similar to: Faint background noise/crackle on FXS port on TDM400P

We are running asterisk with a single POTS line for local calls and a voip line for long distance. Whenever we receive a call on the POTS line it is more than likely, but not always, going to have significant distracting echo. In addition to that there is occasional heavy crackle or static. I have tried to follow the guidelines at :

I recently plugged in Phone to my TDM400P Card to test out something I mostly use sip phones to interface with *. All of sudden I'm getting lot of line static noise coming of the card is there any settings I should look at or anything I need to do on the command line at this point I'm open to any ideas I'm running 0.7.1 on Redhat 9.0 machine. Any insight would be greatly appreciated.

Hi, In regards to the previous thread about static and snapping on incoming calls to the TE410P card when using a Dell 1850 server I now seem to be getting significantly better call quality with two E100P cards. So far I haven't been able to make any calls with detectable static on the line. Regards, Aaron

Hi Folks, Does anybody have experiences with plugging 3 TDM400P cards in one PC?? I think about a Asterisk box handling 8 incoming analogue lines and providing 4 lines to an old analogue PBX. I read a lot about trouble with the TDM400P cards so this idea seams to be not really god, or? Ciao Joerg -- _____________________ Don't PANIC

All, Experiencing some issues on my FXO lines. If a call comes in on an FXO and then get transferred to another FXO (say to call someones cell phone), those two lines will stay tied together indefinitely. This happens to us when we transfer an incoming call to our on call guys after hours and on weekends. We have installed 3 other * boxes and they do the same thing. We use a Adit Channel bank

Server A has 4 PRIs coming into it, and provides a SIP connection to Server B. When we call into server B via a DID from one of Server A's PRIs, we get crackly sound on recorded playback only (prompts, IVR, voicemail instructions, etc.), but not on actual live voices once you're talking. Server A provides this same service to other servers and problem does not appear in those cases.

Good Day List, I'm having some issues with echo cancel on my Asterisk box, and have done extensive reading and have gained some useful pointers from this list but have a couple of hopefully fairly simple questions. The Asterisk box is connected via 20 FXO ports on a TDM2400 with the Hardware echo cancel module. Echo cancel almost works, but the users hear what they describe as a

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,

For my Smart 2200, a charged battery string is anywhere between 25.6 and 26.8 volts. I'm seeing nut report 13.0 volts, which seems suspicious. Even if the UPS were reporting half of the actual battery voltage to keep protocol compatibility with models having a 12V battery system, the value of exactly 13.0 seems unlikely. According to the page at

Anyone? :-) > > ----- Original Message ----- > From: "Matt Klein" <mklein@nmedia.net> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Sent: Wednesday, December 29, 2004 7:37 PM > Subject: Re: [Asterisk-Users] Cisco 7690 Voicemail Problem > > >> >> a faint scratching sound

Leslie, This message is most appropriate for the NUT user list. I kept Rik Faith's and Nicholas Kain's copyright information because the tripplite_usb driver borrows heavily from the tripplite (serial) driver, but they are not responsible for the mess that I made in the USB driver. See below. On Dec 28, 2008, at 5:28 PM, Lelsie Rhorer wrote: > Gentlemen, > >

Hi, I am sorry for possible offtopic as this question is not quite NUT-related but I cannot imagine any other place with people of such deep and intimate knowledge of UPS behavior. One day all of a sudden my APC Back-UPS RS 1000 refused to go online (green led) even though mains was connected. I used upsc to see what is going on, and ups.status was "OB DISCHRG", battery.charge was

I received an additional module for my TDM400P yesterday. I installed it last night. Before I installed it, I had a working 1x1 PBX. After installing the physical hardware, I spent a couple of hours unsuccessfully attempting to get dial-tone on the new FXS part. I gave up, made sure the old FXS still rang and went to bed. I should have tested further. This afternoon, I got a call at work

Hello, I want to use Asterisk PBX in front of my old, legacy PBX. The legacy PBX can be outfitted with caller-ID and is already able to handle Calling Party Control Signal Detection (this is a Panasonic KX-TD1232 Super Hybrid PBX. My question is how would one enable Asterisk to control the TDM400P/FXS port to provide to the /FXO CO port on the legacy PBX, support for proper answer supervision/CPC

I thought I had posted this, but I didn't see it in the archives, so I guess I hadn't. I've got FXS lines going to a legacy IVR. When I Dial into one of these lines and then hang up, FXS plays the Congestion tone until the IVR drops voltage. I would like the IVR to hang up sooner. I could do this by either making the IVR recognize the standard Congestion tone, or changing the

Hi; I'm trying to connect a TDM400P with an FXS module to a Valcom V-9940 Paging adaptor. This port on the TDM400P was connected to a 2500 Set and was working I just re-connected it to the Valcom (which is known to work on a Telco POTS line) and its not picking up. The Valcom docs say it need a minimum of 75 Volts at 20-30 Hz to recognize a call... So the question is what ring voltage

I have a Digium TDM400p (1 FXO, 1 FXS) with the FXS module connected to a standard analog handset with caller id display (US caller ID). Although it appears that caller id information is coming into asterisk (it shows up in voicemail), I can not get it to display on the analog handset. Is there anything special I need to do to send the caller id info out the FXS port? I've tried a few

I have a TDM400P with one FXO module and two FXS modules in it. I also have a Wildcard X101P. After trying hard to get things working on various Intel computers, but having echo problems that made it not really usable, I decided to try it on some older PowerPC (Macintosh) hardware running Yellow Dog Linux. Things started off smoothly. Both zaptel and asterisk seemed to compile okay, and both

Hy, I would want to "test" the FXO and FXS channels of a TDM400p with the asterisk. For example, when I want to Know if a sip peers is correctly register, I use ChanisAvail, but if a module FXS hasn't a phone connected, How could I Know it?. Note, that I don't want to Know if the channel ZAP is correctly configurated. This information is given for Chanisavail. Thanks for

Hellow everyone, here's an other newby question. I've got a * configured with the card in the subject line. At times Asterisk fails to notice a disconet from the incoming line going into one of the FXO ports. Consequently it just keeps the line off-hook for ever and that causes my provider to mark the line aut of order. Is there any way to "help" Asterisk notice the disconect?