Update documentation

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172270 65c4cc65-6c06-0410-ace0-fbb531ad65f3

CHANGES

@@ -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
 --------------

configs/sip.conf.sample

@@ -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