Update documentation

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

diff --git a/CHANGES b/CHANGES
index b4810a34ce79f112b28113b40ee196ec780b9885..89a02179304b23e7dd912ba16625465f0d6b27e3 100644 (file)
--- a/CHANGES
+++ b/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
 ------------

diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index e0a06ed34ba33a2c7e3d8901507e01b93bc03c96..61cdf114a80b978cdcd68fcc6809467c1abd7313 100644 (file)
--- a/configs/sip.conf.sample
+++ b/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.