QUESTION A2Billing - An Adventure in PBX Billing - Part 1

tshif

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The Further Adventures in A2Billing – Conference Rooms - Part 2

The Further Adventures in A2Billing – Conference Rooms - Continued

Installation:
The install is as easy as can be. Logged in as root, from putty (or your favorite tool) execute these lines:

Code:
cd /root
wget http://www.fonicaprojects.com/downloads/piaf/piaf-wmm.sh
chmod +x piaf-wmm.sh
Now, if you have changed your MySQL password (as you may have done if you use Joe Ropers PBX management tool), you need to edit the piaf-wmm.sh script located in /root. You will find it on line 37 of the script.

After editing the password – if needed – execute this line:

./piaf-wmm.sh

As the installation finishes, you will see this message, indicating a successful installation.

This installation is covered by the same htaccess and by database authentication as the previous version. Just as you would have before, continue to use your meetme, wwwadmin or maint password to gain access to the front page, and the passwd-meetme script in PBXIAF to manage the password.

When you arrive at the Web MeetMe main screen, you may log in with the default id/pw of root@localhost / password.

If you have elected to keep your previous WebMeetme database, your previous log ins and ID’s still work, and can be used normally as before.

If this is a new setup, you will probably want to log in as administrator and establish ID/passwords for all users that you wish to be able to set up conferences in the system.

Rememebr, In Web MeetMe, non administrative ID’s only have access to their own conferences, and User ID. So many users can share the system without running into each others meeting data.

That’s it – we’ve completed the install or upgrade of Web MeetMe. Now we can move on to configure it for use in the Family Phone system.

Configuring DID for WebMeetme Conferences
We have already performed similar steps before in this article series, so I’ll give you the short list. Refer to previous articles in this thread for more detailed instructions for each of these steps:

In FreePBX:
1. Acquire a New DID for this purpose, and add an inbound route for it pointing to the A2B-DID Custom Destination.

In A2Billing:
2. Create the DID, Assign Sell Cost, etc.
3. Add an Inbound Rate for it, assigned to the Local Destination rate card.
4. Create a Destination for it. The destination is the feature code of the WebMeetMe conference rooms.

Remember – if you change the feature code of the Web MeetMe system in the Feature Code page of FreePBX, you will need to update A2Billing’s destination to reflect the new feature code.

Consider: Under more typical circumstances, Web Meetme would have only a single DID pointed at it, and all outside users would access the system using that DID. In the family phone system however, we need to bill for the inbound minutes – and for that to work out – each family that uses the conference system must have its own DID for doing so. In other words, for each family (or financially responsible party) using Web MeetMe, we must perform the 4 steps above.

Installing the Web MeetMe PIAF FreePBX Module
Now its time to download and install the FreePBX Module that completes this improved Web MeetMe installation. The Module creates the Feature Code in FreePBX, and allows you to manage it from the FreePBX Feature Code page. It also provides access to the Web MeetMe interface from within FreePBX.


Step 1: Download the module to your desktop from here:
Code:
[URL="http://www.fonicaprojects.com/downloads/piaf/webmeetme-1.0.0.tgz"] http://www.fonicaprojects.com/downloads/piaf/webmeetme-1.0.0.tgz[/URL]
Don’t decompress the archive!

Step 2: Use a web browser to access FreePBX on your Asterisk PBX. Choose Tools, Module Admin, Upload Module. Now Browse to the webmeetme-1.0.0.tgz file on your Desktop and click the Upload button. Now click local module administration, scroll down and click on Web MeetMe and then the Install radio button. Now click Process, and complete the usual FreePBX install and reload process.

Step 3: While still in FreePBX, navigate to Setup, Feature Codes. Near the bottom of the screen, locate Web MeetMe. Use these controls to change the Feature code to anything you like (providing it doesn’t conflict with anything else in your PBX). You can also enable or disable the Web MeetMe function entirely from this location. Submits your changes, approve and process when you are satisfied with the settings.

Again, Remember – if you change the feature code of the Web MeetMe system in the Feature Code page of FreePBX, you will need to update A2Billing’s destination to reflect the new feature code.

Once you’ve completed these steps, your “outside access” conference room direct dial number is ready to work, and be properly billed.

Until next time -
 

manugmedia

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tshif,
:hurray::hurray::hurray::hurray:
Hat's off, Hat's off to you. Hat's off to tshif. Ladies and Gentlemen, Please welcome the ever greatest........ tshif. Eihhhhh!
I must say i have followed your tutorials and learned some things i never knew and i certainly say i would never have learned Asterisk and A2B can do, in a very short time. I have been following your tutorial and Man! you rock! You left no stones unturned in my quest to get Asterisk and A2B in my nerves. tshif? what can i say? very clear and concise tutorial. i have seen my system work to my liking. I must say i have printed the tutorial and laminated it because it answers so many questions i have come across in my imagined system, and am sure to refer to it in future. Waoh! its all :lol: smiles to me as i head to create that grand conference room. No, i won't forget Joe and Tom, and all the nerds i couldn't mention. Incredible work. Let's keep them phones ringing.

