similar to: just on my LAN

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 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 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 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 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 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 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 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, 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

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! 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

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 installed Asterisk on a Debian machine by apt-get install asterisk I only want to call a user inside the LAN, what files I have to edit??? sip.conf??? thanks for all -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070327/355be0cc/attachment.htm

Hello again, I have a little problem, every time I switch on the Asterisk server I must load two modules: modprobe zaptel and modprobe wctdm Is there any way to load there automatically when the server start? I have a Debian Etch. One more cuestion, it's posible to start Asterisk (asterisk -vvvc)as well? What metod do you prefer? "asterisk" or "asterisk -vvvc"? Thanks

Hello everybody. I have two AT530 telephones and one X-Lite extension conected to my Asterisk. This is part of my extensions.con. exten => 105,1,Answer exten => 105,2,Background(/home/user/suport) exten => 1,1,Dial(SIP/101,30,Ttm) exten => 2,1,Dial(SIP/102,30,Ttm) When I call to 105 extension from the AT530 telephones and I select option "1" it doesn't redirect to

2014-02-23 17:57 GMT+01:00 Charles Lepple <clepple at gmail.com>: > On Feb 22, 2014, at 9:51 AM, Josu Lazkano wrote: > >> The problem is that I don't know where to configure it. >> >> Anyone with a UPS connect to a OpenWRT device? > > What version of OpenWRT? > > I haven't tried this myself, but it looks like recent versions of OpenWRT use UCI,

Hello everybody, I am new on this mail-list. I want to try to config a HD video stream server on my LAN. I use Debian Squeeze OS and the idea is to use H.264 video codec. I just install Icecast2 on a Debian Lenny server and it works great with a OGG audio playlist. I have lost of questions about it: 1. Is possible to configure Icecast2 to stream video as LiveTV? 2. Which will be the best client