similar to: tdm400p static - out of ideas

Ryan, from the console what does "zap show channel 1" or 2/3/4 in your case say. I have X100P's and I seem to be having similar sounding problems. I noticed that the above command shows the channel to be off-hook at all times when a phone line is plugged in. I don't know why or if it is a bug in the application reporting the status. dbc. Ryan Courtnage wrote: > On July 8,

Hi all, A while back, there was a short thread on using the FXS interface from a Primus TalkBroadBand device (a DLink ATA) as a incoming line for the FXO interface on the TDM400P: Primus <--> DLink ATA FXS <--> TDM400P FXO <--> Asterisk In that thread, a couple of people suggested that this does not work reliabley, and the ATA FXS <--> TDM FXO link 'goes

Hello. I've seen several posts talking about line quality using Digium cards that are sharing IRQs or on machines where X is running but after trying all of those fixes I am still having a problem with line static on outoing calls. BTW, calls that are from one extension to another extension have no static, however, they have occasional clicks and pops. At any rate, I was wondering if

Hi all, We have an analog line from telco, on which 3-way calling is subscribed to. This line is plugged into an FXO module on a tdm400p. If an incoming call comes in on this line, can */zaptel send Flash to telco via the FXO module? If it could, then an incoming call could be 'transfered' to a cell-phone, for example, with a single analog line. (where 'transfer' is really

Hello all We have tested for a mounth or two an asterisk PBX using one T1 channel bank with 24 fxs and one TDM400P digium card with 4 FXO modules. This worked with minor problems, the most notorious being some sporadic static noice or failure in the first FXO module on the wildcard. Now we have a client with 12 pstn lines and 48 extensions and we are trying to deploy an Asterisk PBX server

I have Asterisk CVS-HEAD-05/25/04-17:13:22, Copyright (C) 1999-2004 Digium. Usign exclusively digium hardware. 3 TDM400P cards. 1 4xFXO 1 4xFXS 1 1xFX0 & 3xFXS When * is attending FXO calls, bridged to FXS calls, natively ofcourse, at a random time, the call hangus up. Also, for example, if a call is done, and an other extension hangup, there are some probability that the other extension

Hi all, Several months ago we built an * box with a quad-FXO tdm400p (REV e/f). >From the get-go, there has been a problem where occasionally (2-3 times a week) zaptel/* will not detect the ringing on a line. (The call will ring through to telco voicemail). The problem is not specific to a single line or FXO port on the tdm400p. I have 2 theories: #1 - the ring voltage for some calls is

I have an intermittent problem with the one FXS line that I have. On most calls, the first ~5 seconds of the call has a loud buzzing noise on the line. After 5 seconds or so, it fades off to nothing, and the sound quality is great. Searching for "buzzing" on the list doesn't give a whole lot to work with. The buzzing happens on calls that are routed over both my FXO line and

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

Hello list... 1) Is there an IAX softphone that supports any kind of voicemail indicator? 2) I have 2 TDM400Ps installed in a system. I need the audio on the analog phone (FXS modules) to be amplified somewhere between 10 and 15db so I set rxgain and txgain to 15 for the FXO interfaces and FXS interfaces. When a call comes in on the FXO at this setting, the call sometimes has about 20 seconds of

Hi all, I curious to know if other UIP200 users have this same issue: You flash (XFER button) to consult-transfer a caller to another extension. If the transfer target party is unavailable (ie: voicemail), there appears to be no way to get the original caller back. If it's a known limitation, has anyone come up with a functional work around? Thank -- ..................................

I have a TDM400P card all installed, and I can call out from a sip phone, but when I call from the PSTN into the asterisk machine, as soon as the Answer() gets called, the dial tone changes and is sounds like there is a lot of static on the line. Below is the part of the dial plan for answering the call. exten => s,1,Wait,1 exten => s,2,Answer exten => s,3,Dial(Sip/pfriedel,20,tT)

Has anyone had success with the TDM400 in production? I have multiple boxes where these cards lock up and the only thing that will fix them is to unload *, modprobe -r wctdm, modprobe wctdm, load asterisk. Does not matter if it is a FXS/FXO module. I know this topic has been discussed many times before, but my questions is not "is anyone else having this problem" since I know that

Hi all, I've run into 2 separate problems relating to fax: 1) Using tdm400p + fxo, Asterisk is unable to detect the fax from some fax machines (from others it can). Using zap barge, I can confirm that these troublesome calling fax machines _do_ send the fax tone loud and clear. Are there certain circumstances where I should expect a Digium card to fail in detecting a fax? 2) Using

Two days ago, I was talking on the phone from the FXO, to a SIP phone. After some time (like 1h30m), all of a sudden, there's a huge noise, like a buzz... Really loud. So I hungup, and called my asterisk box again... All I could hear was that sound. Someone called me from the internet, and as Asterisk dialed the FXO, all she heard was that noise too. So, I logged in my Asterisk server,

Hello all, The "Secret Agent" final release of the Asterisk Management Portal is now available for download: http://amp.coalescentsystems.ca/ This exciting new release adds a great deal of functionality and flexibility. Thank you for all the contributions and feedback! 1.10.007 - Added AMP Users (multi-department, basic multi-tenant) - Added incremental upgrade script

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!

I couldn't be happier with the simplicity of this - but it's a hack! Hi all, I'm currently using a SIP client (BT101) to connect via DSL to a remote instance of Asterisk. - Asterisk has a private IP behind my OFFICE router. - The SIP client has a private IP behind my HOME router. I'm doing this _without_ the use of STUN or proxy servers. Here's how it works: -

I just got a TDM400P with 2 FXO cards. I got it all configured and I can place and receive calls. I seem to be getting static on the call, mainly when I speak. E.g, if I call someone, I can hear them just fine, but they would hear static. Not a lot...more like a constant background "hissing" noise. What can I look at to try and troubleshoot what is going on, or is this normal? I

Ok all here is a strange one...... I have a TDM400P with 3 fxo modules. I can very rarely make an outbound call to the PSTN....about once every 10 tries. However if I use a analog phone pluged into the same phone line as one of the tdm channels say channel 4, and I place the analog phone off hook and then place a call via asterisk , it work everytime. It seems like the TDM400p is having