similar to: please help !! - context for an incoming call

I have installed Digium TDM04B with the latest CVS. However I have encountered following problems: 1. When it dials out, many times the digits are not properly recognized by telco as I hear the announcement "please check the number and dial again" although I see on the screen that the dialed number is correct. 2. When the call is forwarded outside, with something like exten =>

Hi I am interested in setting up an Asterisk PBX in my office with digium hardware, and I just have a few questions in regards to what I would need. It is my understanding that an FXO card is used to interface with an incoming/outgoing phone line, and an FXS card is used for interfacing with a phone within the system. Currently we have 4 incoming/outgoing phone lines and would like to have

Just did a complete cvs checkout (Sep 4, 9:30am cdt). Seems any attempt to dial out via a tdm04b (fxo) now fails. Does not seize the pstn line. CLI> zap show channels Chan Extension Context Language MusicOnHold pseudo inbound-bus-x10 en default 1 inbound-home en default 2

I have a normal backup phone (and an alarm panel) sharing the POTS line with the Digium X101P FXO: | | Wall |>---+------X101P FXO as Zap/5 | | | Phone & Alarm Whenever the Phone is used, Asterisk sees a 'false ring' signal immediately when the phone is hung up. The Alarm panel dials out nightly at around 1AM, and each time it completes the call, Asterisk

OK, here is the scenario, Asterisk @ Home 1.0 with TDM04B and TDM40B. I can receive and place calls with no issues, however, when I receive a call, the CID only shows "Analog Line" on the Grandstream 2000XP phone. Does anyone have any ideas even where to look to change this?? Is it a setting in the phone, Asterisk, or both?? Thanks, Marc

Greetings, I need if possibile an explanation on how to manage the incoming callerid for an incoming call. Let me explain the situation: We have two different companies in this office that shares the same PBX (* box). Each company have its own number for the incoming calls. What i'd like to implement is something that, depending on the incoming line that is involved in the call, plays a

FYI... Just installed one of the new Sangoma A200D analog pstn cards with the hardware echo canceller on a trial basis. The card has four fxo interfaces. Excellent audio quality, excellent echo cancelling, and excellent audio levels. The four pstn lines at this location are rather long analog loops that have rather long echo trails. I started with a pair of x100p's a couple of years ago,

I have * installed as a voicemail only solution attached to a Toshiba Strata phone system. I am testing * at the current time, so it is not attached to ports designated as VM on the Toshiba. I am not sure if the VM ports would handle the call any different than an analog port. Currently if I call one of the DIDs that I have connected to the *, my greeting plays and I am able to dial my

Hello, I'm interested in implementing DID functionality with the Digium TDM04B adapter. Is DID supported with POTS? Are there any caveeats or drawbacks that I should be aware before proceeding? This pbx will be implemented in eastern europe. Thanks in advance. -- Martin Spasov <mspasov@techno-link.com>

Hello all. My company currently has an older Executone PBX system that we are outgrowing. Rather than wait until the last minute to make a hasty decision, I thought it would be a good idea to do some research and compare options first. My expertise is in computers and networking, and telephony systems are mostly foreign to me. What we currently have are 5 incoming POTS lines and 25

Hi folks I have a X100P in my home asterisk box and I have it setup as a default context of 'incoming-pstn' in my extensions.conf i have [incoming-pstn] exten => s,1,Goto(incoming,01225<myofficenumber>,1) then: [incoming] exten => 01225<myofficenumber>,1,Answer exten => 01225<myofficenumber>,2,Dial(SIP/data|m) etc etc Anywho back to the plot. It all works

I am having a problem with echo during the begining of incoming phone calls on our TDM04B. The echo is only noticable on our end and will eventually go away. It is also noteworthy to mention that during the day incoming calls ring 3 phones first before going into an automated menu system. If we answer the call immediately, the echo is present. If we wait until the call goes through the

Hello I have installed a TDM04B and disabled any devices not required in my PC. (TDM04B is analog card with 4 ports to plug into telephone co lines). I am running this version of * Asterisk CVS-HEAD-01/10/05-02:11:15-AstLinux built by root@ls on a i686 running Linux As you see below the wctdm module is loaded: pbx root # lsmod Module Size Used by binfmt_misc 12296 1 - Live

OK, here's the setup, AAH 0.8, Grandview 2000 phone, Digium TDM04B interface to POTS lines. Everything seems to be working just fine, but I have some questions on how to access voicemail options. I can leave a message for an extension, but when I try to retrieve it by using *97 it asks for the password and even though I type in the same password I gave the extension for vmail, it tells

We are installing a number of systems with 2 TDM04B cards. Have done all the isolation to unique IRQs, etc. All inbound calls seem to work fine. However, outbound calls are hit or miss. Sometimes they work fine and other times we get a "you must first dial a 1 or 0" message back from telco when dialing out standard POTS lines. We are running AAH 1.0 which is Asterisk 1.0.7. Six

Hi. Am new to this concept but have been requested to add VOIP capability to a small office phone system. They currently have 4 standard analog lines running into a PBX feeding 16 phones, with all the usual features, call transfer call hold internal calls etc. would the following seem reasonable ? asterisk server:- ( what specs ) cat5 > broadband (VOIP) 4 FXO's for incoming PSTN

Hi guys/girls, We are running a TDM04B card with Asterisk in a Linux box that has 15 GS102 extensions and 1 SNOM190 phone which we are using as an operator console. The FXO ports in the TDM04B are plugged directly into our telecoms provider's analogue lines. Something I've picked up with the SNOM is that sometimes when there are two active incoming calls via Zap channels and the first

All, I have a machine with a TDM04B and a TDM13B card in it. For over a week it was working find placing a call out every 10 minutes to the intercom system. Today that stopped working. The call was still getting placed just as it was in the outgoing spool and "looked" like everyting was working just no audio on the intercom. I picked up the analog phone on the TDM13B (the one port that

I am trying to integrate a Nortel Norstar system with an Asterisk service using a TDM04B card (4 fxo). So far everyting works from Asterisk to Nortel. The problem is when someone dials from the Nortel PBX to the Asterisk server. Asterisk answers the call and provides a dialtone (with DISA) but appartently the DTMF tones are not passed to asterisk and the call cannot proceed. This only

I'm installing my first Asterisk server. I have a TDM04B card installed in my asterisk server (4x FXO ports). I have 5 Cisco IP phone 7960 working fine on asterisk using SIP. My configuration to receive call is working as expected meaning anyone calling on one of the 4 FXO ports is answer by asterisk and asked to enter the extension of the person to reach and then it is transfer on the