VoIP: OpenSIPS route ke arah Asterisk - Revision history

Onnowpurbo: /* Lebih Dalam */

2014-02-25T12:57:00Z

Lebih Dalam

← Older revision		Revision as of 12:57, 25 February 2014
Line 55:		Line 55:
	==Lebih Dalam==		==Lebih Dalam==

			* [[VoIP: Trunk]]
	* [[VoIP: Asterisk menerima Anonymous Call]]		* [[VoIP: Asterisk menerima Anonymous Call]]
	* [[VoIP: Asterisk route ke arah IP PBX lain dengan kode area]]		* [[VoIP: Asterisk route ke arah IP PBX lain dengan kode area]]
	* [[VoIP: Cara Mengkonfigurasi Trunk di Asterisk]]		* [[VoIP: Cara Mengkonfigurasi Trunk di Asterisk]]
	* [[VoIP: Cara Mengkonfigurasi Dial Out Melalui 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===		===OpenSIPS===
Line 64:		Line 71:
	* [[VoIP: OpenSIPS route ke arah Asterisk]]		* [[VoIP: OpenSIPS route ke arah Asterisk]]
	* [[OpenSIPS: Rewrite URI]]		* [[OpenSIPS: Rewrite URI]]
			* [[OpenSIPS: Rewritehostport]]

			===OpenSIPS round robin===

			* [[OpenSIPS: dispatcher]]


	==Pranala Menarik==		==Pranala Menarik==

Onnowpurbo: /* Referensi */

2014-01-20T04:35:32Z

Referensi

← Older revision		Revision as of 04:35, 20 January 2014
Line 51:		Line 51:

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


			==Lebih Dalam==

			* [[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]]

			===OpenSIPS===

			* [[VoIP: OpenSIPS route ke arah Asterisk]]
			* [[OpenSIPS: Rewrite URI]]





			==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]]

Onnowpurbo at 01:50, 20 January 2014

2014-01-20T01:50:15Z

← Older revision		Revision as of 01:50, 20 January 2014
Line 3:		Line 3:
	Contoh hook ke Asterisk		Contoh hook ke Asterisk

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

Onnowpurbo: New page: Sumber: http://www.opensips.org/Documentation/Tutorials-OpenSIPSAsteriskIntegration-1-8 Contoh hook ke Asterisk # ASTERISK HOOK - BEGIN # media service number? (digits starting with *)...

2014-01-20T01:40:41Z

New page: Sumber: http://www.opensips.org/Documentation/Tutorials-OpenSIPSAsteriskIntegration-1-8 Contoh hook ke Asterisk # ASTERISK HOOK - BEGIN # media service number? (digits starting with *)...

New page

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