diff --git a/src/mod/endpoints/mod_sofia/conf/sip_profiles/external.xml b/src/mod/endpoints/mod_sofia/conf/sip_profiles/external.xml
deleted file mode 100644
index a709cbb94c..0000000000
--- a/src/mod/endpoints/mod_sofia/conf/sip_profiles/external.xml
+++ /dev/null
@@ -1,95 +0,0 @@
-<profile name="external">
-  <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --> 
-  <!-- This profile is only for outbound registrations to providers -->
-  <gateways>
-    <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
-  </gateways>
-
-  <aliases>
-    <!-- 
-    <alias name="outbound"/>
-    <alias name="nat"/>
-    -->
-  </aliases>
-
-  <domains>
-    <domain name="all" alias="false" parse="true"/>
-  </domains>
-
-  <settings>
-    <param name="debug" value="0"/>
-	<!-- If you want FreeSWITCH to shutdown if this profile fails to load, uncomment the next line. -->
-	<!-- <param name="shutdown-on-fail" value="true"/> -->
-    <param name="sip-trace" value="no"/>
-    <param name="sip-capture" value="no"/>        
-    <param name="rfc2833-pt" value="101"/>
-    <!-- RFC 5626 : Send reg-id and sip.instance -->
-    <!--<param name="enable-rfc-5626" value="true"/> -->
-    <param name="sip-port" value="$${external_sip_port}"/>
-    <param name="dialplan" value="XML"/>
-    <param name="context" value="public"/>
-    <param name="dtmf-duration" value="2000"/>
-    <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
-    <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
-    <param name="hold-music" value="$${hold_music}"/>
-    <param name="rtp-timer-name" value="soft"/>
-    <!--<param name="enable-100rel" value="true"/>-->
-    <!--<param name="disable-srv503" value="true"/>-->                
-    <!-- This could be set to "passive" -->
-    <param name="local-network-acl" value="localnet.auto"/>
-    <param name="manage-presence" value="false"/>
-
-    <!-- used to share presence info across sofia profiles 
-	 manage-presence needs to be set to passive on this profile
-	 if you want it to behave as if it were the internal profile 
-	 for presence.
-    -->
-    <!-- Name of the db to use for this profile -->
-    <!--<param name="dbname" value="share_presence"/>-->
-    <!--<param name="presence-hosts" value="$${domain}"/>-->
-    <!--<param name="force-register-domain" value="$${domain}"/>-->
-    <!--all inbound reg will stored in the db using this domain -->
-    <!--<param name="force-register-db-domain" value="$${domain}"/>-->
-    <!-- ************************************************* -->
-
-    <!--<param name="aggressive-nat-detection" value="true"/>-->
-    <param name="inbound-codec-negotiation" value="generous"/>
-    <param name="nonce-ttl" value="60"/>
-    <param name="auth-calls" value="false"/>
-    <!--
-	DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS!
-    -->
-    <param name="rtp-ip" value="$${local_ip_v4}"/>
-    <param name="sip-ip" value="$${local_ip_v4}"/>
-    <param name="ext-rtp-ip" value="auto-nat"/>
-    <param name="ext-sip-ip" value="auto-nat"/>
-    <param name="rtp-timeout-sec" value="300"/>
-    <param name="rtp-hold-timeout-sec" value="1800"/>
-    <!--<param name="enable-3pcc" value="true"/>-->
-
-    <!-- TLS: disabled by default, set to "true" to enable -->
-    <param name="tls" value="$${external_ssl_enable}"/>
-    <!-- Set to true to not bind on the normal sip-port but only on the TLS port -->
-    <param name="tls-only" value="false"/>
-    <!-- additional bind parameters for TLS -->
-    <param name="tls-bind-params" value="transport=tls"/>
-    <!-- Port to listen on for TLS requests. (5081 will be used if unspecified) -->
-    <param name="tls-sip-port" value="$${external_tls_port}"/>
-    <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
-    <param name="tls-cert-dir" value="$${external_ssl_dir}"/>
-    <!-- Optionally set the passphrase password used by openSSL to encrypt/decrypt TLS private key files -->
-    <param name="tls-passphrase" value=""/>
-    <!-- Verify the date on TLS certificates -->
-    <param name="tls-verify-date" value="true"/>
-    <!-- TLS verify policy, when registering/inviting gateways with other servers (outbound) or handling inbound registration/invite requests how should we verify their certificate -->
-    <!-- set to 'in' to only verify incoming connections, 'out' to only verify outgoing connections, 'all' to verify all connections, also 'in_subjects', 'out_subjects' and 'all_subjects' for subject validation. Multiple policies can be split with a '|' pipe -->
-    <param name="tls-verify-policy" value="none"/>
-    <!-- Certificate max verify depth to use for validating peer TLS certificates when the verify policy is not none -->
-    <param name="tls-verify-depth" value="2"/>
-    <!-- If the tls-verify-policy is set to subjects_all or subjects_in this sets which subjects are allowed, multiple subjects can be split with a '|' pipe -->
-    <param name="tls-verify-in-subjects" value=""/>
-    <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
-    <param name="tls-version" value="$${sip_tls_version}"/>
-
-  </settings>
-</profile>
diff --git a/src/mod/endpoints/mod_sofia/conf/sip_profiles/external/example.xml b/src/mod/endpoints/mod_sofia/conf/sip_profiles/external/example.xml
deleted file mode 100644
index 70668a9ce5..0000000000
--- a/src/mod/endpoints/mod_sofia/conf/sip_profiles/external/example.xml
+++ /dev/null
@@ -1,38 +0,0 @@
-<include>
-  <!--<gateway name="asterlink.com">-->
-  <!--/// account username *required* ///-->
-  <!--<param name="username" value="cluecon"/>-->
-  <!--/// auth realm: *optional* same as gateway name, if blank ///-->
-  <!--<param name="realm" value="asterlink.com"/>-->
-  <!--/// username to use in from: *optional* same as  username, if blank ///-->
-  <!--<param name="from-user" value="cluecon"/>-->
-  <!--/// domain to use in from: *optional* same as  realm, if blank ///-->
-  <!--<param name="from-domain" value="asterlink.com"/>-->
-  <!--/// account password *required* ///-->
-  <!--<param name="password" value="2007"/>--> 
