whitespace cleanup

conf/insideout/sip_profiles/external.xml

@@ -30,9 +30,9 @@
     <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
+         manage-presence needs to be set to passive on this profile
-	 if you want it to behave as if it were the internal profile 
+         if you want it to behave as if it were the internal profile
-	 for presence.
+         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}"/>

conf/insideout/sip_profiles/internal-ipv6.xml

@@ -82,7 +82,7 @@
     <!--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-->
+        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>
-

conf/insideout/sip_profiles/internal.xml

@@ -27,13 +27,13 @@
 
   <settings>
     <!--
-	When calls are in no media this will bring them back to media
+        When calls are in no media this will bring them back to media
-	when you press the hold button.
+        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 
+        This will allow a call after an attended transfer go back to
-	 bypass media after an attended transfer.
+        bypass media after an attended transfer.
     -->
     <!--<param name="media-option" value="bypass-media-after-att-xfer"/>-->
     <!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
@@ -117,7 +117,7 @@
     <!--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-->
+        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"/>-->
@@ -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. 
+        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_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.
+        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.
+        <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 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
+        If you use any other value besides auto_to or auto_from you'll loose
-	the ability to do multiple domains.
+        the ability to do multiple domains.
 
-	Note: comment out to restore the behavior before 2008-09-29
+        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>
-

conf/sbc/sbc_profiles/external.xml

@@ -31,9 +31,9 @@
     <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
+         manage-presence needs to be set to passive on this profile
-	 if you want it to behave as if it were the internal profile 
+         if you want it to behave as if it were the internal profile
-	 for presence.
+         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}"/>

conf/sbc/sbc_profiles/internal-ipv6.xml

@@ -82,7 +82,7 @@
     <!--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-->
+        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>
-

conf/sbc/sbc_profiles/internal.xml

@@ -26,13 +26,13 @@
 
   <settings>
     <!--
-	When calls are in no media this will bring them back to media
+        When calls are in no media this will bring them back to media
-	when you press the hold button.
+        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 
+        This will allow a call after an attended transfer go back to
-	 bypass media after an attended transfer.
+        bypass media after an attended transfer.
     -->
     <!--<param name="media-option" value="bypass-media-after-att-xfer"/>-->
     <!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
@@ -123,7 +123,7 @@
     <!--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-->
+        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. 
+        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_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.
+        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.
+        <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 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
+        If you use any other value besides auto_to or auto_from you'll loose
-	the ability to do multiple domains.
+        the ability to do multiple domains.
 
-	Note: comment out to restore the behavior before 2008-09-29
+        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>
-

conf/vanilla/sip_profiles/external.xml

@@ -7,8 +7,8 @@
 
   <aliases>
     <!--
-    <alias name="outbound"/>
+        <alias name="outbound"/>
-    <alias name="nat"/>
+        <alias name="nat"/>
     -->
   </aliases>
 
@@ -18,8 +18,8 @@
 
   <settings>
     <param name="debug" value="0"/>
-	<!-- If you want FreeSWITCH to shutdown if this profile fails to load, uncomment the next line. -->
+    <!-- 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="shutdown-on-fail" value="true"/> -->
     <param name="sip-trace" value="no"/>
     <param name="sip-capture" value="no"/>
     <param name="rfc2833-pt" value="101"/>
@@ -40,9 +40,9 @@
     <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
+         manage-presence needs to be set to passive on this profile
-	 if you want it to behave as if it were the internal profile 
+         if you want it to behave as if it were the internal profile
-	 for presence.
+         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>

conf/vanilla/sip_profiles/internal-ipv6.xml

@@ -83,7 +83,7 @@
     <!--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-->
+        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.
+        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.
+        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 -->
+    <!-- the following can be used as workaround with bogus SRV/NAPTR records -->
-	 <!--<param name="disable-srv" value="false" />-->
+    <!--<param name="disable-srv" value="false" />-->
-	 <!--<param name="disable-naptr" value="false" />-->
+    <!--<param name="disable-naptr" value="false" />-->
 
   </settings>
 </profile>
-

conf/vanilla/sip_profiles/internal.xml

@@ -10,7 +10,7 @@
   <!--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 -->
@@ -33,19 +33,19 @@
     <!-- <param name="rtp-digit-delay" value="40"/>-->
 
     <!--
-	When calls are in no media this will bring them back to media
+        When calls are in no media this will bring them back to media
-	when you press the hold button.
+        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 
+        This will allow a call after an attended transfer go back to
-	 bypass media after an attended transfer.
+        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. -->
+    <!-- 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="shutdown-on-fail" value="true"/> -->
     <param name="sip-trace" value="no"/>
     <param name="sip-capture" value="no"/>
 
@@ -106,8 +106,8 @@
 
     <!--<param name="aggressive-nat-detection" value="true"/>-->
     <!--
-	There are known issues (asserts and segfaults) when 100rel is enabled.
+        There are known issues (asserts and segfaults) when 100rel is enabled.
-	It is not recommended to enable 100rel at this time.
+        It is not recommended to enable 100rel at this time.
     -->
     <!--<param name="enable-100rel" value="true"/>-->
 
