similar to: Question about TDM11B Configuration

Hello my brothers and sisters, Is "X100P" card suitable to VoIP? and if "yes", am i need to only "X100P" and "Asterisk" Package? or i need also to other cards or packages? and if "X100P" card not suitable to VoIP, please recommend a another card, (please take in your account that i would like to connect standard analog line to the card

Hello, my config : aah 2.6 (asterisk 1.2.4) , centos 4.2, 1 TDM11B (1 Fxo / 1 fxs ), 1 phone, 1 softphone I'm in France When someone from PSTN calls and hangs up before the call is answered, internal extension keeps ringing until timeout occurs. PSTN line keeps busy. Hangup detection doesn't work. I've played with different paremeters (callprogress, busydetect, busycount,

Hi, I got my TDM11B and am trying to get it to work on my PC. However, I'm having difficulty getting the wcfxs driver to load. I've Googled this problem, and while there are others who have ran into the same problem, none of the solutions work for me. I would very much appreciate it if you could suggest things I could try to resolve this problem. Details follow. The PC is somewhat

Hi Asterisk Community, I have a development with a TDM11B in it. I am trying to connect this to a exchange line as well as a UK telephone and are looking for some pinout information for the FXS port. Is it the centre 2 pins that at the tip and ring. I have been digging around the Digium site but cannot seem to pick up this info. Any help appreciated. Thanks -- Gary

I recently purchased a TDM11B so I could hopefully hook flash the FXO from either the FXS (on the TDM11B) or a SIP device. From the FXS, I've tried hitting # then transferring to an extension that flashes the line then dials the FXS again (3020). This seems to send me to a busy signal and the console tells me no such host of 3020 (the number I'm on). The call on call waiting gets sent

Dear All, All these days I was ahpppily using Asterisk with TDM11B, but from today all of a sudden asterisk has started acting strange. The telephone device connected to channel 1 rings continously, following info is displayed on console -- Starting simple switch on 'Zap/4-1' Jan 28 15:38:58 ERROR[77024176]: callerid.c:261 callerid_feed: fsk_serie made mylen < 0 (-8) Jan 28

Hi list! I'm not getting incoming CallerID in The Netherlands on my TDM11B. Everything was configures according to the docs at digium.com. The error on the console is this: Feb 13 16:49:40 ERROR[16123]: callerid.c:260 callerid_feed: fsk_serie made mylen < 0 (-84) Feb 13 16:49:40 WARNING[16123]: chan_zap.c:5396 ss_thread: CallerID feed failed: Success Feb 13 16:49:40 WARNING[16123]:

Hi all, I need help in disconnect supervision. Im running on AAH ver.2.5 at home with TDM400P with 1 FXO and 1 FXS (TDM11B). I have implemented DISA on AAH for origination (PSTN to VOIP bridging). I'm facing problems with disconnection supervision. My calls are not getting disconnected at times and it causes a lot of loss as the provider is charging me. After some serious study i have found

Hi list! I have a TDM11B (DevKit). When I start asterisk everyting is working fine. However as soon as I do a RELOAD from the console the FXS port stops working. I have a phone connected to it, when I pick up the phone after issuing a RELOAD I get this on the console: Jun 1 20:27:38 WARNING[14316]: pbx.c:2357 __ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's'

I have a TDM400P with 1 fxo and 1 fxs module(TDM11B) i just wanted to know how to configure the zaptel and zapata & extension conf to work well Please help me on that

Hello all, This is my first post to this mailing list. I've been following the development (not as a developer, but as an interested user) of Asterisk for over a year now. It's only in the last month that we've decided to give it a try (hey, might as well...everything else we use is open source). We ordered a TDM11B (1 FXS, and 1 FXO). The card and Asterisk install was uneventful.

I get this error no matter what I do. I've tried switching to different locations on the TDM400p to no avail. I've got the card setup with the green module in slot1 and red in slot2. My config files are exactly like the "Configure Dev-Kit PCI (TDM11B)" example. http://www.digium.com/index.php?menu=configuration#TDM11B When I run ztcfg -v, I get: 2 channels configured.

I looked through this forum and other lists trying to get a definitive answer to the VMware question... I have a Windows 2003 Server box that is my development machine. It is fully loaded with many s/w packages, dev tools, etc. It's not very feasible to convert this box to Linux without losing an awful lot of stuff. On it, I'm running VMware GSX 3.1.0. Under VMware, I have a

Today I received a TDM11B (1 FXO and 1 FXS) and got it installed just fine. I bought the card mainly to get caller ID to work properly in Sweden, and that works just fine. However, if the called or calling party hangs up after I hangup my SIP channel, polarity CID detection kicks in and dials a couple of signals to my incoming context. This happens with Asterisk 1.0.6 and CVS-HEAD. I have tried

I have fixed this before, but I cannot for the life of me remember how I did it. I have a TDM400P with one fxo module and one fxs module. I setup Asterisk @Home and everything works fine, except when I try and call out. If I call out with a SIP phone it calls the zap extension and not the pstn line? If I take the zap extension offhook and call with the SIP phone it dials out the pstn line

Hi all, I am working wit a Mandriva 2007 and have installed the asterisk svn 1.4 with success. I also used the patch for cellphones and it works perfectly. I was that happy that I decided to buy a TDM11B and it works. Now, I want to study a bit the code used by this people. Does anybody know how can I go deeper in this code with funcitonal bloqs in order to understand how is possible to

Hello I'm in Egypt and i would like to provide VoIP Service. what i (as hardware and software )need to provide this service? only Digium telephony card and Asterisk (Configuration Package)? am i need to any other getway , hardware or software? if any 1 have complete solution, please provide me it. thanks in advance Regards Eng. Yousri Farouk Linux System administrator www.egypt.com

Hi all!! I have downloaded the asterisk from svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk-trunk (is the asterisk 1.4 subversion). I also downloaded the patch for cellphone and make it work fine. Then I bought the tdm11b board to have phone connection in my computer. I installed the hardware for zapte and the libpri modules in my Mandriva 2007 and the lights of the pci card

The information (or lack of it) on upgrading from zaptel to that @&*^QW%&^%!!! dahdi is very frustrating. I cannot find anything on how to uninstall zaptel, i found an earlier post to this list which suggested make uninstall and make remove in the zaptel directory which just generates errors and does nothing (on zaptel 12.1). Then i install dahdi-linux and dahdi-tools and i want to

Hi, I saw there was an active discussion about pci passthrough support in KVM. I'm not a Dev, only a sys admin that need this support. As i saw on the KVM home page that this feature is plan for 2H2008 in the roadmap, I wondered if you already have a more precise time frame for this release? Xen already have this feature but for the rest i prfer KVM. As this is a key point for the final