-  <!--/// extension for inbound calls: *optional* same as username, if blank ///-->
-  <!--<param name="extension" value="cluecon"/>-->
-  <!--/// proxy host: *optional* same as realm, if blank ///-->
-  <!--<param name="proxy" value="asterlink.com"/>-->
-  <!--/// send register to this proxy: *optional* same as proxy, if blank ///-->
-  <!--<param name="register-proxy" value="mysbc.com"/>-->
-  <!--/// expire in seconds: *optional* 3600, if blank ///-->
-  <!--<param name="expire-seconds" value="60"/>-->
-  <!--/// do not register ///-->
-  <!--<param name="register" value="false"/>-->
-  <!-- which transport to use for register -->
-  <!--<param name="register-transport" value="udp"/>-->
-  <!--How many seconds before a retry when a failure or timeout occurs -->
-  <!--<param name="retry-seconds" value="30"/>-->
-  <!--Use the callerid of an inbound call in the from field on outbound calls via this gateway -->
-  <!--<param name="caller-id-in-from" value="false"/>-->
-  <!--extra sip params to send in the contact-->
-  <!--<param name="contact-params" value="tport=tcp"/>-->
-  <!--send an options ping every x seconds, failure will unregister and/or mark it down-->
-  <!--<param name="ping" value="25"/>-->
-  <!--</gateway>-->
-  <!--rfc5626 : Abilitazione rfc5626 ///-->
-  <!--<param name="rfc-5626" value="true"/>-->
-  <!--rfc5626 : extra sip params to send in the contact-->
-  <!--<param name="reg-id" value="1"/>-->
-</include>
diff --git a/src/mod/endpoints/mod_sofia/conf/sip_profiles/internal-ipv6.xml b/src/mod/endpoints/mod_sofia/conf/sip_profiles/internal-ipv6.xml
deleted file mode 100644
index ee27a241a0..0000000000
--- a/src/mod/endpoints/mod_sofia/conf/sip_profiles/internal-ipv6.xml
+++ /dev/null
@@ -1,130 +0,0 @@
-<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="$${domain}"/>
-    <!-- disable register and transfer which may be undesirable in a public switch -->
-    <!--<param name="disable-transfer" value="true"/>-->
-    <!--<param name="disable-register" value="true"/>-->
-    <!--<param name="enable-3pcc" value="true"/>-->
-    <!-- use stun when specified (default is true) -->
-    <!--<param name="stun-enabled" value="true"/>-->
-    <!-- use stun when specified (default is true) -->
-    <!-- set to true to have the profile determine stun is not useful and turn it off globally-->
-    <!--<param name="stun-auto-disable" value="true"/>-->
-
-	 <!-- the following can be used as workaround with bogus SRV/NAPTR records -->
-	 <!--<param name="disable-srv" value="false" />-->
-	 <!--<param name="disable-naptr" value="false" />-->
-
-  </settings>
-</profile>
-
diff --git a/src/mod/endpoints/mod_sofia/conf/sip_profiles/internal.xml b/src/mod/endpoints/mod_sofia/conf/sip_profiles/internal.xml
deleted file mode 100644
index b8390c7934..0000000000
--- a/src/mod/endpoints/mod_sofia/conf/sip_profiles/internal.xml
+++ /dev/null
@@ -1,385 +0,0 @@
-<profile name="internal">
-  <!--
-      This is a sofia sip profile/user agent.  This will service exactly one ip and port.
-      In FreeSWITCH you can run multiple sip user agents on their own ip and port.
-      
-      When you hear someone say "sofia profile" this is what they are talking about.
-  -->
-  
-  <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
-  <!--aliases are other names that will work as a valid profile name for this profile-->
-  <aliases>
-    <!--
-    <alias name="default"/>
-    -->
-  </aliases>
-  <!-- Outbound Registrations -->
-  <gateways>
-    <X-PRE-PROCESS cmd="include" data="internal/*.xml"/>
-  </gateways>
-  
-  <domains>
-    <!-- indicator to parse the directory for domains with parse="true" to get gateways-->
-    <!--<domain name="$${domain}" parse="true"/>-->
-    <!-- indicator to parse the directory for domains with parse="true" to get gateways and alias every domain to this profile -->
-    <!--<domain name="all" alias="true" parse="true"/>-->
-    <domain name="all" alias="true" parse="false"/> 
-  </domains>
-  
-  <settings>
-    <!--
-	When calls are in no media this will bring them back to media
-	when you press the hold button.
-    -->
-    <!--<param name="media-option" value="resume-media-on-hold"/> -->
-    <!--
-	This will allow a call after an attended transfer go back to 
-	 bypass media after an attended transfer.
-    -->
-    <!--<param name="media-option" value="bypass-media-after-att-xfer"/>-->
-    <!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
-    <param name="debug" value="0"/>
-	<!-- If you want FreeSWITCH to shutdown if this profile fails to load, uncomment the next line. -->
-	<!-- <param name="shutdown-on-fail" value="true"/> -->
-    <param name="sip-trace" value="no"/>
-    <param name="sip-capture" value="no"/>
-
-    <!-- Use presence_map.conf.xml to convert extension regex to presence protos for routing -->
-    <!-- <param name="presence-proto-lookup" value="true"/> -->
-    
-
-    <!-- Don't be picky about negotiated DTMF just always offer 2833 and accept both 2833 and INFO -->
-    <!--<param name="liberal-dtmf" value="true"/>-->
-
-
-    <!-- 
-        Sometimes, in extremely rare edge cases, the Sofia SIP stack may stop
-        responding. These options allow you to enable and control a watchdog
-        on the Sofia SIP stack so that if it stops responding for the
-        specified number of milliseconds, it will cause FreeSWITCH to crash
-        immediately. This is useful if you run in an HA environment and
-        need to ensure automated recovery from such a condition. Note that if
-        your server is idle a lot, the watchdog may fire due to not receiving
-        any SIP messages. Thus, if you expect your system to be idle, you
-        should leave the watchdog disabled. It can be toggled on and off
-        through the FreeSWITCH CLI either on an individual profile basis or
-        globally for all profiles. So, if you run in an HA environment with a
-        master and slave, you should use the CLI to make sure the watchdog is
-        only enabled on the master. 
