similar to: TDM400 (?) zap hangup

Afternoon, I was hoping someone could point me in the right direction. I have an asterisk PBX deployed in China using a TDM400P based card. The incoming calls are being picked up correctly, but are not being hung up. I suspect that this might be an issue with the signaling that has been selected. If anyone here has deployed asterisk in china using an analog card, it would be a great help

Let me know if I should post this on the asterisk-dev list instead. I am building a Linux-Vserver (http://www.linux-vserver.org) host system that will have several guests running Asterisk. Since the guests can't load kernel modules or do other dangerous stuff, but can access them I built zaptel 1.4 and it is now loaded by the host. The issue I see is there will be no "Zaptel"

I am having a problem with and Asterisk installation where two ports connected to a TDM400 card will have a very loud noise when you try to dial. The server has an OpenVox D110P, a TDM04B and a Xorcom Astribank 8 fxs. It is running Zaptel 1.4.11 and Asterisk 1.4.18. The problem always happens with two ports (34 and 35) which are connected to two GSM gateways. They will work fine for a week

I have a TDM400 with an FXS and FXO interface. I have adjusted the TX and RX gain for both the FXS and FXO channels so that everything is balanced. I hear myself very loudly with no lag. I also get alot of background noise when the bridge is formed between the FXS and FXO channels. I have read the following articles: http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation

Hi, I'm curious if there is any other option beside relaxdtmf in zapata , or any where else to tune dtmf detection on TDM400 fxo boards. in one of our sites provider is giving us 4 analog lines out of Adtran router and Asterisk often recognize DTMF wrong. Obviously playing with relaxdtmf was not helpfull. What do we know anout 1.2 and 1.4 DTMF handling diffrences? At this time i'm using

Hi, I'm curious what chip Digium is using in the latest TDM400 cards. Specifically, to my recollection, they used to use the TigerJet 320G, however somewhat recently, Tigerjet was bought out, and now the 320G is no longer produced. Maybe a better question is: is there a way I can take the latest DAHDI source and get a list of supported chipsets from it? Thanks. MCH -------------- next part

I've been having trouble with making outbound calls to my TELCO from a TDM400 card (FXS KS signalling) after upgrading from 1.6-beta9 to 1.6.0. The problem is completely intermittent. When it fails, I get this message: [Nov 9 19:12:26] WARNING[18916] app_dial.c: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) At some point, it starts working, but I don't know what

I have a TMD400 card installed in a PC with one fxs (installed in slot 2) and two fxos (installed in slots 3 & 4).? fxos work fine but I am unable to get a dial tone for any devices connected to the fxs.? I have correctly connected the power supply to the card and I have even tried moving the card from slot 1 to 2 on the board. Below is from the console when I try to route a call from FXO on

I had a working Asterisk 1.4.24.1 installation on SUSE 9 Linux but SIP only. I then downloaded and installed latest Zaptel and could not get Zaptel working. So I downloaded Asterisk again and re-installed. But still problems: Here is my ztcfg output: asterisk:/etc/asterisk # ztcfg -vvvvv Notice: Configuration file is /etc/zaptel.conf line 4: Unable to read Zaptel version information. Zaptel

Just a little pun there! I've been mostly lurking for a couple of weeks and realize how little I know and understand about this PBX and phone stuff. I did a little looking about and came across a glossary but they terms are -- for me -- kind of out of context. I'm wondering if there is (much as I hate the term "Dummy") a "PBX for Dummy's" or similar. I've

Hi people, I would like to read your suggestions as to where the issue might be. ATA286=Grandstream Budgetone ATA 286 in SIP mode. One Lan Port and one FXS port. TDM04B= 4 FXO signal fxls There is a 8FXO-to-SIP unit in this scenario that works perfectly so i will not make mention of it. PSTN----VOIPprovider---Internet---ATA286------tdm04b---Asterisk1.2.-13 Asterisk is being used as a meetme

Hi, all I have installed TDM400 card. I can see it is there (lspci). But Asterisk does not find it. phonebox2*CLI> zap show status No Zaptel interface found. I assume that driver is not loaded, but I am sure I have installed it (I compiled zaptel and then re-build asterisk and did make install for both zaptel and asterisk). When asterisk is started I get: Jan 2 06:28:08 WARNING[3473]:

Hi all, I have a TDM400 with all FXO on it. When I make an outgoing call, I can hear callee but callee claims the volume is too low so that he/she can't hear very clear. Can I adjust to increase the volume in callee side? Is it increase the value of txgain can solve the problem? ango

Hi I have had a Centos 5 installed with asterisk and zaptel for a couple of weeks, I had to reboot eh machine today, and when it rebooted it got stuck at "Starting udev" if I remove thew tdm400 it boots OK, but no zaptel has anyone seen this , and can offer any advice? Thanks Robb

I have the following dialplan (segment) that isn't working as I expected it to: exten => s,n,Dial(Zap/1&SIP/202&SIP/203,18) exten => s,n,Dial(Zap/1&SIP/201&SIP/202&SIP/203,42) The plan was to have SIP/201 added to the group of ringing phones after 3 or so rings. What ends up happening, though, is the Zap/1 phone STOPs ringing when the dialplan falls through to

Hi, Some weird problem (or I'm too sleepy) happens with a tdm04B revision G (4fxo) Steps: modprobe zaptel modprobe wctdm ztcfg -vv /etc/zaptel.conf fxsls=1-4 # TDM04B defaultzone=us loadzone=us /etc/asterisk/zapata.conf signalling=fxs_ls group=1 context=incoming channel => 1-4 modprobe zaptel and wctdm load fine, however ztcfg -vv shows: 0 channels configured Im using centos 4.4 with

I have a TDM400 card with 3x FXO and 1x FXS ports on it. At the moment I'd prefer (till I can get it working more reliable with iaxmodem), for a faxmodem to answer one of the lines instead of the linecard. I've tried changing the context of that line so that the exten => s does nothing, but that stops the line from being able to receive calls (get a recorded This number is not

I have had a TDM400 with 2 FXO and 2 FXS working for ages (>12 months). It has stopped working. All four green lights are still lit. I have rebuilt zaptel and asterisk and restarted but the problem persists. /sbin/ztcfg -vvvv Zaptel Configuration ====================== Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03:

Hi all, I have an Astribank connected to Asterisk 1.4. I'm setting up extensions and I have a problem with inbound calls to zap extensions. The phone at 65 rings once and then the line gets hung up. If I pick up the phone really fast, it works. Any suggestions? I have the following setup: [from-pstn] exten => 207582401,1,Dial(Zap/65,30) CLI shows me this: -- Accepting call from

Hi, I'm trying to get up and running a TDM400 with a standard italian pots line but i'm having problems at getting asterisk to detect when the line get answered on outgoing calls. I'm using asterisk 1.6 beta 9 with zaptel 1.4.11. I tried with and without answeronpolarityswitch=yes but it didn't change anything at all. With callprogress=yes answer get never detected. With