similar to: Echo and crackle

Hi, I have had this issues for ages but have ignored it, but my handsets get faint background crackle on the FXS port. When I connect the handset directly into my PSTN, it goes away complete, so I am generating this from somewhere in my asterisk box. I get it immediately when the phone goes off-hook and it stays through the dialing and call progress. I can't think of any other devices

On 23/11/17 19:49, Sorin Srbu wrote: > The Skype flatpak at home worked beautifully, that is to say for about ten > minutes, then the sound (using a Koss SB/45 headset) it started crackling to > the point of not being usable anymore. I hung up and reconnected, then sound > was fine, for another ten minutes at which point the crackling returned. > Checking the sound prefs, I saw

Hi all, I have setup a Dell PowerEdge 2550 with a Sangoma A200 card with 2xFSO and 1XFS modules. The PowerEdge specs are 1 x P3 1133MHz, 512MB RAM. Sangoma A200 has 3 analogue PSTN lines connected. This server is based in Office 1, with 5 users all with a Linksys SPA942 VoIP Handset. There is another Office (Office 2) connected to here using VPN. There are two users in Office 2 with the

I don't really know the name of what I want to look for but maybe someone could tell me if it would be available. I have a number of old analogue cell phones laying about here and I was thinking it would be useful if I could set up a short range base station for them that would cover maybe an acre or so. What I would like to be able to do is use it to connect into Asterisk and this way

I have two call waiting problems. I have a POTS line into and FXO port and telephones on an FXS port 1) I can't seem to use the flash button(on the phone) to answer a call waiting call. I see the callerid coming though and here the call waiting tone, but I just can't seem to answer it. The flash button seems to have no effect. I have: callwaiting=yes in zapata.conf

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

I have a normal backup phone (and an alarm panel) sharing the POTS line with the Digium X101P FXO: | | Wall |>---+------X101P FXO as Zap/5 | | | Phone & Alarm Whenever the Phone is used, Asterisk sees a 'false ring' signal immediately when the phone is hung up. The Alarm panel dials out nightly at around 1AM, and each time it completes the call, Asterisk

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

I just bought an TDM400P card with one FXS port and a X101P FXO card to try and put together the beginnings of a PBX here. The computer they are going in should arrive in a week. So I thought I would start learning how to config the card in an IBM Aptiva we have here, an old 400Mhz P2 box. However when I put the TDM400P in the box the machine won't POST. Does this mean that the card

I am finding that I have to increase the txgain in zapata.conf to 8 when my X101P is connected to my BT phone line, otherwise people can hardly hear me. This then gives echo issues. Do other people have the same problem on BT lines. I was wondering if I should give BT a call and get them to increase the gain on the line. Strange though as the rxgain is OK and I don't have this problem with

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,

I've been following the list for months, and I have a working Asterisk setup, but it'd be *really* useful to me at this point if someone could summarize when Asterisk has echo problems and when it doesn't. For instance, I usually hear a far-end echo when talking on my 7940, but not when using a POTS phone plugged into a TDM400 FXO port. It doesn't seem to matter if the call

Greetings all, I am interested in setting up a lab in my house to run my home phone system. Can some one point me in the right direction hardware wise for achieving this goal? I happen to have a few 3com NBX phones, and have been told that they are SIP compliant but am not exactly sure if I believe 3com...I would be interested to know if anyone has these things working on *. In any

I am looking to replace all of the old "Bell" (POTS) phones in my home and office with IP phones. As you can imagine I don't have a huge budget to work with but I want phones that will provide acceptable voice quality and durability. There are basically three categories as I see it 1. satellite phones (low cost, low function) 2. primary domestic phone (good quality, POE capable,

We have a S100U which we were using as our sole internal phone extension. Today it died. Picking up the phone now results in a brief dialtone followed by bursts of random static and crackling noises and then a second or two of fast-busy followed by silence and more crackling noises. I've tried the hardware attached to a couple of different machines as well as a couple of different phones.

In features.conf, I have uncommented the transfer features under feature map, but I still cannot transfer using a POTS phone on an IAXy adapter. I think I am missing something here.... Any help is appreciated. Here is features.conf: ; ; Sample Parking configuration ; [general] parkext => 700 ; What extension to dial to park parkpos => 701-720 ;

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

Here is the setup: POTS -> PRI -> Asterisk -> ATA (Fax) The ATA is set to only 711. Asterisk's sip.conf sets this device to only 711. Yet, faxing works less than 50% of the time. I cannot possibly be the only person with this problem. I can't even get app_rxfax to work. It answers but rxfax drops the call 5 seconds into it. If anyone out there has near 100% success with

I have two IAX trunked *, there are loud crackles and pops, they are dialing out a T-1 and are sip devices, it also drops words, any help or Ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060516/dc68274f/attachment.htm

I just installed two new X100P's today in my RH7.3 box and was poking around /proc What does the RED refer to on board 2? /proc/zaptel/1 Span 1: WCFXO/0 "Wildcard X101P Board 1" 1 WCFXO/0/0 /proc/zaptel/2 Span 2: WCFXO/1 "Wildcard X101P Board 2" RED 2 WCFXO/1/0 Any clues on how to setup channels on the two interfaces?? I am up to here in