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authorMark Michelson <mmichelson@digium.com>2008-12-29 18:04:52 +0000
committerMark Michelson <mmichelson@digium.com>2008-12-29 18:04:52 +0000
commitff20b9116a4a5f7522d802d5cfc35bb40974cebe (patch)
tree93d5216ca92828b1336bde6fd40f69ccf183c4ae /res/res_http_post.c
parentc04417b47712e1d495c29ca478f5d7503a36426e (diff)
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: https://origsvn.digium.com/svn/asterisk/trunk@166861 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'res/res_http_post.c')
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