Konfigurasi asterisk pada server Kantor Pusat Sip.conf [general] bindport=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=ulaw allow=gsm insecure=very register=>311:311@192.168.3.2 [101] type =friend context=kantorA callerid=ronald username=101
secret=101 host=dynamic dtmfmode=inband [102] type =friend context=kantorA callerid=ronald username=102 secret=102 host=dynamic dtmfmode=inband [103] disallow=all allow=gsm type =friend
context=kantorA callerid=ronald username=103 secret=103 host=dynamic [104] disallow=all allow=gsm type =friend context=kantorA callerid=ronald username=104 secret=104 host=dynamic ;untuk gsm gateway
[110] type =friend context=kantorA callerid=ronald username=110 secret=110 host=dynamic dtmfmode=inband ;untuk ivr gsm [1100] type =friend context=kantorA callerid=ronald username=1100 secret=1100 host=dynamic
dtmfmode=inband
;untuk dikenal diserver cabang
[111] disallow=all allow=gsm type =friend context=kantorA callerid=ronald username=111 secret=111 host=dynamic dtmfmode=inband [tokantor_B] disallow=all allow=gsm
type=friend
context=kantorA
username=311
secret=311
host=192.168.3.2
;nomor yang tidak bisa telpon ke gsm/pstn 112-115
[112] type =friend context=kantorAB callerid=ronald username=112 secret=112 host=dynamic dtmfmode=inband [113] type =friend
context=kantorAB callerid=ronald username=113 secret=113 host=dynamic dtmfmode=inband [114] disallow=all allow=gsm type =friend context=kantorAB callerid=ronald username=114 secret=114 host=dynamic
[115] disallow=all allow=gsm type =friend context=kantorAB callerid=ronald username=115 secret=115 host=dynamic
;untuk program sipp
[sipp] type=friend context=tesserver host=192.168.1.1 port=5060 user=sipp
extention.conf [kantorA] include=> parkedcalls include=> kantorAB ;untuk ke gsm exten=>_988.,1,Dial(SIP/${EXTEN:3}@110,30,t)
;untuk ke gsm dikantor cabang
exten=>_999.,1,Dial(SIP/tokantor_B/${EXTEN},30,t)
;untuk ke gsm dengan timer
exten=>_978.,1,Dial(SIP/${EXTEN:3}@110,30,tL(300000:60000))
;untuk ke gsm dikantor cabang dengan timer
exten=>_979.,1,Dial(SIP/tokantor_B/${EXTEN},30,t)
[kantorAB]
include=> parkedcalls
exten=>101,2,VoiceMail(101@vm) exten=>101,3,Hangup exten=>102,1,Dial(sip/102,30,t) exten=>102,2,VoiceMail(102@vm) exten=>102,3,Hangup exten=>103,1,Dial(sip/103,30,t) exten=>103,2,VoiceMail(103@vm) exten=>103,3,Hangup exten=>104,1,Dial(sip/104,30,t) exten=>104,2,VoiceMail(104@vm) exten=>104,3,Hangup ;panggilan dari gsm exten=>1100,1,Answer()
exten=>1100,2,Set(TIMEOUT(absolute)=120) exten=>1100,3,Background(selamat-datang) exten=>1100,4,Wait(1) exten=>1100,5,Goto(3) exten=>T,1,Wait(1) exten=>T,2,Goto(101,1) exten=>112,1,Dial(sip/112,30,t) exten=>112,2,VoiceMail(112@vm) exten=>112,3,Hangup exten=>113,1,Dial(sip/113,30,t) exten=>113,2,VoiceMail(113@vm) exten=>113,3,Hangup exten=>114,1,Dial(sip/114,30,t) exten=>114,2,VoiceMail(114@vm)
exten=>114,3,Hangup
exten=>115,1,Dial(sip/115,30,t)
exten=>115,2,VoiceMail(115@vm)
exten=>115,3,Hangup
;untuk mendengar voice mail
exten=>8101,1,VoiceMailMain(101@vm) exten=>8102,1,VoiceMailMain(102@vm) exten=>8103,1,VoiceMailMain(103@vm) exten=>8104,1,VoiceMailMain(104@vm) exten=>8112,1,VoiceMailMain(112@vm) exten=>8113,1,VoiceMailMain(113@vm) exten=>8114,1,VoiceMailMain(114@vm) exten=>8115,1,VoiceMailMain(115@vm) ;untuk converence
exten=>1111,1,Answer() exten=>1111,2,MeetMe(1111) exten=>1111,3,Hangup ;untuk ke cabang exten=>_3.,1,Dial(SIP/tokantor_B/${EXTEN},30,t) [tesserver] exten=>2005,1,Answer exten=>2005,2,Wait(60) exten=>2005,3,Hangup features.conf [general] parkext=>700 parkpos=>701-720 context=>parkedcalls
