similar to: Help! Zap echo on bridged calls

I was just checking out the dids for all of my fail over providers and noticed that neither DID that I have with SixTel work. Both pause for a long long time The local number gives a recording: 'The number you have dialed is not in service or is assigned in a different area code. Please check your number and dial again'. The 800 number just rings busy. Anyone else having this issue or

The registry's are stored in DB. Just export your database with 'database show' Schedule it with cron to run every 5 minutes or so. You can do that with -rx command line switch for asterisk. Send the file across to other node and pipe it through awk/perl/cut or whatever you like and import it when you bring the other node up. You will have to stop and start asterisk I think. I

Anyone use livevoip? opinions? -- JD Austin Twin Geckos Technology Services LLC email: jd@twingeckos.com http://www.twingeckos.com phone/fax: 480.422.1250

Hello I saw Ver1.2 is out. Whats new? Thanks for the hard work, David

I'm still relatively a novice with asterisk and am having issues with echo. The calling party that calls a PSTN number doesnt hear the echo, but the answered side via sip or forwarded to another PSTN number over voip hears excessive echo that makes it difficult to communicate. I've been playing with the zapata.conf settings for echocancel, echotraining, rxgain, txgain, etc and am

Hello everyone, So, this isn't exactly what it seems. I am not looking to integrate Asterisk and Mambo. I am the maintainer/creator of AstLinux, and I have recently decided that I should really have a better web site for it. I would like to use Mambo so that I can do updates easily, from anywhere, without having to waste time learning PHP/HTML/etc. Mambo CMS seems the best and most

Hello - I've been using Broadvoice with Asterisk for a couple of months with no issues. Today, it has stopped registering. The Sip Debug from CLI is below. It tries to register five times and then gives up. Any suggestions? As you might suspect, I have not been able to get Broadvoice on the phone and usually get cut off after being on hold about 5 minutes. Strategic portions of IP

Hello i would like to receive incoming faxes thru' asterisk as tiff files thru' the rxfax application. I setup extensions 101 like this exten=> 101,1,rxfax(/tmp/fax.tif) then from CLI i run: dial 101 and rxfax send me his "scream" about the fax ^^ instead when i send a real fax from a faxmachine to that extension my 101+rxfax is executed but it just "does

This is frustrating. Asterisk has crashed now twice today and neither crash has produced a core file. My ulimit is unlimited. I'm using safe_asterisk so asterisk is restarting immediatly, but how the hell am I suposed to find out wtf happened with no core file? Debug log doesn't say anything either. AGRHHHHHHHH -Matthew --

Here is a letter I sent them for my $150 paper weight. Dear Voipsupply, As a small service provider, using you company for the first time, I'm very disappointed that you have removed the configuration CD that should have been shipped with the Mediatrix 2102 just to get a few more bucks. I have contacted mediatrix and they have informed me that the CD's is shipped in every 2102. If I

This will sound weird but the command 'asterisk -r -x reload' fails to work when issued by Cron. But it works when I issue it from a bash session. What is not configured correctly? I need to refresh the configuration every a short amount of time. rom root@localhost.localdomain Wed Jun 15 18:42:00 2005 Date: Wed, 15 Jun 2005 18:42:00 -0400 From: root@localhost.localdomain (Cron Daemon)

hello how to solve these errors /var/lib/asterisk/agi-bin/agi-rad-auth.pl line 10 use Asterisk::AGI; vi /etc/asterisk/extensions.conf exten => _X.,1,agi,agi-rad-auth.pl|Routing=SIP&AuthorizeBy=SIP vi /etc/asterisk/modules.conf load => res_agi.so <---------------errors------------------------> *CLI> Can't locate Asterisk/AGI.pm in @INC (@INC contains:

Hi all, Before I setup an account with them, I'd like to hear other people's impression of LiveVoip. I'm considering using them for 800 numbers, and I'd like to feel comfortable that others here on the list have had good experiences with them. Thanks, sorry if this is the wrong list for this. :) Sena

911 Help! I accidentially deleted all directories under /var/spool/asterisk I did use the backup facility not too long ago but cannot find the process for restore. However, I don't believe a full restore is needed -- I just need to know the names of the directories under /var/spool/asterisk and re-create them (I hope!). Can some kind soul give me some direction or tell me the directory

For people who are using GS 101, what do you think the sidetone generated by the phone. I find mind a bit annoying. It has a delay and you notice it as an echo. The volume of the sidetone is also quite hight. I am distracted when both caller and called party talking over each other occasssionally. The volume of the sidetone can be turned down using the volume button but it also control the

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Outward dialing is a no brainer. VoipJet is the best outbound call provider I have come across. Period. It always works for me and the call quality is always very very good. So far that seems to be the norm for them. I am still working on getting my inward DIDs solidified so no opinion there... W -----Original Message----- From: asterisk-users-bounces@lists.digium.com

This is the guy that has a ton of email addresses. Almost as many as he has phone numbers. google "kvj" He doesn't like our president either: Here's look at a MISERABLE FAILURE and I use facts: George W. Bush (herein referred to as 'bushwhack') is the village idiot and he pushed a series of Trojan horses at Americans: 1) The Overtime Pay act is nothing more than a

Hi guys, I've run into a problem that I can't figure out on a bunch of handsets I have running into a Rhino Equipment 24-port FXS channel bank hooked up to a T100P and running asterisk-0.9.0 and the associated stable Zaptel release. The sidetone (your own voice that you hear in your handset, built in for comfort) is noticeably louder than it should be, and it doesn't seem to

Hi everyone, Is there a simple way of answering a different ringing extension from a sip phone using AAH? I have absolutely zero technical know-how when it comes to modifying conf files etc. Still working on figuring it all out. ;) That brings me to my second question... where the hell does one find an extensive manual of sorts that explains all conf files and what the strings all mean etc?