similar to: Frustrated with echo...

Hi I am fairly new to Asterisk. I will have few questions on one Asterisk system: Description: I looked at Asterisk Mall says that "The TrueLine SMB PBX is perfect for the small office providing service for up to 28 Telco lines". This is a 1U Rack Mount Single T1 Asterisk Appliance. Question: Does it mean it has 7 PCI bus and takes 7x4=28 channels (if they used 7 TDM400P card

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

I just picked up a couple CAC Access Bank 1s loaded with FXS that should be arriving shortly. Does anyone have one that they use with Asterisk? If so, would you be willing to shoot me a note with your current configs? I'm not very familiar with CAC/etc, and it would save me countless hours of muddling through .. :) Any others tips you can give me on these banks would be very

Having ISDN FXOs and a bunch of ISDN DECT base stations, I went looking for a CAPI compatible active BRI adapter capable of NT mode. I have found that the 'Eicon Diva Server 4BRI' fulfills theses requirements. However, at EUR 1387 for four BRI interfaces, it does not come cheap. Are there other users of similar hardware that can point me to alternatives ?

I have a TMD400 card installed in a PC with one fxs (installed in slot 2) and two fxos (installed in slots 3 & 4).? fxos work fine but I am unable to get a dial tone for any devices connected to the fxs.? I have correctly connected the power supply to the card and I have even tried moving the card from slot 1 to 2 on the board. Below is from the console when I try to route a call from FXO on

Does anyone on-list have direct experience with the new analog cards from Sangoma? I'm thinking about FXOs with echo cans. Need 2-4 ports but don't want to go through another TDM400 style experience. Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc.

Been using a few different FXO interfaces (X100P, Voicetronix, and SPA-3000) at a couple sites and they've all run into nasty issues at times. I'm looking at moving one of my offices and am in the market for a solution for connecting 4 FXOs. I got excited about recent offerings from Digium and Sangoma in the way of new cards but I hear they're causing troubles for more than a few.

helleo ,I have one wildcard installed on the server,and configured it.there are two modules which both FXOs .all configurations complete,and when we start asterisk use "asterisk -vvvvc",it shows asterisk ready,no warnings.afterwards when the sever connected to the PSTN in the one port which configured,and have a phone line connected to the other port then run

Hi, I have pushed my radio player for FxOS to GitHub: https://github.com/emilis/worldradioplayer Currently it uses a precompiled data from http://dir.xiph.org/yp.xml: https://raw.github.com/emilis/worldradioplayer/master/static/js/data/xiph.org.stations.json I will have to switch to asynchronous station list loading and filtering in the near future and then will nag you again about the JSON API

Hi, I am trying to do the world's most simple install. I have a Wildcard TDM400P with 3 ports: 1 FXS on port 1 and 2 FXOs on ports 3 and 4. (i'm not using port 3 for now, put want it for expansion purposes) I simply (to start with) am looking to have the FXS phone ring when an FX0 port is dialed. I would also like to be able to place outgoing calls on the FXS through the FXO. Right

Hey all, I've been seeing this repeatedly and am wondering if anyone has a clue what's causing it.. at least once a day I see two zap fxo channels being bridged, and hanging..now, these two channels should never bridge, but they keep doing it.. any leads on where to look for what's causing it? here's what I see when I 'show channels' Zap/37-1 s@from-pstn:1

Hi all, I just installed an TDM02B. My system is a dell pc with linux 2.6.12-1.1456_FC4 asterisk-1.2.0-beta1 zaptel-1.2.0-beta1 libpri-1.2.0-beta1 in /etc/zaptel.conf I have (all others are default): fxsks=3-4 <--- I saw light in the ports channels=1-2 <--- change it to 3-4 has same result but... [root@nmsd0 asterisk]# /etc/rc.d/init.d/zaptel

Whoo hoo! I just received my WIP300 from voipsupply. I have to let it charge before I can play with it. A few quick comments: - I started a Wiki page at voip-info to post issues, firmware news, etc. I really like the wealth of info on the GXP-2000 page, so I wanted to start something similar for this phone. http://www.voip-info.org/wiki/index.php?page=Linksys%20WIP300 - My kit

Hi all, We have an analog line from telco, on which 3-way calling is subscribed to. This line is plugged into an FXO module on a tdm400p. If an incoming call comes in on this line, can */zaptel send Flash to telco via the FXO module? If it could, then an incoming call could be 'transfered' to a cell-phone, for example, with a single analog line. (where 'transfer' is really

I am trying to reconfigure an asterisk box that was using an HFC-S card with bristuff but is now using 2 analog lines therefore I want to use the TDM02B to connect to two POTS lines. The TDM02B has 2 red modules. I have this in /etc/zaptel.conf loadzone=nl defaultzone=nl fxsks=1-2 I have /etc/asterisk/zapata.conf signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes echotraining=400

I'm rolling out an installation with snom 360s in the near future. Simple SOHO configuration, 3 FXOs hanging off a TDM400B, 4 snom 360s, a snom 200, some variant of IAX softphone, and an IAXy or Sipura 2002. I have the 360's set up to subscribe and notify for the line use lights, which works like a charm for interoffice calling (between the 360's, anyway. The IAXy, 200 and,

Hi everyone, I just installed a TDM02B and surprisingly, I had really no problems except one. If I place an outbound call on the Zap line (Zap/3), everything works fine except when the called party hangups before I do. I do get congestion, but that is expected. However, when I try to make another outbound call using that Zap line, the CLI shows that the call is being dialed, but nothing

-----Original Message----- My question concern outgoing calls. How can I configure my extensions.conf to get a PSTN line on my TDM04B card in the following order : first trying on the channel 4 then if 4 is busy then switch to 3 if 3 is busy then switch to 2 and if 2 is busy then say there's no more line available. I don't want to dial on the first channel as it's my main number

Hi, I am looking to integrate Asterisk with a Siemens HiCom 150e via BRI and wondered if anyone is able to offer any advice. In simplistic terms, my goal is to pass calls from the HiCom to the Asterisk box. e.g: HiCom user dials access code and can call Asterisk extension or establish SIP call over Internet. Likewise, I'd like Asterisk to be able to present a call to the Hicom, either

Hi all, I've tried researching this, and so far, have struggled to find any contemporary information on the issue, so I do apologise if asking this irritates people who have answered this before. I have managed to set up Asterisk 1.8 on the web server here. I have two softphones (Ekiga) able to communicate with it. So far so good. I'm now curious to see if I can link it with the PSTN