VoIP: OpenSIPS route ke arah Asterisk: Difference between revisions

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

Latest revision as of 12:57, 25 February 2014

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


Lebih Dalam

Asterisk Round Robin

OpenSIPS

OpenSIPS round robin

Pranala Menarik

Latar Belakang

Untuk Pemula

Untuk Peneliti / Pencoba

Untuk Operator

Topik Lanjut

Buku Teknologi VoIP

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