similar to: Problem with bridging two external lines

I'm working on getting two SIP servers to talk. An Asterisk box and a Zultys MX250 system. So far, things have been working pretty well. I can call from one of my Asterisk-managed phones to an MX250-managed phone and vice versa. However, there are some strange issues. If I call from an MX250 phone to an Asterisk phone, the conversation is ok, but there is a noticeable delay in the voice

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:

All, Experiencing some issues on my FXO lines. If a call comes in on an FXO and then get transferred to another FXO (say to call someones cell phone), those two lines will stay tied together indefinitely. This happens to us when we transfer an incoming call to our on call guys after hours and on weekends. We have installed 3 other * boxes and they do the same thing. We use a Adit Channel bank

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

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

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

I spoke the other day about my preliminary tests with office-wide paging with Cisco phones using the new SIP 6.1 image which supports auto-answer. I've got a small and crude recipe for those of you who want to experiment and hopefully create some better and more complete examples than the one I've thrown together below. Create a new line on each of the Cisco phones, and put the

Check 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

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

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

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

Hi, I have been trying to set up a network bridge on my FreeBSD 3.4 machine (hades), and as soon as I enable the bridge, Samba stops servicing requests from both sides of the network. The following is a 'diagram' to give you an idea of my setup. |------| |----+Charon| |-------| |----| | |------| |Neptune+----+Hades+----|

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:

Hi all, Is it just me and not reading the docs right, or has anybody else had problems with the AbsoluteTimeout application and the 'T' extension when used inside a macro? [macro-attended] ; ARG1 is the device to dial out on, SIP or Zap, or whatever ; ARG2 is the extension to dial using 'attended' dialing exten => s,1,AbsoluteTimeout(30) exten =>

I've got an * system that is having some real problems with 1.0.2. The biggest problem is that calls going through my T100P get choppy for about 10 seconds every 1 or 2 minutes. Asterisk is running on a debian stable system with current packages. The T100P is plugged into a Adit Channelbank with 8 POTS lines hooked up to the Channelbank. I've watched the vritual memory and CPU status on

Hi, I am trying to get priority + 101 to work with Dial application. My dial plan is like this: [dial-mobile-peak] exten => s,1,AbsoluteTimeout(${ABSOLUTETIMEOUT}) exten => s,2,Dial(${TRUNKONE}${CALLEDNO:1}) exten => s,103,AbsoluteTimeout(${ABSOLUTETIMEOUT}) exten => s,104,Dial(${TRUNKTWO}${CALLEDNO:1}) I have changed password for first trunk to simulate trunk failure. Trunk one

Ok I tried but it does not work: now the settings are as follow exten => _123.,1,Answer exten => _123.,2,AGI(test.agi) exten => T,1,hangup the AbsoluteTimeout(5) is in test.agi (PHP) I put "AbsoluteTimeout" before "Answer": when i call for e.g 123456 it tries upto timing out. So I put again "Answer" and then "AbsoluteTimeout" then the last AGI

Hello everybody. Recently I've been trying to limit the duration of some calls for a simple application I'm writing. Unfortunately all of the documented methods are failing and I'm not sure what else to try. Here are some samples of what I've done: 1) The AbsoluteTimeout application. - exten => 1,1,AbsoluteTimeout (30) 2) The new version of AbsoluteTimeout. - exten

Hello, I'm runing Debian Stable linux with flac version 1.2.1. Weekly, I run a cron job to test all my flac files for problems using the -ts options. I have several computers storing the same information over various raid arrays, and occasionally I do find the odd file that has had problems and can then change hard drives and re-synchronize. Anyways, I have recently encountered a couple of

I'm using the M() option to Dial() and having problems. In the following dialplan example ANY digit exits the macro. When the callee presses 1 the Noop(Reset AbsoluteTimeout(0)) does not get run. Does anyone have any ideas as to what I'm doing wrong? Asterisk 1.2.x [extensions] exten => 2998,1,Dial(Zap/1/5551212,,wM(answer-confirmation^20)) [macro-answer-confirmation] exten