Eyed Of Office Boy: January 2014

Topologi

Dalam proyek ini akan dibangun sambungan line PABX baru melalui jalur wireless dari Main Office ke Remote Office A dan Remote Office B untuk itu dibutuhkan SPA400, (Internet Telephony Gateway) dan PAP2T (Internet Phone Adapters) serta tentu saja Asterisk. SPA400 ini menghubungkan jaringan telpon analog (PSTN atau PABX) ke jaringan VOIP (Asterisk). PAP2T menghubungkan pesawat telpon ke jaringan VOIP. Asterisk dikonfigurasi agar nomor extension PABX tertentu selalu melalui port tertentu di SPA400 (incoming dan outgoing).

Topologi

Router dan Server

Mikrotik RB750 : 192.168.51.190
Server Linux (CentOS 5.4) : 192.168.51.253

Perangkat VOIP

SPA400 (Internet Telephony Gateway w/ 4 FXO ports)

No Nama IP Address Line 1 Line 2 Line 3 Line 4 UserID

1 SPA400-1 192.168.51.185 310 311 320 321 9000

2 SPA400-2 192.168.51.172 - 312 - - 9001
PAP2T (Internet Phone Adapters with 2 VoIP Ports)

No Nama IP Address Phone 1 (VOIP 1) Phone 2 (VOIP 2)

1 PAP2T-1 192.168.51.186 321 (2000) 311 (2001)

1 PAP2T-2 192.168.51.187 310 (2002) 320 (2003)

3 PAP2T-3 192.168.51.173 312 (2004) -

Instalasi Asterisk

Berikut ini instalasi asterisk di centos 5.4.

Buat file /etc/yum.repos.d/centos-asterisk.repo:

[asterisk-tested] 
name=CentOS-$releasever - Asterisk - Tested 
baseurl=http://packages.asterisk.org/centos/$releasever/tested/$basearch/ 
enabled=0 
gpgcheck=0 
#gpgkey=http://packages.asterisk.org/RPM-GPG-KEY-Digium 

[asterisk-current] 
name=CentOS-$releasever - Asterisk - Current 
baseurl=http://packages.asterisk.org/centos/$releasever/current/$basearch/ 
enabled=1 
gpgcheck=0 
#gpgkey=http://packages.asterisk.org/RPM-GPG-KEY-Digium

Buat file /etc/yum.repos.d/centos-digium.repo:

[digium-tested] 
name=CentOS-$releasever - Digium - Tested 
baseurl=http://packages.digium.com/centos/$releasever/tested/$basearch/ 
enabled=0 
gpgcheck=0 
#gpgkey=http://packages.digium.com/RPM-GPG-KEY-Digium 

[digium-current] 
name=CentOS-$releasever - Digium - Current 
baseurl=http://packages.digium.com/centos/$releasever/current/$basearch/ 
enabled=1 
gpgcheck=0 
#gpgkey=http://packages.digium.com/RPM-GPG-KEY-Digium

Instal asterisk

Konfigurasi Asterisk

Buat file /etc/asterisk/sip.conf

[general] 
port = 5060 
bindaddr = 0.0.0.0 
context = others 
register => 9000@192.168.51.185/9000 
register => 9001@192.168.51.172/9001 

[2000] 
type=friend 
defaultuser=2000 
context=ph2000 
secret=1234 
host=dynamic 
dtmfmode=rfc2833 
canreinvite=no 
disallow=all 
allow=alaw 
allow=ulaw 

[2001] 
type=friend 
defaultuser=2001 
context=ph2001 
secret=1234 
host=dynamic 
dtmfmode=rfc2833 
canreinvite=no 
disallow=all 
allow=alaw 
allow=ulaw 

[2002] 
type=friend 
defaultuser=2002 
context=ph2002 
secret=1234 
host=dynamic 
dtmfmode=rfc2833 
canreinvite=no 
disallow=all 
allow=alaw 
allow=ulaw 

