similar to: Q: Does anyone have a WE multi-line card dialer phone working with *?

Rob at draughon.org writes > 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

In my * server I have a T100P, connected with a T1 crossover cable to an Adtran TA750 with 2 FXO cards. After I do 'modprobe wct1xxp;ztcfg' the alarm on the card starts blinking red. Running zttool confirmed that yes, it is a red alarm. I've read a few previous posts on this type of setup and I'm not too much closer. What should I be checking next? A different cable? The

All, I need help. I have an (actually 2) Adtran TA750's with 8 FXO ports. I get a terrible buzz on every FXO port. If I unplug the Adtran and put an analog phone on each incoming line, I have no buzz. I also have 2 Carrier Access Access Bank I's with 12 FXO ports. When I plug the same analog lines into either one of those, no noise or buzz whatsoever. I went so far as to move the TA750

I recently set up Asterisk with an Adtran TA750. All is well except the phones do not show the MWI. I have configured zapata.conf properly, as all phones will receive a stutter dial tone if there is a message waiting in it's assigned mailbox. Does anybody know how I might fix this problem? Thank you for your time __________________________________ Do you Yahoo!? Yahoo! SiteBuilder - Free,

Hello. I have been going thru the wiki and asterisk related sites and have not been able to find any documentation about how to configure an Adtran TA750 channel bank. The remote disconnect supervision doesn't seem to be working, when the remote caller hangs up asterisk takes up to 30-45 seconds to hangup the call. Can somebody help? Thank's

All, Thanks for taking time to read. I am wanting someone to tell me that I have a clue here and that the following proposed setup would (possibly) work. Currently existing are 4 pots lines with qty. 10 AT&T 954/854 phones. Would like to convert to asterisk and at the same time do a mild feature upgrade. Proposing to get: PIII 800MHz or better >=256MB RAM

Hello all, I've got a T100P connected to an Adtran TA750 with a T1 crossover... This connects to a patch panel with phone ports. The Adtran is fully populated with FXS cards. All I get on any phone port is a fast clicking noise... No dialtone. Asterisk 'sees' the card, (the channels show up in /proc/zaptel). Incoming calls are routed to the zap/x channel, but no ring. I'm

I get seg. fault with my * box. at the crash time i had about 35 Bridged Channel. i have: - dual xeon box (3.2Ghz) - 2Gb of memory - E7501 chipset motherboard. - U320 scsi disks - intel Gb ethernet device. - i only use sip for clients (no fxs in box) - TE405P for fxo (with 4 atran TA750). - ulaw is used as codec and echo cancellationo is enabled. but the core dump file has nothing to show with

Folks, Thanks to Jerry Jones, David Josephson, John Novack, and George Cohn for their posts about how a key system phone works. Now I'm starting to scrounge around for a KSU so I can look into the Collector's Net work :-) Most of the phones in the house are (fairly) new single-line analog jobs; but, since "retro-telephony" is an amusing part of this hobby, it seems only

Hello, I need help in creating a simple PSTN Gateway. This is the scenario: -I have one client sending me VoIP traffic (they don't have asterisk, so IAX is out of the picture for me) and I need to validate that traffic (only accept calls coming from his IP). After that I would terminate the calls to the PSTN network and keep logs for billing purposes. -I have a TE405P board and

Some default setting have changed. Use: testparm -v from your various versions of samba to detect which parameters may be causing you issues. ------------------------------------------------------------------------ Tony Hoover, Network Administrator KSU - Salina, College of Technology and Aviation (785) 826-2660 "Don't Blend in..."

Hey gang, I'm looking at escaping from a Nortel Meridian CISC system to Asterisk/Digium/SIP phones. I'm currently in the testing and proof of concept phase. I'm going to need a SIP phone and don't want to re-purchase and have "orphans" around. We currently run Nortel 7310 phones and they work great. I'm sort of overwhelmed by all of the different IP phones. I

Can anyone share any advice / documentation on how to configure a Tellabs 2572 T1 echo canceller? I connected one between a T100P and an Adtran TA750 FXO/FXS channelbank, but when echo cancellation is active I get a LOT of snap-crackle-pop (and other problems) on the line. The 2572 has a bunch of configuration options and they're impossible to guess without documentation, which I cannot find

Hello, I have been researching Echo Can's for a while now, and wanted to post this out to the list to solicit feedback (and if my assumptions are correct, hopefully help others out)... If anyone out there knows anything about wiring up an Echo Can, and my outline below is incorrect, please let the me/list know.. I have posted the Documents I received from Tellabs at:

I'm trying to figure out how to start *..... Rh7.3,CVS,TE410P,TA750 If I just try the way the docs spell it out "/usr/sbin/asterisk -vvvc" it fails...... /var/log/asterisk/messages Oct 3 22:23:34 WARNING[1024]: File chan_zap.c, Line 610 (zt_open): Unable to open '/dev/zap/channel': No such device Oct 3 22:23:34 ERROR[1024]: File chan_zap.c, Line 4930 (mkintf): Unable

I am running Asterisk and using Answer machine detection with call files on a virtual Vcloud server running Centos 5.3 and LAMP. I am finding that AMD is only detecting HUMAN or MACHINE for about 30% of the calls (I sent over 50,000 outbound calls last week, and 70% said NOTSURE). I have a suspicion that the problem may be due to the timing source on virtual server when its under load delivering

Rich, Thanks a bunch, totally understand now and that actually makes total sense. (no need for schematics). This also explains why I used an TA750 to go into a Nortel MICS system, using FXO and no buzz. Totally balanced load from the analog ports on the Nortel across the 5 feet of CAT5 to the FXO on the adtran. Now I need to get rid of some Adtrans --- Anyone lookin to buy? :) Thanks

Does anyone know the secret to get the GXP-2000 Message waiting lamp to illuminate? Or can point me toward some docs that might explain it? Thanks! --Jeffrey -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060501/7ffc668f/attachment.htm

> I read your excellent post on the subject. I too am hearing the 60 Hz > hum on the FXO ports of my TA750. One Interesting thing is that it > goes away if there's an analog phone on the same POTS line, and you go > off-hook on it. I guess the analog phone has the repeat coils built > into it, and they balance the line for the FXO port. That's exactly correct; the

Folks, I'd like to change the value of ${CALLERIDNAME} for incoming PSTN calls from certain numbers, but haven't found a way that works. The goal is to provide more informative names on my phones' caller ID displays--e.g., I would prefer to display "ROB CELL" instead of "CELLULAR CALL" when I call home from my cell phone. This is what I tried in the context