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| author | tilghman <tilghman@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-12-03 14:09:35 +0000 |
|---|---|---|
| committer | tilghman <tilghman@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-12-03 14:09:35 +0000 |
| commit | 77c19bced93f9648f794dcac9f1919267d63d6fc (patch) | |
| tree | b5ec0715dfd5a59f900b2c467f676a1159e3ae6a /configs/modules.conf.sample | |
| parent | 75d05abdfd7717e591c1fe9d91bd5e89c8e60a67 (diff) | |
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/branches/1.4@160480 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'configs/modules.conf.sample')
0 files changed, 0 insertions, 0 deletions