Merged revisions 160480 via svnmerge from
authorTilghman Lesher <tilghman@meg.abyt.es>
Wed, 3 Dec 2008 14:11:53 +0000 (14:11 +0000)
committerTilghman Lesher <tilghman@meg.abyt.es>
Wed, 3 Dec 2008 14:11:53 +0000 (14:11 +0000)
commitc9f471ac77a8f8e2b4094299b35f90d1b401adaf
treebe428465bb4b7ef5f3c8c23e5880789aa84c81a3
parentbfe0c6c714388668a091d7d07c6d110688213078
Merged revisions 160480 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  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.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160481 65c4cc65-6c06-0410-ace0-fbb531ad65f3
channels/chan_sip.c