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-------------- 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

...ortant are per minute rates 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...

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.

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,

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

...ll have Digium quad T1 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?...

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

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

...transport=udp> Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 147.135.4.128:5060;branch=jkjk245kjelkjelkj2435sadflkj435.1sr From: "TEST PHONE"<sip:5555551313@147.135.4.129;user=phone;bvoice=ACME-06t5tpji5ub7e>;ta g=SD50vt601-662260634-1105742161664 To: "Deon Rodden"<sip:5555551212@sip.broadvoice.com;user=phone>;tag=as59299ec2 Call-ID: SD50vt601-8b297b5d0b4543648439f18f9eba5903-js19002 CSeq: 967783297 INVITE User-Agent: CSCO/7 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:5555551212@123.123.123.123> Content-Length: 0 Ans...

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

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

you can not ping that address because ICMP is turned off. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Deon Rodden Sent: Monday, July 26, 2004 2:22 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Broadvoice problems again Attn: James Greetings, C:\>ping 147.135.8.129 Pinging 147.135.8.129 with 32 bytes of data: Request timed out. Request timed out. Request timed out. Request timed o...

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

G'Day, I have a bit of FireFly problem that hopefully someone has seen before. What happens is if I make to or receive a call from the FireFly network the call will connect successfully. However, around 10 seconds after I answer the call I am disconnected. The weird thing is same thing happens if I make a call. I've had a look at the * console and I can't see that my * PBX drops

I had my asterisk configuration working very well with broadvoice, but it stopped working this afternoon. I plugged the Cisco 7960 phone I used for my origional signup (just a few days before they offered generic BYOD) and it works fine. I did notice it seems to do all of its comunication through proxy.broadvoice.com (I used tcpdump). I have never contacted broadvoice about using asterisk

...rom: Scott Laird <scott@sigkill.org> Subject: Re: [Asterisk-Users] RAID/SCSI/IDE/SATA and a TE405P (or T100P) card. Should I expect problems? Date: Thu, 22 Jul 2004 11:14:24 -0700 To: asterisk-users@lists.digium.com Reply-To: asterisk-users@lists.digium.com On Jul 22, 2004, at 9:41 AM, Deon Rodden wrote: > I'm confused. In the end, overall, which is best for a T100P (or even a > TE405P) card? IDE or SCSI? Raid or No Raid? > > I was anticpating putting a single Quad-Port TE405P inside a Dell > PowerEdge, > Dual 1.3ghz Processors, SCSI Hard Drive (No Raid). Was going t...

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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