search for: telnor

Hi All, Just installed our very first asterisk system, and we love it! I cant believe the different things you can do with it, just great =] My question is: How do I configure my system to play an mp3 file when a caller gets put on hold? Thanks in advance, Steve.

Hello all. What is ztcfg for ?, what does it do ?

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

...time) :) -------- Original Message -------- Subject: Re: [Asterisk-Users] call pickup fails. Date: Thu, 27 May 2004 07:38:44 -0600 From: Rich Adamson <radamson@routers.com> Reply-To: asterisk-users@lists.digium.com To: asterisk-users@lists.digium.com References: <40B59209.6040202@telnor.net> > I saw a few weeks ago a discussion about cal pickup, *8, not working > but did not find a message about it being resolved, I look for a bug on > the bug list but did not find anything about it not working, nor a bug open. > I installed asterisk 0.9.0, have one sip...

Hi, i just received an E100P, this is the first one I have ever seen, and notice that the board reads T100P. Is this right ? The antistatic bag had a small label that has E100P written on it, and the card is a bit different than the T100P I already have, Does Digium use the same boards for both cards ? I don't have an E1 link here, can I test the card just by loading the driver and run

Anyone can share me with Music Onhold Configuration sample? Thanks in advance for your help, Kang

....digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > --__--__-- > > Message: 4 > Date: Mon, 19 Jan 2004 21:22:39 -0800 > From: "Ing. Angel Gomez Garcia" <angom@telnor.net> > Subject: Re: [Asterisk-Users] SIP: Register that isn't a register? > To: asterisk-users@lists.digium.com > Reply-To: asterisk-users@lists.digium.com > > Walter Doerr wrote: > > >On Mon, Jan 19, 2004 at 12:33:41PM +0100, Philipp von Klitzing wrote: > > &gt...

Hi all. Is there a way to collect the digits dialled in asterisk and stored them in a variable ? I'm setting a submenu for the user to change his extension dial in treatment from a standard extension to something like 'automatic transfer' and I need to ask for the number where to transfer the calls, pass it to an AGI and store it in a DB for later use. I also need to ask

Hi all. Just got my 'Developer Kit Lite', installed it, and made the changes to load the modules in kernel and in the configuration files. Call thru-from fxo and the fxs sound great. Even fxs-iax-sip sound ok. When a answer a call coming into asterisk from the PSTN thru the fxo i have a loud echo. I have echocancel=yes. Is there another parameter I can change in the

Hello. I have two snom200 ip phones and 1 mp108fxs (audiocodes 8 fxs) and i dont get a ring in the caller phone when I dial from a snom200 to the other snom200 or the mp108fxs, I made a debug with ethereal, and I can see a "Ringing" packet being return from the called snom200 or mp108fxs to the asterisk box, but it is not being re-transmitted to the caller snom200. Altough

Good evening all. I'm having this strange behavior when dialing two or more simultaneus calls via IAX to other * boxes. Sound starts to have more latency, wich increments until it's almost impossible to talk (6 or more seconds), I try this calling with two grandstreams, one grandstream one tdm410p, one tdm410p and sjphone, sjphone and one grandstream, the result are similar.

Hi all. Is there a problem with digium cvs ? I can't connect to it, it just keeps giving a... cvs [login aborted]: connect to cvs.digium.com(216.207.245.20):2401 failed: Coonection refused Thank's

Hello all. I saw a few weeks ago a discussion about cal pickup, *8, not working but did not find a message about it being resolved, I look for a bug on the bug list but did not find anything about it not working, nor a bug open. I installed asterisk 0.9.0, have one sip fxo gateway and only sip phones, all of them have callgroup=1 and pickupgroup=1 but I can not get a call that is

Hi all. I have this configuration: Telco <-----(E1)----->TE410P//Dual Xeon Server 2.4Ghz<-----(Ethernet)----->Switch<----->GS//BT The Server is running RedHat Linux 8.0 with kernel 2.4.18-14-smp and we are having the following 2 issues: 1.- When making calls from the GrandStream to the PSTN the audio is choopy, plus theres is a pulsing sound, but when the GS

Hello. Has someone been able to make work faxes over sip, i have one mp108 fxo and one mp108 fxs, my setup is : telco analog line -----> mp108fxo -----> Asterisk ------> mp108fxs -------> fax machine 1) Asterisk detects the tone from the sending fax ( i am receiving ) but looks for extension 'ff' not 'fax', ( at least that's what * complaint, invalid

Hello. I have this issue, when I pickup a call that is ringing in a SIP Phone, it keeps ringing. There is bug #116 that mention something about these, but it does not seem to be resolved , at least, not yet. Anybody else has seen it behavior ? Thank's.

Hi. I have the next configuration... I dial from my analog phone in the TDM400P to extension 102, and the second agi works about 1 out of 10 times, the other nine it gives me these error on the asterisk console: -- Starting simple switch on 'Zap/2-1' -- Executing Macro("Zap/2-1", "receivecall") in new stack -- Executing AGI("Zap/2-1",