Zed-3 GS8 Manual de usuario

Busca en linea o descarga Manual de usuario para Gateways / Controladores Zed-3 GS8. Asterisk 1.6.1 on openSUSE Mohammad Edwin Zakaria Manual de usuario

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Asterisk 1.6.1 on openSUSE
Mohammad Edwin Zakaria
Notes:
This article is derived freely from http://medwinz.blogsome.com. I use Bahasa Indonesia in explaining
the dialplan so I hope everyone will understand it clearly. This is not an academic paper, I'm not an
expert, but only a practitioner so it may not suit for academic presentation.
PART 1
I use Asterisk 1.6.1.5 from openSUSE repository. Actually I built a custom 64 bit appliance using KDE
4.3 from factory repositories through SUSE Studio and took Asterisk from openSUSE Build Service
repositories. I also included DAHDI (Digium Asterisk Hardware Device Interface), but during the
implementation I have a problem with Indonesia PSTN telephone signaling so I should download dahdi
trunk version from digium subversion server to make the digium card works.
Here are the hardware I use:
1. 2 HP tower based server with 8 GB memory (it is overkill actually, but the owner insist it)
running in high availability. See the pictures here and here.
2. 10 PSTN lines
3. 3 Digium TDM 410P cards (with 4 FXO ports per card and hardware echo canceler) per server
4. several RJ12 coupler
5. RJ 12 cables
6. 2 Zed-3 GS8 GSM gateway, each with 2 GSM modules
7. Several Polycom IP-330 with PoE
8. Polycom KIRK Wireless Server 600V3
9. Several Polycom DECT 3040 Wireless Handset
Digium and Polycom prices are expensive but the quality of the sound is very good. There are some
alternatives for the IP Phone like Grandstream and Aastra that also can be used.
In this project, Asterisk will be use to setup the voip communication between this site in Denpasar/Bali
with the headquarter (HQ) in Jakarta as well as with other regional center in Java and Sumatera. Also
Asterisk will act as traditional PBX to connect this site to PSTN lines as well as to GSM/CDMA lines.
Every conversation through the PABX will be recorded by monitor application in Asterisk.
Before we go any further lets discuss a logical design about our setup. There is one HQ and several
remote sites including Bali. These sites is special because it’s also act as second node beside HQ that
can receive and transmit voip traffic to other center. The setup of every site is similar like the diagram
below.
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Indice de contenidos

Pagina 1 - Mohammad Edwin Zakaria

Asterisk 1.6.1 on openSUSEMohammad Edwin [email protected]:This article is derived freely from http://medwinz.blogsome.com. I use Bahasa

Pagina 2

exten => _000811.,2,Dial(SIP/9031/${EXTEN:1}) exten => _0

Pagina 3

{STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)},m) exten => _000855.,2,Dial(SIP/9031/${EXTEN:1}) ext

Pagina 4

{STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)},m) exten => _000878.,2,Dial(SIP/9032/${EXTEN:1}) ext

Pagina 5

exten => t,1,Playback(en/maafmohonulangi) exten => t,2,Goto(500,5) exten => i,1,Playback(en/pesanandasalah) exten => i,2,Goto(500,5) exte

Pagina 6

language=encontext=internal-fxosignalling=fxs_ks rxwink=300 cidstart=polarity ; jangan ada line yang ngutang akan mengacaukan DTM

Pagina 7

allow=allallow=ulawallow=gsmcontext=internal-sip [9001]type=friendhost=dynamic dtmfmode=rfc2833 language = en context=reco

Pagina 8

dtmfmode=rfc2833 nat=no [9032]type=peerinsecure=verydisallow=all allow=ulaw allow=alaw allow=gsm contex

Pagina 9

autokill=yes register => ncpabxsv:[email protected]:4569register => dppabxsv:0000@10

Pagina 10

qualify=yesrequirecalltoken=no[ygpabxsv]type=friendauth=md5secret=0000context=localhost=dynamicdefaultip=10.8.1.120qualify=yesrequirecalltoken=no[jbpa

