Difference between revisions of "VoIP: Asterisk menerima Anonymous Call"
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==Lebih Dalam== | ==Lebih Dalam== | ||
| + | * [[VoIP: Trunk]] | ||
* [[VoIP: Asterisk menerima Anonymous Call]] | * [[VoIP: Asterisk menerima Anonymous Call]] | ||
* [[VoIP: Asterisk route ke arah IP PBX lain dengan kode area]] | * [[VoIP: Asterisk route ke arah IP PBX lain dengan kode area]] | ||
* [[VoIP: Cara Mengkonfigurasi Trunk di Asterisk]] | * [[VoIP: Cara Mengkonfigurasi Trunk di Asterisk]] | ||
* [[VoIP: Cara Mengkonfigurasi Dial Out Melalui Trunk di Asterisk]] | * [[VoIP: Cara Mengkonfigurasi Dial Out Melalui Trunk di Asterisk]] | ||
| + | * [[VoIP: Asterisk forward call ke IP softswitch lain]] | ||
| + | |||
| + | ===Asterisk Round Robin=== | ||
| + | |||
| + | * [[VoIP: Astersk Dial Round Robin]] | ||
| + | * [[VoIP: Asterisk pakai GotoIf]] | ||
===OpenSIPS=== | ===OpenSIPS=== | ||
| Line 49: | Line 56: | ||
* [[VoIP: OpenSIPS route ke arah Asterisk]] | * [[VoIP: OpenSIPS route ke arah Asterisk]] | ||
* [[OpenSIPS: Rewrite URI]] | * [[OpenSIPS: Rewrite URI]] | ||
| + | * [[OpenSIPS: Rewritehostport]] | ||
| + | |||
| + | ===OpenSIPS round robin=== | ||
| + | |||
| + | * [[OpenSIPS: dispatcher]] | ||
==Pranala Menarik== | ==Pranala Menarik== | ||
Latest revision as of 19:55, 25 February 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
- VoIP: Trunk
- 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
- VoIP: Asterisk forward call ke IP softswitch lain
Asterisk Round Robin
OpenSIPS
OpenSIPS round robin
Pranala Menarik
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
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