God Bless you!
 

tshif

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:blush5: Thank you.

It's pretty tough to know how to respond to the kind words here and a couple other places lately.

It's a rare and pleasant experience to be able to give back to a community that has helped me so much. Someone recently said it very well in another thread here someplace - but I have been the recipient of information, time and help from some of the most tremendously generous techno-geeks a person could ever encounter. We all know who they are - they are why we are here.

From the Trio that launched PBXIAF; Ward, Tom, and Joe we got a rock solid distribution. And they continue to set the tone here as one of technical-camaraderie. A perfect place for us all to geek-out.

In fact, It was a little kindness and some nudging from Joe Roper that really convinced me to share what I was working on with the forums in the first place. And he continues to put up with my stupid questions and occasional rantings with good humor - and does his best to keep me accurate with the articles here in the great "Adventure in PBX Billing". This Article series is one project I could never have embarked on without his ready ear.

And as far as other forum users go - well so many have helped me out over the last three years that this response I'm writing could be as long as the A2Billing article itself if I began thanking them all by name.

Too much more praise however, and it could damage my reputation as that "grumpy-old network engineer who works nights". But as that grumpy old network engineer - truly - nothing makes me happier than knowing folks are being successful with stuff I offer to the forums.

It has been an completely unexpected, but absolutely heart warming experience to receive comments from you and others. Again, thank you so much.

To our future projects together!

until next time

-Tony
 

kevinb

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Tony,
Trying to install A2B per your instructions. I get as far as setting up the inbound contexts and routs, but when I call the DID number assigned to that route it answers, tries to go to the custom context and just hands up. I've been over the docs a dozen times and can't see what I'm missing. Any ideas? This is on a PIAX 1.4 box. Thanks.
 

tshif

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Kevin;
Lets have a look at your custom context called [custom-a2billing-did], should be in extension_a2billing.conf.

Also, it would be interesting to see a CLI trace of the inbound call - as it tries to get into a2b.

Hmm. Lets see... Assuming it makes it into a2b - you'd want to check and see that you have the DID registered with a2b exactly as its being conveyed from your provider through FreePBX, and that there is a destination (extension) for the DID.

Let us know if any of this helps -
 

nsukho

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same problem for the incoimng and outgoing calls through A2Billing

Tony;
I have the same problem as Kevin has for incoming calls and also for outgoing calls in my case. I have followed the exact instruction in this thread.

I have the following (all) contexts in extension_a2billing.conf file:
[a2billing]
exten => _X.,1,Answer
exten => _X.,n,Wait(1)
exten => _X.,n,DeadAGI(a2billing.php|1)
exten => _X.,n,Hangup

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

[a2billing-cid-callback]
exten => _X.,1,DeadAGI(a2billing.php|1|cid-callback|34) ;last parameter is the callback area code
exten => _X.,n,Hangup

[a2billing-all-callback]
exten => _X.,1,DeadAGI(a2billing.php|1|all-callback|34) ;last parameter is the callback area code
exten => _X.,n,Hangup

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

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

[custom-a2billing-callback]
exten => _X.,1,DeadAGI(a2billing.php|1|callback)
exten => _X.,n,Hangup

[custom-a2billing]
exten => _X.,1,Answer
exten => _X.,n,Wait(1)
exten => _X.,n,DeadAGI(a2billing.php|1)
exten => _X.,n,Hangup

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

[custom-a2billing-2]
;For call through service
exten => _X.,1,Answer()
exten => _X.,n,Wait(1)
exten => _X.,n,deadAGI(a2billing.php|2)
exten => _X.,n,Hangup
I have attached the CLI trace of outgoing & incoming calls for you to examine.
Also I have made sure that I have the DID registered with a2b exactly as its being conveyed from my provider through FreePBX, and that there is a destination (extension) for the DID for incoming calls.
Please tell me what needs to be done.
Thanks again,
Neil
 

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tshif

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nsukho -

Thanks for the information. Dealing with the inbound call - the CLI trace.

Line 157 tells us much. The call has been received by a2b, and it has been able to associate it with a card number. A2b has tried to dial it on a trunk called SIP/grnVOIP/

Line 164: SIP/grnVOIP/ <== Is this a valid trunk on your system?

