Merged revisions 160480 via svnmerge from

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

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  r160480 | tilghman | 2008-12-03 08:09:35 -0600 (Wed, 03 Dec 2008) | 7 lines
  
  Jon Bonilla (Manwe) pointed out on the -dev list:
  "I guess that having only ip-phones in mind is not a good approach. Since it is
  possible to have a sip proxy connected to asterisk we could receive a 407
  (unauthorized) or 483 (too many hops) as response and dialog ending would not be
  a good behavior."
  So modified.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160481 65c4cc65-6c06-0410-ace0-fbb531ad65f3