VoIP: Asterisk menerima Anonymous Call: Difference between revisions

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* http://blog.lithiumblue.com/2007/07/receiving-incoming-sip-calls-from.html
* http://blog.lithiumblue.com/2007/07/receiving-incoming-sip-calls-from.html
==Lebih Dalam==
* [[VoIP: Asterisk menerima Anonymous Call]]
* [[VoIP: Asterisk route ke arah IP PBX lain dengan kode area]]
* [[VoIP: Cara Mengkonfigurasi Trunk di Asterisk]]
* [[VoIP: Cara Mengkonfigurasi Dial Out Melalui Trunk di Asterisk]]
===OpenSIPS===
* [[VoIP: OpenSIPS route ke arah Asterisk]]
* [[OpenSIPS: Rewrite URI]]
==Pranala Menarik==
* [[VoIP]]
* [[OpenBTS]]
===Latar Belakang===
* [[Menjadikan VoIP dan 4G Legal]]
* [[Sekitar VoIP Rakyat]]
* [[VoIP: Dasar Hukum Internet Telepon]]
* [[VoIP: Beberapa Skenario Topologi]]
* [[VoIP: Pilihan Teknologi Internet Telepon]]
* [[VoIP: Pengkodean Suara di Jaringan Komputer]]
* [[VoIP: Konsep Video Conference]]
===Untuk Pemula===
* [[VoIP: Kebutuhan Peralatan dan Software]]
* [[VoIP: Internet Telepon PC ke PC]]
===Untuk Peneliti / Pencoba===
* [[VoIP: Bandwidth Internet Telepon]]
* [[VoIP: Softswitch / Server Internet Telepon]]
* [[VoIP: Repository Software Internet Telepon]]
* [[VoIP: Menghubungkan PSTN dan Selular]]
===Untuk Operator===
* [[VoIP: Server Video Conference]]
* [[VoIP: Software dan peralatan client Internet Telepon]]
* [[VoIP: Penggunaan DAHDI]]
* [[VoIP: Hardware Client VoIP]]
* [[VoIP: Hardware Server VoIP]]
* [[VoIP: Interkoneksi dan Alokasi Nomor Telepon]]
* [[VoIP: Peering Antar Operator VoIP]]
* [[VoIP: Menghubungkan PSTN dan Selular]]
* [[VoIP: Trunk]]
===Topik Lanjut===
* [[VoIP: ENUM untuk pengenalan nomor telepon di Internet Telepon]]
* [[VoIP: Teknik Evaluasi Internet Telepon]]
* [[VoIP: Troubleshooting]]
* [[VoIP: Video Conference Server]]
===Buku Teknologi VoIP===
* [[Onno W. Purbo]], "VoIP Cikal Bakal Telkom Rakyat", Infokomputer, 2007.
* http://125.160.17.21/speedyorari/index.php?dir=ebook-voip
* http://opensource.telkomspeedy.com/speedyorari/index.php?dir=ebook-voip
[[Category: VoIP]]
[[Category: Internet Telepon]]

Revision as of 04:32, 20 January 2014

Sumber: http://blog.lithiumblue.com/2007/07/receiving-incoming-sip-calls-from.html

Di FreePBX / Briker -> General Settings -> Allow Anonymous Inbound SIP Calls -> Yes

Edit sip.conf 

[general]
bindport=5060 ; UDP Port to bind to
bindaddr=0.0.0.0 ; (0.0.0.0 binds to all)
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc
context=from-sip-external
callerid=Unknown
tos=0x68

;------------- Ryan's Mods --------------
externip=203.214.45.124 ;required behind NAT
localnet=192.168.0.0/255.255.255.0 ;required behind NAT
fromdomain=lithiumblue.com
canreinvite=no ;Required for UM calls to work
insecure=very
srvlookup=yes ;Required for outbound calls

#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf


Referensi


Lebih Dalam

OpenSIPS

Pranala Menarik

Latar Belakang

Untuk Pemula

Untuk Peneliti / Pencoba

Untuk Operator

Topik Lanjut

Buku Teknologi VoIP

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