<?xml version="1.0" encoding="utf-8" ?>
<!DOCTYPE packagegui SYSTEM "./schema/packages.dtd">
<?xml-stylesheet type="text/xsl" href="./xsl/package.xsl"?>
<packagegui>
	<copyright>
	<![CDATA[
/*
	siproxd.xml
	Copyright (C) 2006 Scott Ullrich
	Copyright (C) 2010 Jim Pingle
	All rights reserved.

	Redistribution and use in source and binary forms, with or without
	modification, are permitted provided that the following conditions are met:

	1. Redistributions of source code must retain the above copyright notice,
	   this list of conditions and the following disclaimer.

	2. Redistributions in binary form must reproduce the above copyright
	   notice, this list of conditions and the following disclaimer in the
	   documentation and/or other materials provided with the distribution.

	THIS SOFTWARE IS PROVIDED ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES,
	INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY
	AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE
	AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY,
	OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
	SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
	INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
	CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
	ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
	POSSIBILITY OF SUCH DAMAGE.
*/
	]]>
	</copyright>
	<description>Describe your package here</description>
	<requirements>Describe your package requirements here</requirements>
	<faq>Currently there are no FAQ items provided.</faq>
	<name>siproxdsettings</name>
	<version>0.5.13_pfs2</version>
	<title>siproxd: Settings</title>
	<include_file>/usr/local/pkg/siproxd.inc</include_file>
	<aftersaveredirect>/pkg_edit.php?xml=siproxd.xml&amp;id=0</aftersaveredirect>
	<menu>
		<name>siproxd</name>
		<tooltiptext>Modify siproxd users and settings.</tooltiptext>
		<section>Services</section>
		<url>/pkg_edit.php?xml=siproxd.xml&amp;id=0</url>
	</menu>
	<service>
		<name>siproxd</name>
		<rcfile>siproxd.sh</rcfile>
		<executable>siproxd</executable>
	</service>
	<tabs>
		<tab>
			<text>Settings</text>
			<url>/pkg_edit.php?xml=siproxd.xml&amp;id=0</url>
			<active/>
		</tab>
		<tab>
			<text>Users</text>
			<url>/pkg.php?xml=siproxdusers.xml</url>
		</tab>
		<tab>
			<text>Registered Phones</text>
			<url>/siproxd_registered_phones.php</url>
		</tab>
	</tabs>
	<additional_files_needed>
		<prefix>/usr/local/pkg/</prefix>
		<chmod>077</chmod>
		<item>http://www.pfsense.com/packages/config/siproxdusers.xml</item>
	</additional_files_needed>
	<additional_files_needed>
		<prefix>/usr/local/pkg/</prefix>
		<chmod>077</chmod>
		<item>http://www.pfsense.com/packages/config/siproxd.inc</item>
	</additional_files_needed>
	<additional_files_needed>
		<prefix>/usr/local/www/</prefix>
		<chmod>077</chmod>
		<item>http://www.pfsense.com/packages/config/siproxd_registered_phones.php</item>
	</additional_files_needed>
	<fields>
		<field>
			<fielddescr>Inbound interface</fielddescr>
			<fieldname>if_inbound</fieldname>
			<description>Select the inbound interface.</description>
			<type>interfaces_selection</type>
		</field>
		<field>
			<fielddescr>Outbound interface</fielddescr>
			<fieldname>if_outbound</fieldname>
			<description>Select the outbound interface.</description>
			<type>interfaces_selection</type>
		</field>
		<field>
			<fielddescr>Listening port</fielddescr>
			<fieldname>port</fieldname>
			<description>Enter the port on which to listen for SIP traffic (default 5060). Do not change this unless you know what you're doing.</description>
			<type>input</type>
		</field>
		<field>
			<fielddescr>Default expiration timeout</fielddescr>
			<fieldname>defaulttimeout</fieldname>
			<description>If a REGISTER request dose not contain an Expires header or expires= parameter, this number of seconds will be used and reported back to the UA in the answer.</description>
			<type>input</type>
		</field>
		<field>
			<name>RTP Settings</name>
			<type>listtopic</type>
		</field>
		<field>
			<fielddescr>Enable RTP proxy</fielddescr>
			<fieldname>rtpenable</fieldname>
			<description>Enable or disable the RTP proxy. (default is enabled)</description>
			<type>select</type>
			<options>
				<option>
					<name>Enable</name>
					<value>1</value>
				</option>
				<option>
					<name>Disable</name>
