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| author | mmichelson <mmichelson@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-12-29 18:16:10 +0000 |
|---|---|---|
| committer | mmichelson <mmichelson@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-12-29 18:16:10 +0000 |
| commit | 9897c0b825170dbccf5e16b1a89ee2f3111766ad (patch) | |
| tree | ea3682b28aa0385476c3d6165b09604b5e5dc1d6 /res/res_phoneprov.c | |
| parent | 26bf548c6637cc760a7da4e26c5e83896d7d7357 (diff) | |
Merged revisions 166861 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
........
r166861 | mmichelson | 2008-12-29 12:04:52 -0600 (Mon, 29 Dec 2008) | 14 lines
Update app_queue to deal with the removal of AST_PBX_KEEPALIVE
When placing a call to a queue which ran a gosub on the member's
channel, Asterisk would crash every time, stemming from the fact
that the member's channel was being hung up unexpectedly when the
Gosub completed. The necessary change was pretty much copied and
pasted from app_dial's similar changes made last week.
I also took the opportunity to change a LOG_DEBUG message in
app_dial to use ast_debug. I am guessing this was due to a direct
merge from 1.4 that was not corrected to use trunk's preferred
syntax.
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@166863 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'res/res_phoneprov.c')
0 files changed, 0 insertions, 0 deletions