[featuremap] blindxfer=> # meetme.conf [general] audiobuffers=32 [rooms] conf=>1111,123456 voicemail.conf [vm] 101=>101,101,101@xxxx.ac.id 102=>102,102,102@xxxx.ac.id 103=>103,103,103@xxxx.ac.id 104=>104,104,104@xxxx.ac.id 112=>112,112,112@xxxx.ac.id
113=>113,113,113@xxxx.ac.id
114=>114,114,114@xxxx.ac.id
115=>115,115,115@xxxx.ac.id
Konfigurasi asterisk pada server Kantor Cabang
Sip.conf [general] bindport=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=ulaw allow=gsm insecure=very register=>111:111@192.168.1.1
[301] disallow=all allow=gsm type =friend context=kantorB callerid=ronald username=301 secret=301 host=dynamic [302] disallow=all allow=gsm type =friend context=kantorB callerid=ronald username=302
secret=302 host=dynamic [303] type =friend context=kantorB callerid=ronald username=303 secret=303 host=dynamic dtmfmode=inband [304] type =friend context=kantorB callerid=ronald username=304
secret=304 host=dynamic dtmfmode=inband ;untuk gsm gateway [310] type =friend context=kantorBC callerid=ronald username=310 secret=310 host=dynamic dtmfmode=inband ;untuk ivr gsm [3100] type =friend context=kantorBC
callerid=ronald
username=3100
secret=3100
host=dynamic
dtmfmode=inband
;untuk dikenal di server cabang
[311] disallow=all allow=gsm type =friend context=kantorB callerid=ronald username=311 secret=311 host=dynamic dtmfmode=inband
;nomor yang tidak bisa telepon ke gsm [312] disallow=all allow=gsm type =friend context=kantorBC callerid=ronald username=312 secret=312 host=dynamic [313] disallow=all allow=gsm type =friend context=kantorBC callerid=ronald
username=313 secret=313 host=dynamic [314] type =friend context=kantorBC callerid=ronald username=314 secret=314 host=dynamic dtmfmode=inband [315] type =friend context=kantorBC callerid=ronald
username=315 secret=315 host=dynamic dtmfmode=inband [tokantor_A] disallow=all allow=gsm type=friend context=kantorB username=111 secret=111 host=192.168.1.1
;untuk program sipp
[sipp]
context=tesserver host=192.168.3.2 port=5060 user=sipp extention.conf [kantorB] include=>parkedcalls include=>kantorBC ;untuk ke gsm exten=>_999.,1,Dial(SIP/${EXTEN:3}@310,30,t)
;untuk ke gsm di kantor pusat
exten=>_988.,1,Dial(SIP/tokantor_A/${EXTEN},30,t)
;untuk ke gsm denganm timer
exten=>_979.,1,Dial(SIP/${EXTEN:3}@310,30,tL(300000:60000))
exten=>_978.,1,Dial(SIP/tokantor_A/${EXTEN},30,t) [kantorBC] include=> parkedcalls exten=>301,1,Dial(sip/301,30,t) exten=>301,2,VoiceMail(301@vm) exten=>301,3,Hangup exten=>302,1,Dial(sip/302,30,t) exten=>302,2,VoiceMail(302@vm) exten=>302,3,Hangup exten=>303,1,Dial(sip/303,30,t) exten=>303,2,voiceMail(303@vm) exten=>303,3,hangup exten=>304,1,Dial(sip/304,30,t) exten=>304,2,VoiceMail(304@vm)
exten=>304,3,Hangup ;panggilan dari gsm exten=>3100,1,Answer() exten=>3100,2,Set(TIMEOUT(absolute)=120) exten=>3100,3,Background(selamat-datang) exten=>3100,4,Wait(1) exten=>3100,5,Goto(3) exten=>T,1,Wait(1) exten=>T,2,Goto(301,1) exten=>312,1,Dial(sip/312,30,t) exten=>312,2,VoiceMail(312@vm) exten=>312,3,Hangup exten=>313,1,Dial(sip/313,30,t) exten=>313,2,VoiceMail(313@vm)
exten=>313,3,Hangup exten=>314,1,Dial(sip/314,30,t) exten=>314,2,VoiceMail(314@vm) exten=>314,3,Hangup exten=>315,1,Dial(sip/315,30,t) exten=>315,2,VoiceMail(315@vm) exten=>315,3,Hangup
;untuk melihat voice mail
