Hello,
I am under debian on a Dell, I use a card Digium TDM400P with 1 FXO + 1 FXS module and I have a classical analogue FT. Foremost, it must physically install the card: just plugger card in a PCI, then you must install the card drivers. It is managed by the Zaptel driver.
Here is how you can do this
Code:
# Apt-get install zaptel zaptel-source linux-headers-`uname-r`
you must provide dependencies that must of course accept. Therefore, we have the tools zaptel, source code drivers, and our dear linux-headers.
to compile the driver for the card:
Code:
# Module-assistant auto-install zaptel
module-assistant and starts preparing a package that installs itself thereafter. To verify that everything is ok, do this
Code:
Ztcfg-vv
Zaptel Configuration
======================
Channel map:
Channel 03: FXO Kewlstart (Default) (Slaves: 03)
Channel 04: FXS Kewlstart ( Default) (Slaves: 04)
2 channels configured.
In my case, I see the 2 modules I mentioned above +. It acts in fact the type of signal that travels over the modules. wholesale shipments of FXS FT us from their central, and our standard phone listening FXO (they are a little taken as males and females). Basically, an FXS module will be useful only if you want to connect an analog phone directly on your server (in which case you must connect an additional power module FXS PCI card). If you got up there, your hardware is installed and you can enter the heart of the matter.
Code:
# Apt-get install asterisk asterisk-doc asterisk-prompt
There 's other packages asterisk for a multitude of functionality such as: faxing, voicemail, voicemail, but for now, already use the basic elements
after apt-get to do his work, test that the server is correctly installed
there, you must have a whole bunch of lines that scroll across your screen for a prompt Style:
Code:
Asterisk Ready.
* CLI>
Go out and stop the server, there are several methods:
ctrl + c (a bit brutal for a service)
gracefully stop (stops the server is no longer used)
now stop (stops the server immediately), your server is running, there is more to configure it, I'm in testing period and I discover the functions of a little asterisk + every day. I use Twinkle to phone headset with microphone + (apt-get install twinkle) I have at my disposal a hardphone. I do not use the FXS module for the moment because I do not have enough power cable.
/ etc / asterisk / files are all server configuration, there are a lot ...
I have configured as 3:
- zapata.conf (asterisk indicates how to use the modules of the card)
- sip.conf (states rights and the configuration of each IP phone)
- extensions.conf (Asterisk indicates the actions to follow when it gets a request)
In each file, there are plenty of examples for configuration help. it is best to save before doing anything:
Code:
# Cp / etc / asterisk / zapata.conf / etc / asterisk / zapata.conf.OLD
# cp / etc / asterisk / sip.conf / etc / asterisk / sip.conf.OLD
# cp / etc / asterisk / extensions.conf / etc / asterisk / extensions.conf.OLD
Here are the three files once configured
Code:
; Configuration file zapata.conf
;
[trunkgroups]
; AC is pointless ... ?
[channels]
language = en; definition of the default language
= switchtype national to the analog line
rxwink = 300; Atlas Seems to use long (250ms) winks
usecallerid = yes; use of caller ID
hidecallerid = No, do not hide the caller ID
callwaiting = No; "no" in the config guillaume
usecallingpres = yes; number of active presentation
callwaitingcallerid = yes, communication is waiting caller ID
threewaycalling = yes; Conference to 3
transfer = yes; transfer allows
canpark = yes;???
cancallforward = yes; forward the call is authorized
callreturn = yes; ringback active
echocancel = yes; attenuation of the active echo
echocancelwhenbridged = yes; attenuation of the echo in bridge mode active
; rxgain = 0.0; gain reception, default: no comment
; txgain = 0.0; gain Emmis, default: no comments
immediate = no;???
busydetect = yes; default: comments
busycount = 4; default: comments
; definition of channels
, the configuration of the line analog phone
; = fxo_ks signaling
; callerid = "asterisk fxs"
; context = from-sip
; channel => 3
; configuration FT telephone line
signaling = fxs_ks
= callerid asreceived
context = FXO
channel => 4
as you can see, I have not yet entered into operation, but is commented for later, to respond.
notice any of the same context = FXO. this context is defined in the file
Code:
extension.conf:
; Extension.conf file for asterisk
,
[general]
static = yes; interidiction modify this file
WriteProtect = yes; interest to find, protect in writing
autofallthrough = yes; if = No, problem when no corresponding action
clearglobalvars = No; variables are conserved
[globals]
CONSOLE = Console / dsp; Console interface for demo
; CONSOLE = Zap / 1
TRUNKMSD = 1; MSD digits to strip (Usually 1 or 0)
[from-sip]
exten => 520.1, Dial (SIP/520, 15, tr); hardphone IP
exten => _0., 1, Dial (Zap / 4 / $ (EXTEN), 45, tr); No meeting starts at 0
exten => 521.1, Dial ( SIP/521, 15, tr); IP softphone
[FXO]
; exten => s, 1, Goto (context, extension, priority)
exten => s, 1, Goto (from-sip, 521.1)
- if you have received a request for No. 520 then opens a communication using SIP appointed to the post 520
- if You receive an application for a No. starting with 0 then opens a communication using the zaptel driver to No. $ EXTEN (No. complete).
- If you receive a request for 521 then opens a communication protocol using SIP appointed to the position 521
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