Add documentation on "allowtransfer"
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@25614 f38db490-d61c-443f-a65b-d21fe96a405b
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@ -29,6 +29,8 @@ context=default ; Default context for incoming calls
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; this can also be set to 'osp'
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; if asterisk was compiled with OSP support.)
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allowoverlap=no ; Disable overlap dialing support. (Default is yes)
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;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
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; Default is enabled
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;realm=mydomain.tld ; Realm for digest authentication
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; defaults to "asterisk"
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; Realms MUST be globally unique according to RFC 3261
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@ -334,6 +336,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; restrictcid restrictcid
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; allowoverlap allowoverlap
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; allowsubscribe allowsubscribe
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; allowtransfer allowtransfer
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; subscribecontext subscribecontext
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; videosupport videosupport
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; maxcallbitrate maxcallbitrate
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