similar to: Viable Alternatives to TDM400P

Hi folks, Any Brits on the list got experience of hooking devices with standard BT plugs to the TDM400P? The TDM400P has the RJ11/RJ45 socket but I'm not sure if a standard adapter will do the trick or if we need to consider ring capacitors etc. Any links to buy the stuff would also be appreciated. Cheers, Simon -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Folks, Does anybody have experiences with plugging 3 TDM400P cards in one PC?? I think about a Asterisk box handling 8 incoming analogue lines and providing 4 lines to an old analogue PBX. I read a lot about trouble with the TDM400P cards so this idea seams to be not really god, or? Ciao Joerg -- _____________________ Don't PANIC

I have a CentOS-5.6 remote server in a house in Italy, where there are occasional thunder-storms. There was one yesterday, when the electricity went off 3 times, for a second or so on each occasion. My server, an HP MicroServer, came back (re-booted) on 2 of the 3 occasions, but not on the third. I assume that the problem arises because the machine does not close down properly. (Although it is

Hi, Digium stopped to produce TDM400P and the new TDM410 is too new to find it in our shops. The only alternative available is a "fully-compatible" Openvox product...but is it really "fully-compatible"? Any experience about Openvox products (card and zaptel versions, etc...)? Thank you! Giorgio.

I would like to hear from anyone good or bad as what their experience has been in recent times with STABILITY of current builds of Asterisk and drivers for TDM400P. The sort of configuration is: 6 incoming POTS lines. ie. 2 TDM400P cards. I am not concerned with: price points, or the advantages or disadvantages of using POTS vs ISDN technology, but simply RELIABILITY & stability of the

I'm testing a TDM400P with FXO module to receive incoming calls from an analogue line and send it to a SIP device. To recieve callerid, I need to use cidsignalling=dtmf and cidstart=polarity. The problem is that when a call is finished, the TDM400P seems to require about 20 seconds to prepare for the next incoming call. If a new call comes in within 20 seconds after the previous call was

After working with the Grandstream GXP 2000 series phones, I have decided that I am quite unhappy with their problems, both voice quality, volume, features and others. For their price now, there are plenty of phones to choose from as well. So subjectively what would be the best Hardphone for a small/medium business with multiple line support, BLF, etc. Are the Cisco 7960 the best of the

Hello All, Does anyone know of an expansion module (keypad extension for attendant) that works with the gxp-2000? - Jason -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060508/f49a24e3/attachment.htm

I installed the TDM400P and installed it on a system running on Fedora Core1, please refer to the steps below. I connected an analogue phone to the FXS port. When I pick up the speaker I don't hear the dialtone although when I press a number key I hear a DTMF generated. What could be the problem? I appreciate your help 1- First I installed the TDM400P in the PCI slot and connected

Evening all. I'm looking at building an Asterisk system for one of the projects of the Charity where I'm the SysAdmin. The project has two analogue phone lines - BT Featureline Compact, we're in the UK - that I'd like Asterisk to handle. My current quandary is which FXO interface to use. I've been looking at the Digium TDM400P and Sangoma A200, which are similar prices for

Hello I setup Asterisk@Home with purely VoIP and it worked fine. I then added an X100P card so I could call out / take inbound calls via PSTN and that went fine. But I have just added a TDM400P card (specifically a TDM30B) and now problems. Here is some of the output. Any ideas on what I should be looking at next? When I run genzaptelconf -s -d I get lots of erors on screen - bit I can

I am sorry if this is an FAQ, but there doesn't seem to be searchable archives, and the supported manufacturer's list doesn't include "surge master". Google was also not much help. I have just put a Linux box and the customer has a "surge max " UPS. Anyone any ideas what protocol it uses? There is a Green serial cable. The manufacturers web site is here:

Sorry to post such a newb set of questions but I have been hammering about trying to get Asterisk running on FC2 machine reading everything available (I think that is what stuffed me, shouldn't have read it all :-) ). Config FC2 running Asterisk 1.0.0, with the h323 compiled in and installed correctly. Amazingly enough I have everything compiled correctly and installed. I am running a

I've got the latest zaptel and cvs asterisk software loaded on my phone server running FC3. And yes, It's fully updated and udev is setup correctly. I've got a TDM400P with one FXS and one FXO module installed. When I load zaptel and wctdm and run ztcfg -vvv, I get this: [root@asterisk ~]# ztcfg -vvv ZT_CHANCONFIG failed on channel 1: No such device or address (6) Zaptel

Has anyone got a sensible zaptel.conf and zapata.conf for 2 TDM400P's working with UK set-up. They're set-up with 7 analogue phones and 1 PSTN port. Currently zaptel.conf has fxoks=1-7 fxsks=8 loadzone=uk defaultzone=uk It's really zapata.conf that would be useful. Currently using the zaptel/asterisk that comes with Ubuntu (latest) which needed a bit of tweaking (1.2.16), but could

Dear All The setup is te110p with an 8 channels PRI to make and receive all calls. SIP phones throughout the company. TDM400p with 4 FXS modules to send/receive faxes and make credit card transactions. I have an analogue phone on the tdm400p for testing. I can receive calls to the exten. There is a dialing tone. However, when I try to make a call I get a busy signal. Asterisk stated busy then

Hello everyone. This is a message I've sent before on Sunday, no one replied so I'm reposting it (guess not everyone's at work 7/7) I've got this really annoying and beyond-my-knowledge-to-debug problem. The line connected to my FXO port gets marked "out of order" by my telco operator. I don't know how to explain this further. If I dial my own number from a

I am located in the UK, and I am using Sipura spa-2000 adapters to connect analog phones to a voip network. The network connects to the PSTN as well via the Sipura spa-3000 adapter. I would like to provide surge protection for the spa-2000 and the spa-3000 adapters. 1. For spa-2000, fxs port: What is the maximum tip-to-ring voltage before damage to the the adapter occurs? 2. For spa-2000,

All, I have a fairly old Compaq R3000h which is overdue for new batteries. Before I go and splash out a few hundred francs, I'd just need a quick sanity check - when I disconnect the power (UPS under load), it cuts out pretty much right away, so the batteries are flat (even though the charge indicator says full). When I reconnect the power, the surge triggers a 16A mains fuse. This sounds

As best i could figure it out, I've installed dahdi and rc4. My TDM400P doesn't answer fxo or fxs. /etc/dahdi/system.conf: loadzone = us defaultzone=us fxoks=1,2 fxsks=4 /etc/asterisk/chan_dahdi.conf: [house-phones] context=internal ; Uses the [internal] context in extensions.conf signalling=fxo_ks ; fxo_ks Use FXO signalling for an FXS chanel dahdichan => 1 ;