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