-        If such crash occurs, FreeSWITCH will dump core if allowed. The
-        stacktrace will include function watchdog_triggered_abort().
-    -->
-    <param name="watchdog-enabled" value="no"/>
-    <param name="watchdog-step-timeout" value="30000"/>
-    <param name="watchdog-event-timeout" value="30000"/>
-
-    <param name="log-auth-failures" value="false"/>
-    <param name="forward-unsolicited-mwi-notify" value="false"/>
-
-    <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="$${global_codec_prefs}"/>
-    <param name="rtp-timer-name" value="soft"/>
-    <!-- ip address to use for rtp, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
-    <param name="rtp-ip" value="$${local_ip_v4}"/>
-    <!-- ip address to bind to, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
-    <param name="sip-ip" value="$${local_ip_v4}"/>
-    <param name="hold-music" value="$${hold_music}"/>
-    <param name="apply-nat-acl" value="nat.auto"/>
-
-
-    <!-- (default true) set to false if you do not wish to have called party info in 1XX responses -->
-    <!-- <param name="cid-in-1xx" value="false"/> -->
-
-    <!-- extended info parsing -->
-    <!-- <param name="extended-info-parsing" value="true"/> -->
-
-    <!--<param name="aggressive-nat-detection" value="true"/>-->
-    <!--
-	There are known issues (asserts and segfaults) when 100rel is enabled.
-	It is not recommended to enable 100rel at this time.
-    -->
-    <!--<param name="enable-100rel" value="true"/>-->
-
-    <!-- uncomment if you don't wish to try a next SRV destination on 503 response -->
-    <!-- RFC3263 Section 4.3 -->	    
-    <!--<param name="disable-srv503" value="true"/>-->    
-    
-    <!-- Enable Compact SIP headers. -->
-    <!--<param name="enable-compact-headers" value="true"/>-->
-    <!--
-	enable/disable session timers 
-    -->
-    <!--<param name="enable-timer" value="false"/>-->
-    <!--<param name="minimum-session-expires" value="120"/>-->
-    <param name="apply-inbound-acl" value="domains"/>
-    <!--
-	This defines your local network, by default we detect your local network
-	and create this localnet.auto ACL for this.
-    -->
-    <param name="local-network-acl" value="localnet.auto"/>
-    <!--<param name="apply-register-acl" value="domains"/>-->
-    <!--<param name="dtmf-type" value="info"/>-->
-
-
-    <!-- 'true' means every time 'first-only' means on the first register -->
-    <!--<param name="send-message-query-on-register" value="true"/>-->
-    
-    <!-- 'true' means every time 'first-only' means on the first register -->
-    <!--<param name="send-presence-on-register" value="first-only"/> -->
-      
-
-    <!-- Caller-ID type (choose one, can be overridden by inbound call type and/or sip_cid_type channel variable -->
-    <!-- Remote-Party-ID header -->
-    <!--<param name="caller-id-type" value="rpid"/>-->
-
-    <!-- P-*-Identity family of headers -->
-    <!--<param name="caller-id-type" value="pid"/>-->
-
-    <!-- neither one -->
-    <!--<param name="caller-id-type" value="none"/>-->
-
-
-
-    <param name="record-path" value="$${recordings_dir}"/>
-    <param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
-    <!--enable to use presence -->
-    <param name="manage-presence" value="true"/>
-    <!-- send a presence probe on each register to query devices to send presence instead of sending presence with less info -->
-    <!--<param name="presence-probe-on-register" value="true"/>-->
-    <!--<param name="manage-shared-appearance" value="true"/>-->
-    <!-- used to share presence info across sofia profiles -->
-    <!-- Name of the db to use for this profile -->
-    <!--<param name="dbname" value="share_presence"/>-->
-    <param name="presence-hosts" value="$${domain},$${local_ip_v4}"/>
-    <param name="presence-privacy" value="$${presence_privacy}"/>
-    <!-- ************************************************* -->
-    
-    <!-- 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"/>-->
-    <!-- Can be 'true' or 'contact' -->
-    <!--<param name="multiple-registrations" value="contact"/>-->
-    <!--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="$${internal_ssl_enable}"/>
-    <!-- Set to true to not bind on the normal sip-port but only on the TLS port -->
-    <param name="tls-only" value="false"/>
-    <!-- additional bind parameters for TLS -->
-    <param name="tls-bind-params" value="transport=tls"/>
-    <!-- Port to listen on for TLS requests. (5061 will be used if unspecified) -->
-    <param name="tls-sip-port" value="$${internal_tls_port}"/>
-    <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
-    <param name="tls-cert-dir" value="$${internal_ssl_dir}"/>
-    <!-- Optionally set the passphrase password used by openSSL to encrypt/decrypt TLS private key files -->
-    <param name="tls-passphrase" value=""/>
-    <!-- Verify the date on TLS certificates -->
-    <param name="tls-verify-date" value="true"/>
-    <!-- TLS verify policy, when registering/inviting gateways with other servers (outbound) or handling inbound registration/invite requests how should we verify their certificate -->
-    <!-- set to 'in' to only verify incoming connections, 'out' to only verify outgoing connections, 'all' to verify all connections, also 'in_subjects', 'out_subjects' and 'all_subjects' for subject validation. Multiple policies can be split with a '|' pipe -->
-    <param name="tls-verify-policy" value="none"/>
-    <!-- Certificate max verify depth to use for validating peer TLS certificates when the verify policy is not none -->
-    <param name="tls-verify-depth" value="2"/>
-    <!-- If the tls-verify-policy is set to subjects_all or subjects_in this sets which subjects are allowed, multiple subjects can be split with a '|' pipe -->
-    <param name="tls-verify-in-subjects" value=""/>
-    <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
-    <param name="tls-version" value="$${sip_tls_version}"/>
-
-    <!-- turn on auto-flush during bridge (skip timer sleep when the socket already has data) 
-	 (reduces delay on latent connections default true, must be disabled explicitly)-->
-    <!--<param name="rtp-autoflush-during-bridge" value="false"/>-->
-    
-    <!--If you don't want to pass through timestamps 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"/>-->
-    <!-- Handle 302 Redirect in the dialplan -->
-    <!--<param name="manual-redirect" value="true"/> -->
-    <!-- Disable Transfer -->
-    <!--<param name="disable-transfer" value="true"/> -->
-    <!-- Disable Register -->
-    <!--<param name="disable-register" 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="$${internal_auth_calls}"/>
-    <!-- Force the user and auth-user to match. -->
-    <param name="inbound-reg-force-matching-username" value="true"/>
-    <!-- on authed calls, authenticate *all* the packets not just invite -->
-    <param name="auth-all-packets" value="false"/>
-    
-    <!-- external_sip_ip
-      Used as the public IP address for SDP.
