search for: tdm01b

Hello, I am trying the setup the TDM01B card. 1 FXO. I connected it to a regular home line. in the /etc/zaptel.conf, I have fxsls=4 In the /etc/asterisk/zapata.conf I have: signaling=fxs_ls language=en group=1 context=default channel => 4 When I start asterisk, I get this error: ERROR[10376]: chan_zap.c:6584 mkintf; Signaling reques...

Hi all, Your help would be greatly appreciated, I have been struggling for days with inst / config of TDM01B. I have installed TDM01B. Using Asterisk 1.2.9.1 on RH Ent 4.00 cat /proc/interrupts shows card wctdm on int 10. Green LED is on (module is in port 4). I have my /etc/zaptel.conf which includes fxsks=4 zttool show my card. ztcfg -vv shows my card. zap show status shows my card but I see only...

The TDM01B is 4 port capable but has only 1 FXO module. I'm running asterisk 1.2.7 and zaptel 1.2.5. I cannot seem to get the TDM01B working. When I do the zttool it shows 4/1/0. I can dial out from a POTS phone up to the point that the cable plugs into the card. Here is my /etc/zaptel.conf loadz...

Hello If someone had the opportunity of trying those two analog cards, how do they compare? Digium's sells for $150 while OpenVox's sells for $95. Thanks.

I'm attempting to use an asterisk box with a Digium TDM01B as voicemail for an existing Meridian/Norstar PBX with an ATA-2 adapter. We're having problems where hangup is not always (but sometimes) detected. It's not detected probably 70% of the time or more. (The users transfer callers to an extension--caller then has to navigate a menu to g...

I'm trying to setup a demo where my Asterisk box with a TDM01B (FXO) answers an incoming call and forwards that call to a SIP softphone (X-lite.) Seems all is built/installed okay: # ztcfg -vv Zaptel Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. I'm pretty new at this and the...

...eally new to PBX systems so all this is totally foreign to me. They currently have 5 inbound trunk lines and about 20 analog phones. >From what I can gather they are using the Altigen Quantum cards that support 8 extensions and 4 trunks. >From what I can gather the solution is a TDM04B and TDM01B to bring in the lines from the CO. A T100P to connect to a channel bank for the analog phones. I am looking to do this as cheap as possible since the main reason we want to upgrade is to take advantage of the VOIP functionality in Asterisk. We also want to use IAX to connect another small office...

...my zapata.conf is: context=incoming signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no busydetect=no callprogress=no musiconhold=default usecallerid=yes callerid=asreceived channel => 1 I am using an SNOM200 SIP phone and a TDM01B (1-port FXO bundle). When I run asterisk and dial from the SIP phone I get this error: *CLI> -- Executing Dial("SIP/555-83ee", "ZAP/1/92262802") in new stack Jul 23 13:50:24 WARNING[-267056208]: channel.c:1860 ast_request: No channel type registered for 'ZAP' J...

I'm setting up a small office with about 8 SIP phones. Incoming and outgoing lines will be through IAX. We would also like to use an analog line for 911. Is the TDM01B a good option for this kind of configuration? Are there gotchas I'm missing? Finally, we would like to be able to use analog fax machines in the office. Would it make more sense to purchase the TDM400 card with 1 FXO and 1 FXS port and use the FXS for the analog faxes, or to split the analog...

I have a couple SIP phones on a PIII 1Ghz 256MB * server with a TDM01B connected to the PSTN. Calls between SIP phones are clear. Calls to the PSTN are quite noisy. The other person does not hear noise but I hear quite a bit. It is not an annoying sound but definitely much noisier than typical PSTN or even cell phone calls. I believe I have a TDM400P REV H car...

Dear list, We have an weird problem with our FXO card (TDM01B). When I made a call using this channel, all goes well, the called phone rings but when the called phone answers the call. In me handset I can hear an weird sound like a "Clack". I tryed diferents TDM cards and modules, and my zapata.conf is like, language=en context=from-zaptel switchty...

Hello, I have configured my TDM01B Card (1 FXO Port ) as follows (below) but it will not pick up an incoming call. Any suggestions/tools to see what the problem is? I have looked at zttool where this line changes but I don't understand what it means (The last digit changed from 0 to 1) Total/Conf/Act: 4/ 1/ 1 /etc...

Hi i have buy a Digium TDM400P with 1 fxo modules TDM01B for connect my asterisk to a french analog line. In my zaptel.conf, i have: loadzone=fr defaultzone=fr fxols=3 when i load the module, i have in logs: Nov 24 06:13:40 gw kernel: Freed a Wildcard Nov 24 06:13:40 gw kernel: Zapata Telephony Interface Unloaded Nov 24 06:13:40 gw zaptel: R...

...> You can reach the person managing the list at > asterisk-users-owner@lists.digium.com > > When replying, please edit your Subject line so it > is more specific > than "Re: Contents of Asterisk-Users digest..." > > > Today's Topics: > > 1. Re: TDM01B card configuration (Dave Cotton) > 2. Re: [SPAM:***** SpamScore] RE: > [Asterisk-Users] Call Transfer > using SIP clients (Frank Schoep) > 3. RE: presence and IM again, want to develop a > working"hack" > (Florian Overkamp) > 4. calling shell scri...

Folks, For the last few days I've been trying to experiment with a Panasonic PBX and an X100P but have run into quite a few problems which I am not sure if they can be solved with this type of card (how about TDM01B?) 1) I wanted to use *'s IVR capabilities, so I routed the calls to the extension where the x100p was connected to. Asterisk should answer the call, playback a message, dial another PBX extension and if no one answers dial another extension (via IAX). The first problem I ran into was that th...

I am using asterisk version 1.2.6 with Zaptel version 1.2.5. I have a POTs line coming into a Digium TDM01B. It appears to not be getting CID at all. If I hook up a POTS phone to the line CID comes through fine. Inbound and outbound calls work fine but there is just no CID on inbound for this channel.The incoming route for the channel is Zaptel Channel 0. No DID or CID settings applied. My IP routes inbo...

...o create a production system. Since lots of people are apparently using this in production environments now I just assumed that echo IS avoidable. As others have recommended, I created a test system with the proposed production parts. I bought a couple different SIP phones to try and a Digium TDM01B card. I am using an older PIII 1Ghz system with 815chipset (PCI Rev2.2) with 256MB for my test system. The only thing that will be different on a production system is that I will be using a newer chipset PC with faster processor and 512MB. Probably Intel 7505, 7210, or 7211 chipsets which seem t...

Guys, I need to use Asterisk to call out PSTN numbers via an analogue line. I understand Digium manufactures these kinds of cards, but can someone tell me which model number it is. I really only need a card with one or 2 analogue ports max. Walid -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, My asterisk setup was working when using a Wildcard T100P. I recieved incoming calls and was able to make outgoing calls. I recently decided to change my card to a TDM01B one (one FXO). However when I dial a working extension, I can see on the console that the call goes through sucessfully, but it is never actually executed (i.e if I call my mobile phone, Asterisk shows the call as being answered, but there is no actual call or ringing on my mobile.) This also occu...

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