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| author | Rusty Newton <rnewton@digium.com> | 2014-01-17 17:16:14 +0000 |
|---|---|---|
| committer | Rusty Newton <rnewton@digium.com> | 2014-01-17 17:16:14 +0000 |
| commit | f6647d2362964539f3d7aaf23479fe284e9442ff (patch) | |
| tree | ec7615ab108a3fe0654ba8e7faff698de65d3759 /apps/app_transfer.c | |
| parent | 3fb2906955dab8081d164969a6789a99d7bbf1a6 (diff) | |
Documentation: doc fixes across various parts of the code for ASTERISK issues 23061,23028,23046,23027
Fixes typos of "transfered" instead of "transferred" in various code. Fixes incorrect gosub param help text for app_queue.
Fixes Asterisk man pages containing unquoted minus signs. Adds note about the "textsupport" option in sip.conf.sample.
(issue ASTERISK-23061)
(issue ASTERISK-23028)
(issue ASTERISK-23046)
(issue ASTERISK-23027)
(closes issue ASTERISK-23061)
(closes issue ASTERISK-23028)
(closes issue ASTERISK-23046)
(closes issue ASTERISK-23027)
Reported by: Eugene, Jeremy Laine, Denis Pantsyrev
Patches:
transferred.patch uploaded by Jeremy Laine (license 6561)
hyphen.patch uploaded by Jeremy Laine (license 6561)
sip.conf.sample.patch uploaded by Eugene (license 6360)
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Merged revisions 405791 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 405792 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 405829 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405830 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'apps/app_transfer.c')
| -rw-r--r-- | apps/app_transfer.c | 2 |
1 files changed, 1 insertions, 1 deletions
diff --git a/apps/app_transfer.c b/apps/app_transfer.c index 3ed66f120..1b8108278 100644 --- a/apps/app_transfer.c +++ b/apps/app_transfer.c @@ -54,7 +54,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$") <description> <para>Requests the remote caller be transferred to a given destination. If TECH (SIP, IAX2, LOCAL etc) is used, only - an incoming call with the same channel technology will be transfered. + an incoming call with the same channel technology will be transferred. Note that for SIP, if you transfer before call is setup, a 302 redirect SIP message will be returned to the caller.</para> <para>The result of the application will be reported in the <variable>TRANSFERSTATUS</variable> |