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...ates to you? how many minutes do you expect to use per month? We are in Tampa Florida and have 15 T1s from several different providers so I may be able to refer you to one if it's a match to what you're looking for. MATT--- -----Original Message----- From: Deon Rodden [mailto:drodden@webunited.net] Sent: Thursday, July 22, 2004 8:53 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Can anybody recommend a good T1/PRI provider? We're in South Florida, right now we primarily use Xpedius PRI and 2 IDS PRI's. We were looking at getting a MCI PRI but upon reviews from...

-------------- next part -------------- An embedded message was scrubbed... From: Deon Rodden <drodden@webunited.net> Subject: Re: [Asterisk-Users] Strange Quality problems with Asterisk, Gentoo, Redhat and Kernels - /dev/dsp Date: Thu, 30 Sep 2004 09:05:39 -0400 Size: 5509 Url: http://lists.digium.com/pipermail/asterisk-users/attachments/20040930/289c69cc/dsp.eml

Hello, I am looking for a user test automation tool. When I say ''user'', I mean it should mimic the user interacting with the app. That ultimately boils down to a tool that drives a browser or is able to make http requests and allow us to deal easily with the responses. Our requirements are to be able to write the tests in Ruby (no need to switch to other language for test

Anybody have any experience with blocking numbers in the U.S's Do Not Call list? We have a customer that will be getting their own Asterisk server from us, and they want it to be check outbound numbers against the do not call list; this is for a backup, in case there's a slip up and one of their people try to dial somebody on the do not call list. The list has millions of numbers, and

...cards and they all have from 2 to 4 T1s hooked up to them. We have had no noticable problems with dropped calls/poor quality. What are you looking to do with this system? what kind of traffic will be going through these 4 T1s? MATT--- -----Original Message----- From: Deon Rodden [mailto:drodden@webunited.net] Sent: Thursday, July 22, 2004 12:41 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] RAID/SCSI/IDE/SATA and a TE405P (or T100P) card. Should I expect problems? I'm confused. In the end, overall, which is best for a T100P (or even a TE405P) card? IDE or SCSI? Raid or No Ra...

Good day all I have figured out most/basics of asterisk.I went with sip and made my own sip.conf and extensions.conf No I have 2 servers running sip in different towns.Both is connected with static ip so thats fine,but now. Lets say someone want to call someone else in the other town.How do I get asterisk to know,for instance sip extension 101 is on another sip server on a different ip. And I

I've tried to compile rate-engine 0.5.2 on Fedora Core 1, Redhat 9 and OpenNA Linux 1.0 and all give me an "Error 1" after typing "make" but with no real reason given. Just a few standard/non-critical warning messages, and then suddenly "Error 1" Anybody successfully compile Rate Engine? The least cost routing module for Asterisk? Thanks in advance.

For almost 6 months now I've upgraded Asterisk every couple of weeks or so and I've never had this problem. When I'm at the asterisk console (asterisk -r) it shows me live status. Who called who, what it's playing and when, etc. It logs to the screen. When I type reload, it says "added so and so to so and so context" gives me some long display as it reloads. But

If I relay through my Cisco 7940/7960, does it do QOS, even with a dumb switch? I know you can set quality/qos but only if you have a layer2/layer3 switch that supports the tagging. A simple little linksys 5 port switch wouldn't know about QOS, it'd give everybody equal priority. If a computer plugged into the phone, and the phone into the dumb 5 port switch and then to the internet,

Hi I want to play a bit with Asterisk. I currentlly install a new system for that and I would like to get your recommendations regarding the linux distro to use there. This is NOT intended to become a general distro flame war. My favorite distro is ******** and no argument that you flame will convince me here (probably because I've heard it before). However I would like to minimize the OS

How do I limit the length of an extension? In my test IVR/Automated Attendant (whatever it's called), at the beginning it plays "if you know your parties 3 digit extension, you may enter it now) and then it gives a list of options. If the caller puts the 3 digit extension, it goes through fine, if they press 1, or 2 it goes to the selected menu option, but if they dial 91235551212 it

I've compiled Asterisk on Redhat 9 and Fedora Core 1 in the past, generally without any problems. Especially w/ the stock kernel, which I generally loathe. When I tried to upgrade my Redhat 9 Server to the 2.4.27 kernel, doing a manual/clean compile, I had massive quality issues. I was forced to go back down to a stock 2.4.24 kernel. Never figured out why. Now, I've installed Gentoo

Anyone have any thoughts on why versions of asterisk I try (4 so far) after CVS-07/18/03 always end up locking up on me... which means no sip clients can register/re-register and if I type "reload" or "stop now" at the cli it just returns and does nothing. I have experienced this same issue on three separate boxes. Two running RedHat 9 and one running Redhat 8. I don't

Is it possible to restrict the number of concurrent calls made to a SIP peer? Or maybe the number of concurrent calls made to a particular extension. This way I can avoid asterisk trying to make more voice calls to my remote SIP gateway then I have bandwidth to handle. /davidh

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Last night I updated to a custom 2.4.27 kernel, I also upgraded asterisk. This morning I discovered Asterisk is no longer playing sounds to users. ie when they go to the voicemail, asterisk says it's playing vm-login but the user never hears it. It's not a firewall issue or anything like this, as it worked before the upgrade. I thought maybe the latest CVS was the problem

We originally purchased a TE405P so we could connect our Asterisk server directly to the T1 PRI from our provider, however due to all the problems reported with T1 PRI's interacting directly with Asterisk, we scrapped that idea and decided to stick with our Cisco router making the conversion to SIP. Anyways, we found a new potential use for it. We have an old location with an existing PBX,

I've been working with Asterisk for about 2 months now and am doing well. However I decided to switch platforms from Fedora Core 1, that my predacessor was using, to Gentoo, for obvious reasons. It just seems faster and less "bloated" everything I need, nothing I don't. Anyways, I've read what the Wiki had to say about it and I was only confused on one thing, putting

Are there any sip providers out there providing full business telephone service. Not just single line/residential service like I have seen with vonage etc. For example take a company currently using a legacy pbx connected to the PSTN with a PRI. I would like to replace this setup with a data T1, an asterisk box, and some SIP Phones then pass all calls (local and long distance) directly

I don't know if this is a problem with my cisco sip IP Phones or asterisk but I thought I would post here in case someone else has experienced this issue. When I make a call from my SIP cisco IP Phone to some remote IVRs I never get the rest of my soft keys, only the "End Call" soft key, and also DTMF doesn't work... its like the phone is acting like the remote end hasn't