Difference between revisions of "OpenSIPS: Rewrite URI"
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Onnowpurbo (talk | contribs) (New page: Sumber: http://www.opensips.org/Documentation/Script-CoreFunctions#toc40 ==Contoh perintah== rewriteuri("sip:test@opensips.org"); ==Perlu dicoba== # call ke PABX lain dengan kode are...) |
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Sumber: http://www.opensips.org/Documentation/Script-CoreFunctions#toc40 | Sumber: http://www.opensips.org/Documentation/Script-CoreFunctions#toc40 | ||
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+ | |||
+ | |||
+ | '''WARNING: Perlu di cek cara yang benar utk memakai rewriteuri''' | ||
+ | |||
==Contoh perintah== | ==Contoh perintah== | ||
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route(1); | route(1); | ||
}; | }; | ||
+ | |||
+ | |||
+ | ==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=== | ||
+ | |||
+ | * [[VoIP: Astersk Dial Round Robin]] | ||
+ | * [[VoIP: Asterisk pakai GotoIf]] | ||
+ | |||
+ | ===OpenSIPS=== | ||
+ | |||
+ | * [[VoIP: OpenSIPS route ke arah Asterisk]] | ||
+ | * [[OpenSIPS: Rewrite URI]] | ||
+ | * [[OpenSIPS: Rewritehostport]] | ||
+ | |||
+ | ===OpenSIPS round robin=== | ||
+ | |||
+ | * [[OpenSIPS: dispatcher]] | ||
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* http://www.opensips.org/Documentation/Script-CoreFunctions#toc40 | * http://www.opensips.org/Documentation/Script-CoreFunctions#toc40 | ||
+ | |||
+ | ==Pranala Menarik== | ||
+ | |||
+ | * [[OpenSIPS]] | ||
+ | * [[OpenSIPS Softswitch]] | ||
+ | * [[Compile OpenSIPS]] | ||
+ | * [[OpenSIPS: Cek Konfigurasi]] | ||
+ | * [[OpenSIPS: Demo Dial Plan]] | ||
+ | * [[Menggunakan opensipsdbctl]] | ||
+ | * [[Menggunakan opensipsctl]] | ||
+ | * [[Konfigurasi minimal OpenSIPS]] | ||
+ | * [[OpenSIPS: Demo User]] | ||
+ | * [[Relay ke PSTN Menggunakan OpenSIPS]] | ||
+ | * [[Relay ke Selular Menggunakan OpenSIPS]] | ||
+ | * [[OpenSIPS: Rewrite URI]] | ||
+ | * [[ENUM Query di OpenSIPS]] | ||
+ | * [[OpenSIPS: Menjalankan Softswitch]] |
Latest revision as of 19:57, 25 February 2014
Sumber: http://www.opensips.org/Documentation/Script-CoreFunctions#toc40
WARNING: Perlu di cek cara yang benar utk memakai rewriteuri
Contoh perintah
rewriteuri("sip:test@opensips.org");
Perlu dicoba
# call ke PABX lain dengan kode area 3 if (uri=~"^sip:3[0-9]*@*") { ## rewriteuri( "^sip:[0-9]*@192.168.0.200:5060"); ## 192.168.0.200:5060 route(1); };
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
Referensi
Pranala Menarik
- OpenSIPS
- OpenSIPS Softswitch
- Compile OpenSIPS
- OpenSIPS: Cek Konfigurasi
- OpenSIPS: Demo Dial Plan
- Menggunakan opensipsdbctl
- Menggunakan opensipsctl
- Konfigurasi minimal OpenSIPS
- OpenSIPS: Demo User
- Relay ke PSTN Menggunakan OpenSIPS
- Relay ke Selular Menggunakan OpenSIPS
- OpenSIPS: Rewrite URI
- ENUM Query di OpenSIPS
- OpenSIPS: Menjalankan Softswitch