Difference between revisions of "OpenSIPS: Rewritehostport"
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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 19:57, 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