Difference between revisions of "Relay ke PSTN Menggunakan OpenSIPS"
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==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]] | ||
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Latest revision as of 06:04, 31 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); };