diff --git a/conf/insideout/sip_profiles/external.xml b/conf/insideout/sip_profiles/external.xml
index 6f6045255d..3fcb969f7f 100644
--- a/conf/insideout/sip_profiles/external.xml
+++ b/conf/insideout/sip_profiles/external.xml
@@ -1,4 +1,4 @@
-<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --> 
+<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
 <profile name="external">
   <!-- This profile is only for outbound registrations to providers -->
   <gateways>
@@ -29,10 +29,10 @@
     <!-- This could be set to "passive" -->
     <param name="manage-presence" value="passive"/>
 
-    <!-- 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.
+    <!-- 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="$${domain}"/>
@@ -48,7 +48,7 @@
     <param name="auth-calls" value="false"/>
     <param name="rtp-timeout-sec" value="1800"/>
     <!--
-	DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS!
+        DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS!
     -->
     <param name="rtp-ip" value="$${local_ip_v4}"/>
     <param name="sip-ip" value="$${local_ip_v4}"/>
diff --git a/conf/insideout/sip_profiles/external/example.xml b/conf/insideout/sip_profiles/external/example.xml
index 7ac8db1186..c6b79e8fe3 100644
--- a/conf/insideout/sip_profiles/external/example.xml
+++ b/conf/insideout/sip_profiles/external/example.xml
@@ -9,7 +9,7 @@
   <!--/// domain to use in from: *optional* same as  realm, if blank ///-->
   <!--<param name="from-domain" value="asterlink.com"/>-->
   <!--/// account password *required* ///-->
-  <!--<param name="password" value="2007"/>--> 
+  <!--<param name="password" value="2007"/>-->
   <!--/// extension for inbound calls: *optional* same as username, if blank ///-->
   <!--<param name="extension" value="cluecon"/>-->
   <!--/// proxy host: *optional* same as realm, if blank ///-->
diff --git a/conf/insideout/sip_profiles/internal-ipv6.xml b/conf/insideout/sip_profiles/internal-ipv6.xml
index 856ac4905d..649246ad36 100644
--- a/conf/insideout/sip_profiles/internal-ipv6.xml
+++ b/conf/insideout/sip_profiles/internal-ipv6.xml
@@ -53,36 +53,36 @@
     <param name="tls-cert-dir" value="$${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-->
+    <!--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"/>-->
@@ -128,4 +128,3 @@
 
   </settings>
 </profile>
-
diff --git a/conf/insideout/sip_profiles/internal.xml b/conf/insideout/sip_profiles/internal.xml
index a6010b2945..3ea2ef155c 100644
--- a/conf/insideout/sip_profiles/internal.xml
+++ b/conf/insideout/sip_profiles/internal.xml
@@ -1,7 +1,7 @@
 <!--
     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.
 -->
 
@@ -16,24 +16,24 @@
   <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="$${domain}" parse="true"/>-->
     <!-- indicator to parse the directory for domains with parse="true" to get gateways and alias every domain to this profile -->
     <!--<domain name="all" alias="true" parse="true"/>-->
-    <domain name="all" alias="true" parse="false"/> 
+    <domain name="all" alias="true" parse="false"/>
   </domains>
-  
+
   <settings>
     <!--
-	When calls are in no media this will bring them back to media
-	when you press the hold button.
+        When calls are in no media this will bring them back to media
+        when you press the hold button.
     -->
     <!--<param name="media-option" value="resume-media-on-hold"/> -->
     <!--
-	This will allow a call after an attended transfer go back to 
-	 bypass media after an attended transfer.
+        This will allow a call after an attended transfer go back to
+        bypass media after an attended transfer.
     -->
     <!--<param name="media-option" value="bypass-media-after-att-xfer"/>-->
     <!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
@@ -63,7 +63,7 @@
     <!--<param name="dtmf-type" value="info"/>-->
     <param name="record-template" value="$${base_dir}/recordings/${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
 
-    
+
     <!-- This setting is for AAL2 bitpacking on G726 -->
     <!-- <param name="bitpacking" value="aal2"/> -->
     <!--max number of open dialogs in proceeding -->
@@ -88,36 +88,36 @@
     <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 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-->
+    <!--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"/>-->
@@ -138,7 +138,7 @@
     <!-- <param name="vad" value="out"/> -->
     <!-- <param name="vad" value="both"/> -->
     <!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
- 
+
     <!--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 -->
@@ -158,24 +158,24 @@
     <!--<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 -->
+    <!-- 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.
+        Choose the realm challenge key. Default is auto_to if not set.
 
