OpenSIPS: Rewrite URI: Difference between revisions
From OnnoCenterWiki
Jump to navigationJump to search
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... |
Onnowpurbo (talk | contribs) |
||
| (2 intermediate revisions by the same user not shown) | |||
| Line 1: | Line 1: | ||
Sumber: http://www.opensips.org/Documentation/Script-CoreFunctions#toc40 | Sumber: http://www.opensips.org/Documentation/Script-CoreFunctions#toc40 | ||
'''WARNING: Perlu di cek cara yang benar utk memakai rewriteuri''' | |||
==Contoh perintah== | ==Contoh perintah== | ||
| Line 12: | Line 17: | ||
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]] | |||
| Line 17: | Line 47: | ||
* 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 12:57, 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
