Update deprecated config parameters (signalwire#42)

Author: Len-PGH

docs/FreeSWITCH-Explained/Configuration/Sofia-SIP-Stack/Sofia-Configuration-Files_7144453.mdx

@@ -962,7 +962,7 @@ _Default: false_
 
 If true, it passes [RFC 2833](https://tools.ietf.org/html/rfc2833) DTMF's from one side of a bridge to the other, untouched. Otherwise, it decodes and re-encodes them before passing them on.
 
-<param name="pass-rfc2833" value="true"/>
+`<param name="pass-rfc2833" value="true"/>`
 
 ######  liberal-dtmf
 
@@ -971,21 +971,21 @@ TODO [RFC 2833](https://tools.ietf.org/html/rfc2833) is obsoleted by [RFC 4733](
 _Default: false_  
 For DTMF negotiation, use this parameter to just always offer 2833 and accept both 2833 and INFO. Use of this parameter is not recommended since its purpose is to try to cope with buggy SIP implementations.
 
-&lt;param name="liberal-dtmf" value="true"/>
+`<param name="liberal-dtmf" value="true"/>`
 
 #####  SIP Related options 
 
 ###### enable-timer 
 
 This enables or disables support for [RFC 4028](http://tools.ietf.org/html/rfc4028) SIP Session Timers.
 
-&lt;param name="enable-timer" value="false"/>
+`<param name="enable-timer" value="false"/>`
 
 ######  session-timeout 
 
 session timers for all call to expire after the specified seconds Then it will send another invite (re-invite). If not specified defaults to 30 minutes. Some gateways may reject values less than 30 minutes. This values refers to Session-Expires in [RFC 4028](http://tools.ietf.org/html/rfc4028) (The time at which an element will consider the session timed out, if no successful session refresh transaction occurs beforehand)
 
-&lt;param name="session-timeout" value="1800"/>
+`<param name="session-timeout" value="1800"/>`
 
 Note: If your switch requires the timer option; for instance, Huawei SoftX3000, it needs this optional field and drops the calls with "Session Timer Check Message Failed", then you may be able to revert back the commit that took away the Require: timer option which is an optional field by:
 
@@ -999,21 +999,21 @@ make mod_sofia-install
 
 This enable support for 100rel (100% reliability - PRACK message as defined in [RFC3262](http://tools.ietf.org/html/rfc3262)) This fixes a problem with SIP where provisional messages like "180 Ringing" are not ACK'd and therefore could be dropped over a poor connection without retransmission. \*2009-07-08:\* Enabling this may cause FreeSWITCH to crash, see [FSCORE-392](http://jira.freeswitch.org/browse/FSCORE-392).
 
-&lt;param name="enable-100rel" value="true"/>
+`<param name="enable-100rel" value="true"/>`
 
 ######  minimum-session-expires 
 
 This sets the "Min-SE" value (in seconds) from [RFC 4028](http://tools.ietf.org/html/rfc4028). This value must not be less than 90 seconds.
 
-&lt;param name="minimum-session-expires" value="120"/>
+`<param name="minimum-session-expires" value="120"/>`
 
 ######  sip-options-respond-503-on-busy 
 
 When set to true, this param will make FreeSWITCH respond to incoming SIP OPTIONS with 503 "Maximum Calls In Progress" when FS is paused or maximum sessions has been exceeded.
 
 When set to false or when not set at all (default behavior), SIP OPTIONS are always responded with 200 "OK".
 
-&lt;param name="sip-options-respond-503-on-busy" value="false"/> 
+`<param name="sip-options-respond-503-on-busy" value="false"/>` 
 
 Setting this param to true is especially useful if you're using a proxy such as OpenSIPS or Kamailio with dispatcher module to probe your FreeSWITCH servers by sending SIP OPTIONS.
 
