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Posts tagged with "dial plane"

konfigurasi server voip

, , , ...

pada konfigurasi Trixbox (asterisk@home) semuad file di pisah kan berdasarkan fungsi2 nya antara lain :

1. /etc/asterisk => semua file konfigurasi asterisk di simpan dalam foldr ini kecuali konfigurasi zaptel file tersebut di letakan dalam folder /etc/zaptel.conf

2. /usr/sbin => pada folder ini terdapat file2 exucutable dan scripts termasuk asterisk,astmon, astgenkey dan safe asterisk

3. /usr/lib/asterisk => arsitektur asterisk dan specific binary asterisk di simpan pada direktory ini

4. /usr/lib/modules => Runtime for applicatian asterisk ,channel driver ,codec ,file format di letekan di sini

5. /usr/include/asterisk => header file 2 yang di butuhkan asterisk untuk mejalankan apalikasi channel driver application dan juga loadavle modules ada disini

6. /var/lib/asterisk => dalam direktory ini tersimpan file variable data yang di gunakan dalam operasi normal asterisk ..

konfigurasi pada file asterisk.conf (/etc/asterisk.asterisk.conf)
[directories]
astetcdir => /etc/asterisk
astmoddir => /usr/lib/asterisk/modules
astvarlibdir => /var/lib/asterisk
astagidir => /var/lib/asterisk/agi-bin
astspooldir => /var/spool/asterisk
astrundir => /var/run/asterisk
astlogdir => /var/log/asterisk

[options]
transmit_silence_during_record = yes

sedangkan untuk konfigurasi general protokol sip sebagai berikut : (etc/asterisk/sip.conf)

; Note: If your SIP devices are behind a NAT and your Asterisk
; server isn't, try adding "nat=1" to each peer definition to
; solve translation problems.

[general]

bindport = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
Videosupport=yes


domain=uchan1.com[,from-internal]




; If you need to answer unauthenticated calls, you should change this
; next line to 'from-trunk', rather than 'from-sip-external'.
; You'll know this is happening if when you call in you get a message
; saying "The number you have dialed is not in service. Please check the
; number and try again."
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
tos=0x68

; #, in this configuration file, is NOT A COMMENT. This is exactly
; how it should be.
#include sip_nat.conf
#include sip_registrations_custom.conf
#include sip_registrations.conf
#include sip_custom.conf
#include sip_additional.conf



dan untuk mensetting user SIP yang terdaftar dapat dilihat pada settingan berikut
:frown:/etc/asterisk/sip_additonal.conf)

[1001]
type=friend
secret=1001
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
nat=no
mailbox=1001@default
host=dynamic
dtmfmode=rfc2833
disallow=all
dial=SIP/1001
context=from-internal
canreinvite=no
callerid=device <1001>
allow=all

sedangkan untu konfigurasi dial plane /perutingan server sbb :frown:/etc/asterisk/extentions_additonal.conf)

[ext-local]
include => ext-local-custom
exten => 1001,1,Macro(exten-vm,1001,1001)
exten => 1001,n,Hangup
exten => 1001,hint,SIP/1001
exten => ${VM_PREFIX}1001,1,Macro(vm,1001,DIRECTDIAL)
exten => ${VM_PREFIX}1001,n,Hangup
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