Files
asterisk/channels
Tilghman Lesher 2c371ff2b7 Merged revisions 160481 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r160481 | tilghman | 2008-12-03 08:11:53 -0600 (Wed, 03 Dec 2008) | 14 lines
  
  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/branches/1.6.0@160482 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-03 14:13:27 +00:00
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2008-01-09 18:03:40 +00:00
2007-12-11 22:20:22 +00:00