<profile name="internal-ipv6">
	<!--
		This is an example of a sofia profile setup to listen on IPv6.
	-->
	<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
	<!--aliases are other names that will work as a valid profile name for this profile-->
	<settings>
		<!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
		<param name="debug" value="0"/>
		<param name="sip-trace" value="no"/>
		<param name="context" value="public"/>
		<param name="rfc2833-pt" value="101"/>
		<!-- port to bind to for sip traffic -->
		<param name="sip-port" value="$${internal_sip_port}"/>
		<param name="dialplan" value="XML"/>
		<param name="dtmf-duration" value="2000"/>
		<param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
		<param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>
		<param name="use-rtp-timer" value="true"/>
		<param name="rtp-timer-name" value="soft"/>
		<!-- ip address to use for rtp -->
		<param name="rtp-ip" value="$${local_ip_v6}"/>
		<!-- ip address to bind to -->
		<param name="sip-ip" value="$${local_ip_v6}"/>
		<param name="hold-music" value="$${hold_music}"/>
		<!--<param name="enable-100rel" value="false"/>-->
		<!--<param name="disable-srv503" value="true"/>-->
		<param name="apply-inbound-acl" value="domains"/>
		<!--<param name="apply-register-acl" value="domains"/>-->
		<!--<param name="dtmf-type" value="info"/>-->
		<param name="record-template" value="$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
		<!--enable to use presence and mwi -->
		<param name="manage-presence" value="true"/>
		<!-- This setting is for AAL2 bitpacking on G726 -->
		<!-- <param name="bitpacking" value="aal2"/> -->
		<!--max number of open dialogs in proceeding -->
		<!--<param name="max-proceeding" value="1000"/>-->
		<!--session timers for all call to expire after the specified seconds -->
		<!--<param name="session-timeout" value="1800"/>-->
		<!--<param name="multiple-registrations" value="true"/>-->
		<!--set to 'greedy' if you want your codec list to take precedence -->
		<param name="inbound-codec-negotiation" value="generous"/>
		<!-- if you want to send any special bind params of your own -->
		<!--<param name="bind-params" value="transport=udp"/>-->
		<!--<param name="unregister-on-options-fail" value="true"/>-->

		<!-- TLS: disabled by default, set to "true" to enable -->
		<param name="tls" value="$${internal_ssl_enable}"/>
		<!-- additional bind parameters for TLS -->
		<param name="tls-bind-params" value="transport=tls"/>
		<!-- Port to listen on for TLS requests. (5061 will be used if unspecified) -->
		<param name="tls-sip-port" value="$${internal_tls_port}"/>
		<!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
		<param name="tls-cert-dir" value="$${internal_ssl_dir}"/>
		<!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
		<param name="tls-version" value="$${sip_tls_version}"/>

		<!--If you don't want to pass through timestampes from 1 RTP call to another (on a per call basis with rtp_rewrite_timestamps chanvar)-->
		<!--<param name="rtp-rewrite-timestamps" value="true"/>-->
		<!--<param name="pass-rfc2833" value="true"/>-->
		<!--If you have ODBC support and a working dsn you can use it instead of SQLite-->
		<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->

		<!--Uncomment to set all inbound calls to no media mode-->
		<!--<param name="inbound-bypass-media" value="true"/>-->

		<!--Uncomment to set all inbound calls to proxy media mode-->
		<!--<param name="inbound-proxy-media" value="true"/>-->

		<!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok-->
		<!--<param name="inbound-late-negotiation" value="true"/>-->

		<!-- this lets anything register -->
		<!--  comment the next line and uncomment one or both of the other 2 lines for call authentication -->
		<!-- <param name="accept-blind-reg" value="true"/> -->

		<!-- accept any authentication without actually checking (not a good feature for most people) -->
		<!-- <param name="accept-blind-auth" value="true"/> -->

		<!-- suppress CNG on this profile or per call with the 'suppress_cng' variable -->
		<!-- <param name="suppress-cng" value="true"/> -->

		<!--TTL for nonce in sip auth-->
		<param name="nonce-ttl" value="60"/>
		<!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec 
		that the originator is using-->
		<!--<param name="disable-transcoding" value="true"/>-->
		<!-- Used for when phones respond to a challenged ACK with method INVITE in the hash -->
		<!--<param name="NDLB-broken-auth-hash" value="true"/>-->
		<!-- add a ;received="<ip>:<port>" to the contact when replying to register for nat handling -->
		<!--<param name="NDLB-received-in-nat-reg-contact" value="true"/>-->
		<param name="auth-calls" value="$${internal_auth_calls}"/>
		<!-- on authed calls, authenticate *all* the packets not just invite -->
		<param name="auth-all-packets" value="false"/>
		<!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> -->
		<!-- <param name="ext-sip-ip" value="$${external_sip_ip}"/> -->
		<!-- rtp inactivity timeout -->
		<param name="rtp-timeout-sec" value="300"/>
		<param name="rtp-hold-timeout-sec" value="1800"/>
		<!-- VAD choose one (out is a good choice); -->
		<!-- <param name="vad" value="in"/> -->
		<!-- <param name="vad" value="out"/> -->
		<!-- <param name="vad" value="both"/> -->
		<!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
		<!--
		These are enabled to make the default config work better out of the box.
		If you need more than ONE domain you'll need to not use these options.

		-->
		<!--all inbound reg will look in this domain for the users -->
		<param name="force-register-domain" value="$${domain}"/>
		<!--all inbound reg will stored in the db using this domain -->
		<param name="force-register-db-domain" value="$${domain}"/>
		<!-- disable register and transfer which may be undesirable in a public switch -->
		<!--<param name="disable-transfer" value="true"/>-->
		<!--<param name="disable-register" value="true"/>-->
		<!--<param name="enable-3pcc" value="true"/>-->
		<!-- use stun when specified (default is true) -->
		<!--<param name="stun-enabled" value="true"/>-->
		<!-- use stun when specified (default is true) -->
		<!-- set to true to have the profile determine stun is not useful and turn it off globally-->
		<!--<param name="stun-auto-disable" value="true"/>-->

		<!-- the following can be used as workaround with bogus SRV/NAPTR records -->
		<!--<param name="disable-srv" value="false" />-->
		<!--<param name="disable-naptr" value="false" />-->

	</settings>
</profile>