@@ -1031,8 +1031,8 @@ Result will be that clients will not re-register at the same time-interval thus
 
 For example, if you set:
 
-&lt;param name="sip-force-expires" value="1800"/>
-&lt;param name="sip-expires-max-deviation" value="600"/>
+`<param name="sip-force-expires" value="1800"/>`
+`<param name="sip-expires-max-deviation" value="600"/>`
 
 then the expires that is responded will be between 1800-600=1200 and 1800+600=2400 seconds.
 
@@ -1042,25 +1042,25 @@ This param can be overridden per individual user by setting a _sip-expires-max-d
 
 Setting this param will send all outbound transactions to the value set by outbound-proxy.
 
- &lt;param name="outbound-proxy" value="127.0.0.1"/>
+ `<param name="outbound-proxy" value="127.0.0.1"/>`
 
 ######  send-display-update 
 
 Tells FreeSWITCH not to send display UPDATEs to the leg of the call.
 
- &lt;param name="send-display-update" value="false"/>
+ `<param name="send-display-update" value="false"/>`
 
 #####  RTP Related options 
 
 ######  auto-jitterbuffer-msec 
 
 Set this to the size of the [jitterbuffer](../../Codecs-and-Media/JitterBuffer_6587407.mdx#about) you would like to have on all calls coming through this profile.
 
-&lt;param name="auto-jitterbuffer-msec" value="120"/>
+`<param name="auto-jitterbuffer-msec" value="120"/>`
 
 ######  rtp-timer-name 
 
-&lt;param name="rtp-timer-name" value="soft"/>
+`<param name="rtp-timer-name" value="soft"/>`
 
 ######  rtp-rewrite-timestamps 
 
@@ -1070,7 +1070,7 @@ The result is that FreeSWITCH will regenerate and rewrite the timestamps in all
 
 This could be necessary to fix audio issues when sending calls to some paranoid and not RFC-compliant gateways (Cirpack is known to require this).
 
-&lt;param name="rtp-rewrite-timestamps" value="true"/>
+`<param name="rtp-rewrite-timestamps" value="true"/>`
 
 
 ######  media_timeout 
@@ -1081,15 +1081,17 @@ This channel variable configures the number of milliseconds of RTP inactivity (n
 
 Example: set the media timeout to 30 seconds
 
-&lt;param name="media_timeout" value="30000"/>
+`<X-PRE-PROCESS cmd="set" data="media_timeout=300000"/>`
 
 ###### media_hold_timeout   
 
-was: rtp-hold-timeout-sec (deprecated config parameter)  
+was: rtp_hold_timeout_sec (deprecated config parameter)
+
+`rtp_hold_timeout_sec deprecated use media_timeout variable.`
 
 This channel variable configures the number of seconds of RTP inactivity (media silence) for a call placed on hold by an endpoint before FreeSWITCH considers the call disconnected, and hangs up. It is recommended that you use session timers instead, as some phones stop sending media when placed on hold. If this setting is omitted, the default value is "0", which disables the timeout.
 
-&lt;param name="media_hold_timeout" value="1800"/>
+`<X-PRE-PROCESS cmd="set" data="media_hold_timeout=1800000"/>`
 
 ######  rtp-autoflush-during-bridge 
 
@@ -1099,7 +1101,7 @@ When set to "true" (the default), FreeSWITCH will notice when more than one RTP
 
 If rtp-autoflush-during-bridge is set to false, FreeSWITCH will instead preserve all RTP packets on bridged calls, even if it increases the latency or "lag" that callers hear.
 
-&lt;param name="rtp-autoflush-during-bridge" value="true"/>
+`<param name="rtp-autoflush-during-bridge" value="true"/>`
 
 ######  rtp-autoflush 
 
@@ -1109,7 +1111,7 @@ Unlike "rtp-autoflush-during-bridge", the default is false, meaning that high-la
 
 Setting "rtp-autoflush" to true will discard packets to minimize latency when possible. Doing so may cause errors in DTMF recognition, faxes, and other processes that rely on receiving all packets.
 
-&lt;param name="rtp-autoflush" value="true"/>
+`<param name="rtp-autoflush" value="true"/>`
 
 #####  Auth