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| author | tilghman <tilghman@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-12-03 14:13:27 +0000 |
|---|---|---|
| committer | tilghman <tilghman@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-12-03 14:13:27 +0000 |
| commit | 0cfa1fc820bea91f1a5cc3723eb0c2b3aa8fda83 (patch) | |
| tree | e4aeb0663e13d96c6dc09e98b760e77ad7a5fedd /funcs/func_realtime.c | |
| parent | a14ce24397822338713f77a06b8599ee69768ab1 (diff) | |
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@160482 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'funcs/func_realtime.c')
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