@@ -118,14 +118,14 @@
     <!-- 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
+        This defines your local network, by default we detect your local network
-	and create this localnet.auto ACL for this.
+        and create this localnet.auto ACL for this.
     -->
     <param name="local-network-acl" value="localnet.auto"/>
     <!--<param name="apply-register-acl" value="domains"/>-->
@@ -204,7 +204,7 @@
     <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)-->
+         (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)-->
@@ -235,7 +235,7 @@
     <!--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-->
+        that the originator is using-->
     <!--<param name="disable-transcoding" value="true"/>-->
     <!-- Handle 302 Redirect in the dialplan -->
     <!--<param name="manual-redirect" value="true"/> -->
@@ -254,14 +254,14 @@
     <param name="auth-all-packets" value="false"/>
 
     <!-- external_sip_ip
-      Used as the public IP address for SDP.
+         Used as the public IP address for SDP.
-      Can be an one of:
+         Can be an one of:
-           ip address            - "12.34.56.78"
+         ip address            - "12.34.56.78"
-           a stun server lookup  - "stun:stun.server.com"
+         a stun server lookup  - "stun:stun.server.com"
-           a DNS name            - "host:host.server.com"
+         a DNS name            - "host:host.server.com"
-           auto                  - Use guessed ip.
+         auto                  - Use guessed ip.
-           auto-nat              - Use ip learned from NAT-PMP or UPNP
+         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.
+        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.
+        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 -->
@@ -298,27 +298,26 @@
     <!--<param name="disable-register" value="true"/>-->
 
     <!--
-	 enable-3pcc can be set to either 'true' or 'proxy', true accepts the call
+        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
+        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. 
+        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_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.
+        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.
+        <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 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
+        If you use any other value besides auto_to or auto_from you'll
-	the ability to do multiple domains.
+        loose the ability to do multiple domains.
-	
-	Note: comment out to restore the behavior before 2008-09-29
 
+        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
+    <!-- clear clears them all or supply the name to add or the name
-	 valid values:
+         prefixed with ~ to remove valid values:
 
-	 clear
+           clear
-	 CISCO_SKIP_MARK_BIT_2833
+           CISCO_SKIP_MARK_BIT_2833
-	 SONUS_SEND_INVALID_TIMESTAMP_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 -->
+    <!-- the following can be used as workaround with bogus SRV/NAPTR records -->
-	 <!--<param name="disable-srv" value="false" />-->
+    <!--<param name="disable-srv" value="false" />-->
-	 <!--<param name="disable-naptr" value="false" />-->
+    <!--<param name="disable-naptr" value="false" />-->
 
-	<!-- The following can be used to fine-tune timers within sofia's transport layer 
+    <!-- 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 -->
+         Those settings are for advanced users and can safely be left as-is -->
 
-	<!-- Initial retransmission interval (in milliseconds).
+    <!-- Initial retransmission interval (in milliseconds).
-		Set the T1 retransmission interval used by the SIP transaction engine. 
+         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. 	 -->
+         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" /> -->
+    <!-- <param name="timer-T1" value="500" /> -->
 
-	<!--  Transaction timeout (defaults to T1 * 64).
+    <!--  Transaction timeout (defaults to T1 * 64).
-		Set the T1x64 timeout value used by the SIP transaction engine.
+         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 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. -->
+         The timeout value T1x64 can be adjusted separately from the initial retransmission interval T1. -->
-	<!-- <param name="timer-T1X64" value="32000" /> -->
+    <!-- <param name="timer-T1X64" value="32000" /> -->
 
 
-	<!-- Maximum retransmission interval (in milliseconds).
+    <!-- Maximum retransmission interval (in milliseconds).
-		Set the maximum retransmission interval used by the SIP transaction engine. 
+         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. 
+         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 
+         Note that the timer A is not capped by T2. Retransmission interval of INVITE requests grows exponentially
-		until the timer B fires.  -->
+         until the timer B fires.  -->
-	<!-- <param name="timer-T2" value="4000" /> -->
+    <!-- <param name="timer-T2" value="4000" /> -->
 
-	<!--
+    <!--
-		Transaction lifetime (in milliseconds).
+        Transaction lifetime (in milliseconds).
-		Set the lifetime for completed transactions used by the SIP transaction engine. 
+        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. 
+        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. -->
+        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" /> -->
+    <!-- <param name="timer-T4" value="4000" /> -->
 
-	<!-- Turn on a jitterbuffer for every call -->
+    <!-- Turn on a jitterbuffer for every call -->
-	<!-- <param name="auto-jitterbuffer-msec" value="60"/> -->
+    <!-- <param name="auto-jitterbuffer-msec" value="60"/> -->
 
 
-	<!-- By default mod_sofia will ignore the codecs in the sdp for hold/unhold operations 
+    <!-- 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.
+         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.
+         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"/>-->
+    <!--<param name="renegotiate-codec-on-hold" value="true"/>-->
 
   </settings>
 </profile>
-