similar to: How to configure asterisk to emulate FXO signaling ?

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

Hi, I have HT488 with it's FXO connected to Israeli PSTN (bezeq) when dialing to that PSTN line asterisk see gets the call and direct it to the right extension but if the extension doesn't answer and the dialer is hanging the call the extension will keep on ringing. I'm not an expert but it seems like my asterisk doesn't recognize the hangup signal from the HT488 -or it's the

Hello, I'm trying to set the 3rd span of a new digium quad card as a E&M T1 for Faxes to a Hylafax server. The 1st and 2nd spans are working as PRIs. When I start asterisk, the logs show a signaling error and chan_zap.c dies. I also get an error that it can't read the gains but they are the standard shown below. 2.6 kernel, Debian Stable, * 1.2 svn from feb 2007 my procedure: make

Hi, I installed a 1-port FXO on my Ubuntu 8.4. I was earlier only hearing a fast clicking sound and now I am not hearing any dial-tone. The FXO card has 2 slots: [phone | line ]. I hve connected the wall-phone-input to the "line" slot and "phone" to my home-phone. I do not hear any dial-tone on my home-phone. Asterisk seems to recognize my hardware....here are the relevant

I need to select a line from the Zap group channel using the SIP Phone (not FXO and not FXS ports). ignorepat does not work? Also, what is the method to let the second dial tone has another tone frequency? Regards Bilal ---------------- No, ignorepat is for FXS ports (FXS ports use FXO signaling). Also, ignorepat does not apply to SIP phones, because SIP phones provide their own dialtone,

New system install. At what point, in bootup, should I be able to get a dial tone on the phone ports on a tdm400p? There are two fxo and two fxs ports. I know which to plug into <g>. At boot up, as soon as wctdm is loaded, all the ports "go green, yet I do not get a dial tone on the phone ports. I thought as long as zapata.conf is correct, the board should be

Dear all I have TDM808B 8 port FXO it is configure perfectly but i got some problem of incomming phone Hangup and callerid display problem i am going to explain you the issue i have install asterisk 1.4 and i have 100 of SIP phone now everything is fine but problem is when i incoming call on FXO and dial sip extention SIP phone is rining but when i disconnect my incoming

Hi Moy, Bottom line up front (yes I'm military):How do I now configure some of my spans as normal Euro ISDN and a few with R2? Do I use a combination of /etc/asterisk/unicall.conf and /etc/zaptel.conf or just unicall.conf for everything? See below for the rest of the details and thanks in advance for advice! I downloaded your astunical-1.4.16 and have been studying it. I have a machine

Hello All, Does anyone know of a software emulator that can be used to simulate hardware such as an E1? I need to play with AstUnicall in a test environment and don't have access to these circuits from the US. If there is an alternate way to test/play with AstUnicall, please let me know! Thanks, Mark -------------- next part -------------- An HTML attachment was scrubbed...

I installed Trixbox and a TDM400P with 2 FXO and 2 FXS ports and am having an annoying issue with the FXO ports. As soon as I plug either one into the phone line it's as though the line is disconnected i.e. get disconnected tone when trying to dial out, line is busy when dialling in. The CLI shows the following: trixbox1*CLI> zap show channel 4 Channel: 4 File Descriptor: 18 Span: 11*

I am attempting a fresh install of ASTCC on Ubuntu. Getting install invalid user as bellow. Has any one seen this? Can some one help out? /usr/src/astcc# make install mkdir -p /var/www mkdir -p /var/www/html/_astcc mkdir -p /var/www/cgi-bin/astcc-admin chmod 755 ./astcc.agi chmod 755 ./astcc-admin.cgi echo | ./astcc.agi >/dev/null Detected dry run! ./astcc-admin.cgi >/dev/null DBI

Running Asterisk 1.2.9. I just installed a TE110P card and configured zaptel.conf & zapata.conf. The config files look right to me but I'm getting the following error when trying to start asterisk: Asterisk died with code 1. Automatically restarting Asterisk. Does anyone have any idea what is wrong with this configuration?? Thanks in advance!!! Here's my config files: zaptel.conf

I have an Asterisk box running Fedora Core 4, Asterisk 1.4, Lippri, 1.4, and Zaptel 1.4 The Digium cards installed are TDM2400 and TE110P. Everything was working fine until I upgraded to zaptel 1.2.12 from 1.2.9 Now when I run ztcfg I get the following error message: (CAS signalling on span 2 conflicts with Clear channel on channel 40) --NOTE: signaling was spelled wrong in the error

Hi All, Does anyone know, if there is a tool, which is doing the follwing: - Testprogram on host A establishes a sip connection to testprogramm on host B - Testprogram on host A plays a tone and Testprogram B verifies, if tone is playing correctly (without any interruptions) Thank You.

I am having a problem detecting incoming FAX. TMD22p (tdm400p 2 fxo, 2fxs) As I understand it, I must have faxdetect = incoming to enable detection of the fax tone. Then, I must have a [fax] context to pickup the line and send it to whatever extension the FAX device is on. In my case, I ask it to answer immediately and do a distinctive ring (r3) to alert that is its a FAX call so no one picks

I have a Digium TDM400b11, 1FXO [port2] & 1FXS [port 1] When I reload the chan_zap I get: [chan_zap.so] => (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Apr 9 22:39:36 ERROR[3541]: chan_zap.c:10388 setup_zap: Signalling must be specified before any channels are. Apr 9 22:39:36 WARNING[3541]: loader.c:414 __load_resource: chan_zap.so: load_module

I've installed fail2ban, it seems to be working as it identified my failed test logins, BUT, my question is: what can I do when I see same invalid name trying to login to dovecot, different IP each time, how can I say block each IP as used by this name ? or it that a bad idea ? I can see two persistent attempts as so: I don't have such user 'ignacio' or 'julian' # grep

For some reason Asterisk is producing a US ring tone when it should be an Australian ring tone. I am using ztdummy and do not have any cards installed. My configuration is as follows. I am using Trixbox 1.2.2. Can someone please guide me into the right direction? zaptel.conf loadzone = au defaultzone = au zapata.conf [channels] language=au indications.conf [general]

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 have a small problem with tdm2400 with tone detection. About 5% of all calls doesn't end because Asterisk dosen't detect "busy" tone on remote hangup. I know that board does support current reversal and current off, but how do I configure asterisk to use tone detection first and if that should fail, current off detection which my operator support (40 sec after remote