[2003] 
type=friend 
defaultuser=2003 
context=ph2003 
secret=1234 
host=dynamic 
dtmfmode=rfc2833 
canreinvite=no 
disallow=all 
allow=alaw 
allow=ulaw 

[2004] 
type=friend 
defaultuser=2004 
context=ph2004 
secret=1234 
host=dynamic 
dtmfmode=rfc2833 
canreinvite=no 
disallow=all 
allow=alaw 
allow=ulaw 

[9000] 
;type=peer 
type=friend 
user=9000 
host=192.168.51.185 
dtmfmode=rfc2833 
canreinvite=no 
context=from-spa400-1 
insecure=port,invite 
disallow=all 
allow=alaw 
allow=ulaw 

[9001] 
;type=peer 
type=friend 
user=9001 
host=192.168.51.172 
dtmfmode=rfc2833 
canreinvite=no 
context=from-spa400-2 
insecure=port,invite 
disallow=all 
allow=alaw 
allow=ulaw

Buat File /etc/asterisk/extension.conf:

[globals] 
TRUNK1=SIP/9000 
TRUNK2=SIP/9001 

[others] 

[trunklocal] 
include => fun-stuff 
exten => _200[0-4],1,Dial(SIP/${EXTEN},15) 
exten => _200[0-4],2,VoiceMail(${EXTEN},u) 
exten => h,1,Hangup 

; 321 
[ph2000] 
include => trunklocal 
exten => _XXX,1,Dial(${TRUNK1}/L1${EXTEN},15) 
exten => _70X.,1,Dial(${TRUNK1}/L1${EXTEN},15) 
exten => _83X.,1,Dial(${TRUNK1}/L1${EXTEN},15) 
exten => h,1,Hangup 

; 311 
[ph2001] 
include => trunklocal 
exten => _XXX,1,Dial(${TRUNK1}/L2${EXTEN},15) 
exten => _70X.,1,Dial(${TRUNK1}/L2${EXTEN},15) 
exten => _83X.,1,Dial(${TRUNK1}/L2${EXTEN},15) 
exten => h,1,Hangup 

; 310 
[ph2002] 
include => trunklocal 
exten => _XXX,1,Dial(${TRUNK1}/L3${EXTEN},15) 
exten => _70X.,1,Dial(${TRUNK1}/L3${EXTEN},15) 
exten => _83X.,1,Dial(${TRUNK1}/L3${EXTEN},15) 
exten => h,1,Hangup 

; 320 
[ph2003] 
include => trunklocal 
exten => _XXX,1,Dial(${TRUNK1}/L4${EXTEN},15) 
exten => _70X.,1,Dial(${TRUNK1}/L4${EXTEN},15) 
exten => _83X.,1,Dial(${TRUNK1}/L4${EXTEN},15) 
exten => h,1,Hangup 

; 312 
[ph2004] 
include => trunklocal 
exten => _XXX,1,Dial(${TRUNK2}/L2${EXTEN},15) 
exten => _70X.,1,Dial(${TRUNK2}/L2${EXTEN},15) 
exten => _83X.,1,Dial(${TRUNK2}/L2${EXTEN},15) 
exten => h,1,Hangup 

[from-trunk-1] 
exten => 321,1,Dial(SIP/2000,15) 
;exten => 321,2,VoiceMail(2000,u) 
;exten => 321,2,Hangup 
exten => 311,1,Dial(SIP/2001,15) 
;exten => 311,2,VoiceMail(2001,u) 
;exten => 311,2,Hangup 
exten => 310,1,Dial(SIP/2002,15) 
;exten => 310,2,VoiceMail(2002,u) 
;exten => 310,2,Hangup 
exten => 320,1,Dial(SIP/2003,15) 
;exten => 320,2,VoiceMail(2003,u) 
;exten => 320,2,Hangup 
exten => h,1,Hangup 
;exten => _X.,1,Congestion() 

