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	Add documentation on "allowtransfer"
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@25614 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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		| @@ -29,6 +29,8 @@ context=default			; Default context for incoming calls | ||||
| 				; this can also be set to 'osp' | ||||
| 				; if asterisk was compiled with OSP support.) | ||||
| 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 | ||||
| 				; defaults to "asterisk" | ||||
| 				; Realms MUST be globally unique according to RFC 3261 | ||||
| @@ -334,6 +336,7 @@ srvlookup=yes			; Enable DNS SRV lookups on outbound calls | ||||
| ; restrictcid		      restrictcid | ||||
| ; allowoverlap		      allowoverlap | ||||
| ; allowsubscribe	      allowsubscribe | ||||
| ; allowtransfer	      	      allowtransfer | ||||
| ; subscribecontext	      subscribecontext | ||||
| ; videosupport		      videosupport | ||||
| ; maxcallbitrate	      maxcallbitrate | ||||
|   | ||||
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