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Update documentation 

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Olle Johansson 2009-01-29 13:24:01 +00:00
parent b79a12e929
commit 0685c4b281
2 changed files with 5 additions and 3 deletions
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6
CHANGES
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@ -28,8 +28,8 @@ SIP Changes
* Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
option is enabled, a SIP channel will go to the fax extension (if it exists)
after T38 is negotiated. This option is disabled by default.
* If ATTENDED_TRANSFER_COMPLETE_SOUND is set, the sound will be played to the
target of an attended transfer
* If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
the sound will be played to the target of an attended transfer
* Added two new configuration options, "qualifygap" and "qualifypeers", which allow
finer control over how many peers Asterisk will qualify and the gap between them
when all peers need to be qualified at the same time.
@ -46,6 +46,8 @@ SIP Changes
information
* Added a function to remove SIP headers added in the dialplan before the
first INVITE is generated - SIPRemoveHeader()
* Channel variables set with setvar= in a device configuration is now
set both for inbound and outbound calls.
Skinny Changes
--------------

2
configs/sip.conf.sample
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@ -1057,7 +1057,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;defaultip=192.168.0.4 ; IP address to use until registration
;defaultuser=goran ; Username to use when calling this device before registration
; Normally you do NOT need to set this parameter
;setvar=CUSTID=5678 ; Channel variable to be set for all calls from this device
;setvar=CUSTID=5678 ; Channel variable to be set for all calls from or to this device
;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
; cause the given audio file to
; be played upon completion of