[from-trunk-2] 
exten => 312,1,Dial(SIP/2004,15) 
;exten => 312,2,VoiceMail(2004,u) 
;exten => 312,2,Hangup 
exten => h,1,Hangup 
;exten => _X.,1,Congestion() 

[from-spa400-1] 
exten => _X.,1,Goto(from-trunk-1,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1) 

[from-spa400-2] 
exten => _X.,1,Goto(from-trunk-2,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1) 

[fun-stuff] 
exten => 500,1,Answer() 
exten => 500,2,Playback(demo-echotest) 
exten => 500,3,Echo 
exten => 500,4,Playback(demo-echodone) 
exten => 500,5,Hangup

Buat file /etc/asterisk/voicemail.conf:

[general] 
format = wav 
[default] 
2000 => 4711,voicemail,2000@domain.com
2001 => 0815,voicemail,2001@domain.com
2002 => 0915,voicemail,2002@domain.com
2003 => 1015,voicemail,2003@domain.com
2004 => 1115,voicemail,2004@domain.com

Jalankan asterisk.
```
/etc/init.d/asterisk start
```

Konfigurasi SPA400

SPA400-1
- User ID : 9000
- IP Address : 192.168.51.185/255.255.255.192
- Gateway : 192.168.51.190
- Port ID
  - Port ID 1 : 321
  - Port ID 2 : 311
  - Port ID 3 : 310
  - Port ID 4 : 320
SPA400-2
- User ID : 9001
- IP Address : 192.168.51.172/255.255.255.192
- Gateway : 192.168.51.190
- Port ID
  - Port ID 1 : OFF1
  - Port ID 2 : 312
  - Port ID 3 : OFF3
  - Port ID 4 : OFF4
Konfigurasi Voice atau PSTN
Setelah itu hubungkan line pabx sesuai nomor ext ke line1 s/d line4 di SPA400-1 dan line2 di SPA400-2. Status masing-masing SPA400 diperlihatkan pada gambar-gambar di bawah ini.

Pastikan bahwa SIP registration status menunjukkan Registered (apabila asterisk sudah running). Kemudian lihat Battery Level di Line 1 s/d 4 (SPA400-1), apabila semua line extension PABX telah disambungkan ke port-port line 1 s/d 4 di SPA400-1 maka akan ditunjukkan besar tegangan tertentu seperti 27V.Line status menunjukkan apakah ada komunikasi voip atau tidak. Jika ada komunikasi dan pembicaraan sedang berlangsung maka status dari line yang bersangkutan akan berisi Off-Hook dan tegangan line tersebut akan senantiasa < 10V. Misal jika ada panggilan melalui line 1 kemudian terjadi pembicaraan telpon maka status line 1 akan menunjukkan Off-Hook dan tegangan line 1 akan senantiasa < 10V.Apabila USB Capacity status menunjukkan error atau tidak sesuai nilai di atas, itu berarti usb flash disk belum dicolokkan ke port usb di SPA400. Matikan dahulu SPA400 sebelum mencolokkan usb.Saat memasukkan extension PABX ke port-port line di SPA400 selalu monitor battery level seperti gambar di atas. Apabila ada keanehan seperti semua line menunjukkan level tegangan namun yang dimasukkan hanya 3 line atau kurang dari itu maka matikan SPA400, tunggu sesaat kemudian hidupkan kembali.

Konfigurasi PAP2T

Konfigurasi IP Address
Untuk konfigurasi IP Address PAP2T gunakan Interactive Voice Response Menu dengan cara hubungkan pesawat telpon ke salah satu port phone di PAP2T kemudian dial ****. Jangan tekan tombol apa pun sampai terdengar suara “Configuration menu. Please enter option followed by the # (pound) key or hang up to exit.”

