Files
asterisk/channels
Joshua Colp 60de470a77 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
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