similar to: problems after adding FXS module to TDM400P

We have been having a problem with sshd on our shell server. This has been happening since March 4, 2003 or before IIRC. Initially I thought the next OS upgrade, to 4.8 would fix this. I am accustomed to haveing little things go away in a month or two. I think we jumped to 4.7-STABLE on Feb 28, 2003. Some exploit fix wasn't being MFSd to RELENG_4_7 fast enough for my nerves (cvsd?). It

Richard Schilling wrote: Do you notice wether or not it takes a certain number of connections for the bug to show up? I'm not seeing this problem with just a few people connecting via sftp (about 2-4 times per week). --Richard On 2003.06.13 22:36 Scott Lambert wrote: > We have been having a problem with sshd on our shell server. > > This has been happening since March 4,

John Todd's Onlamp article mentions an OpenBSD version as of June 2003. Have I been sleeping while reading asterisk-users? Is it a seperate project or is it just making Asterisk portable? Who is working on this and is it in the main CVS yet? Do they have device drivers ported or just the software parts of Asterisk? I think the software parts would be relatively simple but time consuming.

Hi, I had ordered a TDM40B and developers kit a few months ago. I have everything installed and working, with one exception - sound quality. When placing a call it sounds like a very bad cordless phone - lots of hiss / static in the background. This even happens with the dialtone, though it is much worse one the call is connected. This does not occur when the phone is directly connected to the

Hi all, We have just set up trixbox latest with a Rhino r1t1 card, hooked up to a plain E1 PRI line. We call fine SIP to SIP, but as soon as we make a call from SIP to PSTN all sounds become unintelligible screeching or static kind of noise on both ends, when we call PSTN to SIP the PSTN side seemingly OK at least we hear no screeching sound, but the SIP side is a even worse screeching

I've got the latest zaptel and cvs asterisk software loaded on my phone server running FC3. And yes, It's fully updated and udev is setup correctly. I've got a TDM400P with one FXS and one FXO module installed. When I load zaptel and wctdm and run ztcfg -vvv, I get this: [root@asterisk ~]# ztcfg -vvv ZT_CHANCONFIG failed on channel 1: No such device or address (6) Zaptel

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 TDM21B (A new TDM20B that just arrived plus a X100M that I already had). This card sometimes works, and sometimes only the FXO module is seen. (correction to previous message that incorrectly listed X400M/S400M when I'm using only the single port versions of these modules (X100M/S110M). By "works", I mean all 3 modules have the green lights on the ports on and when

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

In my attempt to setup a single FXS line I have been following the instructions for "Telephony Card Drivers" on the asteriskdocs.org site. I have managed to checkout, make and install the zaptel code and can load the zaptel module but when I attempt to load the wcfxs module it tells me: ZT_CHANCONFIG failed on channel 1: No such device or address (6) FATAL: Error running install

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.

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 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

The HP DL380 G5 (like many rack servers) has no AMP Mate-n-Lok connector available to attach to a card that needs more power than the PCI bus can provide, like the TDM400P when FXS modules are used. HP has confirmed that there is no part they sell to give you such a connector, and Digium says their business edition folks got it to work, but only by doing nasty warranty-voiding things to the

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

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

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?

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

I just received two TDM400P cards, but I'm having problems with them. The full info is at: http://pastebin.com/824079 Extra, I'm using : libpri-1.2.3 zaptel-1.2.10 On a x86 stable Gentoo box. Kernel: 2.6.17 gcc-4.1.1, glibc-2.4-r4 Is that an hardware problem? Should I try the other card? -- Gustavo Felisberto (HumpBack) Web: http://dev.gentoo.org/~humpback Blog:

I think I have a bad FXS module on my TDM400P. Channel map: >> >> Channel 01: FXO Kewlstart (Default) (Slaves: 01) >> Channel 02: FXS Kewlstart (Default) (Slaves: 02) >> >> 2 channels configured. >> ZT_CHANCONFIG failed on channel 1: No such device or address (6) The zaptel loads ok, but the wctdm reports No such device or address and stays loaded. The