exten=>8301,1,VoiceMailMain(301@vm) exten=>8302,1,VoiceMailMain(302@vm) exten=>8303,1,VoiceMailMain(303@vm) exten=>8304,1,VoiceMailMain(304@vm) exten=>8312,1,VoiceMailMain(312@vm) exten=>8313,1,VoiceMailMain(313@vm)
exten=>8314,1,VoiceMailMain(314@vm) exten=>8315,1,VoiceMailMain(315@vm) ;untuk converence exten=>3333,1,Answer() exten=>3333,2,MeetMe(3333) exten=>3333,1,Hangup ;untuk ke cabang exten=>_1.,1,Dial(SIP/tokantor_A/${EXTEN},30,t) [tesserver] exten=>2005,1,Answer exten=>2005,2,Wait(60) exten=>2005,3,Hangup
features.conf [general] parkext=>700 parkpos=>701-720 context=>parkedcalls [featuremap] blindxfer=> # meetme.conf [general] audiobuffers=32 [rooms] conf=> 3333,123456 voicemail.conf [vm]
301=>301,301,301@xxxx.ac.id 302=>302,302,302@xxxx.ac.id 303=>303,303,303@xxxx.ac.id 304=>304,304,304@xxxx.ac.id 312=>312,312,312@xxxx.ac.id 313=>313,313,313@xxxx.ac.id 314=>314,314,314@xxxx.ac.id 315=>315,315,315@xxxx.ac.id
Konfigurasi untuk membuat fitur Voice Mail
Pada server Kantor Pusat
Pada extention.conf
;untuk mendengar voice mail
exten=>8101,1,VoiceMailMain(101@vm)
exten=>8102,1,VoiceMailMain(102@vm)
exten=>8103,1,VoiceMailMain(103@vm)
exten=>8112,1,VoiceMailMain(112@vm) exten=>8113,1,VoiceMailMain(113@vm) exten=>8114,1,VoiceMailMain(114@vm) exten=>8115,1,VoiceMailMain(115@vm) pada VoiceMail.conf [vm] 101=>101,101,101@xxxx.ac.id 102=>102,102,102@xxxx.ac.id 103=>103,103,103@xxxx.ac.id 104=>104,104,104@xxxx.ac.id 112=>112,112,112@xxxx.ac.id 113=>113,113,113@xxxx.ac.id 114=>114,114,114@xxxx.ac.id 115=>115,115,115@xxxx.ac.id
Pada server Kantor Cabang
Pada extention.conf
;untuk melihat voice mail
exten=>8301,1,VoiceMailMain(301@vm) exten=>8302,1,VoiceMailMain(302@vm) exten=>8303,1,VoiceMailMain(303@vm) exten=>8304,1,VoiceMailMain(304@vm) exten=>8312,1,VoiceMailMain(312@vm) exten=>8313,1,VoiceMailMain(313@vm) exten=>8314,1,VoiceMailMain(314@vm) exten=>8315,1,VoiceMailMain(315@vm) Pada voicemail.conf [vm] 301=>301,301,301@xxxx.ac.id 302=>302,302,302@xxxx.ac.id 303=>303,303,303@xxxx.ac.id
304=>304,304,304@xxxx.ac.id
312=>312,312,312@xxxx.ac.id
313=>313,313,313@xxxx.ac.id
314=>314,314,314@xxxx.ac.id
315=>315,315,315@xxxx.ac.id
Konfigurasi untuk membuat fitur Conference Call
Pada server Kantor Pusat
Pada extention.conf ;untuk converence exten=>1111,1,Answer() exten=>1111,2,MeetMe(1111) exten=>1111,3,Hangup Pada meetme.conf [general] audiobuffers=32
[rooms]
conf=>1111,123456
Pada server Kantor Cabang
Pada extention.conf ;untuk converence exten=>3333,1,Answer() exten=>3333,2,MeetMe(3333) exten=>3333,3,Hangup Pada meetme.conf [general] audiobuffers=32 [rooms] conf=>3333,123456
Konfigurasi untuk membuat fitur IVR
Pada server Kantor Pusat
;panggilan dari gsm exten=>1100,1,Answer() exten=>1100,2,Set(TIMEOUT(absolute)=120) exten=>1100,3,Background(selamat-datang) exten=>1100,4,Wait(1) exten=>1100,5,Goto(3) exten=>T,1,Wait(1) exten=>T,2,Goto(101,1)
Pada server Kantor Cabang
Pada extention.conf ;panggilan dari gsm exten=>3100,1,Answer() exten=>3100,2,Set(TIMEOUT(absolute)=120) exten=>3100,3,Background(selamat-datang) exten=>3100,4,Wait(1) exten=>3100,5,Goto(3)
exten=>T,1,Wait(1)
exten=>T,2,Goto(301,1)
Konfigurasi untuk call parking
Pada server kantor pusat dan kantor cabang
Pada features.conf [general] parkext=>700 parkpos=>701-720 context=>parkedcalls [featuremap] blindxfer=> #
The Asterisk(R) Open Source PBX
by Mark Spencer <markster@digium.com> and the Asterisk.org developer community Copyright (C) 2001-2006 Digium, Inc. and other copyright holders.
