Re: [asterisk-users] (OT) DS3 Barrel/T-connector/Adtran MX2800 Problems

2006-08-04 Thread David Coulson
BJ Weschke wrote: It's a fairly common issue, and unfortunately, there isn't a best practice solution that I've seen people use that isn't ugly. In a prior job they zip tied the cables down to the connectors and this fairly reliable. At least on my MX2800s there is a loop for a zip tie on the

Re: [Asterisk-Users] Clustering

2006-03-11 Thread David Coulson
From what I can find online, OSPF seems to be a technology or method, not necessarily a program. What are you using to perform OSPF? OSPF is a routing protocol. Quagga (quagga.net) is a good open source implementation of OSPF for Unix. David

Re: [Asterisk-Users] Can IAX be used without going thre a IAX server

2005-10-18 Thread David Coulson
Chadwick E. Labno wrote: Is it possible to route a call from an Asterisk box through the Internet to a IAX device (in this case Digium IAXy) without using an IAX service like IAXTel? I have it working on my local Ethernet LAN so it should be possible to use VPN to cross the internet. Anyone

Re: [Asterisk-Users] MCI vs. XO/Allegiance

2005-06-13 Thread David Coulson
Wiley Siler wrote: Anyone out there using ISDN PRI from either MCI or XO/Allegiance? We have a DS-3 full of PRI from X/O. They work great, mostly, but their tech support sucks. They screw up number ports all the time and about every week there is some local number I can't dial to via XO which

Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?

2005-06-13 Thread David Coulson
Nir Simionovich wrote: Now, E1 and T1 lines are based upon a channel based connection, which means you get a line with X number of data lines and a single control/signalling line. On T1 it means that you have 23 lines dedicated for Voice/Data (each is 64kbps) and a single signaling line

Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?

2005-06-13 Thread David Coulson
Leon Sun wrote: Not really true about T1 description. When you apply for T1, you need tell vendor if it's channelized or non-ch. If you are going to use it for 1.5M network, you need use unchannelized T1. T1 is T1. How you use the DS0s delivered across it is up to you. You can mux them out

Re: [Asterisk-Users] PLEASE HELP Adit 600 went kaput?

2005-02-12 Thread David Coulson
Andrew Kohlsmith wrote: Unplug it and plug in a loopback plug (pin 1-5, pin 2-6) -- if the T1 alarm doesn't go away, the T1 controller itself is kaput. If it goes green (or off), then your wire is suspect. If he gets a green light with a loopback plug wired like that, his controller is

Re: [Asterisk-Users] MeetMe ztdummy

2005-02-04 Thread David Coulson
[EMAIL PROTECTED] wrote: I've heard that 2.6 kernel does not need usb hardware for ztdummy to function. Maybe someone else can confirm... although this would require a complete reinstall for you. I have ztdummy loaded under 2.6 without USB hardware and it works fine. David

Re: [Asterisk-Users] Asterisk on 64bit ?

2004-06-27 Thread David Coulson
Dr. Rich Murphey wrote: How do you balance the number of active connections per server? www.linuxvirtualserver.org Does weighted least connections balancing of connections - Not sure how you'd make sure the SIP RTP session hit the same physical box though. David

Re: [Asterisk-Users] NuFone?

2004-03-18 Thread David Coulson
Linus Surguy wrote: ob: Magrathea offers A-Z IAX termination, origination blah blah blah blah. I asked a while ago, and you passed me to a reseller who never answered my question - How much to terminate a call in the UK? David -- David Coulsonemail

Re: [Asterisk-Users] Digium connectivity issue?

2004-02-15 Thread David Coulson
-- David Coulsonemail: [EMAIL PROTECTED] Linux Developer / web: http://davidcoulson.net/ Network Engineer phone: (216) 533-6967 ___ Asterisk-Users mailing list

[Asterisk-Users] * and Cisco 7910

2003-12-18 Thread David Coulson
I've looked on Google, but I get mixed results. I understand the 7910 does not support SIP, instead using SCCM, but does it work reliably with Asterisk? Is there a specific firmware revision for the device that makes it work? If anyone has this phone working with Asterisk, I would be

Re: [Asterisk-Users] Dial / Ring multiple sip channels

2003-12-11 Thread David Coulson
Paul Oster wrote: exten = 101,Dial(Sip/101,10) Dial(Sip/102,10) Almost there: exten = 101,Dial(Sip/101Sip/102,10) David -- David Coulsonemail: [EMAIL PROTECTED] Linux Developer / web: http://davidcoulson.net/ Network Engineer

[Asterisk-Users] Incoming IAX2 problems with NuFone

2003-12-07 Thread David Coulson
works just fine, but I can't get incoming to work at all. Any ideas? I googled for the error, but I couldn't find anything. David -- David Coulsonemail: [EMAIL PROTECTED] Linux Developer / web: http://davidcoulson.net/ Network Engineer