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I believe it is not fiber, but I am not sure. I am going to take one of them home tonight and hook it to my POTS line there, which for sure is not fiber. Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Michael Welter Sent: Thursday, April 22, 2004 5:02 PM To: asterisk-users@lists.digium.com Subject:...

...duplicates of every part for the TA750's and swapped every component and cannot get rid of the hum on either unit. I have the power supply of the Adtran grounded. I am out of ideas :-(. Any assistance would be greatly appreciated. Thanks in advance.................>>Greg Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com <http://www.golden-tech.com/> 219-462-7200 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040422/4945af8c/attachment.htm

...the Redirection on the web interface of your Snom 200. If it says "When Busy" that's your problem. It should say "Never". Also make sure on Sip->Lines your line appearance says "All" or else you will have the same problem. Hope this helps :) Greg Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Frederic Steinfels Sent: Saturday, April 24, 2004 12:30 PM To: asterisk-users@lists.digium.com Sub...

Jay, I have had a lot of trouble with the FXO ports on Adtran TA750. Unless the incoming POTS lines have a balance impedance, they will buzz very bad. Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jay Milk Sent: Sunday, April 25, 2004 12:16 AM To: asterisk-users@lists.digium.com Subject: [SPAM]...

I am perplexed by this so I how someone can help me out. On one of my servers the users began complaining that if they picked up a parked call they could not use the # key to transfer the call. This is a particualarly annoying issue since everyone has been taught to use #700 to park calls. At first I thought it was a DTMF issue with the polycom phones, since rebooting seemed to fix the problem.

I am surprised you needed to turn the rxgain down so much, usually it is just the opposite. I experienced the same problem you did when my txgain was too low. Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of bam Sent: Friday, May 07, 2004 12:31 PM To: asterisk-users@lists.digium.com Subject: [SPAM] - RE:...

Run /usr/src/zaptel/ztmonitor 32 -v And adjust your gains in /etc/asterisk/zapata.conf accordingly. Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of bam Sent: Friday, May 07, 2004 3:35 AM To: asterisk-users@lists.digium.com Subject: [SPAM] - [Aster...

Hi All I have a snom 200 phone here which works perfectly when using the handset to playback the voicemail messages etc. However when I play back the voice using the speakerphone it sounds choppy. Anyone had this problem before? Regards Mark

...thing. We use a Adit Channel bank for all incoming FXO and the other installs use multiple 1 port digium FXO cards or a combination of the 1 port and 4 port FXO cards. Has anyone else experienced this and if so how did you fix it? Thanks in advance....................>Greg Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com <http://www.golden-tech.com/> 219-462-7200 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040618/3a1f2711/attachment.htm

...s 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 again.............Greg Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Rich Adamson Sent: Thursday, April 22, 2004 6:03 PM To: asterisk-users@lists.digium.com Cc: mike@i...

That does work, I use that same approach to get analog extensions in a norstar system to dial a specific sip phone in *. Works really well. We then also tie the calleridname to which channel they dial out from as well. Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jerimiah Cole Sent: Friday, May 07, 2004 1:01 PM To: asterisk-users@lists.digium.com Subject: [SPA...

Greetings, I have a new * system installed and everything works as it should except for one annoying little problem: I can't play any sound files. What this means is that when an extension script gets to the point where it should play a sound file (voicemail greeting, auto-attendant, whatever), the caller hears a click and then silence. According to the * log, the sound file is being

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Gordon Henderson > Sent: Friday, February 09, 2007 9:47 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Conferencing Phones ... > > > Anyone got any experiences of good quality

I'm running Asterisk CVS-HEAD-05/24/04-17:37:48 on kernel 2.4.25-gentoo-r3. I have a Digium TDM-400P card with 4 FXO ports. Here are the pertinent files: zaptel.conf: fxsks=1-4 loadzone = us defaultzone=us zapata.conf: [channels] context=north_in_pots_vip group=1 signalling=fxs_ks usecallerid=no hidecallerid=no callwaiting=no restrictcid=no threewaycalling=no echocancel=1

Set busydetect=no in your zapata.conf file. That should stop the random hang-ups. If you really need busy detection, try setting busycount=8 or even 10. If you still get random hang-ups, turn off busy detection and turn on call progress. May help the situation. Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Shahid Mahmood Sent: Friday, April 23, 2004 2:39 PM To: asterisk-users@lists.digium.com Subject: [...

Hey all, I have a situation where I have 8 lines from the phone company in a hunt group coming in to my asterisk box. These are the same lines I'm using for outgoing calls ( named g0 ). The problem arises when someone dials our number at the same time asterisk tries to put a call out on one of the zap channels in the g0 group. This has happened twice that I know of so far, once to

I need a good reliable source for channel banks. Ebay is nice but too many variations of quality. I need one that can do 8 FXO and 16 FxS does anyone have a particular model, other than adtran that they can recommend??? And if possible include a source to buy from... Thanks in advance, Alex -------------- next part -------------- An HTML attachment was scrubbed... URL: