Update documentation

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93160 65c4cc65-6c06-0410-ace0-fbb531ad65f3

CHANGES

@@ -101,6 +101,9 @@ SIP changes
   * A new option called "callcounter" (global/peer/user level) enables call counters needed
     for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
     used to enable this functionality).
+  * New settings for timer T1 and timer B on a global level or per device. This makes it 
+    possible to force timeout faster on non-responsive SIP servers. These settings are
+    considered advanced, so don't use them unless you have a problem.
 
 IAX2 changes
 ------------

configs/sip.conf.sample

@@ -81,13 +81,6 @@ srvlookup=yes			; Enable DNS SRV lookups on outbound calls
 				; and subscriptions (seconds)
 ;minexpiry=60			; Minimum length of registrations/subscriptions (default 60)
 ;defaultexpiry=120		; Default length of incoming/outgoing registration
-;t1min=100			; Minimum roundtrip time for messages to monitored hosts
-				; Defaults to 100 ms
-;timert1=500		; Default T1 timer
-				; Defaults to 500 ms
-;timerb=32000		; Call setup timer. If a provisional response is not received
-						; in this amount of time, the call will autocongest
-				; Defaults to 64*timert1
 ;notifymimetype=text/plain	; Allow overriding of mime type in MWI NOTIFY
 ;buggymwi=no			; Cisco SIP firmware doesn't support the MWI RFC
 				; fully. Enable this option to not get error messages
@@ -191,6 +184,19 @@ srvlookup=yes			; Enable DNS SRV lookups on outbound calls
 				; this setting will enforce inactivation of the regexten
 				; extension for the peer
 ;
+;--------------------------- SIP timers ----------------------------------------------------
+; These timers are used primarily in INVITE transactions. 
+; The default for Timer T1 is 500 ms or the measured run-trip time between
+; Asterisk and the device if you have qualify=yes for the device.
+;
+;t1min=100			; Minimum roundtrip time for messages to monitored hosts
+				; Defaults to 100 ms
+;timert1=500		        ; Default T1 timer
+				; Defaults to 500 ms
+;timerb=32000		        ; Call setup timer. If a provisional response is not received
+				; in this amount of time, the call will autocongest
+				; Defaults to 64*timert1
+
 ;--------------------------- RTP timers ----------------------------------------------------
 ; These timers are currently used for both audio and video streams. The RTP timeouts
 ; are only applied to the audio channel.