					<value>0</value>
				</option>
			</options>
		</field>
		<field>
			<fielddescr>RTP port range (lower)</fielddescr>
			<fieldname>rtplower</fieldname>
			<description>Enter the bottom edge of the port range siproxd will allocate for incoming RTP traffic. This range must be one not blocked by the firewall (default 7070).</description>
			<type>input</type>
		</field>
		<field>
			<fielddescr>RTP port range (upper)</fielddescr>
			<fieldname>rtpupper</fieldname>
			<description>Enter the top edge of the port range siproxd will allocate for incoming RTP traffic. This range must be one not blocked by the firewall (default 7079).</description>
			<type>input</type>
		</field>
		<field>
			<fielddescr>RTP stream timeout</fielddescr>
			<fieldname>rtptimeout</fieldname>
			<description>After this number of seconds, an RTP stream is considered dead and proxying it will be stopped (default 300sec).</description>
			<type>input</type>
		</field>
		<field>
			<name>Dejittering Settings</name>
			<type>listtopic</type>
		</field>
		<field>
			<fielddescr>Input Dejitter</fielddescr>
			<fieldname>rtp_input_dejitter</fieldname>
			<description>Artificial delay to be used to de-jitter RTP data streams. This time is in microseconds. 0 - completely disable dejitter (default)</description>
			<type>input</type>
		</field>
		<field>
			<fielddescr>Output Dejitter</fielddescr>
			<fieldname>rtp_output_dejitter</fieldname>
			<description>Artificial delay to be used to de-jitter RTP data streams. This time is in microseconds. 0 - completely disable dejitter (default)</description>
			<type>input</type>
		</field>
		<field>
			<name>SIP over TCP Settings</name>
			<type>listtopic</type>
		</field>
		<field>
			<fielddescr>TCP inactivity timeout</fielddescr>
			<fieldname>tcp_timeout</fieldname>
			<description>Inactivity timeout (seconds). After that an idling TCP connection is disconnected. NOTE: Making this too short may cause multiple parallel registrations for the same phone. This timeout must be set larger than the used registration interval.</description>
			<type>input</type>
		</field>
		<field>
			<fielddescr>TCP Connect Timeout</fielddescr>
			<fieldname>tcp_connect_timeout</fieldname>
			<description>Defines How many msecs siproxd will wait for an successful connect when establishing an outgoing SIP signalling connection. This should be kept as short as possible as waiting for an TCP connection to establish is a BLOCKING operation - while waiting for a connect to succeed no SIP messages are processed (RTP is not affected).</description>
			<type>input</type>
		</field>
		<field>
			<fielddescr>TCP Keepalive</fielddescr>
			<fieldname>tcp_keepalive</fieldname>
			<description>For TCP SIP signalling, if > 0 empty SIP packets will be sent every 'n' seconds to keep the connection alive. Default is off.</description>
			<type>input</type>
		</field>
		<field>
			<name>Proxy Settings</name>
			<type>listtopic</type>
		</field>
		<field>
			<fielddescr>Enable proxy authentication</fielddescr>
			<fieldname>authentication</fieldname>
			<description>If this is checked, clients will be forced to authenticate themselves at the proxy (for registration only).</description>
			<type>checkbox</type>
		</field>
		<field>
			<fielddescr>Outbound proxy hostname</fielddescr>
			<fieldname>outboundproxyhost</fieldname>
			<description>Enter the hostname of an outbound proxy to send all traffic to. This is only useful if you have multiple masquerading firewalls to cross.</description>
			<type>input</type>
		</field>
		<field>
			<fielddescr>Outbound proxy port</fielddescr>
			<fieldname>outboundproxyport</fieldname>
			<description>Enter the port of the outbound proxy to send all traffic to. This is only useful if you have multiple masquerading firewalls to cross.</description>
			<type>input</type>
		</field>
		<field>
			<name>DSCP Settings</name>
			<type>listtopic</type>
		</field>
		<field>
			<fielddescr>Expedited RTP Forwarding</fielddescr>
			<fieldname>expeditedforwarding</fieldname>
			<description>This service is designed to allow ISPs to offer a service with attributes similar to a "leased line". This service offers the ULTIMATE IN LOW LOSS, LOW LATENCY AND LOW JITTER by ensuring that there is always sufficient room in output queues for the contracted expedited forwarding traffic.
