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| author | tilghman <tilghman@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-12-03 14:11:53 +0000 |
|---|---|---|
| committer | tilghman <tilghman@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-12-03 14:11:53 +0000 |
| commit | 56ff6e011ec4f10a55234a6e9cebfbb6dc656b4e (patch) | |
| tree | be428465bb4b7ef5f3c8c23e5880789aa84c81a3 /pbx | |
| parent | c874e12308ba26e4c96097d729822653feb646c3 (diff) | |
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: http://svn.digium.com/svn/asterisk/trunk@160481 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'pbx')
0 files changed, 0 insertions, 0 deletions