similar to: Wildcard remote looping

Has anyone had successfully installed more then two digium wildcards in the same machine? I'm going for four. thanks Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd. Office: +972-3-9230686 ext. 179 Fax: +972-3-9216642 Mobile: +972-54-8000200

I am hoping to replace my Nortel 8x24 switch with Asterisk. Right now my cabling comes from my outside phone box into my office and into a punchdown block and leaves the punchdown block as an amphenol connector which currently plugs into the Nortel swicth. A second amphenol connector them plugs into the switch and extends to another punchdown block that I believe carries the lines throughout the

I had a hard time coming up with a subject. I have some SP IP500s that I'm using with *. When I access voicemail or get the busy announcement, I'm only hearing the last half of the first word or coming in on the second word. As example, when accessing voicemail, I hear ' have 1 new message.' It's like * is starting to play the announcement before opening the channel. This is

Anybody have a guide to the Cisco 79xx phones? One that I can give the 7 or 8 ppl in my office so that they can stop asking me questions. I was going to type up a basic guide but then decided I don't want to reinvent the wheel, one of you may already have one. I tried to use Cisco's guide but it's for their own protocol, a lot of options are different or rearranged. I need a basic

I'm working on an Asterisk system with all of it's PRI ports configured with the LBO setting at 5, like this: span=3,0,5,esf,b8zs As of yet, I am unwilling to change the LBO to 0 to where it probably should be because the system is working and I'm not sure exactly what the LBO does. I'm aware some changes were made to deal with low audio levels. Does anybody know what audio

Hello, I am currently pricing out various T1 and PRI options for a client of mine. We need voice and data - we want T's. Whether it be two seperate T's, two superate fractional T's, or one combined fractional T, we need it done. We are getting pricing and one provider is telling us that they have quality issues with the "Integrated Access" product. From what they

Below you will find, what I believe to be a typical setup with a T100P card. My question is - 1. Is this correct? 2. What kind of phones would be needed here... (Would you have to use Digital phones) And if so what would you recommend. PRI/T1----- | | | ---------------- | | | Channel Bank | | |

Hello, I was wondering if anyone had any ideas to a problem I am having. I have asterisk installed, and running. All extension phones are analog on FXS ports. The phones call each other and even transmit caller ID information but they DO NOT RING. Here is my setup: T400P Quad span T1 card currently connected to a CAC I w/ 12 channels of FXO for incoming lines and 12 channels of FXS for

Anyone have a suggestion on where I can get a decent new MB with 5v capable PCI slots. It seems like every decent server MB on the market has 3.3V slots only. Is diving into the junkbin my only choice if I can't afford to replace the 5v quad-T1 wildcard? Thanks Mark Farver

Can anyone clue me in about these T400 T1 cards I see advertised? I hear they are Digium Clones. Is there some reason to avoid these? How do they compare to TE410P's for example. Bart

It appears that my X100P card is only answering after two rings. Ideally, I'd like it to answer on the first ring. Here is the incoming section of my extensions.conf file: [incoming] exten => s,1,Answer exten => s,2,BackGround(demo-congrats) ; Play a congratulatory message exten => 1234,1,Goto(jgunther,1234,1) exten => 4321,1,Goto(mgunther,4321,1) exten =>

I know a few ppl have those CAC Adit 600's with t400 I can't seem to get my second span up on the T400 connected to the second spand on the adit (A:2) A:1 seems ok Can someone post they zaptel.conf span defintions And maybe a "print config" from the adit 600 cli I think my issue is timing srcs the coding, framing. bld out are all matched thx -------------- next part

Hi guys, I wanted to fetch some info from webpage and use it in my db. I read hpricot is one way to do it. But the problem is I''m unable to use it with my app. For some reason after successfully installing hpricot it fails when I try to use it in the rails console. I''ve been trying to fix this for the last four-five days. I''m stumped ! I tried re-installing everything

OK. Now I show my ignorance. What's the difference between a T-1 and a channelised T-1? I see that Covad's voip service (formerly GoBeam) requires a channelised T-1. Then I read recently on the list that many T-1s being installed are actually HDSL....which would be not a T-1 at all...right? Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist

I see there is the DigitTimeout application that sets the maximum time between digits before asterisk will interpet. Is there any way to control the minimum? We are having problems with incoming calls on our X100P where callers try to dial 10, but the 1 gets detected twice and they end up on extension 11. Thanks Mark Farver

> The documentation mentions that the Digium channels can be split into some > voice channels and the remainder of the channels used for routing IP > traffic. > > Does any one have this in use in conjunction with Asterisk? Does it work > well? Would you recommend it for a production server? > > Obviously, if this works, this makes for a cost effective platform where

Could someone explain to a non-US dummy the following phrases I have seen on the list. "I can provide you with tier 1 termination 6/6. I can blend or NPANXX breakout." "We provide US48 termination, blended rate for 1 MOU and above is .008 with 6/6." What is 6/6? What is US48? What is blended? What is MOU? What is NPANXX breakout? -------------- next part --------------

I have a pair of sites tied together with a T1 line running zaptel PPP on either end. The system works, but I keep getting these messages in the kernel logs, and users are reporting connection problems (TCP timeouts, and hangs) especially under high usage. --snip-- HDLC Receiver overrun on channel WCT1/0/1 (master=WCT1/0/1) HDLC Receiver overrun on channel WCT1/0/1 (master=WCT1/0/1) HDLC

Is it possible to configure * so that if a caller reaches voicemail for someone in Engineering, but doesn't want to leave a message they can press zero (0) and reach the Engineering Secretary or if they are calling someone in Accounting and reach voicemail, pressing '0' would reach the Accounting secretary, not the Engineering secretary? Don Pobanz

Modem connect speeds on calls through * seem to be lower than calls made through the telephone company lines or our old Rolm PBX. All data calls have 2 wire analog modems on both ends. For my set up I have channels of a Zhone channel bank tied to 2 modems. The Zhone channel bank interfaces my * server with a T400P card. modem --- Zhone Channel bank - * via T400P card - Zhone channel bank -