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That change is wrong, wrong, wrong.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370821 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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All voicemails now have a 'msg_id' included in their metadata. The ODBC
message storage backend now requires this column; as such, the MySQL contrib
script that creates the voicemail_data table has been updated with the appropriate
column information.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370820 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Presence support has been added. This is accomplished by
allowing for presence hints in addition to device state
hints. A dialplan function called PRESENCE_STATE has been
added to allow for setting and reading presence. Presence
can be transmitted to Digium phones using custom XML
elements in a PIDF presence document.
Voicemail has new APIs that allow for moving, removing,
forwarding, and playing messages. Messages have had a new
unique message ID added to them so that the APIs will work
reliably. The state of a voicemail mailbox can be obtained
using an API that allows one to get a snapshot of the mailbox.
A voicemail Dialplan App called VoiceMailPlayMsg has been
added to be able to play back a specific message.
Configuration hooks have been added. Configuration hooks
allow for a piece of code to be executed when a specific
configuration file is loaded by a specific module. This is
useful for modules that are dependent on the configuration
of other modules.
chan_sip now has a public method that allows for a custom
SIP INFO request to be sent mid-dialog. Digium phones use
this in order to display progress bars when files are played.
Messaging support has been expanded a bit. The main
visible difference is the addition of an AMI action
MessageSend.
Finally, a ParkingLots manager action has been added in order
to get a list of parking lots.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This patch is based on the one by David Vossel, developer extrodinaire, at
https://reviewboard.asterisk.org/r/344/. If multiple peers are defined with the
same host/port, but differing callbackextensions, it chooses the peer with the
matching callbackextension. Since callbackextension creates an outbound
registration with the callbackextension as the Contact address, matching an
incoming request by that (in addition to the host/port) makes a lot of sense.
This patch also adds support for callbackextension to realtime by querying all
peers with callbackextensions on reload and adding registrations for them.
(closes issue ASTERISK-13456)
Review: https://reviewboard.asterisk.org/r/344/
Review: https://reviewboard.asterisk.org/r/1717/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354458 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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1. Set lastms to 0 when clearing instead of ""
2. Don't set ipaddr or port to the string "(null)" when they are empty
3. Add missing required fields, set default for lastms to 0, and modify
the length of the ipaddr field to 45 in the Postgresql realtime.sql
file.
(closes issue ASTERISK-19172)
Review: https://reviewboard.asterisk.org/r/1703/
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Merged revisions 354348 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 354349 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354360 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This patch allows the imapserver, imapport, and imapflags settings to be
overridden for any voicemail user. It also documents the settings in
the sample voicemail.conf file, and updates the voicemail schema to
allow storage of those columns.
(closes issue ASTERISK-16489)
Reporter: Hubert Mickael
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1614/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349106 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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in r342871
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345601 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This patch adds two new menu features to app_confbridge, admin_toggle_menu_
participants and participant_count. The admin action will globally mute /
unmute all non-admin participants on a converence, while the participant
count simply exposes the existing participant count function to the
conference bridge menu.
This also adds configuration options to change the sound played when the
conference is globally muted / unmuted, as well as the necessary config
hooks to place these functions in the DTMF menus.
(closes issue ASTERISK-18204)
Reported by: Kevin Reeves
Tested by: Matt Jordan
Patches:
app_confbridge.c.patch.txt, conf_config_parser.c.patch.txt,
confbridge.h.patch.txt uploaded by Kevin Reeves (license 6281)
Review: https://reviewboard.asterisk.org/r/1518/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345560 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This option is not only useless, but has been broken since inception since
the flag was never copied from the peer where it is set to the pvt where
it was checked. RFC 3261 specificially states that you should not send a
provisional response to a non-INVITE request, and if we did fix the code
so that it worked, it would cause the same kind of user enumeration
vulnerability that we've discussed with the nat= setting. This patch
removes registertrying option and any code that would have sent a 100
response to a register.
Review: https://reviewboard.asterisk.org/r/1562/
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Merged revisions 343220 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 343221 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343222 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Somewhere between 1.4 and 1.8 the sipusers family has become completely
unused. Before that, the sipfriends family had been obsoleted in favor
of separate sipusers and sippeers families. Apparently, they have been
merged back again into a single family which is now called "sippeers".
Reviewed by: irroot, oej, pabelanger
Review: https://reviewboard.asterisk.org/r/1523
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Merged revisions 342869 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 342870 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342871 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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keyword.
This permits the list of codecs to be specified in one configuration line,
instead of two or more, generally with the aim of either allowing all codecs
with the exception of a few or disallowing most but permitting a few.
Review: https://reviewboard.asterisk.org/r/1411/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334574 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r317805 | russell | 2011-05-06 14:14:39 -0500 (Fri, 06 May 2011) | 7 lines
Add a new sipfriends.sql for MySQL that has more fields in it.
(closes issue #16399)
Reported by: pabelanger
Patches:
sipfriends.sql.v3 uploaded by pabelanger (license 224)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317807 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r317486 | russell | 2011-05-05 18:15:53 -0500 (Thu, 05 May 2011) | 9 lines
Fix some more realtime MySQL schema issues.
(closes issue #18537)
Reported by: denzs
Patches:
sipfriends.sql.svndiff uploaded by denzs (license 1182)
queue_log.sql.svndiff uploaded by denzs (license 1182)
meetme.sql.svndiff uploaded by denzs (license 1182)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317487 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r317484 | russell | 2011-05-05 18:12:35 -0500 (Thu, 05 May 2011) | 8 lines
Fix some errors in sample MySQL realtime schema files.
(closes issue #18915)
Reported by: Dovid
Patches:
sipfriends.patch uploaded by Dovid (license 652)
meetme.patch uploaded by Dovid (license 652)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317485 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r286588 | tilghman | 2010-09-14 00:07:16 -0500 (Tue, 14 Sep 2010) | 9 lines
Merged revisions 286587 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r286587 | tilghman | 2010-09-14 00:06:05 -0500 (Tue, 14 Sep 2010) | 2 lines
Add documentation on missing backend tables for Voicemail
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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to be used, even when realtime is used.
(closes issue #17082)
Reported by: coolmig
Patches:
20100720__issue17082.diff.txt uploaded by tilghman (license 14)
Tested by: coolmig
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278307 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277452 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Also, update meetme to the full list of supported fields.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276869 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r275909 | tilghman | 2010-07-13 09:47:30 -0500 (Tue, 13 Jul 2010) | 2 lines
Move SQL scripts into their own database-specific directories.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275910 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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