Joined: Sat Jan 13, 2007 11:30 am
Posts: 3
I just found something interesting. With the same configuration and routing logic, ser (0.9.6) works very well but openser (1.1.0) has above problems. But I do need the features of round-robin dispatching of openser.


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 Post subject: How many concurrent are you talking about at 200,000 min /mo
PostPosted: Wed Jul 25, 2007 12:13 pm 
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Joined: Wed Jul 25, 2007 11:41 am
Posts: 1
glitchsys

Just wonder.
Correct me if I'm wrong.
yon are running:
2 servers for asterisk
1 server for MySQL
1 server for openser (or else) to do the load balancing.

How many concurrent are you talking about? Is it stable?

My hardware is Celeron 2GHz + 512 ram
I'm testing my system (A2billing + asterisk) with Sipp.
I get around 30 concurrent with RTP load on g711.

I plan to build another system to get around 100-120 concurrents with Conroe 3060 + 4GB. Is that possible? I heard that Conroe is much faster (almost 2 times) than Double Xeon 2.8Ghz.

Anyway, the question is ... glitchsys ... how many concurrent you can reach with your system?

Thanks


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 Post subject:
PostPosted: Thu Aug 02, 2007 11:22 pm 
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Joined: Thu Aug 02, 2007 11:20 pm
Posts: 2
Is it possible to load balance asterisk with LVS (linux virtual server) using sticky logic for the SIP loadbalancing?


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