Command	Description
100#	Check DHCP feature status (wheter or not unit obtains ip via dhcp)
101#	Change DHCP feature (1 to enable, 0 to disable)
110#	Announce IP Address of the unit
111#	To set a static IP address, the DHCP feature must be disabled . If the DHCP feature is enabled, you will hear, “Invalid Option”. Enter the IP address using the numbers on your telephone keypad. Use the * (star) key to enter a decimal point.
120#	Announce netmask of the unit
121#	To set the network mask, the DHCP feature must be disabled . If the DHCP feature is enabled, you will hear, “Invalid Option”. Enter the network mask using numbers on your telephone keypad. Use the * (star) key to enter a decimal point.
130#	Announce default route of the unit
131#	To set a static gateway IP address, the DHCP feature must be disabled. If the DHCP feature is enabled, you will hear, “Invalid Option”. Enter the IP address using numbers on your telephone keypad. Use the * (star) key to enter a decimal point.
150#	Gives firmware version
732668#	Reboot the unit

No	Nama	IP Address	Netmask	Gateway
1	PAP2T-1	192.168.51.186	255.255.255.192	192.168.51.190
2	PAP2T-2	192.168.51.187	255.255.255.192	192.168.51.190
3	PAP2T-3	192.168.51.173	255.255.255.192	192.168.51.190

Line Configuration
Untuk setup line configuration login ke alamat web PAP2T http://192.168.51.187.
Lalu klik link Admin Login:

Line Configuration Global untuk PAP2T

Proxy : 192.168.51.253

Register : Yes

Register Expires : 3600

Make Call Without Reg : No

Ans Call Without Reg : No

Preferred Codec : G711a

Line Configuration PAP2T-1

Line Line Enable SIP Port Display Name User ID Password Use Auth ID

1 yes 5060 2000 2000 1234 no

2 yes 5061 2001 2001 1234 no

Line Configuration PAP2T-2

Line Line Enable SIP Port Display Name User ID Password Use Auth ID

1 yes 5060 2002 2002 1234 no

2 yes 5061 2003 2003 1234 no

Line Configuration PAP2T-3

Line Line Enable SIP Port Display Name User ID Password Use Auth ID

1 yes 5060 2004 2004 1234 no

Berikut ini gambar-gambar contoh konfigurasi line 1 dan line 2 PAP2T-2.

Test VOIP

Setelah semua unit PAP2T dipasang dengan benar sesuai seting di atas maka selanjutnya jalankan browser dan buka status PAP2T-1 di http://192.168.51.186
Pastikan bahwa status registrasi line 1 dan line 2 sudah Online seperti gambar di bawah:
Lakukan langkah yang sama untuk mencek status PAP2T yang lain.
Angkat pesawat telpon yang tersambung ke PAP2T dan pastikan mendengar nada dialtone.
Lakukan test dial ke nomor VOIP. Misal jika dari pesawat 2000 dial ke 2001, 2002, 2003 dan 2004.
Lakukan test dial ke nomor EXT. Misal jika dari ext 310 dial ke ext 311, 312, 320 dan 321. Lakukan pula ke nomor EXT di Main Office.
Minta teman di Main Office untuk telpon ke salah satu nomor extension (310, 311, 312, 320 atau 321).
Apabila ada masalah seperti sambungan sibuk yang ditunjukkan dengan tegangan pada sambungan drop di angka < 10V padahal sudah tidak ada percakapan di sambungan tersebut maka reboot SPA400 di Menu Administration / Reboot.