-	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
+        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"/>
@@ -186,4 +186,3 @@
     <!--<param name="outbound-use-uuid-as-callid" value="true"/>-->
   </settings>
 </profile>
-
diff --git a/conf/insideout/sip_profiles/internal/example.xml b/conf/insideout/sip_profiles/internal/example.xml
index 7ac8db1186..c6b79e8fe3 100644
--- a/conf/insideout/sip_profiles/internal/example.xml
+++ b/conf/insideout/sip_profiles/internal/example.xml
@@ -9,7 +9,7 @@
   <!--/// 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"/>--> 
+  <!--<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 ///-->
diff --git a/conf/sbc/sbc_profiles/external.xml b/conf/sbc/sbc_profiles/external.xml
index d16813960f..9a5686bd91 100644
--- a/conf/sbc/sbc_profiles/external.xml
+++ b/conf/sbc/sbc_profiles/external.xml
@@ -1,4 +1,4 @@
-<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --> 
+<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
 <profile name="external">
   <!-- This profile is only for outbound registrations to providers -->
   <gateways>
@@ -7,7 +7,7 @@
 
   <aliases>
     <alias name="outbound"/>
-    <alias name="nat"/> <!-- for backwards compatibility --> 
+    <alias name="nat"/> <!-- for backwards compatibility -->
   </aliases>
 
   <domains>
@@ -30,10 +30,10 @@
     <!-- This could be set to "passive" -->
     <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.
+    <!-- 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"/>-->
@@ -49,7 +49,7 @@
     <param name="auth-calls" value="false"/>
     <param name="rtp-timeout-sec" value="1800"/>
     <!--
-	DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS!
+        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}"/>
diff --git a/conf/sbc/sbc_profiles/external/example.xml b/conf/sbc/sbc_profiles/external/example.xml
index 7ac8db1186..c6b79e8fe3 100644
--- a/conf/sbc/sbc_profiles/external/example.xml
+++ b/conf/sbc/sbc_profiles/external/example.xml
@@ -9,7 +9,7 @@
   <!--/// 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"/>--> 
+  <!--<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 ///-->
diff --git a/conf/sbc/sbc_profiles/internal-ipv6.xml b/conf/sbc/sbc_profiles/internal-ipv6.xml
index 856ac4905d..649246ad36 100644
--- a/conf/sbc/sbc_profiles/internal-ipv6.xml
+++ b/conf/sbc/sbc_profiles/internal-ipv6.xml
@@ -53,36 +53,36 @@
     <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-->
+    <!--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"/>-->
@@ -128,4 +128,3 @@
 
   </settings>
 </profile>
-
diff --git a/conf/sbc/sbc_profiles/internal.xml b/conf/sbc/sbc_profiles/internal.xml
index d06e3a76cd..7e1ceeb9df 100644
--- a/conf/sbc/sbc_profiles/internal.xml
+++ b/conf/sbc/sbc_profiles/internal.xml
@@ -1,7 +1,7 @@
 <!--
     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.
 -->
 
@@ -15,24 +15,24 @@
   <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="$${domain}" parse="true"/>-->
     <!-- indicator to parse the directory for domains with parse="true" to get gateways and alias every domain to this profile -->
     <!--<domain name="all" alias="true" parse="true"/>-->
-    <domain name="all" alias="true" parse="false"/> 
+    <domain name="all" alias="true" parse="false"/>
   </domains>
-  
+
   <settings>
     <!--
-	When calls are in no media this will bring them back to media
-	when you press the hold button.
+        When calls are in no media this will bring them back to media
+        when you press the hold button.
     -->
     <!--<param name="media-option" value="resume-media-on-hold"/> -->
     <!--
-	This will allow a call after an attended transfer go back to 
-	 bypass media after an attended transfer.
+        This will allow a call after an attended transfer go back to
+        bypass media after an attended transfer.
     -->
     <!--<param name="media-option" value="bypass-media-after-att-xfer"/>-->
     <!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
@@ -69,7 +69,7 @@
     <!--<param name="dbname" value="share_presence"/>-->
     <!--<param name="presence-hosts" value="$${domain}"/>-->
     <!-- ************************************************* -->
-    
+
     <!-- This setting is for AAL2 bitpacking on G726 -->
     <!-- <param name="bitpacking" value="aal2"/> -->
     <!--max number of open dialogs in proceeding -->
@@ -94,36 +94,36 @@
     <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 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-->
+    <!--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"/>-->
@@ -154,24 +154,24 @@
     <!--<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 -->
+    <!-- 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.
+        Choose the realm challenge key. Default is auto_to if not set.
 
-	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
+        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"/>
@@ -182,4 +182,3 @@
     <!--<param name="outbound-use-uuid-as-callid" value="true"/>-->
   </settings>
 </profile>
-
diff --git a/conf/sbc/sbc_profiles/internal/example.xml b/conf/sbc/sbc_profiles/internal/example.xml
index 7ac8db1186..c6b79e8fe3 100644
--- a/conf/sbc/sbc_profiles/internal/example.xml
+++ b/conf/sbc/sbc_profiles/internal/example.xml
@@ -9,7 +9,7 @@
   <!--/// 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"/>--> 
+  <!--<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 ///-->
diff --git a/conf/vanilla/sip_profiles/external.xml b/conf/vanilla/sip_profiles/external.xml
index a709cbb94c..39965e9421 100644
--- a/conf/vanilla/sip_profiles/external.xml
+++ b/conf/vanilla/sip_profiles/external.xml
@@ -1,14 +1,14 @@
 <profile name="external">
-  <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --> 
+  <!-- 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"/>
+    <!--
+        <alias name="outbound"/>
+        <alias name="nat"/>
     -->
   </aliases>
 
