]> git.ipfire.org Git - thirdparty/asterisk.git/commit
Merged revisions 206939 via svnmerge from
authorDavid Vossel <dvossel@digium.com>
Fri, 17 Jul 2009 16:16:35 +0000 (16:16 +0000)
committerDavid Vossel <dvossel@digium.com>
Fri, 17 Jul 2009 16:16:35 +0000 (16:16 +0000)
commit88dc0e47d74f19d7f5f9c9b8957ad957e8154332
tree1242209ce10629a1e398322e87d5e4d5bc790439
parentd233eb8d03fb29556e07f07cb886db1d0a8ccf74
Merged revisions 206939 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r206939 | dvossel | 2009-07-17 11:13:22 -0500 (Fri, 17 Jul 2009) | 20 lines

  Merged revisions 206938 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.4

  ........
    r206938 | dvossel | 2009-07-17 11:05:06 -0500 (Fri, 17 Jul 2009) | 14 lines

    SIP incorrect From: header information when callpres is prohib

    Some ITSP make use of the "Anonymous" display name to detect a
    requirement to withhold caller id across the PSTN. This does
    not work if the display name is "Unknown".

    (closes issue #14465)
    Reported by: Nick_Lewis
    Patches:
          chan_sip.c-callerpres.patch uploaded by Nick (license 657)
          chan_sip.c-callerpres_trunk.patch uploaded by dvossel (license 671)
    Tested by: Nick_Lewis, dvossel
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@206941 65c4cc65-6c06-0410-ace0-fbb531ad65f3
channels/chan_sip.c