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Merged revisions 103385 via svnmerge from
authorJoshua Colp <jcolp@digium.com>
Tue, 12 Feb 2008 15:13:06 +0000 (15:13 +0000)
committerJoshua Colp <jcolp@digium.com>
Tue, 12 Feb 2008 15:13:06 +0000 (15:13 +0000)
commit60de470a7738f98a2c64623981bcfa380077a5c5
tree2fee5bfdf9186367f5b316fb8256e5b92c6b843atree | snapshot
parentf98f90af49a593134a61ff1ccabb54bb803e6af2commit | diff
Merged revisions 103385 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r103385 | file | 2008-02-12 11:09:24 -0400 (Tue, 12 Feb 2008) | 4 lines

Even if no CallerID name or number has been provided by the remote party still use the configured sip.conf ones.
(closes issue #11977)
Reported by: pj

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103386 65c4cc65-6c06-0410-ace0-fbb531ad65f3
channels/chan_sip.c diff | blob | blame | history
Mirror of https://github.com/asterisk/asterisk.git