Monitor Asterisk

Setelah logon ke server linux sebagai user root jalankan perintah di bawah:
```
asterisk -rvvvv
```

Asterisk akan menampilkan output sebagai berikut:

Asterisk 1.6.2.19, Copyright (C) 1999 - 2010 Digium, Inc. and others. 
Created by Mark Spencer  
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. 
This is free software, with components licensed under the GNU General Public 
License version 2 and other licenses; you are welcome to redistribute it under 
certain conditions. Type 'core show license' for details. 
========================================================================= 
  == Parsing '/etc/asterisk/asterisk.conf':   == Found 
  == Parsing '/etc/asterisk/extconfig.conf':   == Found 
Connected to Asterisk 1.6.2.19 currently running on server (pid = 3612) 
Verbosity is at least 4

Jalankan perintah sip show peers di command line asterisk untuk melihat sip peers.

server*CLI> sip show peers 
Name/username              Host            Dyn Nat ACL Port     Status     
2000/2000                  192.168.51.186   D          5060     Unmonitored 
2001/2001                  192.168.51.186   D          5061     Unmonitored 
2002/2002                  192.168.51.187   D          5060     Unmonitored 
2003/2003                  192.168.51.187   D          5061     Unmonitored 
2004/2004                  192.168.51.173   D          5060     Unmonitored 
9000                       192.168.51.185              5060     Unmonitored 
9001                       192.168.51.172              5060     Unmonitored 
7 sip peers [Monitored: 0 online, 0 offline Unmonitored: 7 online, 0 offline] 
server*CLI>

Jalankan sip show registry untuk melihat status sip registration.

server*CLI> sip show registry
Host                  dnsmgr Username  Refresh State       Reg.Time                 
192.168.51.172:5060   N      9001          105 Registered  Thu, 30 Aug 2012 08:23:36 
192.168.51.185:5060   N      9000          105 Registered  Thu, 30 Aug 2012 08:24:07 
2 SIP registrations.

Tes dial ke no extension 312 dan perhatikan log di asterisk saat panggilan masuk (ringing), panggilan dijawab (answered) dan panggilan berakhir (hangup).

server*CLI> 
  == Using SIP RTP CoS mark 5 
    -- Executing [9001@from-spa400-2:1] Goto("SIP/9001-0000001a", "from-trunk-2,312,1") in new stack 
    -- Goto (from-trunk-2,312,1) 
    -- Executing [312@from-trunk-2:1] Dial("SIP/9001-0000001a", "SIP/2004,15") in new stack 
  == Using SIP RTP CoS mark 5 
    -- Called 2004 
    -- SIP/2004-0000001b is ringing 
    -- SIP/2004-0000001b answered SIP/9001-0000001a 
    -- Packet2Packet bridging SIP/9001-0000001a and SIP/2004-0000001b 
    -- Executing [h@from-trunk-2:1] Hangup("SIP/9001-0000001a", "") in new stack 
  == Spawn extension (from-trunk-2, h, 1) exited non-zero on 'SIP/9001-0000001a' 
  == Spawn extension (from-trunk-2, 312, 1) exited non-zero on 'SIP/9001-0000001a' 
    -- Executing [h@from-trunk-2:1] Hangup("SIP/9001-0000001a", "") in new stack 
  == Spawn extension (from-trunk-2, h, 1) exited non-zero on 'SIP/9001-0000001a' 
server*CLI>

Catatan: Panggilan yang melalui SPA400 akan diteruskan ke asterisk setelah 3 nada dering (default).

No	Nama	IP Address	Line 1	Line 2	Line 3	Line 4	UserID
1	SPA400-1	192.168.51.185	310	311	320	321	9000
2	SPA400-2	192.168.51.172	-	312	-	-	9001

No	Nama	IP Address	Phone 1 (VOIP 1)	Phone 2 (VOIP 2)
1	PAP2T-1	192.168.51.186	321 (2000)	311 (2001)
1	PAP2T-2	192.168.51.187	310 (2002)	320 (2003)
3	PAP2T-3	192.168.51.173	312 (2004)	-

Proxy	:	192.168.51.253
Register	:	Yes
Register Expires	:	3600
Make Call Without Reg	:	No
Ans Call Without Reg	:	No
Preferred Codec	:	G711a

Line	Line Enable	SIP Port	Display Name	User ID	Password	Use Auth ID
1	yes	5060	2000	2000	1234	no
2	yes	5061	2001	2001	1234	no

Wednesday, January 1, 2014