similar to: AT530 Telephone

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 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, 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

Anyone here used these phones? I'm getting more and more frustrated by todays modern crop of routers with their so-called SIP ALGs which are invariably broken, or routers with built-in ATAs which block internal SIP phones from working, so looking to use IAX for some end-users. I already support it for people who want to use (eg) Zoiper and use IAX a lot to plumb boxes together, but never

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 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 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 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 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:

Hi, I am a newbie in asterisk. trying to configure firefly third party edition to connect to aserisk 1.0.3 im able to authenticate but cannot dial extensions. I have been reading the documentation cant seem to find the correct configs. Attached the error message and configs. What am I missing? *CLI> Urgent handler Feb 12 15:52:05 NOTICE[16537]: chan_iax2.c:5718 socket_read: Rejected connect

Hello friends, in aCentOS with a A400P01 OpenVox PCI I have a analog line connected. I am new in Linux and Asterisk, my steps are theese: 1. Install CentOS 4.4 (basic instalation). 2. Command line: yum -y update yum install gcc kernel-devel bison openssl-devel yum install openssl-devel 3. Download the source: wget http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.17.tar.gz

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Hi list! My skinny phone can make outgoing calls but incoming calls just keep ringing for the calling end but the phone that actually should ring doesn't ring at all. I guess I have something messed up with the dial command. I have this in skinny.conf: [z4040] device=SEP000000000000 (actual number removed) nat=0 callerid="z4040" <105> callwaiting=0 mailbox=199 transfer=1

I am new to asterisk and I'm setting up a test box to flesh out a switchover we're going to do at work. Right now I'm working on voicemail. I can leave a message fine, but when I attempt to listen to messages, I am having trouble. I can dial an extension for voice mail main, login, and it'll tell me how many messages I have. I press one and it will give me the date/time for the

Hello fellows!!! I'm having problems with Ericsson Dialog 3221 phone and DISA. I've configured an extension to test DISA and it work properly with all other phones, but freeze with the mentioned phone. Here is my extension: exten => 105,1,Answer exten => 105,2,Set(TIMEOUT(digit)tting =5) exten => 105,3,Set(TIMEOUT(response)=10) exten =>

I am trying to setup an interactive menu where a caller hits the main menu and can then dial an extension. As far as I can tell the "Waitexten" app is failing to read 3 digits and just reading the first and then announcing that it is invalid since all extensions are 3 digits. How do I make Waitexten wait for 3 digits? I have setup the extension "100" for users to reach the

Hi Asterisk-users, I am working with Aterisk Manager API's. I can login successfuly with the following. char buff[256]; strcpy(buff, "Action: Login\r\nUsername: admin\r\nSecret: unix\r\n\r\n"); send(msock, buff, 255); Now I want to try Action: Originate, therefore I tried the following char buff1[256]; strcpy(buff1, "Action: Originate\r\nChannel:

Hello all: For fun, I am learning about Asterisk, and trying to get Asterisk working at my house. I installed Asterisk@Home. It seems to be functioning fine. I installed a couple of softphones, and have them registered with Asterisk. I actually work for a CLEC, and I have registered my Asterisk box with SER (which I don't begin to understand yet) at the office. In order to try to

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 =>