			</description>
			<type>checkbox</type>
		</field>
		<field>
			<fielddescr>Expedited SIP Forwarding</fielddescr>
			<fieldname>expeditedsipforwarding</fieldname>
			<description>This service is designed to allow ISPs to offer a service with attributes similar to a "leased line". This service offers the ULTIMATE IN LOW LOSS, LOW LATENCY AND LOW JITTER by ensuring that there is always sufficient room in output queues for the contracted expedited forwarding traffic.</description>
			<type>checkbox</type>
		</field>
		<field>
			<name>Plugin Settings - Default Target</name>
			<type>listtopic</type>
		</field>
		<field>
			<fielddescr>Enable Default Target Plugin</fielddescr>
			<fieldname>plugin_defaulttarget</fieldname>
			<description>Redirect unknown calls to a specified target.</description>
			<type>checkbox</type>
		</field>
		<field>
			<fielddescr>Log redirected calls</fielddescr>
			<fieldname>plugin_defaulttarget_log</fieldname>
			<description>Log redirected calls.</description>
			<type>checkbox</type>
		</field>
		<field>
			<fielddescr>Default Target</fielddescr>
			<fieldname>plugin_defaulttarget_target</fieldname>
			<description>Target must be a full SIP URI with the syntax sip:user@host[:port]</description>
			<type>input</type>
		</field>
		<field>
			<name>Plugin Settings - Bogus VIA Networks</name>
			<type>listtopic</type>
		</field>
		<field>
			<fielddescr>Enable Fix Bogus Via Networks Plugin</fielddescr>
			<fieldname>plugin_fix_bogus_via</fieldname>
			<description>Incoming (from public network) SIP messages are checked for broken SIP Via headers. If the IP address in the latest Via Header is part of the list below, it will be replaced by the IP where the SIP message has been received from.</description>
			<type>checkbox</type>
		</field>
		<field>
			<fielddescr>Bogus Via Networks</fielddescr>
			<fieldname>plugin_fix_bogus_via_networks</fieldname>
			<description>Comma separated list of networks which should have their via headers rewritten. Example: 10.0.0.0/8,172.16.0.0/12,192.168.0.0/16</description>
			<type>input</type>
		</field>
		<field>
			<name>Plugin Settings - STUN</name>
			<type>listtopic</type>
		</field>
		<field>
			<fielddescr>Enable STUN Plugin</fielddescr>
			<fieldname>plugin_stun</fieldname>
			<description>Uses an external STUN server to determine the public IP address of siproxd. Useful for "in-front-of-NAT-router" scenarios.</description>
			<type>checkbox</type>
		</field>
		<field>
			<fielddescr>STUN Hostname</fielddescr>
			<fieldname>plugin_stun_server</fieldname>
			<description>External STUN server hostname.</description>
			<type>input</type>
		</field>
		<field>
			<fielddescr>STUN Port</fielddescr>
			<fieldname>plugin_stun_port</fieldname>
			<description>External STUN server port.</description>
			<type>input</type>
		</field>
		<field>
			<fielddescr>STUN Period</fielddescr>
			<fieldname>plugin_stun_period</fieldname>
			<description>Period in seconds to request IP info from STUN server.</description>
			<type>input</type>
		</field>
		<field>
			<name>Debug Options</name>
			<type>listtopic</type>
		</field>
		<field>
			<fielddescr>Debug Level</fielddescr>
			<fieldname>debug_level</fieldname>
			<type>select</type>
			<size>1</size>
			<options>
				<option><value>0x00000000</value><name>No Debug Info</name></option>
				<option><value>0x00000001</value><name>babble (like entering/leaving func)</name></option>
				<option><value>0x00000002</value><name>network</name></option>
				<option><value>0x00000004</value><name>SIP manipulations</name></option>
				<option><value>0x00000008</value><name>Client registration</name></option>
				<option><value>0x00000010</value><name>non specified class</name></option>
				<option><value>0x00000020</value><name>proxy</name></option>
				<option><value>0x00000040</value><name>DNS stuff</name></option>
				<option><value>0x00000080</value><name>network traffic</name></option>
				<option><value>0x00000100</value><name>configuration</name></option>
				<option><value>0x00000200</value><name>RTP proxy</name></option>
				<option><value>0x00000400</value><name>Access list evaluation</name></option>
				<option><value>0x00000800</value><name>Authentication</name></option>
				<option><value>0x00001000</value><name>Plugins</name></option>
				<option><value>0x00002000</value><name>RTP babble</name></option>
				<option><value>-1</value><name>Everything</name></option>
			</options>
		</field>
		<field>
			<fielddescr>TCP Debug Port</fielddescr>
			<fieldname>debug_port</fieldname>
			<description>You may connect to this port from a remote machine and receive debug output. This allows better creation of debug output on embedded systems that do not have enough memory for large disk files. Port number 0 means this feature is disabled.</description>
			<type>input</type>
		</field>
	</fields>
	<custom_php_global_functions>
	</custom_php_global_functions>
	<custom_add_php_command>
		sync_package_siproxd();
	</custom_add_php_command>
	<custom_php_resync_config_command>
		sync_package_siproxd();
	</custom_php_resync_config_command>
	<filter_rules_needed>siproxd_generate_rules</filter_rules_needed>
	<custom_php_validation_command>
		validate_form_siproxd($_POST, &amp;$input_errors);
	</custom_php_validation_command>
</packagegui>