OpenSIPS: Rewritehostport: Difference between revisions
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Onnowpurbo (talk | contribs) New page: if (uri=~"^sip:021[0-9]*@*") { rewritehostport( "192.168.0.200:5061"); ## 192.168.0.200:5061 adalah Analog Telepon Adapter (ATA) route(1); }; |
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if (uri=~"^sip:021[0-9]*@*") { | if (uri=~"^sip:021[0-9]*@*") { | ||
rewritehostport( "192.168.0.200:5061"); ## 192.168.0.200:5061 adalah Analog Telepon Adapter (ATA) | rewritehostport( "192.168.0.200:5061"); ## 192.168.0.200:5061 adalah Analog Telepon Adapter (ATA) | ||
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]] | |||
Revision as of 12:57, 25 February 2014
if (uri=~"^sip:021[0-9]*@*") { rewritehostport( "192.168.0.200:5061"); ## 192.168.0.200:5061 adalah Analog Telepon Adapter (ATA) 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
