similar to: X100P ring detection failure

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 I've got a cheap card from x100p.com for my pots line. I haven't found a definate answer if it is possible to get danish(DTMF without signaling before it). I have had a look at bug #9 but that is written longtime ago. I am running zaptel 1.4.7 and asterisk 1.4.14 BRIstuffed 0.4.0-test4 both from the xorcom debian packages. I believe they should contain some kind functionality for DTMF

Hi, I recently decided to change my setup from AsteriskNow to plain-asterisk 1.4, which I wanted to set up and configure myself on a server running Debian Etch 64bit version. Hardware: Asrock motherboard, model 775Dual880-Pro, with a Celeron D running at 2.8GHz, 1GB memory, standard Nvidia GF4MX videocard, and one X100P clone card. Running the AsteriskNow, everything worked fine, except for

Hardware is an SM56 card (X100P clone). When the line hangs up, ztmonitor displays full bar (or whatever maximum allowed by rxgain) in RX. It only drops zero when the line picks up (and remote was silent). Is this something of concern? The zap channel seems to work despite echo. Additionally, what's the objective of tuning with ztmonitor? I mean, what would indicate an optimal level?

Hey guys, I'm having some trouble with gain settings using a Wildcard X100P and Zaptel 1.4 Beta 2 on Asterisk 1.4 beta 3. My transmit volume between an IAX client (idefisk) and a POTS device through my 1.4 box is great, but my receive volume is terrible. I can hardly hear a word they say. I've cranked my gains up to 100, inserted the wcfxo module with boost=1, but haven't had

Hi all! I'm new to asterisk and I'm trying a simple config with: - Debian GNU/Linux (unstable) - last version of Asterisk - a X100P card I have a problem with dial out from a SIP software phone (XLITE) to a public number (ex. my mobile phone), asterisk start the call, but nothing happen... If I run "ztmonitor 1" I can see the right RX level and if I try to make a call with an

I have a line from BT (UK) connected to my asterisk system, on a TDM400P. I am able to see either distinctive ring cadences or caller ID but not both. If I try to enable both, all drings show up as 0,0,0. This is a pain because, if I make a call out over that line and the number I call is busy, I can elect to camp on it (ringback), which results in a different cadence of ring when the called

Hi, everyone: I am developing a system using Asterisk, TDM-400 analog cards, analog lines, and Polycom SIP phones for internal extensions. Initially there was bad echo but after a series of efforts, I've managed to reduce it to a negligible level (it only happens when both parties speak simultaneously, and even there, only for a few hundred milliseconds). From an echo standpoint, things are

Hello all, I've been lobbying for some time at the #asterisk IRC channel. Until now, I still can't find a solution to my one way audio problem. I rebuilt the Asterisk-1.4.21.2 from the Debian Testing repository on my Debian Etch. I got a Digium TDM400P with 1 FXO (channel 4) and 1 FXS (channel 1). My SIP extension phone located inside the LAN is a SNOM 300 IP phone. This one way audio

I have just purchased a 2nd hand X100P, if I do a ztcfg -vv I get: "Zaptel Version: 1.4.0 Echo Canceller: MG2 Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured." My understanding of the above is that the zaptel driver has detected the card. What do I now need to do, in order to get an incoming call to work with

The SM56 MODEM manual says it does. But when used with zaptel 1.2.12, nothing shows up. Yuan Liu

Hi Tzafrir, I am not sure what to look for, so I haveattached both the contents of /var/log/kern.log as well as the outputof dmesg. If you are looking for something specific, perhaps I can grepfor it and send you a smaller version. I have attached these as a tar ball to get below the 40K limit at which moderator approval is required. Thanks for the help. > Date: Fri, 7 Sep 2007 10:26:11

I have an ubuntu 8.10 machine. Linux 2.6.27-14-generic i want to configure a x100p card an use it with asterisk, so i download, compile and install: asterisk-1.4.24 dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2 libpri-1.4.9 i try almost everything i found on the net but without success: if i run lspci: 04:06.0 Communication controller: Motorola Wildcard X100P when i run dahdi_hardware appears this:

I am finding that with the combination above, the zap channel will hang up, though the call has hung up. I am make all changes in my zapata.conf file that I can find on the internet with nothing fixing this issue. I have added busydetect=yes, busycount=6, and made sure that the signalling is set for fxs_ks these setting worked fine with zaptel 1.2 but not with the newest zaptel drivers.

Hi! I have a very strange question. I'm using trixbox with Asterisk 1.2.23-BRIstuffed-0.3.0-PRE-1y-j. I configured and installed the HFC ISDN card with a script, as here: http://www.trixbox.org/forums/trixbox-forums/help/how-install-hfc-card-trixbox Now i have 6 SIP hardphone, and softphone, using 1 SIP trunk to call out the world, and 1 ZAP ISDN trunk to receive calls from the world. The

Hi, Im getting mad with this error, I have a x100p installed with wcfxo module loaded perfectly, I can receive incoming calls and detect very good the hangup for incoming calls. But for outgoing calls its a mess. When I place a call for outgoing, i heard the ringing, my cell or phone rings and when I pick up the phone it hangs: -- Called g1/91xxxxxxx -- Hungup

Hi all I have a small problem here. I asked this question on another asterisk mailing list, but nobody seemed to be able to help me there. We are running * Asterisk 1.4.17 * Libpri 1.4.3 * Zaptel 1.4.8 on a 1.6 dual core, 2GB ram and a digium TDM800P wildcard, hardware echo cancelation and a quad FXO card. We have 4 analog lines, one of which is a Cellphone line for least cost

install asterisk with x100p clone; the problem is that call me and hangup but the interface zap not detect the hangup and the line open. the error is : The 'reload' command is deprecated and will be removed in a future release. Please use 'module reload' instead. -- Reloading module 'chan_zap.so' (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf':

Is there a way I can change the ring cadence on FXS lines on a system using a Digium Wildcard TDM2400 card? I recently deployed a new phone system, and the customer has a few POTS phones in areas where they don't have data network services, so we're using the FXS lines to provide dialtone at those outbuildings. The old phone system would ring those phones with a short ring-short

So, tweaking configs, rebuilding this and that... restarting, twiddling, it works (yeah!), but fails on re-boot to work at all. Consistently, though. I believe it comes down to this: I can call out only *after* I've received a call. So, cold boot. Then: modprobe dahdi modprobe wctc4xxp modprobe wcfxo dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.1.0.3