Zed-3 GS8 Instrukcja Użytkownika

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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)

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

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

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

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

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

“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/

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

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

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

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

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

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

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

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

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

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