-      Can be an one of:
-           ip address            - "12.34.56.78"
-           a stun server lookup  - "stun:stun.server.com"
-           a DNS name            - "host:host.server.com"
-           auto                  - Use guessed ip.
-           auto-nat              - Use ip learned from NAT-PMP or UPNP
-       -->
-    <param name="ext-rtp-ip" value="auto-nat"/>
-    <param name="ext-sip-ip" value="auto-nat"/>
-
-    <!-- rtp inactivity timeout -->
-    <param name="rtp-timeout-sec" value="300"/>
-    <param name="rtp-hold-timeout-sec" value="1800"/>
-    <!-- VAD choose one (out is a good choice); -->
-    <!-- <param name="vad" value="in"/> -->
-    <!-- <param name="vad" value="out"/> -->
-    <!-- <param name="vad" value="both"/> -->
-    <!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
-    <!--
-	These are enabled to make the default config work better out of the box.
-	If you need more than ONE domain you'll need to not use these options.
-
-    -->
-    <!--all inbound reg will look in this domain for the users -->
-    <param name="force-register-domain" value="$${domain}"/>
-    <!--force the domain in subscriptions to this value -->
-    <param name="force-subscription-domain" value="$${domain}"/>
-    <!--all inbound reg will stored in the db using this domain -->
-    <param name="force-register-db-domain" value="$${domain}"/>
-
-    <!--<param name="delete-subs-on-register" value="false"/>-->
-
-    <!-- enable rtcp on every channel also can be done per leg basis with rtcp_audio_interval_msec variable set to passthru to pass it across a call-->
-    <!--<param name="rtcp-audio-interval-msec" value="5000"/>-->
-    <!--<param name="rtcp-video-interval-msec" value="5000"/>-->
-
-    <!--force suscription expires to a lower value than requested-->
-    <!--<param name="force-subscription-expires" value="60"/>-->
-    <!-- disable register and transfer which may be undesirable in a public switch -->
-    <!--<param name="disable-transfer" value="true"/>-->
-    <!--<param name="disable-register" value="true"/>-->
-
-    <!-- 
-	 enable-3pcc can be set to either 'true' or 'proxy', true accepts the call
-	 right away, proxy waits until the call has been answered then sends accepts
-    -->
-    <!--<param name="enable-3pcc" value="true"/>-->
-	
-    <!-- use at your own risk or if you know what this does.-->
-    <!--<param name="NDLB-force-rport" value="true"/>-->
-    <!--
-	Choose the realm challenge key. Default is auto_to if not set. 
-	
-	auto_from  - uses the from field as the value for the sip realm.
-	auto_to    - uses the to field as the value for the sip realm.
-	<anyvalue> - you can input any value to use for the sip realm.
-
-	If you want URL dialing to work you'll want to set this to auto_from.
-	
-	If you use any other value besides auto_to or auto_from you'll loose
-	the ability to do multiple domains.
-	
-	Note: comment out to restore the behavior before 2008-09-29
-
-    -->
-    <param name="challenge-realm" value="auto_from"/>
-    <!--<param name="disable-rtp-auto-adjust" value="true"/>-->
-    <!-- on inbound calls make the uuid of the session equal to the sip call id of that call -->
-    <!--<param name="inbound-use-callid-as-uuid" value="true"/>-->
-    <!-- on outbound calls set the callid to match the uuid of the session -->
-    <!--<param name="outbound-use-uuid-as-callid" value="true"/>-->
-    <!-- set to false disable this feature -->
-    <!--<param name="rtp-autofix-timing" value="false"/>-->
-
-    <!-- set this param to false if your gateway for some reason hates X- headers that it is supposed to ignore-->
-    <!--<param name="pass-callee-id" value="false"/>-->
-
-    <!-- clear clears them all or supply the name to add or the name prefixed with ~ to remove
-	 valid values:
-
-	 clear
-	 CISCO_SKIP_MARK_BIT_2833
-	 SONUS_SEND_INVALID_TIMESTAMP_2833
-
-    -->
-    <!--<param name="auto-rtp-bugs" data="clear"/>-->
-
-	 <!-- the following can be used as workaround with bogus SRV/NAPTR records -->
-	 <!--<param name="disable-srv" value="false" />-->
-	 <!--<param name="disable-naptr" value="false" />-->
-
-	<!-- The following can be used to fine-tune timers within sofia's transport layer 
-		 Those settings are for advanced users and can safely be left as-is -->
-		
-	<!-- Initial retransmission interval (in milliseconds).
