similar to: ISDN with Billion

hello friends, I am configurin my Billion ISDN and when I start asterisk (asterisk -vvvc) I have this error message: [chan_zap.so] => (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Apr 23 15:27:23 WARNING[2205]: config.c:525 process_text_line: No '=' (equal sign) in line 29 of zapata.conf Apr 23 15:27:23 WARNING[2205]: config.c:525 process_text_line:

hello friends, is there any way to simplify that extensions.conf file? [miprimerejemplo] exten => 20000,1,Dial(SIP/20000,30,Ttm) exten => 20000,2,Hangup exten => 20000,102,Voicemail(20000) exten => 20000,103,Hangup exten => 20100,1,Dial(SIP/20100,30,Ttm) exten => 20100,2,Hangup exten => 20100,102,Voicemail(20100) exten => 20100,103,Hangup exten =>

hello friends, I want to install Asterisk on a Debian machine. I need to download the sources or just with apt-get install is enought??? thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070320/227a3b32/attachment.htm

hello I want to install Asterisk just to use in my LAN, without a analog or digital devices. I need to install all this packages??? Asterisk 1.2.17 Zaptel 1.2.16 Libpri 1.2.4 Addons 1.2.5 Sounds 1.2.1 thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070328/86e4ad95/attachment.htm

Hello everybody. I want to make a menu with the incoming calls, I want that when someone calls the Asterisk play a record (in gsm) and them the caller must choose a option (1,2 or 3). if he choose 1 it will redirect to 101 extension if he choose 2 it will redirect to 102 extension if he choose 3 it will redirect to 103 extension my extensions.conf is this one: [default] exten =>

Hello everybody. I have a problem with my dialplan. That my extensions.conf: [incoming] exten => 943712666,1,Wait(2) exten => 943712666,2,Answer() exten => 943712666,3,Background(/home/lazkano/welcom) exten => 943712666,4,Wait(1) exten => 943712666,5,Background(/home/lazkano/extension) exten => 943712666,6,Wait(4) exten => 943712666,7,Dial(SIP/104|30|tm) exten =>

Hello, I am trying to install a Billion ISDN on Asterisk I have Debian Etch and I installed theese packages: apt-get install linux-headers-`uname -r` apt-get install make apt-get install ncurses-base ncurses-bin ncurses-term apt-get install libncurses5 libncurses5-dev apt-get install bison apt-get install openssl apt-get install libssl0.9.8 apt-get install libssl-dev apt-get install

Hello everybody. I am trying to configure an Asterisk on Debian with the Billion ISDN card. I am using mISDN. But when I call on the CLI apears this: -- Executing Dial("SIP/101-081805b8", "mISDN/1/943833473|45|tTwW") in new stack -- Called 1/943833473 P[ 1] empty_chan_in_stack: cannot empty channel 255 P[ 1] --> we have already send Release_complete == Everyone is

Hello every boy again I have some problems with modprobe. When I type "modprobe zaphfc", this error happens "FATAL: Module zaphfc not found." And when I tyoe "ztcfg -vv" this error happens: Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected Someone can help me??? Thanks to all.

hello, I have instaled a analog line, and when I call on the console apears that: I want to redirect the call to 101 extension. *CLI> -- Starting simple switch on 'Zap/1-1' == Starting Zap/1-1 at default,s,1 failed so falling back to exten 's' == Starting Zap/1-1 at default,s,1 still failed so falling back to context 'default' Apr 27 08:15:53 WARNING[3494]:

Hello everybody! I have some problems with my Astersk. I have an analogical OpenVox card and A Billion ISDN card (with mISDN). I load the modules with modprobe zaptel and modprobe wctdm. When I run ztcfg -vv I have this: Zaptel Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. ZT_CHANCONFIG failed on channel 1: No

Hello everybody, I am 20 days with the same item and I can't configure it. I want to know if someone has the Billion ISDN card on a Debian Etch, because everybody tells me to do that, then the other one but no one has the same configuration. If some one has the same configuration (Billion + Debian Etc), can you help? What packages install and what steps continue. Thanks to all and have a

Hello, I have some problems with mISDN. I can't send or receive call from the Billion ISDN card Mi configuration files are thoose: extensions.conf: [general] static=yes writeprotect=yes [SOME] exten => 101,1,Dial(SIP/101,30,Ttm) exten => 101,2,Hangup exten => 102,1,Dial(SIP/102,30,Ttm) exten => 102,2,Hangup include => outgoing [outgoing] exten

Hello everybody, when I run "make config" I have this error: install: cannot stat `init.asterisk': No such file or directory make: *** [config] Error 1 I don't understand. For what is "make config"? to put on /etc/init.d/? Thanks for all -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello everybody, I have installed the Billion ISDN on a Debian machine. I proved to call with a ISDN telephone conected to ISDN Box and it is OK. So I connect the Billion ISDN to the ISDN Box and I call from a extension to outside. But it doesn't work, that is what I have in the CLI: *CLI> -- Executing Dial("SIP/101-f9eb", "ZAP/g1/943833473|45|tTwW") in new

Hello everybody! I have a problem recording voices for my Asterisk menu. I used the "Record(/home/lazkano/bienvenido:gsm)" function to record the menu voices, but when I call from outside or from an extension the voice listen so low. is there any software to record my voice properly and convert to gsm format? Someone use an other function for that? Thank a lot to everybody. Enjoy

I've moved from trixbox to AsteriskNow. Does anyone know if there's something like the PBX Status screen for AsteriskNow? A module the shows the status of SIP and IAX2 registry and peers, etc for all individual entries? The FreePBX System Status screen shows when something fails to register but doesn't show who it is. Frank

2014-02-25 10:12 GMT+01:00 Virgo P?rna <virgo.parna at mail.ee>: > On Sun, 23 Feb 2014 19:39:02 +0100, Josu Lazkano <josu.lazkano at gmail.com> wrote: >> >> # dmesg >> ... >> [181041.636190] usb 4-4: new low-speed USB device number 2 using ohci_hcd >> > > I recently read about openwrt (for one of the routers I own). And there was a >

I agree I have not seen any H264 streaming info for icecast2. If you are however interested in using ogg/vorbis (Theora) look up Theora Streaming Studio in the Theora Cookbook and it will give you some info. Additionally if you wish to script this task you can look up the info on the three commands TSS wraps, specifically dvgrab/ffmpeg2theora/oggfwd. if your source file is in a h264 medium you

hello, I have the A400P01 from OpenVox. Is necesary to install all this packages? http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.17.tar.gz http://ftp.digium.com/pub/zaptel/releases/zaptel-1.2.16.tar.gz http://ftp.digium.com/pub/libpri/releases/libpri-1.2.4.tar.gz or just with asterisk and zaptel is enough. thanks a lot -------------- next part -------------- An HTML attachment was