Difference between revisions of "OpenSIPS: Rewrite URI"

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       route(1);
 
       route(1);
 
       };
 
       };
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==Lebih Dalam==
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* [[VoIP: Trunk]]
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* [[VoIP: Asterisk menerima Anonymous Call]]
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* [[VoIP: Asterisk route ke arah IP PBX lain dengan kode area]]
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* [[VoIP: Cara Mengkonfigurasi Trunk di Asterisk]]
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* [[VoIP: Cara Mengkonfigurasi Dial Out Melalui Trunk di Asterisk]]
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* [[VoIP: Asterisk forward call ke IP softswitch lain]]
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===Asterisk Round Robin===
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* [[VoIP: Astersk Dial Round Robin]]
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* [[VoIP: Asterisk pakai GotoIf]]
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===OpenSIPS===
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* [[VoIP: OpenSIPS route ke arah Asterisk]]
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* [[OpenSIPS: Rewrite URI]]
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* [[OpenSIPS: Rewritehostport]]
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===OpenSIPS round robin===
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* [[OpenSIPS: dispatcher]]
  
  
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* http://www.opensips.org/Documentation/Script-CoreFunctions#toc40
 
* http://www.opensips.org/Documentation/Script-CoreFunctions#toc40
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==Pranala Menarik==
 
==Pranala Menarik==

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

Asterisk Round Robin

OpenSIPS

OpenSIPS round robin


Referensi

Pranala Menarik

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