=============================================================== * SECURITY
It is imperative that you read and fully understand the contents of the security
information file (doc/security.txt) before you attempt to configure and run an Asterisk server.
* WHAT IS ASTERISK ?
Asterisk is an Open Source PBX and telephony toolkit. It is, in a sense, middleware between Internet and telephony channels on the bottom, and Internet and telephony applications at the top. For more information on the project itself, please visit the Asterisk home page at:
http://www.asterisk.org
In addition you'll find lots of information compiled by the Asterisk community on this Wiki:
http://www.voip-info.org/wiki-Asterisk
There is a book on Asterisk published by O'Reilly under the Creative Commons License. It is available in book stores as well as in a downloadable version on the
* SUPPORTED OPERATING SYSTEMS == Linux ==
The Asterisk Open Source PBX is developed and tested primarily on the GNU/Linux operating system, and is supported on every major GNU/Linux distribution.
== Others ==
Asterisk has also been 'ported' and reportedly runs properly on other operating systems as well, including Sun Solaris, Apple's Mac OS X, and the BSD variants.
* GETTING STARTED
First, be sure you've got supported hardware (but note that you don't need ANY special hardware, not even a soundcard) to install and run Asterisk. Supported telephony
hardware includes:
* All Wildcard (tm) products from Digium (www.digium.com)
* QuickNet Internet PhoneJack and LineJack (http://www.quicknet.net) * any full duplex sound card supported by ALSA or OSS
* any ISDN card supported by mISDN on Linux (BRI) * The Xorcom AstriBank channel bank
* VoiceTronix OpenLine products
The are several drivers for ISDN BRI cards available from third party sources. Check the voip-info.org wiki for more information on chan_capi and zaphfc.
If you are updating from a previous version of Asterisk, make sure you read the UPGRADE.txt file in the source directory. There are some files and configuration options that you will have to change, even though we made every effort possible to maintain backwards compatibility.
In order to discover new features to use, please check the configuration examples in the /configs directory of the source code distribution. To discover the major new features of Asterisk 1.2, please visit http://edvina.net/asterisk1-2/
* NEW INSTALLATIONS
Ensure that your system contains a compatible compiler and development libraries. Asterisk requires either the GNU Compiler Collection (GCC) version 3.0 or higher, or a compiler that supports the C99 specification and some of the gcc language extensions. In addition, your system needs to have the C library headers available, and the headers and libraries for OpenSSL, ncurses and zlib.
On many distributions, these files are installed by packages with names like 'glibc-devel', 'ncurses-'glibc-devel', 'openssl-devel' and 'zlib-devel' or similar.
So let's proceed:
There are more documents than this one in the doc/ directory. You may also want to check the configuration files that contain examples and reference guides. They are all in the configs/directory.
2) Run "./configure"
Execute the configure script to guess values for system-dependent variables used during compilation.
3) Run "make menuselect" [optional]
This is needed if you want to select the modules that will be compiled and to check modules dependencies.
4) Run "make"
Assuming the build completes successfully: 5) Run "make install"
Each time you update or checkout from the repository, you are strongly encouraged to ensure all previous object files are removed to avoid internal inconsistency in Asterisk. Normally, this is automatically done with the presence of the file .cleancount, which increments each time a 'make clean' is required, and the file .lastclean, which contains the last .cleancount used. If this is your first time working with Asterisk, you may wish to install the sample PBX, with demonstration extensions, etc. If so, run:
6) "make samples"
Doing so will overwrite any existing config files you have.
Finally, you can launch Asterisk in the foreground mode (not a daemon) with:
You'll see a bunch of verbose messages fly by your screen as Asterisk initializes (that's the "very very verbose" mode). When it's ready, if you specified the "c" then you'll get a command line console, that looks like this:
*CLI>
You can type "help" at any time to get help with the system. For help with a specific command, type "help <command>". To start the PBX using your sound card, you can type "dial" to dial the PBX. Then you can use "answer", "hangup", and "dial" to simulate the actions of a telephone. Remember that if you don't have a full duplex sound card (and Asterisk will tell you somewhere in its verbose messages if you do/don't) then it won't work right (not yet). "man asterisk" at the Unix/Linux command prompt will give you detailed information on how to start and stop Asterisk, as well as all the command line options for starting Asterisk. Feel free to look over the configuration files in /etc/asterisk, where you'll find a lot of information about what you can do with Asterisk.
Penyebab jitter disebabkan karena adanya penungguan paket SIP-BYE. Pada panggilan
dari jaringan lokal ke GSM (XL – XL ) tidak terdapat paket SIP-BYE pada call trace
Untuk panggilan dari jaringan lokal ke jaringan GSM untuk Mentari – Mentari terdapat
paket SIP-BYE pada call trace.
