diff options
| author | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-01-29 13:24:01 +0000 |
|---|---|---|
| committer | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-01-29 13:24:01 +0000 |
| commit | 7041314e0391abdde2f756f2b49fdcff5b45985a (patch) | |
| tree | 175e0e9ae0d61a43740a38beb46d133c0f6ff93a | |
| parent | 7ad1a935c89e6f9173866140f81b6f98c8c4c1d1 (diff) | |
Update documentation
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@172270 f38db490-d61c-443f-a65b-d21fe96a405b
| -rw-r--r-- | CHANGES | 6 | ||||
| -rw-r--r-- | configs/sip.conf.sample | 2 |
2 files changed, 5 insertions, 3 deletions
@@ -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 -------------- diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index cc428f682..a33ec7f76 100644 --- a/configs/sip.conf.sample +++ b/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 |