similar to: Benchmark

Hi all, i'd like to know if there is a tool like an injector that permits to increase progressively the load onto a Samba server. My aim is to define a threshold as the number of concurrent users connected and accessing shares simultanously on a samba3 PDC. I don't care about network transfer rate. thanks for your answer

Dear List, i have been thinking of building a calling cards solution based on Asterisk and a2billing.. i have a few questions regarding this solution and was hoping you may have the answers and could be generous enough to offer them. the servers i'm thinking of are with the following Specs: Processor: Intel X3210 Ram: 8Gb HDD: 2x500 GB Sata Internet Link: 100mbps Dedicated was thinking of

Hi all, I'm planning to build a VOIP solution for handling SIP calls coming from endpoints registered on a specific SIP proxy...I made some research regarding network architecture and found out that the best solution is to use OpenSips as SIP proxy for registration and local calls between registered endpoints and use asterisk server with a2billing for PSTN calls, rating, routing and all other

Hi: i have a hosted server with asterisk and a2billing as a billing plattform, when i am trying to enter the server remotely by ssh, memory error message displayed: -bash: fork: Cannot allocate memory i have 1GB RAM on the system ,and there is 15 to 25 concurrent calls on the system is'nt 1GB of RAM sufficient for this volume of calls on Asterisk. and when iam using "top"

Hi James, I tested (as I wrote a few days ago) the latest PV drivers release, and they seem to work correctly apart of the "Safely remove Xen Net Device Driver" tray icon glitch. I made a series of benchmarks with PassMark''s PerformanceTest on 6 twin concurrent VMs with the PV drivers installed, both with the /GPLPV switch on and off, and on one lonely VM, still in both

We have an issue with our Asterisk install where Asterisk produces many Zombie processes (on the order of several hundred per minute) until either the Asterisk server is restarted (and the zombies die a natural death), or the kernel runs out of PID space (happens within hours) and brings the system to a halt. This problem only happens when the server is under some non-trivial load. We were

Dear DovecotORG, In my organization, we are about to implement Qmail Server. * The number of current users will be 800, in future it may increase upto 1200. * The number of concurrent users will be 300. I am the engineer to deploy the Qmail in Linux server. I need to tell the storage team on the IOPs requirement. I requested 8TB usable space for the mail storage (can

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]

Hello, I am looking for information on setting up digium FXO card for use as a PSTN Gateway (H323-PSTN) to work with GNUGk. I am basically looking for the setup and it would be great if anyone can share his experiences with the same. Also, if there are any limitations in going for such a setup and problems that may arise/things that I should keep in consideration. Thanks & Regards, Deepak

Hello everyone. I'm using Asterisk 1.4.31 and A2Billing 1.7.0 to manage a small wholesale operation, so I configured A2Billing for not to answer the call nor play any greetings or balance notifications to the caller. I'm authenticating each customer by it's IP address, and each customer has it's own context, in which I set the following: ;=====in extensions.conf======

Adam, I thought Andrew Kohlsmith gave the individual good advice without intentionally malaciously spitting in the guys face. For the question, " 'Whats the ' Minimum CPU required for 60 calls? " I think a Pentium 3, high end, which is cheap right now, should do fine, but you will need either 3 T-1s or arrange for the calls to come in via SIP, but you will still need more

Hello everyone. I'm using Asterisk 1.4.31 and A2Billing 1.7.0 to manage a small wholesale operation, so I configured A2Billing for not to answer the call nor play any greetings or balance notifications to the caller. I'm authenticating each customer by it's IP address, and each customer has it's own context, in which I set the following: ;=====in extensions.conf======

Why is using ulaw or alaw an unlikely scenario? I wouldn't use anything but ulaw\alaw. The Bells can compete on price and will if they have to. Where they CAN'T compete is quality. If there were something better than 711, I'd offer that. Well, there is 722, but not many things support it. ---- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ----- Original

hi all, I have a little problem. I'm trying to configure a2billing (asterisk2billing) with asterisk. Everything done successfully but when I try to call following error occur "WARNING[9690]: pbx.c:3170 pbx_extension_helper: No application 'DeadAGI,a2billing.php' for extension (a2billing, 456,3) and it hang ups the call. Can someone please tell me why this error occuring. My

Hi! Maybe someone could help me out? When a call is routed via a2billing AGI and user does a transfer, the call is dropped. If the trunk is called directly everyhing works. Here's a direct scenario (working fine): [pbx000001] exten => 101,1,Set(__TRANSFER_CONTEXT=pbx000001) exten => 101,n,Dial(SIP/pozitel/37129238254,45,t) exten => 102,1,Dial(SIP/12345,60) so, when user calls ext

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 Everyone, Trying to run a php script after DeadAGI for A2Billing does it's magic. This is the dialplan: [a2billing] exten => _X.,1,System(php pre-call.php ${CALLERID(num)} ${EXTEN} ${UNIQUEID}) exten => _X.,n,AGI(a2billing.php,1) exten => _X.,n,Hangup() *exten => h,1,Wait(5)* *exten => h,n,System(php post-call.php ${CALLERID(num)} ${UNIQUEID})* As you can see above, I even

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

Hi all, I am trying to set up DND in my asterisk, I am using the following context: [app-naoperturbe] exten => *11,1,Set(DND=${DB(ddisturbe/${CALLERIDNUM})})exten => *11,2,GotoIf($["${DND}" = "YES"]?*11,3:*11,101)exten => *11,3,Set(DB(ddisturbe/${CALLERIDNUM})=NO)exten => *11,4,Playback(beep)exten => *11,5,Hangup()exten =>