[asterisk-dev] [asterisk-commits] oej: trunk r93160 - in /trunk: CHANGES configs/sip.conf.sample

[Pavel Jezek]

shouldn't we have also qualifyfreq configurable?
default 'sip ping' frequency 60s isn't enough in many envrironments, 
because udp nat translations on firewalls often timeouts quicker...
PJ




SVN commits to the Asterisk project wrote:
> Author: oej
> Date: Sun Dec 16 02:19:38 2007
> New Revision: 93160
>
> URL: http://svn.digium.com/view/asterisk?view=rev&rev=93160
> Log:
> Update documentation
>
> Modified:
>     trunk/CHANGES
>     trunk/configs/sip.conf.sample
>
> Modified: trunk/CHANGES
> URL: http://svn.digium.com/view/asterisk/trunk/CHANGES?view=diff&rev=93160&r1=93159&r2=93160
> ==============================================================================
> --- trunk/CHANGES (original)
> +++ trunk/CHANGES Sun Dec 16 02:19:38 2007
> @@ -101,6 +101,9 @@
>    * 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
>  ------------
>
> Modified: trunk/configs/sip.conf.sample
> URL: http://svn.digium.com/view/asterisk/trunk/configs/sip.conf.sample?view=diff&rev=93160&r1=93159&r2=93160
> ==============================================================================
> --- trunk/configs/sip.conf.sample (original)
> +++ trunk/configs/sip.conf.sample Sun Dec 16 02:19:38 2007
> @@ -81,13 +81,6 @@
>  				; 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 @@
>  				; 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.
>
>
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