Relay ke PSTN Menggunakan OpenSIPS: Difference between revisions
From OnnoCenterWiki
Jump to navigationJump to search
Onnowpurbo (talk | contribs) No edit summary |
Onnowpurbo (talk | contribs) |
||
| Line 21: | Line 21: | ||
==Pranala Menarik== | ==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 PSTN Menggunakan OpenSIPS]] | ||
* [[Relay ke Selular Menggunakan OpenSIPS]] | * [[Relay ke Selular Menggunakan OpenSIPS]] | ||
* [[ENUM Query di OpenSIPS]] | * [[ENUM Query di OpenSIPS]] | ||
Latest revision as of 23:04, 30 December 2013
Berikut adalah contoh sederhana cara merelay ke PSTN. Asumsi yang digunakan.
- ATA Berada di IP address 192.168.0.200 port 5061.
Dari semua domain
# attempt handoff to PSTN if (uri=~"^sip:021[0-9]*@*") { rewritehostport( "192.168.0.200:5061"); ## 192.168.0.200:5061 adalah Analog Telepon Adapter (ATA) route(1); };
Hanya dari mydomain.com
# attempt handoff to PSTN if (uri=~"^sip:021[0-9]*@mydomain.com") { ## Asumsinya caller register ke mydomain.com rewritehostport( "192.168.0.200:5061"); ## 192.168.0.200:5061 adalah Analog Telepon Adapter (ATA) route(1); };