-		Set the T1 retransmission interval used by the SIP transaction engine. 
-		The T1 is the initial duration used by request retransmission timers A and E (UDP) as well as response retransmission timer G. 	 -->
-	<!-- <param name="timer-T1" value="500" /> -->
-	
-	<!--  Transaction timeout (defaults to T1 * 64).
-		Set the T1x64 timeout value used by the SIP transaction engine.
-		The T1x64 is duration used for timers B, F, H, and J (UDP) by the SIP transaction engine. 
-		The timeout value T1x64 can be adjusted separately from the initial retransmission interval T1. -->
-	<!-- <param name="timer-T1X64" value="32000" /> -->
-	
-	
-	<!-- Maximum retransmission interval (in milliseconds).
-		Set the maximum retransmission interval used by the SIP transaction engine. 
-		The T2 is the maximum duration used for the timers E (UDP) and G by the SIP transaction engine. 
-		Note that the timer A is not capped by T2. Retransmission interval of INVITE requests grows exponentially 
-		until the timer B fires.  -->
-	<!-- <param name="timer-T2" value="4000" /> -->
-		
-	<!--
-		Transaction lifetime (in milliseconds).
-		Set the lifetime for completed transactions used by the SIP transaction engine. 
-		A completed transaction is kept around for the duration of T4 in order to catch late responses. 
-		The T4 is the maximum duration for the messages to stay in the network and the duration of SIP timer K. -->
-	<!-- <param name="timer-T4" value="4000" /> -->
-
-	<!-- Turn on a jitterbuffer for every call -->
-	<!-- <param name="auto-jitterbuffer-msec" value="60"/> -->
-
-
-	<!-- By default mod_sofia will ignore the codecs in the sdp for hold/unhold operations 
-	     Set this to true if you want to actually parse the sdp and re-negotiate the codec during hold/unhold.
-	     It's probably not what you want so stick with the default unless you really need to change this.
-	-->
-	<!--<param name="renegotiate-codec-on-hold" value="true"/>-->
-    
-  </settings>
-</profile>
-
diff --git a/src/mod/endpoints/mod_sofia/conf/sip_profiles/internal/example.xml b/src/mod/endpoints/mod_sofia/conf/sip_profiles/internal/example.xml
deleted file mode 100644
index fc061f96d1..0000000000
--- a/src/mod/endpoints/mod_sofia/conf/sip_profiles/internal/example.xml
+++ /dev/null
@@ -1,37 +0,0 @@
-<include>
-  <!--<gateway name="asterlink.com">-->
-  <!--/// account username *required* ///-->
-  <!--<param name="username" value="cluecon"/>-->
-  <!--/// auth realm: *optional* same as gateway name, if blank ///-->
-  <!--<param name="realm" value="asterlink.com"/>-->
-  <!--/// username to use in from: *optional* same as  username, if blank ///-->
-  <!--<param name="from-user" value="cluecon"/>-->
-  <!--/// domain to use in from: *optional* same as  realm, if blank ///-->
-  <!--<param name="from-domain" value="asterlink.com"/>-->
-  <!--/// account password *required* ///-->
-  <!--<param name="password" value="2007"/>--> 
-  <!--/// extension for inbound calls: *optional* same as username, if blank ///-->
-  <!--<param name="extension" value="cluecon"/>-->
-  <!--/// proxy host: *optional* same as realm, if blank ///-->
-  <!--<param name="proxy" value="asterlink.com"/>-->
-  <!--/// send register to this proxy: *optional* same as proxy, if blank ///-->
-  <!--<param name="register-proxy" value="mysbc.com"/>-->
-  <!--/// expire in seconds: *optional* 3600, if blank ///-->
-  <!--<param name="expire-seconds" value="60"/>-->
-  <!--/// do not register ///-->
-  <!--<param name="register" value="false"/>-->
-  <!-- which transport to use for register -->
-  <!--<param name="register-transport" value="udp"/>-->
-  <!--How many seconds before a retry when a failure or timeout occurs -->
-  <!--<param name="retry-seconds" value="30"/>-->
-  <!--Use the callerid of an inbound call in the from field on outbound calls via this gateway -->
-  <!--<param name="caller-id-in-from" value="false"/>-->
-  <!--extra sip params to send in the contact-->
-  <!--<param name="contact-params" value="tport=tcp"/>-->
-  <!-- Put the extension in the contact -->
-  <!--<param name="extension-in-contact" value="true"/>-->
-  <!--send an options ping every x seconds, failure will unregister and/or mark it down-->
-  <!--<param name="ping" value="25"/>-->
-  <!--<param name="cid-type" value="rpid"/>-->
-  <!--</gateway>-->
-</include>
diff --git a/src/mod/endpoints/mod_sofia/conf/sofia.conf.xml b/src/mod/endpoints/mod_sofia/conf/sofia.conf.xml
new file mode 100644
index 0000000000..2649a24ae8
--- /dev/null
+++ b/src/mod/endpoints/mod_sofia/conf/sofia.conf.xml
@@ -0,0 +1,473 @@
+<configuration name="sofia.conf" description="sofia endpoint">
+  <global_settings>
+    <param name="log-level" value="0"/>
+    <param name="auto-restart" value="false"/>
+    <param name="debug-presence" value="0"/>
+    <!-- <param name="capture-server" value="udp:homer.example.com:5060"/> -->
+  </global_settings>
+  <profiles>
+    <!--
+        This is a sofia sip profile/user agent.  This will service exactly one
+        ip and port.  In FreeSWITCH you can run multiple sip user agents on
+        their own ip and port.
