similar to: Implementing G729 Passthrough - VM recordings, maybe even a2billing

Hello Everyone, We are currently experiencing some higher load on our servers, and since signaling comes into our servers on G729, we would like to implement G729 pass-through. A few questions arise, do we need to convert all the recording to the codec, and what about voicemail? We are also using A2Billing (hope I am not violating any thread rules), and would like to convert all that recording

Not sure if a2billing can be shared here, but ill give a shot If the credit < min_credit the IVR play: sorry you have 0 credit and hangup, I want it to FW me to the IVR to add voucher, please let me know: here is log: [18/10/2010 07:01:12]:[file:a2billing.php - line:75]:[CallerID:]:[CN:]:[IDCONFIG : 1] [18/10/2010 07:01:12]:[file:a2billing.php - line:76]:[CallerID:]:[CN:]:[MODE : standard]

I am pleased to announce the new version of Asterisk2Billing, V1.3.0 STABLE (Yellowjacket) PROJECT URL : http://trac.asterisk2billing.org A2Billing has completely re-written some its modules such as : Invoicing, template management with Smarty, the call-back, added new methods of online payment integration with Moneybookers and Authorize.net in addition to Paypal. A2Billing have also improved the

Hi All, I am newbie in this asterisk and a2billing technology . i had successfully installed asterisk in my server fedora -8 [server behind NAT/STUN] i after installation i can able to create users and tested the call features with X-Lite . the was working fine . after i installed the A2Billing in my same server with follow the steps from a2billing installation guide. but u cant access the

Hello All; After trying A2Billing and certainly when the trunk is analogue lines (FXO ports), I faced a problem that the channels were not hanged up properly from time to time which cause us to do restart for the dahdi. Without A2Billing, I was able to handle the Dial scenario properly and no hanging for the analogue channels and no need to restart dahdi from time to time.? Really I would if

Hello; I am facing a trouble with A2Billing when using analogue lines because the channels are not closing properly when dialing happen through A2Billing (it seems the dialing scenario including the hangup is not handled properly through A2Billing but I do not have control on this). But when I do dialing from asterisk and using analogue lines, I do not face a trouble because I can write the

I am running into an issue with A2Billing. I will explain first of all that everything else works! the system is 90% complete its just this one small problem I am running into. So my problem is that when I place a call, 1. I dial my number that I want and A2Billing gets activated 2. it asks for my pin, upon successful entry of my pin A2Billing then 3. prompts me for my phone number then 4.

hello, I just installed a2billing, I did all the config, at least I guess .. but I still can not integrate sip accounts that I had created (with sip.conf ) in a2billing to apply their billing .. could someone tell me how to make the junction between asterisk and a2billing?? I also noticed that the file additional_a2billing_sip.conf : was always empty ... -------------- next part --------------

Hello All, I got one quick question on A2Billing. Specs: - - A2Billing v1.3 - OS CentOS 4.5 - Asterisk 1.2 - Zaptel 1.2 Did the installation and everything is working as it suppose to... Using the A2Billing documentation, I created the RateCard, SIP Trunks, and SIP Customers. I was also able to login using XLite Dialer and was able to call out to my SIP Trunk also. Now how can I remove the

Hi all, I have configured asterisk and a2billing.for inbound i have also configured did and its forwarded to sip extensions. But i want to enable queues with inbound numbers(DID).But i could not find a way to do this in a2billing. I want enable that if some did comes to asterisk/a2billing it should be forwarded to queues not sip extensions and their i want to enable hunting so if one

Hello List. I just installed a2billing with asterisk 1.6 and got it working. The only problem is that I'm trying to setup something to manage who's using the most minutes in the house. I noticed a2billing only works for callin cards setups, or maybe I didn't configure it correctly for what I want. Can I use a2billing for "?VoIP residential services"? if yes, how? if no,

Hello, im having trouble with the following: [Asterisk]<------>[ISP]<------>[ADSL Modem]<------>[Linksys Router]<------>[Grandstream ATA]<------>[Analog Phone] On server: - Asterisk 1.6 - A2Billing 1.4 A2Billing have 2 Trunks: - TrExt: Voip Provider - TrInt: Internal Calls This structure works on first day (Asterisk+A2Billing installation/configuration). But on

hi, i have installed a2billing , when i open /admin web pages. errors as follow: Fatal error: Call to undefined function bindtextdomain() in /usr/local/src/a2billing/common/lib/languageSettings.php on line 130 do you know what's wrong? -- Thanks, Sucan

Hey there, I am trying to setup a2billing on asterisk 1.6 , but ,when I try to access its web page I see the a2billing directories:Index of /a2billingNameLast modifiedSizeDescriptionParent Directory -admin/15-Sep-2010 19:19-agent/15-Sep-2010 19:21-common/15-Sep-2010 19:18-customer/15-Sep-2010 19:20-Apache/2.2.9 (Debian) PHP/5.2.6-1+lenny8 with Suhosin-Patch Server at Att, Flavio Roberto

Hello Everyone, We are looking to transition our 23 channels from testing/lab into production. During testing we used the free open source g729 license using the instructions found here: http://blog.manhag.org/2010/05/installing-the-free-g729-codec-for-asterisk/ A little more about our setup. All recordings have been converted to G729, all voicemail messages are also in G729, finally

hi, Does A2Billing has mial list? -- Thanks, Sucan

Hi List; I am going to use a post and prepaid billing with my Asterisk vesion 1.4, can anyone advise me which of the billing software will be more stable and it is opensource to be used: ASTCC or a2billing? Any advise that might help to take the proper decision? Regards Bilal

Hellos, I want to setup Asterisk+a2billing for over 10,000 extensions for voip resale. Has anyone done this before. What are the hardware requirements and challenges? James -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090513/186b2bde/attachment.htm

Hi All; Anyone has tried to use A2Billing with Asterisk 1.6 and 1.8 to confirm that is working fine and it is same as 1.4? Appreciate ur kindly help. Regards Bilal

OOH323 Debugging Enabled -- Executing Answer("SIP/3513-090f7d40", "") in new stack -- Executing Wait("SIP/3513-090f7d40", "1") in new stack -- Executing DeadAGI("SIP/3513-090f7d40", "a2billing.php|1") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php a2billing.php|1: line:58 - IDCONFIG : 1