similar to: IVR Prompt errors (scratchy)

Hello, I have found a "bug", I think in the way TDM400P cards handle FXO interface disconnect/re-connect problems. Normally I do keep all the wires connected from my CO / PABX quite securely, but I had a need to re-route the cable from one side of the desk to another, and I simply disconnected the RJ-45 connector and plugged it back in. THIS PROMPTLY RESULTED IN VERY VERY SCRATCHY AUDIO

On Mon, 2005-01-10 at 08:01 -0600, asterisk-users- request@lists.digium.com wrote: > > I am new to asterisk but learn a lot about it to this mailing list > and > > wiki currently i am facing problem about sip phone i have "PA 1688" > > chipset ip-phone and i have iptel.org sip account i registered > locally > > and through iptel.org comfortably my problem

Hello all, This is my first post to this list, but I was wondering if anyone had any suggestions on reference material for a first time user. I am planning on installing Asterisk on a RH9 box with an X100P and a TDM400P. Right now I just want to get the system up and running with all calls being routed through the FXO card. Once I get that configured and working properly, I would like to

This was submitted to the Dev list last week, but there was no response, and perhaps it wasn't the right group. I am developing an application in which I need asterisk to pass on an incoming call to a separate IVR server. The problem is that asterisk appears to hang up while the IVR is playing back a sequence of recorded voice and systhesized voice prompts. My setup is: Analog line

Yo all, I have a problem with a Dutch (KPN) PSTN-line, connected to an X100P. The call wil sound OK at first, but after 10-20 minutes, the audio will start to crackle. Soon after that, this crackle turns into a continuous noise and the parties won't be able to hear eachother anymore. It also sometimes happens that the party on the TDM400P hears a very loud, short-delay echo of themselves,

Hello Group, I am new to asterisk but learn a lot about it to this mailing list and wiki currently i am facing problem about sip phone i have "PA 1688" chipset ip-phone and i have iptel.org sip account i registered locally and through iptel.org comfortably my problem is that when i called from my sip phone to analog or any number after connection my sip phone generates very load scartchy

I have two grandstream budtetone-100 and cisco 7960g phones. When I talk via speaker phone on either of the phones I get a lot of echo. Any suggestions? Also how do I turn on the mark echo canceller. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Tuesday, April 13,

Dear All. I'm very new to CT, but attracted by asterisk. I plan to start learn to build IVR, based on asterisk. Do I need FXO/FXS card to start building simple IVR box ? Can I use OEM X100P - FXO PCI Card ( http://www.digitnetworks.com/store/product_info.php?products_id=28 ) to start build an IVR box ? What is the cheapest FXO PCI card that can use with asterisk to build my first IVR box ?

Hi Guys, I recently got a TDM400P 4 FXO for use in the UK, this at the time seemed like a good idea as I had good results with an X100P clone. Installation went great and call clarity is excellent no echo like I had on the clone card. My problems start with detecting hanging up the line. If a person calls into the system and speaks to me on a SIP phone when I hang up the call clears

Hi All, I'm attempting to use two X100P cards and one TDM400P card. What driver do I load for the TDM400P? Is their any documentation for this card yet? Also, I know this has come up before but I can't seem to find it in the archieve... how do I configure the second X100P card? Anyone have a working sample? Thanks in advance, Randy

I'm looking to add a single FXO port to my Asterisk box. It looks like my options are a Digium Wildcard X100P off eBay for $6.99, or a Wildcard TDM400P with an FXO Module from Digium for $125. Can anyone explain the tradeoffs (other than the ability to put 4 FXO/FSO modules on the TDM400P). What about RTC for the system - I know the TDM400P provides it. Does the X100P? Thanks!

Hi, I have just built my home Asterisk box into a better PC that became available (still only a P2 350 but it only has to manage 1 analog line).. Anyway I have built it on Fedora Core 1.. I have an X100P and a TDM400P (1 module installed).. These cards were working fine in my older PC that was running my Asterisk at home.. The inbound calls via the X100P to my sip phones are working great..

With one IVR payment system, I noticed quite a difference in DTMF transmission between these two cards. The IVR missed nearly all digits from X100P, while receiving digits from TDM fine. Since neither card process or synthesize audio, what can the difference be? (This particular IVR has problem with some regular phone devices, too.) Yuan Liu

Hi everybody I am having problem detecting fax with my X100P. I have RedHat 8 as OS and an X100P and a TDM400P. The X100P being plugged into PSTN. I have successfully installed tiff-v3.5.7 and spandsp-0.0.1 and also patched Asterisk wthout problem. Here is my zapata.conf file context=cda signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5

Hello. All. I am a bit new to asterisk, started from half a month ago. I am setting up a home asterisk server with analog card. I am using asterisk 1.4.27. At the moment, I bought a X100P card and installed it on my computer. I used it to connect my home phone line. For the moment, it works fine when dial in. Soon I noticed when I dial out through it to my mobile, it can't hang up

I've been trying to get a console channel working without success. The sound card, which is built into the motherboard, is a VIA Technologies, Inc. VT82C686 AC97 Audio Controller. Using the oss drivers (vi82cxxx_audio) in kernel 2.4.23 and chan_oss, I just get beeps and screeches. Using alsa drivers (snd-via82cxx) and chan_oss (using the alsa oss emulation), playing sound works, but

Hi! I have just bought the X100P and TDM400P cards to install on my computer to implement the PBX. I also downloaded the newest softwares asterisk_ver0.5.0, libpri_ver0.4.0, and zaptel_ver0.7.0) to install on my computer (Red Hat Linux 8.0). All packages are compiled well. When I use "modprobe" to load drivers (modprobe zaptel, modprobe wcfxo, modprobe wcfxs), the first two (zaptel,

Hi All, Sorry if this has been around a millions times.. I have been off this list for a few months now.. I have installed the latest asterisknow (upgraded asterisk to 1.6 as well) and I am having a hard time getting my X100P and TDM400P working.. Its all new to me with dahdi because my old server is still running Asterisk 1.0.2 so there have been lost of changes.. Can someone point me in

Hi to all. My asterisk pbx has a tdm400p card with 2 FXO cards on it. I configured the extensions.conf to send all the call incoming from that zap channels to an IVR system. I see in the asterisk CLI the call incoming and the playback of the message custom/myfile but no sound is played on the channel, i cannot hear nothing. If I change the configuration and i send the call to an internal sip

Here is my system: POWER SUPPLY 450W ALLIED ATX450P4 R(41) MB NFORCE2 A7N8X DELUXE ASUS RTL(Standard) CPU AMD|2500/333 ATHLON XP BARTON R(Standard) DDRAM 256M|DDR333 PC-2700 -K %(Standard) HD 40GB|WD 7200RPM 8MB WD400JB%(70) VGA ASUS|V8170MAGICII/T 64M MX440SE(58) CD ROM 56X|AOPEN CD-956 RTL(22) Besides the above parts, i have 2 X100P FXO PCI cards, and 1 TDM40B I bought 2 phones from radio