+    -->
+    <profile name="example">
+      <gateways>
+        <gateway name="example-gateway">
+          <!-- account username (required) -->
+          <param name="username" value="cluecon"/>
+          <!-- auth realm (same as gateway name, if blank) -->
+          <param name="realm" value="example.com"/>
+          <!-- username to use in from (same as username, if blank) -->
+          <param name="from-user" value="cluecon"/>
+          <!-- domain to use in from (same as realm, if blank) /// -->
+          <param name="from-domain" value="example.com"/>
+          <!-- account password (required) -->
+          <param name="password" value="xxxx"/>
+          <!-- extension for inbound calls (same as username, if blank) -->
+          <param name="extension" value="cluecon"/>
+          <!-- proxy host (same as realm, if blank) -->
+          <param name="proxy" value="example.com"/>
+          <!-- send register to this proxy (same as proxy, if blank) -->
+          <param name="register-proxy" value="example.com"/>
+          <!-- expire in seconds (3600, if blank) -->
+          <param name="expire-seconds" value="600"/>
+          <!-- do not register -->
+          <param name="register" value="false"/>
+          <!-- which transport to use for register -->
+          <param name="register-transport" value="tcp"/>
+          <!-- how many seconds before a retry when a failure or timeout occurs
+               -->
+          <param name="retry-seconds" value="30"/>
+          <!-- use the callerid of an inbound call in the from field on outbound
+               calls via this gateway -->
+          <param name="caller-id-in-from" value="false"/>
+          <!-- extra sip params to send in the contact -->
+          <param name="contact-params" value="tport=tcp"/>
+          <!-- put the extension in the contact -->
+          <param name="extension-in-contact" value="true"/>
+          <!-- send an options ping every x seconds, failure will unregister
+               and/or mark it down -->
+          <param name="ping" value="25"/>
+          <!-- callerid header mechanism -->
+          <param name="cid-type" value="rpid"/>
+        </gateway>
+      </gateways>
+      <aliases>
+        <!-- aliases are other names that will work as a valid profile name for
+             this profile -->
+        <alias name="default"/>
+      </aliases>
+      <domains>
+        <!-- indicator to parse the directory for domains with parse="true" to
+             get gateways -->
+        <!-- <domain name="$${domain}" parse="true"/> -->
+        <!-- indicator to parse the directory for domains with parse="true" to
+             get gateways and alias every domain to this profile -->
+        <!-- <domain name="all" alias="true" parse="true"/> -->
+        <domain name="all" alias="true" parse="false"/>
+      </domains>
+      <settings>
+        <!-- When calls are in no media this will bring them back to media when
+             you press the hold button. -->
+        <!-- <param name="media-option" value="resume-media-on-hold"/> -->
+        <!-- This will allow a call after an attended transfer go back to bypass
+             media after an attended transfer. -->
+        <!-- <param name="media-option" value="bypass-media-after-att-xfer"/> -->
+        <!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
+        <param name="debug" value="0"/>
+        <!-- If you want FreeSWITCH to shutdown if this profile fails to load,
+             uncomment the next line. -->
+        <!-- <param name="shutdown-on-fail" value="true"/> -->
+        <param name="sip-trace" value="no"/>
+        <param name="sip-capture" value="no"/>
+
+        <!-- Use presence_map.conf.xml to convert extension regex to presence
+             protos for routing -->
+        <!-- <param name="presence-proto-lookup" value="true"/> -->
+
+
+        <!-- Don't be picky about negotiated DTMF just always offer 2833 and
+             accept both 2833 and INFO -->
+        <!-- <param name="liberal-dtmf" value="true"/> -->
+
+        <!--
+             Sometimes, in extremely rare edge cases, the Sofia SIP stack may
+             stop responding. These options allow you to enable and control a
+             watchdog on the Sofia SIP stack so that if it stops responding for
+             the specified number of milliseconds, it will cause FreeSWITCH to
+             crash immediately. This is useful if you run in an HA environment
+             and need to ensure automated recovery from such a condition. Note
+             that if your server is idle a lot, the watchdog may fire due to not
+             receiving any SIP messages. Thus, if you expect your system to be
+             idle, you should leave the watchdog disabled. It can be toggled on
+             and off through the FreeSWITCH CLI either on an individual profile
+             basis or globally for all profiles. So, if you run in an HA
+             environment with a master and slave, you should use the CLI to make
+             sure the watchdog is only enabled on the master.
+
+             If such crash occurs, FreeSWITCH will dump core if allowed. The
+             stacktrace will include function watchdog_triggered_abort().
