Difference between revisions of "Relay ke PSTN Menggunakan OpenSIPS"
Jump to navigation
Jump to search
Onnowpurbo (talk | contribs) |
Onnowpurbo (talk | contribs) |
||
| Line 21: | Line 21: | ||
==Pranala Menarik== | ==Pranala Menarik== | ||
| + | * [[OpenSIPS dengan Backend Asterisk]] | ||
| + | * [[Relay ke PSTN Menggunakan OpenSIPS]] | ||
| + | * [[Relay ke Selular Menggunakan OpenSIPS]] | ||
| + | * [[ENUM Query di OpenSIPS]] | ||
* [[OpenSIPS Softswitch]] | * [[OpenSIPS Softswitch]] | ||
* [[VoIP]] | * [[VoIP]] | ||
Revision as of 10:10, 21 January 2010
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); };