@@ -18,10 +18,10 @@
 
   <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"/> -->
+    <!-- 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="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"/> -->
@@ -34,15 +34,15 @@
     <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"/>-->                
+    <!--<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.
+    <!-- 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"/>-->
@@ -57,7 +57,7 @@
     <param name="nonce-ttl" value="60"/>
     <param name="auth-calls" value="false"/>
     <!--
-	DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS!
+        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}"/>
@@ -90,6 +90,5 @@
     <param name="tls-verify-in-subjects" value=""/>
     <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
     <param name="tls-version" value="$${sip_tls_version}"/>
-
   </settings>
 </profile>
diff --git a/conf/vanilla/sip_profiles/external/example.xml b/conf/vanilla/sip_profiles/external/example.xml
index 70668a9ce5..2a4df31aaf 100644
--- a/conf/vanilla/sip_profiles/external/example.xml
+++ b/conf/vanilla/sip_profiles/external/example.xml
@@ -9,7 +9,7 @@
   <!--/// domain to use in from: *optional* same as  realm, if blank ///-->
   <!--<param name="from-domain" value="asterlink.com"/>-->
   <!--/// account password *required* ///-->
-  <!--<param name="password" value="2007"/>--> 
+  <!--<param name="password" value="2007"/>-->
   <!--/// extension for inbound calls: *optional* same as username, if blank ///-->
   <!--<param name="extension" value="cluecon"/>-->
   <!--/// proxy host: *optional* same as realm, if blank ///-->
diff --git a/conf/vanilla/sip_profiles/internal-ipv6.xml b/conf/vanilla/sip_profiles/internal-ipv6.xml
index ee27a241a0..5a97f0cd60 100644
--- a/conf/vanilla/sip_profiles/internal-ipv6.xml
+++ b/conf/vanilla/sip_profiles/internal-ipv6.xml
@@ -54,36 +54,36 @@
     <param name="tls-cert-dir" value="$${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-->
+    <!--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"/>-->
@@ -103,8 +103,8 @@
     <!-- <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.
+        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 -->
@@ -121,10 +121,9 @@
     <!-- 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" />-->
+    <!-- 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/conf/vanilla/sip_profiles/internal.xml b/conf/vanilla/sip_profiles/internal.xml
index b9605a192c..afb12ab94c 100644
--- a/conf/vanilla/sip_profiles/internal.xml
+++ b/conf/vanilla/sip_profiles/internal.xml
@@ -2,30 +2,30 @@
   <!--
       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"/>
+        <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"/> 
+    <domain name="all" alias="true" parse="false"/>
   </domains>
-  
+
   <settings>
 
 
@@ -33,31 +33,31 @@
     <!-- <param name="rtp-digit-delay" value="40"/>-->
 
     <!--
-	When calls are in no media this will bring them back to media
-	when you press the hold button.
+        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.
+        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"/> -->
+    <!-- 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
@@ -70,7 +70,7 @@
         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. 
+        only enabled on the master.
         If such crash occurs, FreeSWITCH will dump core if allowed. The
         stacktrace will include function watchdog_triggered_abort().
     -->
@@ -106,26 +106,26 @@
 
     <!--<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.
+        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"/>-->    
-    
+    <!-- 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 
+        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.
+        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"/>-->
@@ -134,10 +134,10 @@
 
     <!-- '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 -->
@@ -164,7 +164,7 @@
     <param name="presence-hosts" value="$${domain},$${local_ip_v4}"/>
     <param name="presence-privacy" value="$${presence_privacy}"/>
     <!-- ************************************************* -->
-    
+
     <!-- This setting is for AAL2 bitpacking on G726 -->
     <!-- <param name="bitpacking" value="aal2"/> -->
     <!--max number of open dialogs in proceeding -->
@@ -203,39 +203,39 @@
     <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
     <param name="tls-version" value="$${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)-->
+    <!-- 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-->
+    <!--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"/> -->
@@ -252,16 +252,16 @@
     <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
-       -->
+         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"/>
 
@@ -274,8 +274,8 @@
     <!-- <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.
+        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 -->
@@ -297,28 +297,27 @@
     <!--<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
+    <!--
+        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.
+        Choose the realm challenge key. Default is auto_to if not set.
 
-	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
+        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"/>-->
@@ -332,59 +331,58 @@
     <!-- 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 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
+           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 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" /> -->
+    <!-- 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 -->
 
-	<!-- Turn on a jitterbuffer for every call -->
-	<!-- <param name="auto-jitterbuffer-msec" value="60"/> -->
+    <!-- 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" /> -->
 
 
-	<!-- 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"/>-->
-    
+    <!-- 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/conf/vanilla/sip_profiles/internal/example.xml b/conf/vanilla/sip_profiles/internal/example.xml
index fc061f96d1..10263b05a5 100644
--- a/conf/vanilla/sip_profiles/internal/example.xml
+++ b/conf/vanilla/sip_profiles/internal/example.xml
@@ -9,7 +9,7 @@
   <!--/// 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"/>--> 
+  <!--<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 ///-->