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Merged revisions 186059 via svnmerge from

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https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r186059 | tilghman | 2009-04-02 12:09:13 -0500 (Thu, 02 Apr 2009) | 9 lines
  
  Merged revisions 186056 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.2
  
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    r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02 Apr 2009) | 2 lines
    
    Fix for AST-2009-003
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186060 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Tilghman Lesher
2009-04-02 17:10:28 +00:00
parent bab6e401ef
commit 08971ce205
2 changed files with 111 additions and 10 deletions
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6
configs/sip.conf.sample
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@@ -261,9 +261,11 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;authfailureevents=no ; generate manager "peerstatus" events when peer can't
; authenticate with Asterisk. Peerstatus will be "rejected".
;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
; for any reason, always reject with '401 Unauthorized'
; for any reason, always reject with an identical response
; equivalent to valid username and invalid password/hash
; instead of letting the requester know whether there was
; a matching user or peer for their request
; a matching user or peer for their request. This reduces
; the ability of an attacker to scan for valid SIP usernames.
;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
; order instead of RFC3551 packing order (this is required