similar to: Line death not recognized on TDM400P?

Hi Can TDM400P detect polarity reversal on FXO module? We have C.O. lines that reverse polarity on Answer and release. Thank You This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you

Hi, I'm no phone guy but I've been having quite a bit of fun messing with Asterisk. I just unwrapped my new Wildcard TDM400P (TDM40B) which is the new 4 port FXS PCI card. It has 4 RJ45 connectors and I'd like to hookup 4 regular phones which are RJ11. If I crip RJ45 connectors to the phone cords do I simple use the center two pins (pins 4 and 5)? Thanks, Randy Smith Tiger

On an Asterisk 1.2.12.1 system using a TDM400P with two FXO and two FXS modules, servicing two POTS lines: When dialing a number, such as a bank, or pharmacy, where it is required to enter a long series of numbers via the phone's keypad, an unexpected hangup occurs. The hangup does *not* occur when entering the numbers, as one might initially expect, but after the called end begins to

All, I've recently installed a TDM400P and am unable to get it working. It has four FXO modules on it. What I would like to do is have the first two ports as inbound from pstn and have them ring the last two ports which will be connected to regular analog phones. In Zaptel.conf I have: fxsks=1-4 In Zaptel.conf [channels] context=default switchtype=national rxwink=300 ;

<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

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:

I'm a complete Asterisk novice and have an installation based on the Asterisk@HomeCD. I've installed my TDM400P with 2 x FXO & 2 x FXS, but every time I try to dial out, I get a message "No circuits available". Can someone confirm the pinouts for connecting the FXO's to a UK BT Line - I have RJ11 connectors on the back of my TDM400P card, so ideally I'd like to

I have a TDM400P with 3 fxs and 1 fxo ports. I need to know which phone connector corresponds to which module and also which port number. If we are looking at the card with the PCI connector at the bottom, fxs/o modules at the top and the phone jacks on the left - do the phone jacks start at the top or the bottom (top is port 1 or bottom is port 1)? And which phone jack belongs to which

Hi all. First thing: I want to thank you all for your help over the past month as I've been learning asterisk. This is one of the more helpful lists. Even when I ask questions that have answers in the wiki ( which I missed because I've been over studying ). Second thing is this: My office is scouting out VoIP solutions, and I have suggested an asterisk solution. We will be

Hello everybody, I am sorry to bring this up again if this kind of echo issue has ever discussed. Phone2 in below call path experiences quite annoying echo: Phone1 --> FXS (TDM400P) --> Asterisk --> SIP GW --> PSTN --> Phone2 It is annoying as on phone2, we can hear the whole words we say with the level of maybe 25% of the original sound. I can reduce the echo to maximum with

Here is my system: POWER SUPPLY 450W ALLIED ATX450P4 R(41) MB NFORCE2 A7N8X DELUXE ASUS RTL(Standard) CPU AMD|2500/333 ATHLON XP BARTON R(Standard) DDRAM 256M|DDR333 PC-2700 -K %(Standard) HD 40GB|WD 7200RPM 8MB WD400JB%(70) VGA ASUS|V8170MAGICII/T 64M MX440SE(58) CD ROM 56X|AOPEN CD-956 RTL(22) Besides the above parts, i have 2 X100P FXO PCI cards, and 1 TDM40B I bought 2 phones from radio

Hi all, Yesterday, in an attempt to take back my phone room, I pulled everything apart as far back as the demarc and rebuilt it. In the process of putting things back together I accidentally connected my incoming lines to my FXS ports and my phones to my FXO ports. I quickly realized the mistake I made and corrected things but not before one of my FXS modules was smoked by incoming ring voltage.

Does asterisk support the intel 537/md3200 chipset? I don't want to start any flames here, I know all about using generic crap in asterisk,[*] which I really don't approve of other than for testing, but I have a customer demanding a generic chipset for his one backup analog line. He will not spend the money for a Digium card and says he will find another company if I can not provide a

I've noticed that some callers listen to our main menu and don't press any keys. I have it set up to restart the menu a few times and eventually hang up. I'm wondering if these are wrong numbers (in that case, why don't they hang up) or they really want to speak to someone here but don't understand the menu (what's so hard about "for the operator, press

Folks, I've just successfully set up Asterisk (as part of the Asterisk Management Portal installation). When I say "successfully", I mean that I have gone through all the steps detailed for the installation of AMP and not hit any snags there. I can connect to my asterisk server via ssh and can also connect via Http to the portal to change settings in AMP. Now I'm trying to

This may sound kind of crazy and I maybe missing something. But are you placing the call on hold so you can hear the hold music. This may not be the case but you may have to place the call on hold to here the music. Also you mentioned sound, you do not need a sound card in the asterisk box to use this hold music feature. Hope this helps. -----Original Message----- From:

I have a gateway, which I call from my mobile phone (free of charge, since it is the same phone company). This gateway gives me a dial tone. I can than dial to any extension number or even other gateways, .... It is getting more a trouble to remember all the numbers, or to key in all the long phone numbers when you got the dialtone. I was thinking of using for this Sphinx2. How can I

I am able to get this script to dial, but I am unable to talk or hear anything. The script asks for the number to call and the the caller id to display (if user is not at their normal extension). Once submitted, the external extension receives a call, once answered the call is then placed to the dentition number. The script works as the call is place, but I cannot hear or say anything. Any one

Hi I was doing some bandwidth testing, and my incomming usage is 36% more than my outgoing bandwidth. The setup is IAX2 trunking using GSM codec. Is there any obvious reason I am overlooking to figure out why there is such a big difference between the two.? I am using CVS-head September 3rd, maybe there is a version skew? Any suggestions will be appreciated. Thanks Clive

Hi, I'm looking for a tutorial or installation guide for SER to be used with asterisk to solve the remote SIP agent problem. All the documents available are for large scale installation. Any help is highly appreciated. Regards. __________________________________ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail