similar to: X-Lite, X100P, and Speex

Hello People, I am a newbie asterisk and happy user, i have configured a x100p card and everything works nice, i can forward incoming connections to a x-lite software client and works out of the box, However when i try to make a connection between two x-lite clients then no audio is transmited, i have followed the instructions on voip-info.org, the tutorials on onlamp and i have read some

sorry to bother with this trivial issue, but i am loosing all my hair ;-) got 2 x100p's and * on a slakware box x-lite to x-lite works fine! i also have: #ztcfg -vvv Zaptel Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. and in extensions.conf i got: [locals] exten

Hi all! I'm new to asterisk and I'm trying a simple config with: - Debian GNU/Linux (unstable) - last version of Asterisk - a X100P card I have a problem with dial out from a SIP software phone (XLITE) to a public number (ex. my mobile phone), asterisk start the call, but nothing happen... If I run "ztmonitor 1" I can see the right RX level and if I try to make a call with an

To all: I am having an echo problem with X-Lite and iaxComm. I am using the monitor speakers and a desktop microphone. My problem is that the sound from the speakers is repeated by the microphone causing an echo that is annoying. Is this correctable? I have searched throught this archives and played with various settings but have not been able to fix this. I have purchased a couple if

Hi All... I'm currently trying to have an extension of my Lucent Partner phone system connected to Asterisk using an X100P. The issue I'm having is that the Lucent Partner analog port connection have different ring and dialtone than the one specified for US in indications.conf. Anyone have accomplished this connection before. I need to know what I should use for the different

My home * system I use for test/dev stuff has recently started to miss all calls. I have two of the "X100P" boards from x100p.com that are about a year old in it. They both have worked in the past no problem. At some point in the past couple of weeks they both stopped answering the phone. I don't have a specific date because only I and a couple of other people call in on those

This seems to have been reported before, but I've seen no resolution: http://lists.digium.com/pipermail/asterisk-dev/2003-August/001470.html http://lists.digium.com/pipermail/asterisk-users/2003-August/019091.html http://lists.digium.com/pipermail/asterisk-users/2003-October/024472.html When forcing use of Speex, no sound comes out at all (Speex-1.0.3 on the Asterisk server) When forcing

Hi I've got a cheap card from x100p.com for my pots line. I haven't found a definate answer if it is possible to get danish(DTMF without signaling before it). I have had a look at bug #9 but that is written longtime ago. I am running zaptel 1.4.7 and asterisk 1.4.14 BRIstuffed 0.4.0-test4 both from the xorcom debian packages. I believe they should contain some kind functionality for DTMF

Hi, I recently decided to change my setup from AsteriskNow to plain-asterisk 1.4, which I wanted to set up and configure myself on a server running Debian Etch 64bit version. Hardware: Asrock motherboard, model 775Dual880-Pro, with a Celeron D running at 2.8GHz, 1GB memory, standard Nvidia GF4MX videocard, and one X100P clone card. Running the AsteriskNow, everything worked fine, except for

Hello I was wondering whether anyone has managed to get an X100P card to receive Caller ID information in the UK (connected to a BT line)? I guess that it probably won't work at present because BT sends the CID information before the first ring - hence there is no event to wake up the channel. I can hear it using ztmonitor, but it is far too soon for anything to receive it. According to the

Hi! I currently have the X100P and TDM400P Asterisk cards. For obtaining Asterisk PBX: I have no direct connection to internet, so I use ftp and download the neweast packages: zaptel-0.7.0, libpri-0.4.0, asterisk-0.5.0. The environment I compiled the tool: PC (HP): Intel PIII 500 MHz OS: Red Hat Linux 8.0 (Upgrade from Red Hat Linux 7.1). When I compile the first package (zaptel-0.7.0), I

I'm playing with codecs at the moment and have found some notices errors when x-lite/pro connects to asterisk with Speex or ILBC. Initially I was getting garbled sound, but after changing magic number for both codecs to 97 (as per http://www.voip-info.org/wiki-Asterisk%20phone%20xten%20xlite and http://bugs.digium.com/bug_view_page.php?bug_id=0000918) I was able to get normal voice. BUT,

Hello All, I have this strange problem, I can dial out with my sip phone and it seems to work relatively well, but when I call in, the line just rings and rings, I get no indication in asterisk that it's detecting an incoming call. The strange thing is that in ztmonitor 1 -vv the rx volume goes from around 99% when the line is idle to 0% as soon as the line starts ringing. I'm located

I've been trying out the newest X-Lite (Build 1012) with iLBC and speex codecs. If I enable only iLBC _or_ SPX on X-Lite and call the echo-test on my asterisk server, the call connects, but I get no sound. If I enable only iLBC _and_ SPX, X-Lite indicated that it has connected with iLBC, and I hear a weird squawking. My sip.conf contains: allow=iLBC allow=SPEEX allow=gsm I've heard

hello all, I m joined this asterisk group few months back. Actually i have installed asterisk on my PC and using X100P PSTN interface card to make this PC work as a single line VoIP gateway. Now i have one problem which is as follows: As i told i m using X100P card. I am getting good quality of speech from it whenever i m making call or receiving call through it. Now I have

Hi, I've compiled the speex codec and installed as per docs, speex works perfectly between Asterisk gateways using IAX protocol. My problem is between X-Lite softphone and a Asterisk Gateway users cant hear anything when the speex codec gets selected. Has anyone else experienced this with X-Lite or any other softphone client? Is their any other Softphone for Windows that makes use of the

Hi folks! I bought a X100P a while ago and know I've tried to get it working here at home again ... but I can't manage to get my X-Lite client working with Asterisk (CVS from a day ago) ... I've downloaded the latest version of X-Lite and I believe that I've set it up correctly ;-) But I cant get it to register with my Asterisk - I only get "Login timed out, contact your

I have just started developing asterisk, and am trying to start simple. I have a X100P device and an S100U device. I am trying to use the examples provided, where I add a few lines to the /etc/zaptel.conf, /etc/asterisk/zapata.conf, and /etc/asterisk/extensions.conf so that I may connect an analog line to the X100P and an analog phone to the S100U. When I dial the analog line, it should ring

Hi all, The Wildcard X100P card or the Asterisk Developer's Kit (LITE) are available somewhere in Europe or must be ordered in US? Thanks, Dan

I'm attempting to configure a machine with a DevKit Lite (X100P & S100U). After I modprobe wcfxo, the machine goes into some kind of loop after about 10 seconds, where it spits out what appears to be 32-bit addresses, ad infinitum. At this point, the machine becomes completely unresponsive to keyboard input (i.e. Ctrl-C, Ctrl-Alt-Del, even CapsLock and NumLock states [lights] cannot be