+        -->
+        <param name="watchdog-enabled" value="no"/>
+        <param name="watchdog-step-timeout" value="30000"/>
+        <param name="watchdog-event-timeout" value="30000"/>
+
+        <param name="log-auth-failures" value="false"/>
+        <param name="forward-unsolicited-mwi-notify" value="false"/>
+
+        <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="rtp-timer-name" value="soft"/>
+        <!-- ip address to use for rtp, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
+        <param name="rtp-ip" value="$${local_ip_v4}"/>
+        <!-- ip address to bind to, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
+        <param name="sip-ip" value="$${local_ip_v4}"/>
+        <param name="hold-music" value="$${hold_music}"/>
+        <param name="apply-nat-acl" value="nat.auto"/>
+
+        <!-- (default true) set to false if you do not wish to have called party
+             info in 1XX responses -->
+        <!-- <param name="cid-in-1xx" value="false"/> -->
+
+        <!-- extended info parsing -->
+        <!-- <param name="extended-info-parsing" value="true"/> -->
+
+        <!-- <param name="aggressive-nat-detection" value="true"/> -->
+        <!-- There are known issues (asserts and segfaults) when 100rel is
+             enabled.  It is not recommended to enable 100rel at this time. -->
+        <!-- <param name="enable-100rel" value="true"/> -->
+
+        <!-- uncomment if you don't wish to try a next SRV destination on 503
+             response -->
+        <!-- RFC3263 Section 4.3 -->
+        <!-- <param name="disable-srv503" value="true"/> -->
+
+        <!-- Enable Compact SIP headers. -->
+        <!-- <param name="enable-compact-headers" value="true"/> -->
+        <!-- enable/disable session timers -->
+        <!-- <param name="enable-timer" value="false"/> -->
+        <!-- <param name="minimum-session-expires" value="120"/> -->
+        <param name="apply-inbound-acl" value="domains"/>
+        <!-- This defines your local network, by default we detect your local
+             network and create this localnet.auto ACL for this. -->
+        <param name="local-network-acl" value="localnet.auto"/>
+        <!-- <param name="apply-register-acl" value="domains"/> -->
+        <!-- <param name="dtmf-type" value="info"/> -->
+
+        <!-- 'true' means every time 'first-only' means on the first register -->
+        <!-- <param name="send-message-query-on-register" value="true"/> -->
+
+        <!-- 'true' means every time 'first-only' means on the first register -->
+        <!-- <param name="send-presence-on-register" value="first-only"/> -->
+
+        <!-- Caller-ID type (choose one, can be overridden by inbound call type
+             and/or sip_cid_type channel variable -->
+        <!-- Remote-Party-ID header -->
+        <!-- <param name="caller-id-type" value="rpid"/> -->
+
+        <!-- P-*-Identity family of headers -->
+        <!-- <param name="caller-id-type" value="pid"/> -->
+
+        <!-- neither one -->
+        <!-- <param name="caller-id-type" value="none"/> -->
+
+        <param name="record-path" value="$${recordings_dir}"/>
+        <param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
+        <!-- enable to use presence -->
+        <param name="manage-presence" value="true"/>
+        <!-- send a presence probe on each register to query devices to send
+             presence instead of sending presence with less info -->
+        <!-- <param name="presence-probe-on-register" value="true"/> -->
+        <!-- <param name="manage-shared-appearance" value="true"/> -->
+        <!-- used to share presence info across sofia profiles -->
+        <!-- Name of the db to use for this profile -->
+        <!-- <param name="dbname" value="share_presence"/> -->
+        <param name="presence-hosts" value="$${domain},$${local_ip_v4}"/>
+        <param name="presence-privacy" value="$${presence_privacy}"/>
+
+        <!-- This setting is for AAL2 bitpacking on G726 -->
+        <!-- <param name="bitpacking" value="aal2"/> -->
+        <!-- max number of open dialogs in proceeding -->
+        <!-- <param name="max-proceeding" value="1000"/> -->
+        <!-- session timers for all call to expire after the specified seconds -->
+        <!-- <param name="session-timeout" value="1800"/> -->
+        <!-- Can be 'true' or 'contact' -->
+        <!-- <param name="multiple-registrations" value="contact"/> -->
+        <!-- set to 'greedy' if you want your codec list to take precedence -->
+        <param name="inbound-codec-negotiation" value="generous"/>
+        <!-- if you want to send any special bind params of your own -->
+        <!-- <param name="bind-params" value="transport=udp"/> -->
+        <!-- <param name="unregister-on-options-fail" value="true"/> -->
+
+        <!-- TLS: disabled by default, set to "true" to enable -->
+        <param name="tls" value="$${internal_ssl_enable}"/>
+        <!-- Set to true to not bind on the normal sip-port but only on the TLS
+             port -->
+        <param name="tls-only" value="false"/>
+        <!-- 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}"/>
+        <!-- Optionally set the passphrase password used by openSSL to
+             encrypt/decrypt TLS private key files -->
+        <param name="tls-passphrase" value=""/>
+        <!-- Verify the date on TLS certificates -->
+        <param name="tls-verify-date" value="true"/>
+        <!-- TLS verify policy, when registering/inviting gateways with other
+             servers (outbound) or handling inbound registration/invite requests
+             how should we verify their certificate -->
+        <!-- set to 'in' to only verify incoming connections, 'out' to only
+             verify outgoing connections, 'all' to verify all connections, also
+             'in_subjects', 'out_subjects' and 'all_subjects' for subject
+             validation. Multiple policies can be split with a '|' pipe -->
+        <param name="tls-verify-policy" value="none"/>
+        <!-- Certificate max verify depth to use for validating peer TLS
+             certificates when the verify policy is not none -->
+        <param name="tls-verify-depth" value="2"/>
+        <!-- If the tls-verify-policy is set to subjects_all or subjects_in this
+             sets which subjects are allowed, multiple subjects can be split
+             with a '|' pipe -->
+        <param name="tls-verify-in-subjects" value=""/>
+        <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not
+             work with TLSv1 -->
+        <param name="tls-version" value="$${sip_tls_version}"/>
+
+        <!-- turn on auto-flush during bridge (skip timer sleep when the socket
+             already has data) (reduces delay on latent connections default
+             true, must be disabled explicitly) -->
+        <!-- <param name="rtp-autoflush-during-bridge" value="false"/> -->
+
+        <!-- If you don't want to pass through timestamps 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"/> -->
+        <!-- Handle 302 Redirect in the dialplan -->
+        <!-- <param name="manual-redirect" value="true"/> -->
+        <!-- Disable Transfer -->
+        <!-- <param name="disable-transfer" value="true"/> -->
+        <!-- Disable Register -->
+        <!-- <param name="disable-register" 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}"/>
+        <!-- Force the user and auth-user to match. -->
+        <param name="inbound-reg-force-matching-username" value="true"/>
+        <!-- on authed calls, authenticate *all* the packets not just invite -->
+        <param name="auth-all-packets" value="false"/>
+
+        <!-- external_sip_ip
+             Used as the public IP address for SDP.
+             Can be an one of:
+             ip address            - "12.34.56.78"
+             a stun server lookup  - "stun:stun.server.com"
+             a DNS name            - "host:host.server.com"
+             auto                  - Use guessed ip.
