Update documentation
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93160 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@ -101,6 +101,9 @@ SIP changes
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* A new option called "callcounter" (global/peer/user level) enables call counters needed
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for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
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used to enable this functionality).
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* New settings for timer T1 and timer B on a global level or per device. This makes it
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possible to force timeout faster on non-responsive SIP servers. These settings are
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considered advanced, so don't use them unless you have a problem.
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IAX2 changes
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------------
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@ -81,13 +81,6 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; and subscriptions (seconds)
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;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
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;defaultexpiry=120 ; Default length of incoming/outgoing registration
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;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
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; Defaults to 100 ms
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;timert1=500 ; Default T1 timer
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; Defaults to 500 ms
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;timerb=32000 ; Call setup timer. If a provisional response is not received
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; in this amount of time, the call will autocongest
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; Defaults to 64*timert1
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;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
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;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
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; fully. Enable this option to not get error messages
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@ -191,6 +184,19 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; this setting will enforce inactivation of the regexten
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; extension for the peer
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;
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;--------------------------- SIP timers ----------------------------------------------------
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; These timers are used primarily in INVITE transactions.
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; The default for Timer T1 is 500 ms or the measured run-trip time between
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; Asterisk and the device if you have qualify=yes for the device.
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;
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;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
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; Defaults to 100 ms
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;timert1=500 ; Default T1 timer
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; Defaults to 500 ms
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;timerb=32000 ; Call setup timer. If a provisional response is not received
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; in this amount of time, the call will autocongest
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; Defaults to 64*timert1
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;--------------------------- RTP timers ----------------------------------------------------
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; These timers are currently used for both audio and video streams. The RTP timeouts
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; are only applied to the audio channel.
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