这段时间一直在研究asterisk，是基于《Asterisk™ The Future of Telephony》这本书展开的，涉及asterisk的安装，调试，SIP，IAX，以及一些基本的配置等，这里对测试的脚本进行留存

因为我们用的asterisk大部分都装了freepbx等，配置文件看起来超复杂，找不到重点，这里的保留最原始的。。

配置SIP分机用的，这个文件其实可以超简单的。。

sip.conf

[general]
register => tontone:123456@192.168.0.105/asaka

[asaka]
type=friend
host=192.168.0.105
context=asaka_incoming
secert=123456

[1000]
type=friend
host=dynamic
context=from-internal

[2000]
type=friend
host=dynamic
context=from-internal
;requirecalltoken=no

配置IAX用的。。

iax.conf

[general]
autokill=yes

register => asaka:123456@192.168.0.105

[tontone]
type=friend
secret=123456
host=dynamic
context=incoming_tontone
trunk=yes
;requirecalltoken=no

[zoiper]
type=friend
host=dynamic
context=from-internal

配置dahdi

chan_dahdi.conf

;# Flash Operator Panel will parse this file for dahdi trunk buttons
;# AMPLABEL will be used for the display labels on the buttons

;# %c Dahdi Channel number
;# %n Line number
;# %N Line number, but restart counter
;# Example:
;# ;AMPLABEL:Channel %c – Button %n

;# For Dahdi/* buttons use the following
;# (where x=number of buttons to dislpay)
;# ;AMPWILDCARDLABEL(x):MyLabel

[channels]
language=en

; include dahdi extensions defined in FreePBX
#include chan_dahdi_additional.conf
#include dahdi-channels.conf

; XTDM20B Port #1,2 plugged into PSTN
;AMPLABEL:Channel %c – Button %n
;context=from-pstn
;signalling=fxs_ks
;faxdetect=incoming
;usecallerid=yes
;echocancel=yes
;echocancelwhenbridged=no
;echotraining=800
;group=0

;channel=1-2

usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
immediate=no
callerid=4001
; define channels

context=from-internal ; Uses the [internal] chntext in extensions.conf
signalling=fxo_ks ; Uses FXO signalling for an FXS channel
channel => 1 ; Telephone attached to port 1

context=from-pstn ; Incoming calls go to [incoming] in extensions.conf
signalling=fxs_ks ; Use FXS signalling for an FXO channel
channel => 2 ; PSTN attached to port 2

dahdi-channels.conf

; Autogenerated by /usr/sbin/dahdi_genconf on Tue Jun 22 16:59:35 2010 — do not hand edit
; Dahdi Channels Configurations (chan_dahdi.conf)
;
; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended
; to be #include-d by /etc/asterisk/chan_dahdi.conf that will include the global settings
;

; Span 1: WCTDM/4 “Wildcard S400P Prototype Board 5” (MASTER)
;;; line=”1 WCTDM/4/0″
signalling=fxo_ls
callerid=”Channel 1″ <4001>
mailbox=4001
group=5
context=from-internal
channel => 1
callerid=
mailbox=
group=
context=default

;;; line=”2 WCTDM/4/1″
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 2
callerid=
group=
context=default

配置拔号方案
extensions.conf

[globals]
OUTBOUNDTRUNK=DAHDI/2
TELE=DAHDI/1
ZOIPER=IAX2/zoiper

[general]
autofallthrough=yes

[default]
exten => s,1,Verbose(1|Unrouted call handler)
exten => s,n,Answer()
exten => s,n,Wait(1)
exten => s,n,Playback(tt-weasels)
;exten => s,n,Dial(sip/1000,20)
;exten => s,n,Record(/var/spool/asterisk/monitor/asterisk-${EXTEN}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}:wav)
exten => s,n,Hangup()

[incoming_tontone]
include => from-internal

exten => _105XXX.,1,Verbose(1|exten is 105XXX)
exten => _105XXX.,n,NoOp()
exten => _105XXX.,n,Dial(IAX2/tontone/${EXTEN:3},20)
exten => _105XXX.,n,Playback(the-party-you-are-calling&is-curntly-unavail)
exten => _105XXX.,n,Hangup()

[asaka_incoming]
exten => _135XXX.,1,Verbose(1|exten is 1055XXXX)
exten => _135XXX.,n,NoOp()
exten => _135XXX.,n,Dial(SIP/asaka/${EXTEN:3},20)
exten => _135XXX.,n,Playback(the-party-you-are-calling&is-curntly-unavail)
exten => _135XXX.,n,Hangup()

include => incoming_tontone
include => internal
include => call-out

[rooms]
#include meetme_additional.conf
conf => 600