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Merge OSS fixes for FreeBSD, implement rtptimeout and rtpholdtimeout

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@3097 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Spencer
2004-05-27 22:12:55 +00:00
parent 4d1706d05f
commit e446f4ca81
6 changed files with 93 additions and 22 deletions
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8
configs/sip.conf.sample
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@@ -48,7 +48,10 @@ bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
;language=en ; Default language setting for all users/peers
; This may also be set for individual users/peers
;relaxdtmf=yes ; Relax dtmf handling
;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity
; when we're not on hold
;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity
; when we're on hold (must be > rtptimeout)
; Asterisk can register as a SIP user agent to a SIP proxy (provider)
; Format for the register statement is:
@@ -128,7 +131,8 @@ bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
; port
; qualify
; defaultip
; rtptimeout
; rtpholdtimeout
;[sip_proxy]
; For incoming calls only. Example: FWD (Free World Dialup)