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Add documentation on "allowtransfer"

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@25614 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Olle Johansson
2006-05-08 15:46:02 +00:00
parent 55c6781c9e
commit ca6cf552f9
1 changed files with 3 additions and 0 deletions
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configs/sip.conf.sample
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@@ -29,6 +29,8 @@ context=default ; Default context for incoming calls
; this can also be set to 'osp' ; this can also be set to 'osp'
; if asterisk was compiled with OSP support.) ; if asterisk was compiled with OSP support.)
allowoverlap=no ; Disable overlap dialing support. (Default is yes) allowoverlap=no ; Disable overlap dialing support. (Default is yes)
;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
; Default is enabled
;realm=mydomain.tld ; Realm for digest authentication ;realm=mydomain.tld ; Realm for digest authentication
; defaults to "asterisk" ; defaults to "asterisk"
; Realms MUST be globally unique according to RFC 3261 ; Realms MUST be globally unique according to RFC 3261
@@ -334,6 +336,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; restrictcid restrictcid ; restrictcid restrictcid
; allowoverlap allowoverlap ; allowoverlap allowoverlap
; allowsubscribe allowsubscribe ; allowsubscribe allowsubscribe
; allowtransfer allowtransfer
; subscribecontext subscribecontext ; subscribecontext subscribecontext
; videosupport videosupport ; videosupport videosupport
; maxcallbitrate maxcallbitrate ; maxcallbitrate maxcallbitrate