similar to: A configuration question

I easily get confused when try to undertstand FXO & FXS ports. Is it possible to use an ATA to connect to a TDM400 card. If so, would I use FXO modules or FXS modules ? My goal is to connect my asterisk server to Vonage (via the ATA they send me) so I can use thier standard plan and do with out the Softphone account feature that only allows a few hundred minutes talk time. Thank you, Steve

I'm running Asterisk (stable branch downloaded 2-Sep-05) on RedHat 9 and I have a TDM22B installed (TDM400 w/ 2 FXO and 2 FXS ports). Everything seems to work except threeway calling. I can establish a threeway call, but it uses up BOTH FXO lines. Note that I DO have threeway calling active with my Bell service. Here's a typical scenario: 1) Call 765-1574, 2) When they answer, press

Hi All - I pulled from a working system a TDM400 with one s110 fxs and three x100 fxos. I put it into a new box and the fxs no longer works. The fxos work just fine. I thought it was odd, but I chalked it up to a random chance failure and ordered another s110. The replacement doesn't work either, and now I'm confused. dahdi_scan recognizes the s110, but it says it fails. Output

Hi, I'm running * 1.2.7.1 on Red Hay 9.0 w/ a TDM 400 2 x FXS, 1 x FXO. I'm using a VTech cordless that makes three short beeps when someone another extension is picked up, presumably this lets you know if someone is trying to listen in.. Everything works, except the VTech now makes the three beeps everytime you try to use it, even if another extension is in use. It seems as though the

I recently swapped 2 FXO modules on to what had previously been a 4 FXS version of the TDM400 board. The FXS ports are recognized - the FXO ports don't appear to be recognized (ie modprobe wcfxo and ztcfg both say channel 1 isn't there). Has anyone experienced this problem? All software is current as of this AM. If the old FXS modules are re-installed all works just dandy (other than

My TDM400/zaptel 1.2.10/Asterisk 1.2.13 suddenly stopped bridging outgoing FXO calls. If I make a call (from an FXS channel) to a PSTN destination, and the other side answers, Asterisk will show continued ringback on the FXS channel, while the PSTN side hears silence. No error message appears. If a call from PSTN terminates on the same FXO, Asterisk can still ring the FXS channel, and when

I have a small system based on one TDM400 card with - 2 FXO modules for incoming lines - 1 FXS module for one phone The system was working fine in the past. The motherboard was exchanged and during the switch, the phones line was rewired with a mistake and a incoming phone line was connected to the FXS module, and there was ringing voltage on the module. Now, the system kinda works but the FXS

Folks, I recently swapped a TDM400 FXS card that was working perfectly into a new server (running recent CVS), and it's either misbehaving (unlikely), or I've missed something obvious (much more probable). Everything seems to be working, but I can't get any dialtone from it when I plug a phone into any of the 4 ports!! All of the jacks are lit (green lights) but they all seem dead

Hi. I have been searching through old threads and various forums, but can't seem to find out what the consensus view is here on the best type of FXO hardware to use for a home asterisk implementation. Specifically, should I buy a FXO ATA like the Sipura 3000, or a PCI form factor port like the Digium TDM400 series. I have read horrible things about problems with the X100P, so I have ruled

(Im' answering cc the list, so the knowledge keeps there, and maybe some more qualified answers become). Am Mittwoch, den 15.10.2008, 18:00 -0700 schrieb Francisco del rosario: > Hey Rodolfo... Need some help from you ... > I need to know what hardware do I need to make SIP calls if I set-up > asterisk > So the situation is that I have a PC and configure the software of my PC to

I have an Asterisk serving 15 people using the X-Lite soft-phone. Currently they all register to the internal IP address of Asterisk (192.168.1.110). I only use VoIP internally. External calls go PSTN. I'd like to arrange it so that they register to our external WAN address (port forwarded to Asterisk) so that they can go mobile and still have Asterisk service. Is it possible to arrange it

I have a TDM400 with an FXS and FXO interface. I have adjusted the TX and RX gain for both the FXS and FXO channels so that everything is balanced. I hear myself very loudly with no lag. I also get alot of background noise when the bridge is formed between the FXS and FXO channels. I have read the following articles: http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation

Hi all, We have an asterisk version v1.6.1.20 with a TDM400 board (2 FXS and 2 FXO). We want to do a transfer "blind" and "attended" from a line external connected to one FXO. We have made configuration, and transfers from internal lines (FXS) work fine but from (FXO) not. We have made 2 test, one work fine from FXS and the other form FXO no. Test 1, work fine: 1) A

iH have a TDM400 w/ 1 FXS & 1FXO card. after about a day or so the FXO stops working. after a reboot i get the following error. to get the card working again i have to shutdown, physically unplug/replug the card (reboot only, power off/on does not work) any ideas what may be going wrong here? thanks - hcir Zaptel Configuration ====================== Channel map: Channel 01: FXO

Hi, I'd like to build a personal PBX connecting 4 or 5 analogic phones with a ADSL line and I'd like to know what is the right card I need I visited digium site and I think TDM400 could be the right choice but I cannot understand how it works...I think it has 4 slots where 4 modules (FXS or FXO) can be inserted. How many cards do I need to connect my ADSL line to 5 phones? I think I

-------------------------------------------------------------------------------------------- Originally posted at http://forums.digium.com/viewtopic.php?t=18045 -------------------------------------------------------------------------------------------- Hi! I'm trying to configure a DISA setup (Asterisk 1.4.11). Only, executing DISA seems to prevent any DTMF detection capability when using

I just deployed * on my home system last Sunday. 2x since then the Zap hardware seems to have malfunctioned on some way. One time it would just screech out one FXS, even though it would ring. The other time * would bridge to my FXO but it never got out on the line. I have a new TDM400 with 3 FXS and 1 FXO. Both times I tried unloading the zaptel drivers (which worked) and reloading them,

Perhaps someone with more experience can help me solve this puzzle: Asterisk 1.4.18 on a Dell PowerEdge SC440, CentOS 5, 2.6.18-53.1.14.el5, Digium TDM400 with 3FXO(1-3) and 1FXS(4) Phone service is provided by XO Communications via their XOptionsFlex service (5 analog lines, 3 of which are used by Asterisk). For simplicity I refer to them as xx43, xx44, and xx45 - based on their phone numbers.

Hi all, I would like to make a script/program that would be able to show lots of status information from my analog FXO lines (and FXS lines in the near future). Example of interesting status information: - Hook status: is there a call being made with that zap? - Voltage status: cable connected, voltage values (if possible), line ringing? - RX/TX Volume status I'm using a TDM400 card with

And if that ain't confusing I don't know what would be. I bought a TDM400 with two modules (FXO, FXS) a couple or so years ago and ended up never using it. Passed it along to a friend who is having some problems with it. (He isn't on this list.) We've both tried searches using Google but haven't been able to find anything that helps. So this is more a question of