similar to: genzaptool from "xorcom"

Helo, I have problem with Digium TE110P connected to CISCO 3640 (port on NM-HDV-2E1-60) wth PRI E1 link. I use CISCO now for testing but when I tried with real PBX problem was exactly same. I have this messages in Asterisk conole and log sometimes: NOTICE[1115] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Usually 2-5 such messages in series, can be repeated after 10

re Hi, I m looking for a good document that allow me to install zaptel libpri & asterisk without errors, i ve a TDM400 & TE110P, so please can you help me Kind Regards Younss AZZAYANI KASTERISK.COM

hello my friends, when i make a genzaptelconf i get this message ******************** CAS signalling on span 2 conflicts with HDLC with FCS check on channel ******************* Any idea Please? I m installing zaptel 1.4 i checked in "http://bugs.digium.com/view.php?id=7860" that it's a bug but beacause i m a newbie in asterisk i can't undrestand what exactly mean Thank You

Hello everyone, I want to connect a fax to an FXS port (TDM420P). For testing purposes, I connected an analogue phone to it first. However, when I pick it up, I cannot hear anything at all. The power cable is plugged into the card, the port is configured to use fxo-signalling. Also, immediate=no. Here's the files: /etc/zaptel.conf: # Autogenerated by /usr/sbin/genzaptelconf -- do not hand

hi, when i type ******** asterisk-1.4# ./configure ****************** i got this error **************** configure: error: C++ preprocessor "/lib/cpp" fails sanity check See `config.log' for more details. ***************** # vi config.log *************************** ...........; cpp: installation problem, cannot exec 'cc1plus' : No such file or directory ..............

please can someone send to me his files like zaptel & zapta if he si using TE110P thank you

hello, i ve a lot of probl?me with zaptel 1.4 when i tried to complile it under debian kernel 2.4; so i need to compile a new kernel version 2.6.x but id don't witch kernel version is stable van you help me please? :) Younss AZ KASTERISK.COM skype: younssiga

hi you, I'm having problem with voice quality on my trixbox using TDM2400B.The trixbox is connected via 20 FXO ports on a TDM2400 with the hardware echo cancel module. Echo cancel almost works, but the users hear what they describe as a 'background crackle/buzz' coming back when they talk. anyone have the same problem? pls help me. thanks a lot. my trixbox and config

when i compile zaptel make linux26 make install i got these errors: make[1]: Leaving directory `/usr/src/zaptel-1.4/wct4xxp' make -C datamods clean make[1]: Entering directory `/usr/src/zaptel-1.4/datamods' make -C /lib/modules/2.4.27-3-386/build SUBDIRS=/usr/src/zaptel-1.4/datamods clean make[2]: Entering directory `/usr/src/kernel-headers-2.4.27-3-386' make: *** arch/i386/boot: No

Hi every body, can someone please tell me about a SIP phone that support more than 10 extension (free or not free ;) ) wich will be used in my company, i've bought a SNOM but it just support 5 sip extension Kind regards

What is involved in provisioning Asterisk to use a multiline analog service from our local telco? I will only have one twisted pair entering in on a OpenVox card but am not sure how Asterisk interprets and deals with two incoming calls and/or two outgoing calls? Thanks! jlc

Hi All, I have the problem setting up TDM22B card. Steps what I have followed are: [1] compiled zaptel-1.0.9.2 & installed the same. [2] modprobe wcfxo /lib/modules/2.4.20-8/misc/wcfxo.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg

Hi everybody, it's possible to configure freepbx 2.2 with asterisk 1.4? Have a nice day Younss AZ KASTERISK.COM

I have an Asterisk@Home 1.3 server (Asterisk 1.0.9) and recently added a TDM400P with (1) FXO card on port 4. Inbound calls are always successful but outbound calls fail 75% of the time with intercept messages from my dial tone provider that include "we're sorry, your call did not go through", and "we're sorry, when placing a local call it is now necessary to dial an area

Hi every body, I've set up a Trixbox Server with TE110P,all things seem to work fine(Thank You Malling lists & irc & Forums), but i need your help, i ve 30 numbre from 60 to 89, i need to specify for each sip extension a Zap number for example to call the sales service the caller must call 555-4570 and automaticly the caller will be redirected to the 202 ( sales service ) so nobody

Hi, I am trying to configure * 1.0.7 box with a Digium Wildcard TE110P T1/E1 and libunicall latest code. All libs compiled successfully and the E1 have a green light! I am able to receive a call (or at least) testcall shows some information when an incoming call is received so the drivers and basic configuration is working. My * box has 2 cards, one TDM400B (2 fxs and 2 fxo) and one TE110P

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

Hello, I can't find asterisk-sounds in the svn.digium server, i ve been got asterisk, zaptel,libpri,asterisk-addons (1.2 stable version) Thank You Younss AZ KASTERISK.COM

Hi All This may be a silly question but... I have this dahdi_genconf generated file: ; Autogenerated by /usr/sbin/dahdi_genconf on Fri Jul 4 22:05:29 2014 ; If you edit this file and execute /usr/sbin/dahdi_genconf again, ; your manual changes will be LOST. ; Dahdi Channels Configurations (chan_dahdi.conf) ; ; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended ; to

Hello all, I've got a T100P connected to an Adtran TA750 with a T1 crossover... This connects to a patch panel with phone ports. The Adtran is fully populated with FXS cards. All I get on any phone port is a fast clicking noise... No dialtone. Asterisk 'sees' the card, (the channels show up in /proc/zaptel). Incoming calls are routed to the zap/x channel, but no ring. I'm