similar to: RealWorld Stats; Not achieving expected results

Hey guys, I'm looking for some realworld specs on somebodys machine that will work with the Digium 4-port T1/PRI card and that will support 96 simultaneous calls. Dell is soon to release the PowerEdge 1850: 2U, Dual 3.6Ghz Xenon, 1Gb DDR2 RAM, Dual 36GB Ultra320 SCSI RAID, Hot swap Powersupply, one 64bit 133Mhz PCI and one 64bit 100Mhz PCI for about $3,000. Tack on a 4 port Digium card and

Hey gang, We currently have a class 3 switch (CSX) that..well..it sucks. It does terrible CDR writes, doesn't support LCR, the list goes on and on. We want to replace this with several asterisk boxes each running one or two 4 port PRI cards. The problem is: I can plug in 20 PRI lines into the CSX (from PSTN) and have 1 come from CSX into asterisk. If 1 call comes in on each of the 20 pris,

Server A has 4 PRIs coming into it, and provides a SIP connection to Server B. When we call into server B via a DID from one of Server A's PRIs, we get crackly sound on recorded playback only (prompts, IVR, voicemail instructions, etc.), but not on actual live voices once you're talking. Server A provides this same service to other servers and problem does not appear in those cases.

I'm trying to determine if this is a zaptel issue or a sangoma issue. When I start our server, it gets to the point where it says "starting wanrouter" (thats the sangoma drivers) and it just hangs there. Nothing happens. Can't even use console. I unplugged the 4 PRI lines and rebooted. This time machine comes up. I logged in. While logged in, I plug the 4 spans back in. I try a

I am currently working on an Asterisk test system, and will be presenting a demo to the Board of Directors tomorrow night. I want to make sure I have all of my ducks in a row. The Asterisk system will be used to replace a Norstar MICS. The location has two PRI's coming in, with a few hundred DIDs. I know how to make * use the DIDs incoming, and I know how Nortel uses the DIDs. Now for the

Hello, I have been trying to get my coders to work without a conversion. I have read all the available asterisk documentation and support groups without any luck. Here is my issue. (Please feel free to ask questions if you do not understand what I am talking about.) I am using Cisco ATA-186 set to g729 codec. (But it will switch to g711 if sip-server request g711) I have 2 SIP-services to

I am having classic "frame slip" symptoms on 4 different systems with 4 different providers (all full PRIs, Qwest, XO, Xspedius, and First Digital are the providers). By classic I mean pages cut short. I do not hear clicks on calls however, and faxing from a fax machine plugged into the asterisk box via an fxs port (no VoIP) faxes just fine through the PRI. Also, faxing directly from

Hi all. What is the best hardware configuration to handle this following scenario? - 4 IVR menu with conference applications for each option; - Only SIP/g711 user access - 3500 simultaneous users(800 at the beginning) - No ZAP channels Where is the most important point of failure? CPU? Ethernet? RAM? Im planning to separate in three servers: Server01: 01 Xeon 3Ghz getting the 1st level of

I've got a system I'm putting together to handle IVR calls with * I have one head system that terminates two PRIs. It routes the calls from the PRIs to * boxes using IAX I'm planning on having four or five * boxes. The * boxes run AGI scripts to process the IVR calls. Can I load balance the routing if I have five calls each of the IVR * boxes gets two call and the next call would go

Does anyone have any recommendations on how to connect 160 analogue phones to an asterisk PBX? Background information: A client wishes to replace their current PBX with a new VoIP system. Currently they have 2 PRIs. I intent to set up 2 asterisk PBXs with Debian GNU/Linux on raided drives. These drives will be mounted only read-only to recover gracefully from power-cycles. I am considering 2

Here is the setup: Phone A (in NYC) on own bandwidth. Phone B (in LA) on own bandwidth. Asterisk box in Houston,TX on own bandwidth. Both phones contact asterisk to register. Not much bandwidth used for this as it is a few packets every hour or so. Phone A calls Phone B. Phone A sends a call request to asterisk and asterisk calls phone B. Both phones are connected and both people are talking.

Hi everyone, I'm looking to buy a server that could handle 100 IAX users (g711)-(about 300 registrations) simultaneously. No zap channels. My budget is 1000$ us, Is a fast (3ghz) single server more reliable than a double cpu (like 1ghz) ? Will asterisk take full profit of two cpus? Isn't better to get a second cpu to handle system processes (like stat generation, backups...)? so that the

Hey all, I have an as5300 I use for dial in customers, we have 4 PRIs on it. We have a few free channels on it. I'm wondering if I setup SIP on the as5300 I can have asterisk use the free channels for dial out. I'd still have to use my TDM04B for incoming calls, but at least I can expand my outgoing. Anyone done anything like this before? I've never messed with VoIP on Cisco

We have recently started routing about 3 PRI's worth of traffic thru our asterisk box. The text on the console now flys by so damn fast, I can't really see what the heck is going on. Even with verbosity 0 and debug 0 it is still so fast. Is there some way I can attach to the console in a way that will allow me to grep or otherwise filter the text so I can focus on something in

I'm looking to use * for a HQ/branch office topology with fairly few calls over the WAN. The questions I have all pertain to the following architectural pic: http://www.45891.com/misc/arch.jpg I'm looking at a distributed architecture so users are somewhat functional when the link to HQ is down, with a centralized voicemail server to allow for transfer of voicemail messages from user to

Check with your telco. We had the same problem on 1 of our PRI's, every day at 5:00 sharp, red alarm, with all calls cut off for 30 seconds exactly. Turns out the equipment at the CO was going into a test loop at that time because of a forgotten setting by a tech. Man, what a finger pointing exercise that was. -----Original Message----- From: Matthew Boehm [mailto:mboehm@cytelcom.com] Sent:

Per my conversation below with digium, are there any legal alternatives to digium's G729? It is out of date, and doesn't support VAD nor silence detection. Digium has stated that they have no plans to update it anytime soon. VAD/Silence is a big deal with major carriers and we are having to fight a battle to get them to make special arrangements to turn off VAD/Silence in their

Thank you for your advise, I'll find something with a lot of memory.... Adrien -- Adrien Laurent - CIO 514-284-2020 adrien@modulis.ca www.modulis.ca -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jason Walker Sent: Saturday, October 01, 2005 12:25 AM To: 'Asterisk Users Mailing List - Non-Commercial

Anybody have suggestions on having a 56K dialpool and VOIP connections with an Asterisk box over the same set of PRIs? We've done the PM3 with PRIs for just dialup, but are looking for a way to integrate our Asterisk box and move our voice calls onto the same PRIs. Ian -- Ian White Victoria Free-Net Association email: iwhite@victoria.tc.ca http://victoria.tc.ca/

Anyone know the gsm encoding mip requirement from g711? Or number of channels can be transcoded from g711 to gsm at a time. Thnx