Lines 177-179
Called grnVOIP/127384007002
-- Got SIP response 480 "Temporarily not available" back from 70.42.72.104
-- SIP/grnVOIP-0a089ee8 is circuit-busy

Conclusion: The outbound leg is messed up. The call was answered, got into a2b, and was recognized, a2b tried using follow me (it appears) to dail out. Dial out failed.

I would trouble shoot this like any other outbound call failure from FreePBX. Start by temporarily changing your outbound route so that it uses the trunk Called grnVOIP. If you cant dial out - it has nothing to do with a2b.

Your outbound call log is even ore interesting.

Line 67/68:
-- Executing [s@macro-dialout-trunk:19] Dial("SIP/7002-b7b3eab0", "SIP/grnVOIP/1273840016624174556|300|") in new stack
-- Called grnVOIP/1273840016624174556

That looks a LOT more like a a2b card number than a phone number as its 19 digits long. According to the dial pattern, you are appending 16624174556 to the front of the phone numbers in FreePBX. Is this on purpose? I also still wonder if that trunk is a valid trunk.

There is also no sign of a2b being involved in this outbound call. Was the outbound trunk used pointed to the a2b-DID destination, or to an actual provider trunk? Cause from what I seem to be reading here - this outbound call is failing without ever touching a2billing.

Have a look around your system and let us know what you find -
 

nsukho

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Thanks

Tony;
Thanks for your help in solving both of the problems.
Inbound Calls: I had the wrong extension number in the DID Destination setup page of a2billing. I changed it to the correct extension number and it solved this problem.
Outbound Calls: The Outbound Route in FreePBX, the first trunk in the Trunk Sequence was grnVOIP instead of A2B/1. I deleted all the trunks in the Trunk Sequence except A2B/1 and it solved the outbound problem.
Again thank you for giving the hints that have solved the problems.:smile5:
Neil
 

tshif

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:cheers2: Excellent job finding the issues Neil!
It can be really daunting to look at all these details when your first getting started working with a2billing. Glad I could help!
-Tony
 

rossiv

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When I run install-a2billing, I get this:
Code:
root@pbx:/usr/local/sbin $ install-a2billing
-bash: /usr/local/sbin/install-a2billing: /bin/sh^M: bad interpreter: No such file or directory
 

jeffmac

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please do a search on "install-a2billing". This problem has been solved several times before.....

Jeff
 

nsukho

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customization of a2billing customer user interface

Is there any instruction (thread) out there that guide you through customizing the a2billing customer user interface (such as inserting logo, modifying header & footer, etc.)?
Thanks,
Neil
 

nsukho

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Tony;
Thanks, I have posted it in the a2billing forum.
Neil
 

manugmedia

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true the guys at Lylix knows their job. opened a ticket after my VPS went down after running updates ,and after 2 minutes in precise they sorted it out. What a customer service? Waooh very impressing.
 

nsukho

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inbound calls not being end up in a2billing CDR

I am working on my second PIAF server using the same a2billing setup instruction that is here in this thread after I had 1st PIAF server setup sucessfully about 3 months ago. On this second PIAF server, I got everything working the way it supposed to be except the inbound calls thru a2billing. The inbound calls pass thru a2billing sucessfully using the trunk, A2B/1, but those calls are not being shown up in the a2billing CDR.
I have followed the thread very carefully, even checked it at least five times and I cannot find anything in the setup. What could be done to fix this issue?
Thanks,
Neil
 

tshif

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Neil -
Have you resolved this?

Also - did you get a response from the a2b forum about customizing the ui? DO you have a link to your thread there?
 

nsukho

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Tony,
It is NOT resolved yet, I am still looking for solution.

Regarding customizing a2billing customer UI, I am still looking for the tutorial. As you suggested, I posted it in a2billing forum but no response yet and you may find that thread by clicking here.

Thanks for asking. Looking forward to hear some good news from you.
 

jbh

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a2billing and asterisk 1.6

Just a little FYI for anyone else setting up a2billing with asterisk 1.6.

I was getting the dreaded "your call could not be completed as dialed" - so looked at the logs and found:


[2009-07-14 13:42:09] WARNING[3337] pbx.c: The application delimiter is now the comma, not the pipe. Did you forget to convert your dialplan? (DeadAGI(a2billing.php|1))
[2009-07-14 13:42:09] WARNING[3337] res_agi.c: DeadAGI has been deprecated, please use AGI in all cases!
[2009-07-14 13:42:09] WARNING[3337] res_agi.c: Failed to execute '/var/lib/asterisk/agi-bin/a2billing.php|1': File does not exist.


I then saw on the a2billing forum that someone had come across this issue. The solution is to change the pipes to commas in both extensions_a2billing.conf and extensions_custom.conf.

It then works.
 

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