+             auto-nat              - Use ip learned from NAT-PMP or UPNP
+        -->
+        <param name="ext-rtp-ip" value="auto-nat"/>
+        <param name="ext-sip-ip" value="auto-nat"/>
+
+        <!-- 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="$${domain}"/>
+        <!-- force the domain in subscriptions to this value -->
+        <param name="force-subscription-domain" value="$${domain}"/>
+        <!-- all inbound reg will stored in the db using this domain -->
+        <param name="force-register-db-domain" value="$${domain}"/>
+
+        <!-- <param name="delete-subs-on-register" value="false"/> -->
+
+        <!-- enable rtcp on every channel also can be done per leg basis with
+             rtcp_audio_interval_msec variable set to passthru to pass it across
+             a call -->
+        <!-- <param name="rtcp-audio-interval-msec" value="5000"/> -->
+        <!-- <param name="rtcp-video-interval-msec" value="5000"/> -->
+
+        <!-- force suscription expires to a lower value than requested -->
+        <!-- <param name="force-subscription-expires" value="60"/> -->
+        <!-- disable register and transfer which may be undesirable in a public
+             switch -->
+        <!-- <param name="disable-transfer" value="true"/> -->
+        <!-- <param name="disable-register" value="true"/> -->
+
+        <!--
+             enable-3pcc can be set to either 'true' or 'proxy', true accepts
+             the call right away, proxy waits until the call has been answered
+             then sends accepts
+        -->
+        <!-- <param name="enable-3pcc" value="true"/> -->
+
+        <!-- use at your own risk or if you know what this does. -->
+        <!-- <param name="NDLB-force-rport" value="true"/> -->
+        <!--
+            Choose the realm challenge key. Default is auto_to if not set.
+
+            auto_from  - uses the from field as the value for the sip realm.
+            auto_to    - uses the to field as the value for the sip realm.
+            <anyvalue> - you can input any value to use for the sip realm.
+
+            If you want URL dialing to work you'll want to set this to auto_from.
+
+            If you use any other value besides auto_to or auto_from you'll loose
+            the ability to do multiple domains.
+
+            Note: comment out to restore the behavior before 2008-09-29
+
+        -->
+        <param name="challenge-realm" value="auto_from"/>
+        <!-- <param name="disable-rtp-auto-adjust" value="true"/> -->
+        <!-- on inbound calls make the uuid of the session equal to the sip call
+             id of that call -->
+        <!-- <param name="inbound-use-callid-as-uuid" value="true"/> -->
+        <!-- on outbound calls set the callid to match the uuid of the session
+             -->
+        <!-- <param name="outbound-use-uuid-as-callid" value="true"/> -->
+        <!-- set to false disable this feature -->
+        <!-- <param name="rtp-autofix-timing" value="false"/> -->
+
+        <!-- set this param to false if your gateway for some reason hates X-
+             headers that it is supposed to ignore -->
+        <!-- <param name="pass-callee-id" value="false"/> -->
+
+        <!-- clear clears them all or supply the name to add or the name
+             prefixed with ~ to remove valid values:
+
+               clear
+               CISCO_SKIP_MARK_BIT_2833
+               SONUS_SEND_INVALID_TIMESTAMP_2833
+
+        -->
+        <!-- <param name="auto-rtp-bugs" data="clear"/> -->
+
+        <!-- the following can be used as workaround with bogus SRV/NAPTR
+             records -->
+        <!-- <param name="disable-srv" value="false" /> -->
+        <!-- <param name="disable-naptr" value="false" /> -->
+
+        <!-- The following can be used to fine-tune timers within sofia's
+             transport layer Those settings are for advanced users and can
+             safely be left as-is -->
+
+        <!-- Initial retransmission interval (in milliseconds).
+
+             Set the T1 retransmission interval used by the SIP transaction
+             engine.
+
+             The T1 is the initial duration used by request retransmission
+             timers A and E (UDP) as well as response retransmission timer G.
+             -->
+        <!-- <param name="timer-T1" value="500" /> -->
+
+        <!--  Transaction timeout (defaults to T1 * 64).
+
+             Set the T1x64 timeout value used by the SIP transaction engine.
+
+             The T1x64 is duration used for timers B, F, H, and J (UDP) by the
+             SIP transaction engine.
+
+             The timeout value T1x64 can be adjusted separately from the initial
+             retransmission interval T1. -->
+        <!-- <param name="timer-T1X64" value="32000" /> -->
+
+
+        <!-- Maximum retransmission interval (in milliseconds).
+
+             Set the maximum retransmission interval used by the SIP transaction
+             engine.
+
+             The T2 is the maximum duration used for the timers E (UDP) and G by
+             the SIP transaction engine.
+
+             Note that the timer A is not capped by T2. Retransmission interval
+             of INVITE requests grows exponentially until the timer B fires.
+             -->
+        <!-- <param name="timer-T2" value="4000" /> -->
+
+        <!--
+            Transaction lifetime (in milliseconds).
+
+            Set the lifetime for completed transactions used by the SIP
+            transaction engine.
+
+            A completed transaction is kept around for the duration of T4 in
+            order to catch late responses.
+
+            The T4 is the maximum duration for the messages to stay in the
+            network and the duration of SIP timer K. -->
+        <!-- <param name="timer-T4" value="4000" /> -->
+
+        <!-- Turn on a jitterbuffer for every call -->
+        <!-- <param name="auto-jitterbuffer-msec" value="60"/> -->
+
+
+        <!-- By default mod_sofia will ignore the codecs in the sdp for
+             hold/unhold operations Set this to true if you want to actually
+             parse the sdp and re-negotiate the codec during hold/unhold.  It's
+             probably not what you want so stick with the default unless you
+             really need to change this.
+        -->
+        <!-- <param name="renegotiate-codec-on-hold" value="true"/> -->
+      </settings>
+    </profile>
+  </profiles>
+</configuration>