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authorRichard Mudgett <rmudgett@digium.com>2011-05-06 16:23:14 +0000
committerRichard Mudgett <rmudgett@digium.com>2011-05-06 16:23:14 +0000
commit307f148adb3fd9400969e6b7e9f6c231920b0fcd (patch)
treea9d9adc58056b333225fdfc51681124fba6a6ee3 /pbx/pbx_lua.c
parent892953466b3fcaaca4fb7b523c9f14274715976c (diff)
Merged revisions 317670 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317670 | rmudgett | 2011-05-06 11:19:18 -0500 (Fri, 06 May 2011) | 22 lines Fix SIP connected line updates. This patch fixes a couple SIP connected line update problems: 1) The connected line needs to be updated when the initial INVITE is sent if there is a peer callerid configured. Previously, the connected line information did not get reported until the call was connected so SIP could not report connected line information in ringing or progress messages. 2) The connected line should not be updated on initial connect if there is no connected line information. Previously, all it did was wipe out any default preset CONNECTEDLINE information set by the dialplan with empty strings. (closes issue #18367) Reported by: GeorgeKonopacki Patches: issue18367_v1.8.patch uploaded by rmudgett (license 664) Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1199/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317671 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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