Files
asterisk/channels
Tilghman Lesher cc3b3e68f0 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/branches/1.4@160480 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-03 14:09:35 +00:00
..
2008-11-22 00:04:36 +00:00
2008-09-30 23:55:24 +00:00