Re: [asterisk-users] (OT) DS3 Barrel/T-connector/Adtran MX2800 Problems
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 back on the left on the amphenol connector. If you connect them so the cable comes out the left side, it's pretty easy to secure them on the right side with a screw and on the left side with the zip tie. I think mine even came with little 4 zip ties that did the trick without looking ugly. If you have them connected the other way around (e.g. screw on the left, cable on the right), make sure you take out the connector for DC too, as these will stop the connector from being in securely. No zip tie on that side though. David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Clustering
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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can IAX be used without going thre a IAX server
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 using VPN or other method to acomplish this? Yes. I do IAX2 to IAX2 all the time using either IPSec or GRE (Usually both). David ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MCI vs. XO/Allegiance
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 once I open a ticket mysteriously gets fixed without a good explanation. Eventually everything works, but you have to beat on them continously to get things done. Better than dealing with SBC though. David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
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 (64kbps). A T1 has no seperate signaling line - You're thinking of PRI. T1 gives you 24 DS0 (64kbit) channels, which you can do whatever you want with. PRI just shanks off one channel for D channel signaling. David -- David J. Coulson email: [EMAIL PROTECTED] web: http://www.davidcoulson.net/ phone: (216) 920-3100 / (216) 258-4942 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
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 to POTS lines, use them all for data or mix it up and run voice and data over the same T1. Telco vendors don't care what you do with it, unless it's terminating for data/voice in their equipment. Even when you use all 24 channels for data, they still function as 24 distinct DS0 channels as far as timing is concerned. Unlike OC-nc circuits (Where you save some overhead for the sake of being unable to channelize the STS channels) , there is no overhead variation when channelizing a DS-1 versus using a full DS-1 for data. David -- David J. Coulson email: [EMAIL PROTECTED] web: http://www.davidcoulson.net/ phone: (216) 920-3100 / (216) 258-4942 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PLEASE HELP Adit 600 went kaput?
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 definatly screwed up :-) 1-4 2-5 That was how I always learned to wire a loop plug anyway. David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MeetMe ztdummy
[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 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on 64bit ?
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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NuFone?
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: [EMAIL PROTECTED] Linux Developer / web: http://davidcoulson.net/ Network Engineer phone: (216) 533-6967 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium connectivity issue?
John Fraizer wrote: There are several entities out there who will do secondary DNS for free. You might want to look into that. If you pull their NS entries from one of the root servers, you get: digium.com. 172800 IN NS bos.nameserver.net. digium.com. 172800 IN NS linux-support.net. digium.com. 172800 IN NS marko.net. digium.com. 172800 IN NS phl.nameserver.net. digium.com. 172800 IN NS rdu.nameserver.net. digium.com. 172800 IN NS sjc.nameserver.net. digium.com. 172800 IN NS sou.nameserver.net. ;; ADDITIONAL SECTION: bos.nameserver.net. 172800 IN A 203.20.52.5 linux-support.net. 172800 IN A 216.207.245.1 marko.net. 172800 IN A 216.207.245.12 phl.nameserver.net. 172800 IN A 203.56.139.102 rdu.nameserver.net. 172800 IN A 64.245.56.205 sjc.nameserver.net. 172800 IN A 205.158.174.201 sou.nameserver.net. 172800 IN A 194.196.163.7 Looks like they just didn't update their digium.com zone to match. David -- David Coulsonemail: [EMAIL PROTECTED] Linux Developer / web: http://davidcoulson.net/ Network Engineer phone: (216) 533-6967 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * and Cisco 7910
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 interested to learn about the configuration. David ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial / Ring multiple sip channels
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 phone: (216) 533-6967 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming IAX2 problems with NuFone
I've been using NuFone with Asterisk for a while, but I've started seeing this error with incoming calls: NOTICE[114696]: File chan_iax2.c, Line 4581 (socket_read): Rejected connect attempt from 216.234.116.189, requested/capability 0x4/0x4 incompatible with our capability 0xff03. Outgoing 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 phone: (216) 533-6967 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