Pagina 11

language=en operator=no envelope=yes attach=yes maxmsg=20 maxsecs=180 minsec

Pagina 12

All the digium card provide 12 lines of PSTN, in this case we only use 10 lines. We then use RJ 12 coupler so that every line goes to 2 PBX server, PA

Pagina 13

Ip phones mempunyai extension 9001 sampai dengan 9027. GSM gateway diperlakukan sebagai sip extension dengan nomer extension 9031 dan 9032. Lihat fil

Pagina 14

• t : time out : apa yang dilakukan kalau timeout sudah lewat Sekarang coba kita perhatikan syntax extensions.conf berikut: [internal-fxo] exten =>

Pagina 15

• s : jika diberikan akan membuat pesan "Please leave your message after the tone. When done, hang up, or press the pound key" tidak dimaink

Pagina 16

CID maka asterisk bisa membaca asterisk yang masuk, tetapi sekiranya anda tidak berlangganan CID maka incoming call akan disimpan dengan nama misalnya

Pagina 17

best tools to tune the card named fxotune. To tune your card first shutdown the asterisk service and then run:# /usr/sbin/fxotune -i 0 I put 0 (zero)

Pagina 18

exten => i,2,Goto(500,5)exten => 1,1,System(/bin/mv /var/lib/asterisk/sounds/en/mymessage.gsm /var/lib/asterisk/sounds/en/autoattendant.gsm)exte

Pagina 19

from here. 14.1,3 means play the sound file /var/lib/asterisk/sounds/en/tekan3.gsm. You can record a custom sound file which contain something like &q

Pagina 20

userid=8001callerid=TELEPH01 <8001>mailbox=8001allow=allqualify=yes[8002]type=friendhost=dynamicdtmfmode=rfc2833language=encontext=internal-sipn

Pagina 21

context=recordingsnat=nocanreinvite=nousername=TELEPH06userid=8006callerid=DPTELEPH06 <8006>mailbox=8006allow=allqualify=yes; KIRK DECT 3040 at

Pagina 22

[8031]type=peerinsecure=verydisallow=allallow=ulawallow=alawallow=gsmcontext=internal-siphost=10.7.1.31username=GS8permit=10.7.1.31/255.255.255.255qua

Pagina 23

“cd dahdi-linux”“make”and follow the instructions on the screen.If all the installation successful, then you will have :/etc/dahdi//etc/asterisk//var/

Pagina 24

host=10.7.1.32username=GS8permit=10.7.1.32/255.255.255.255qualify=yescanreinvite=nocall-limit=4dtmfmode=rfc2833nat=no[8001] and [8006] are the desk ip

Pagina 25

cidsignalling=dtmfbusydetect=yesbusycount=6……echocanceller=mg2,1-12channel => 1-12there is line with "context=internal-fxo". Basically it

Pagina 26

type=friendauth=md5 secret=0000context=localhost=dynamic defaultip=10.1.1.120qualify=yes requirecalltoken=no [dppabxsv]type=friendauth=md5 secret=0000

Pagina 27

host=dynamicdefaultip=10.8.1.120qualify=yesrequirecalltoken=no In all site with the asterisk server we should configure iax.conf so every server can b

Pagina 28

# Span 2: WCTDM/1 "Wildcard TDM410P Board 2"fxsks=5echocanceller=mg2,5fxsks=6echocanceller=mg2,6fxsks=7echocanceller=mg2,7fxsks=8echocancell

Pagina 29

group= context=default ;;; line="2 WCTDM/0/1"signalling

Pagina 30

channel => 6 callerid= group= context=default ;;; line="7 WCTDM/1/2"signalling=fxs_ks calle

Pagina 31

group=0context=from-pstnchannel => 11callerid=group=context=default;;; line="12 WCTDM/2/3"signalling=fxs_kscallerid=asreceivedgroup=0cont

Pagina 32

My extensions.conf is:; extensions.conf - the Asterisk dial plan; Created by M. Edwin Z for xxxxxxxxxxxxxxxx; [email protected]

Pagina 33

exten => _XXXX,103,Voicemail(${EXTEN}@default,b) exten => _XXXX,104,Hangup [internal-fxo] exten => s,1,Answer exten => s,2,Wait(1) ext

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