similar to: Ordering digital trunks?

Hello, Did anybody successfully tried upgrade Adtran TSU 600 to firmware which is working properly with T100P and asterisk ? B.

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031012/59c113df/attachment.htm -------------- next part -------------- Hello, We are planning to buy following Hardware for Asterisk TestBed. Please let me know if this seems fine to you. 1. IP Phones ( 5 in number) CISCO 7940/7960, SNOM 200, Pingtel xpressa 2. Wildcard T100P interface card,

Greetings, According to the FAQ "When you load the module and have no circuit/channel bank the LED's should flash red" I get the "knight rider" lights before the module loads, but after the modules are loaded I don't get any lights, other equipment plugged in or not. The card has been on my shelf a while so I wonder if it needs a firmware upgrade ? Can the cards

anyone have a recommendation for a place I can buy cheap motherboards that supports those 64-bit 3.3 volt PCI slots for the T100P ? I can't find them at Fry's or anywhere locally. All I can find online is dual processor server boards that are overkill for this application. I would like to use a P3/ P4/ AMD single processor. No Xeons or dual processor junk. Anyone know why digium

The quick question: Do the digium drivers for the Digium Wildcard TE410P (4 port T1/E1/PRI 3.3v card) , the T100P (single port T1), and the TDM400P support hot plug PCI? I am also noting that while the TDM400P doesn't state the voltage requirements, it looks like a 5v card. I hope that I am wrong on this. What type of T1 channel bank would people recoment for ISDN (for video

I bought an Aastra PT480 from digium, but I wanted to see if I could get some more help with this before Monday. Any help would be appreciated. I have the phone connected to the TDM400P card, and I also have the T100P and the X100P in the same box. My problem is, it appears as if the phone and asterisk can't understand each other. The port the phone is connected to always remains

Hi list, I have a terrible noise issue with Dell SC1430 + Digium TE110P. The digium card is not sharing interrupts with any other device, as I saw in Dell's BIOS and also with "lspci -vb" command. After changing coax wire, UTP, balum, digium card ... I have found that the problem is in Dell box, so now I'm running the same Asterisk config in other server with the same

Hi! I have just bought the X100P and TDM400P cards to install on my computer to implement the PBX. I also downloaded the newest softwares asterisk_ver0.5.0, libpri_ver0.4.0, and zaptel_ver0.7.0) to install on my computer (Red Hat Linux 8.0). All packages are compiled well. When I use "modprobe" to load drivers (modprobe zaptel, modprobe wcfxo, modprobe wcfxs), the first two (zaptel,

Hello All, I could use a recommendation if anyone has a moment. I have the T100P but I have not gotten my service yet. I want to have at least 12 lines of digital voice with DID. Should I just seek out a PRI ISDN provider or is there something else I should look for? I want to keep cost as low as possible. Also, I want to own my own router for the phones since it is always a hassle to get

Folks, I recently obtained a Western Electric multi-line phone and am seeking help with getting this beast working with *. The interesting stuff in my * implementation consists of a T100P card, a TDM400P card, and an Adtran TA750 channel bank with three quad-port FXS modules and a quad-port FXO. The TA750 is wired to a 24-port Cat 5 patch panel via a 25-pair Amp cable. The phone is a model

For anyone who is interested, I have a working tftpd (modified wvtftpd) capable of serving configuration, dialplans, and ringtones to Cisco 7960/7940 and ATA-186 devices that are located behind NAT firewalls. As TFTP is not a very firewall/NAT friendly protocol, I had to break some rules to get it to work with these cisco devices. It might cause problems for other TFTP clients, but it works with

Greetings again all, With the help of another list member I was able to get the my TDM400P card working properly (and the PRI card loaded too for channels 1-24).

I'm looking for a way to give VoIP capabilities to an existing PBX: it's made by Mitel and it's used in a small/medium environment (a few dozen phones, but the PBX has capabilities for up to 200, if i remember correctly). Any high-level guidelines on how to integrate Asterisk with a PBX that's already in use? Probably that particular PBX is not supported directly, but are there

Hi all, I am experiencing a problem with the quality of the voice communication between an IAX based softphone (WinIAX) and an outside line through a FXO port or even with a regular analog phone connected to a FXS port. The party using the IAX softphone hears his own echo a plit of a second after speaking. The party on the analog end does not experience any echo. I tried to modify the KFLAG

Hey all, I am in the middle of creating a new user wizard which will generate all the .conf's the new Asterisk user will require to get themselves up and running in Asterisk without having to touch a single configuration file. This is what I have come up with as a rough draft. It is far from complete, so I'm asking people to submit things that should be added, changed, removed

I am always getting multiple PCI Master Abort messages when I try to plug in a second TDM400P. I have asked this question before, but I nothing really solved my problem and I just put it on the back burner for a while. I am at a point where this is a crucial issue. I have read that the Zaptel devices share an IRQ, is this causing the problem? Is there a way that I can manually change the IRQs of

I'm unable to get faxes working over tdm400p (4fxs modules) Too many errors sending and receiving faxes with an analog fax 1) echocancel=no on the zap channels 2) ztmonitored the channel for a good/low audio volume I'm trying to send fax between zap fxs channels. No way to get it working right Has someone else the same problem?

We currently have a 15-phone system using Asterisk, a combination of analog phones/Grandstream HandyTone-286 and Grandstream BT101s, and 4 X100Ps connected to analog lines. The system works well except for the occasional echo problem. I have all the echo parameters configured, removed all the extra incoming analog lines except to the PBX, etc. following all the advice on the wiki and on the

I have just received a T100P and an Adtran TSU 600 in the mail. I seem to be having a problem with the T100P card. So far I have done the following: vi zaptel.conf fxoks=1-22 fxsks=23-24 ... vi zapata.conf ... signalling=fxo_ks ... channel => 1-22 ... signalling=fxs_ks ... channel => 23-24 I then run modprobe zaptel modprobe wct1xxp ztcfg -vv There are no errors to report. In

I'm connecting to an NEC t1 card via t100p (working great so far!) however I'm having problems dialing from the NEC system to an asterisk extension (sip-grandstream). If I hit the trunck line and dial REAL quick 103 I get the sip extension ringing; if I don't I get an invalad selection message from asterisk - and I can see on the console only one or two digits arrived. How can I