View source for VoIP: OpenSIPS route ke arah Asterisk

Sumber: http://www.opensips.org/Documentation/Tutorials-OpenSIPSAsteriskIntegration-1-8

Contoh hook ke Asterisk

 	# ASTERISK HOOK - BEGIN
 	# media service number? (digits starting with *)
 	if ($rU=~"^\*[1-9]+") {
 		# we do provide access to media services only to our
 		# subscribers, who were previously authenticated 
 		if (!is_from_local()) {
 			send_reply("403","Forbidden access to media service");
 			exit;
 		}
 		#identify the services and translate to Asterisk extensions
 		if ($rU=="*1111") {
 			# access to own voicemail IVR
 			$ru = "sip:VM_pickup@ASTERISK_IP:ASTERISK_PORT";
 		} else
 		if ($rU=="*2111") {
 			# access to the "say time" announcement 
 			$ru = "sip:AN_time@ASTERISK_IP:ASTERISK_PORT";
 		} else
 		if ($rU=="*2112") {
 			# access to the "say date" announcement 
 			$ru = "sip:AN_date@ASTERISK_IP:ASTERISK_PORT";
 		} else
 		if ($rU=="*2113") {
 			# access to the "echo" service
 			$ru = "sip:AN_echo@ASTERISK_IP:ASTERISK_PORT";
 		} else
 		if ($rU=~"\*3[0-9]{3}") {
 			# access to the conference service 
 			# remove the "*3" prefix and place the "CR_" prefix
 			strip(2);
 			prefix("CR_");
 			rewritehostport("ASTERISK_IP:ASTERISK_PORT");
 		} else {
 			# unknown service
 			$ru = "sip:AN_notavailable@ASTERISK_IP:ASTERISK_PORT";
 		}
 		# after setting the proper RURI (to point to corresponding ASTERISK extension),
 		# simply forward the call
 		t_relay();
 		exit;
 	}
 	# ASTERISK HOOK - END
 


==Referensi==

* http://www.opensips.org/Documentation/Tutorials-OpenSIPSAsteriskIntegration-1-8


==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]]

==Pranala Menarik==

* [[VoIP]]
* [[OpenBTS]]

===Latar Belakang===

* [[Menjadikan VoIP dan 4G Legal]]
* [[Sekitar VoIP Rakyat]]
* [[VoIP: Dasar Hukum Internet Telepon]]
* [[VoIP: Beberapa Skenario Topologi]]
* [[VoIP: Pilihan Teknologi Internet Telepon]]
* [[VoIP: Pengkodean Suara di Jaringan Komputer]]
* [[VoIP: Konsep Video Conference]]

===Untuk Pemula===
* [[VoIP: Kebutuhan Peralatan dan Software]]
* [[VoIP: Internet Telepon PC ke PC]]

===Untuk Peneliti / Pencoba===
* [[VoIP: Bandwidth Internet Telepon]]
* [[VoIP: Softswitch / Server Internet Telepon]]
* [[VoIP: Repository Software Internet Telepon]]
* [[VoIP: Menghubungkan PSTN dan Selular]]

===Untuk Operator===
* [[VoIP: Server Video Conference]]
* [[VoIP: Software dan peralatan client Internet Telepon]]
* [[VoIP: Penggunaan DAHDI]]
* [[VoIP: Hardware Client VoIP]]
* [[VoIP: Hardware Server VoIP]]
* [[VoIP: Interkoneksi dan Alokasi Nomor Telepon]]
* [[VoIP: Peering Antar Operator VoIP]]
* [[VoIP: Menghubungkan PSTN dan Selular]]
* [[VoIP: Trunk]]

===Topik Lanjut===
* [[VoIP: ENUM untuk pengenalan nomor telepon di Internet Telepon]]
* [[VoIP: Teknik Evaluasi Internet Telepon]]
* [[VoIP: Troubleshooting]]
* [[VoIP: Video Conference Server]]

===Buku Teknologi VoIP===
* [[Onno W. Purbo]], "VoIP Cikal Bakal Telkom Rakyat", Infokomputer, 2007.
* http://125.160.17.21/speedyorari/index.php?dir=ebook-voip
* http://opensource.telkomspeedy.com/speedyorari/index.php?dir=ebook-voip


[[Category: VoIP]]
[[Category: Internet Telepon]]