1. 1. ทบทวนกันก่อน

 

จากตอนที่ 5 และ ตอนที่ 8 เรามี settings ดังนี้

Debian 8 + Asterisk 11

     – 2 SIP extensions (2000, 2001)

     – 1 SIP trunk (voip)

     – 4 FXO ports

มี sip.conf ดังนี้

============

[general]

register=adventekvoip:[email protected]     ; Register to provider
port=5060                                                         ; Port to bind to (SIP is 5060)
bindaddr=0.0.0.0                                               ; Address to bind to (all addresses on machine)
allow=all                                                           ; Allow all codecs
match_auth_username=yes

[voip]                                         ; This is for SIP trunk

defaultuser=adventekvoip            ; Username that you get from provider
type=friend
secret=xxxxxxx                           ; Password
nat=force_rport,comedia
host=77.72.174.128
fromuser=adventekvoip                ; Username that you get from provider
fromdomain=77.72.174.128          ; SIP server
dtmfmode=rfc2833
disallow=all
defaultexpirey=20
canreinvite=no
qualify=yes
allow=g729
allow=ulaw
allow=alaw

[2000]                                 ; This is our extension nember

type=friend                          ; This device takes and makes calls
defaultuser=2000                  ; Username on device

qualify=yes
secret=1234                         ; Password for device
host=dynamic                       
context=from-sip                  
mailbox=100@default            ; Define voicemail for extension 2000

[2001]                                  ; Same patterm as of extension 2000

type=friend
defaultuser=2001

qualify=yes
secret=1234
host=dynamic
context=from-sip
mailbox=101@default

มี chan_dahdi.conf ดังนี้

=====================

[trunkgroups] [channels] context = from-pstn
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
faxdetect=incoming
echotraining=800
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1

;Uncomment these lines if you have problems with the disconection of your analog lines
busydetect=yes
busycount=3

immediate=no

#include dahdi-channels.conf

มี dahdi-channels ดังนี้

==================

; Span 1: WCTDM/0 “Wildcard TDM410P” (MASTER)
;;; line=”1 WCTDM/0/0″
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 1
callerid=
group=
context=default

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

;;; line=”3 WCTDM/0/2″
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 3
callerid=
group=
group=
context=default

;;; line=”4 WCTDM/0/3″
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 4
callerid=
group=
context=default

มี extensions.conf ดังนี้

=====================

[general]

static=yes         ; These two lines prevent the command-line interface
writeprotect=yes   ; from overwriting the config file. Leave them here.

[from-pstn]       ; incoming call จาก DAHDI trunk จะเข้าที่นี่แล้วจจะถูกส่งไปที่ extension 2000
exten => s,1,Dial(SIP/2000,30)                                                  
exten => s,2,Hangup();

; If the number dialed by the calling party was “2000”, then
; Dial the user “2000” via the SIP channel driver. Let the number
; ring for 20 seconds, and if no answer, proceed to priority 2.
;

exten => 2000,1,Dial(SIP/2000,20)
exten => 2000,2,Voicemail(100@default)
exten => 2000,3,PlayBack(vm-goodbye)
exten => 2000,4,Hangup

;
; Now, what if the number dialed was “2001”?
;

exten => 2001,1,Dial(SIP/2001,20)
exten => 2001,2,Voicemail(101@default)
exten => 2001,3,PlayBack(vm-goodbye)
exten => 2001,4,Hangup

;
; Define a way so that users can dial a number to reach
; voicemail. Call the VoicemailMain application with the
; number of the caller already passed as a variable, so
; all the user needs to do is type in the password.
;

exten => 2999,1,VoicemailMain()

;
; According to SIP in sip.conf – sip outbound to mm
;
; โทรออกผ่าน SIP trunk

exten => _02XXXXXXX,1,Dial(DAHDI/g0/${EXTEN},30)     ; โทร 02 ผ่าน DAHDI trunk
exten => _02XXXXXXX,2,Hangup                 

 

exten => _0[689]XXXXXXXX,1,Dial(SIP/voip/${EXTEN},30)     ; โทรหามือถือผ่าน SIP trunk
exten => _0[689]XXXXXXXX,2,Hangup

 

exten => _0[3457]XXXXXXX,1,Dial(SIP/voip/${EXTEN},30)     ; โทรต่างจังหวัดผ่าน SIP trunk
exten => _0[3457]XXXXXXX,2,Hangup

               

3. สรุป

ในตอนนี้เราได้ config SIP และ DAHDI trunk รวมทั้งสร้าง dial plan ที่สามารถเลือก trunk

ตามเบอร์ปลายทางที่เราโทรไป แต่ยังไม่ได้รวม special services เช่น 191 (emergency)

181 (เทียบเวลา)   1150 (พิซซ่า)   ลองเพิ่มเติมดูได้ครับ

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