similar to: Problem with TDM400P

Hello, I am wanting to close a specific channel for example; SofthangUp(SIP/EXTEN-UNIQUEID) but the problem is the channel is assigned a unique id as well. The need fits into the idea of receiving a call from a higher status user and thus closing a specific channel to allow the higher priority call to route through the dial plan to the freed extension. Any ideas welcome. Many thanks

We have a TDM400P installed here with four FXS modules. It works well except for a couple of issues: First, I have a Panasonic KX-TG2431 telephone (so others can reach me when I am in o ther parts of the building) hooked up to one of the FXS ports. When the other end hangs up, I get the usual CPC disconnect signal. After the CPC, sometimes it will go to a dialtone, and other times a

Hi I have a server with the last version of asterisk branches, zaptel branches, 2 Digium Card with TDM800P 16 HPEC channels (Echo Cancelation), 40 Grandstream BT200 and 10 Grandstream GXP2000. zapata.conf echocancel=64 rxgain=0 txgain=0 when i place a call o receive a call, I finish a sentence i hear a ssssssss, AND when the both side talks at the same time i have choppy audio. Any

Hi I just want to know if anyone have problems with server DELL 1600, Like: Hangup Call. Thanks Ruben

<div><BR>Hello,</div> <div>&nbsp;I bought a TDM400P, and intended to use it with my analog phone, which is RJ11 ofcourse. So, the question now, how do I plug in my RJ11 phone to the TDM400P card, which has an RJ45 interface? Also, since it's an 11B card, I also intend to bring in an analog line into the RJ45, so i am still left with the same question....how do I go

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031230/a1136ef2/attachment.htm -------------- next part -------------- Hello All, I managed to Configure TDM400P, now I can call Analog Phones connected to TDM400P from SIP Phones (CISCO and SNOM). But when I try to dial any number from Analog Phones I get following message -- Starting simple

Hi, I have a running asterisk 1.2.1 (bristuffed) with a TDM400 Board. Connected to 1 port is am ordanary Fax Machine. Everything 'seems' to work, however receiving faxes is very unreliable. Sometimes I receive a normal page, without problems. Sometimes half of a page and the rest is scrambled, but most often, I receive nothing and the other site reports a Fax error... The Fax machine

I have been having some troubles with the zaptel channel on what appears to be the inbound process. The box is running the stable CVS code and has a TDM400P 4-port FXO card in it for analog connectivity. Channel 1 is the only active port on the card at the moment as we only have one analog line. What has been happening is that it looks like Asterisk has been detecting an inbound call even though

Hi All, Sorry if this has been around a millions times.. I have been off this list for a few months now.. I have installed the latest asterisknow (upgraded asterisk to 1.6 as well) and I am having a hard time getting my X100P and TDM400P working.. Its all new to me with dahdi because my old server is still running Asterisk 1.0.2 so there have been lost of changes.. Can someone point me in

Hi, I have just built my home Asterisk box into a better PC that became available (still only a P2 350 but it only has to manage 1 analog line).. Anyway I have built it on Fedora Core 1.. I have an X100P and a TDM400P (1 module installed).. These cards were working fine in my older PC that was running my Asterisk at home.. The inbound calls via the X100P to my sip phones are working great..

I have a Digium TDM400p (1 FXO, 1 FXS) with the FXS module connected to a standard analog handset with caller id display (US caller ID). Although it appears that caller id information is coming into asterisk (it shows up in voicemail), I can not get it to display on the analog handset. Is there anything special I need to do to send the caller id info out the FXS port? I've tried a few

I've setup * with TDM400P w/1 FXS, 3 FXO modules. I've one analog phone connected to TDM400P FXS module, 1 PSTN line to one of the FXO module(ZAP) , and 2 analog phones connected to Sipura 2000 (SIP). The calls between SIPs and zap phone (fxs) are OK. But 2 issues cannot be solved: 1. To dial to PSTN via zap phone, the setup in extensions.conf with the following exten =>

Hi I am trying to use Asterisk as a backend to send and receive faxes over analog channels connected to a Siemens HiPath 3550 switch and a TDM400P card. Receiving faxes, including multipage ones, works really fine, we have no issues at all. But when it comes to send faxes using the app_txfax application, spandsp can't send multipage TIFF files for some reason, it stops sending in the

Hi! I am using asterisk with two tdm400p cards. Sometimes (one call out of ten), when a call comes in and is taken, there is some terrible noise for a short time in the line (for about a second). Both partys can hear the noise. And sometimes the call has to be hung up, because the noise doesn't disappear. Has anyone any idea where the problem could be? cheers, tom

I need to ask, to refresh, is the aux power connector on the TDM400P card *only* to power the ringer on any analog phones/devices on the system? Can I still use this board, to "terminate" POTS lines and use all SIP Phones? Due to circumstances, I end up with a 1u server that has no aux power connectors available. I "have to" use this server, so am considering abandoning

Hi, I bought the Digium Asterisk Developers kit a few weeks ago, got the x101p and a tdm400p cards, both seemed to be working fine up until today, now the tdm400p seems to have gone to the great PBX in the sky ...... If I left the reciever on the analog phone I don't get a dial tone and nothing appears on the asterisk console, If I use my cellphone and dial in I see Asterisk Ready.

Hi, I'm in the process of reinstalling an asterisk server with the following configurations: Asterisk 1.6.1.4 Asterisk-Addons 1.6.1.1 Asterisk-Gui 2.0.4 Dahdi-Linux-Complete 2.2.0.2+2.2.0 TDM400P Rev 1 Board 5 Dahdi "sees" the TDM400P (dahdi_cfg -vv) Asterisk "sees" the TDM400P (dahdi show status) The gui, unfortunately. does not see any analog card, which is rather

I'm having real problems with my connection to BT, it is a home line, but after a while it sets an alarm and only a restart of asterisk resets it could some one look at the below configs and suggest any changes to make this more reliable Thanks for your help Robb asterisk version 1.6.1.10 dahdi version SVN-trunk-r7445 dahdi_scan active=yes alarms=OK description=Wildcard TDM400P REV I

Hello, Over the past several weeks, we have been having an intermittant problem with our deployment of a TDM400P card (4 fxo). ?We have tried many things, and the problem still re-occurs. The Problem: Occasionally (every 48 hours), the TDM400p card will stop answering incoming calls on ALL fxo ports. ?Attempts to send outbound calls on any Zap channel will result in hearing a loud

Hi list, I have a "Wildcard TDM400P REV I Board 1" with 4 FXO modules and * 1.0.9 up-and-running. Only 2 FXO ports are used for 2 analog phones and are doing fine. I now wanted to use the 3rd and 4th port, but when I insert an analog phone, take it off hook, I do not get a dial tone. With my 1st and 2nd port, I get messages like: -- Starting simple switch on 'Zap/13-1'