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Also adds a new "reporting" permission for manager, since it can be incredibly spammy.
This permission was discussed on the -dev mailing list some months back.
Issue 8613, patch by johann8384, with some minor changes by me.
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have cdr_custom capabilities. Many thanks to eserra for this contribution
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with carriers that do Polarity/DTMF CID signalling.
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SS7 number options (#10000)
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Issue 9932, patch by eliel
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else seemed to think was very funny. Oh well ... :)
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disclaimer along with SIP messages in the header, X-Disclaimer. This is off by
default. Also, the text of the disclaimer can be customized in sip.conf.
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features.conf. This allows you to create a feature one time, and then map it
into groups for various different key mappings for the same feature, as well
as easy access control to groups of features.
(patch from bbryant)
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data is properly initialized.
(issue #9765, reported by MatsK, patch from eliel)
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class in musiconhold.conf, a caller on hold may press this digit to switch
to listening to that music class.
This involved adding a new callback for generators, which allow generators
to get notified of DTMF from the channel they are running on. Then, a callback
was implemented for the music on hold generators.
(patch from bbryant)
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manager user. (issue #8664, reported and original patch by ssokol, patch
updated by bkruse, and further updated by me)
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becomes unreachable
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results.
Issue 9514.
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Combined effort by DEA and mvanbaak.
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r63329 | russell | 2007-05-07 17:28:50 -0500 (Mon, 07 May 2007) | 3 lines
Add a sample configuration file and example tables for use with res_config_pgsql.
(issue #9676, suretec)
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r63047 | pari | 2007-05-04 11:45:29 -0500 (Fri, 04 May 2007) | 1 line
explanation for httptimeout in manager.conf
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(issue #7077, patch by adomjan)
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add some new options to control what happens when you hangup on an attended
transfer before the target extension answers the transferred channel. You
can now have it send the transferee back to the transferer.
(issue #8413, patch from sergee with very minor modifications by me)
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r62497 | russell | 2007-05-01 11:26:48 -0500 (Tue, 01 May 2007) | 11 lines
Merged revisions 62496 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r62496 | russell | 2007-05-01 11:26:23 -0500 (Tue, 01 May 2007) | 3 lines
Add indications.conf information for the Philippines.
(issue #9525, reported and patched by loloski)
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file doc/qos.tex has been updated to document the new functionality.
(issue #9540, patch submitted by IgorG)
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r62371 | qwell | 2007-04-30 09:52:31 -0500 (Mon, 30 Apr 2007) | 2 lines
Remove unused (and potentially confusing) jitterbuffer options from sample config.
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This set of changes introduces a new generic event API for use within Asterisk.
I am still working on a way for events to be shared between servers, but this
part is ready and can already be used inside of Asterisk.
This set of changes introduces the first use of the API, as well. I have
restructured the way that MWI (message waiting indication) is handled. It is
now event based instead of polling based. For example, if there are a bunch
of SIP phones subscribed to mailboxes, then chan_sip will not have to
constantly poll the mailboxes for changes. app_voicemail will generate events
when changes occur.
See UPGRADE.txt and CHANGES for some more information on the effects of these
changes from the user perspective. For developer information, see the text in
include/asterisk/event.h.
As always, additional feedback is welcome on the asterisk-dev mailing list.
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control the
minimum amount of time between queue announcements for use when the caller's queue
position changes frequently.
(issue #9604, patch by Matthew Roth)
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blocks instead of one large monolithic app. Supports multiple templates
and is designed mostly for voicemail delivery over e-mail.
There's a todo with a list of ideas in the source code if you want
to contribute. Feedback is appreciated!
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r60603 | russell | 2007-04-06 15:58:43 -0500 (Fri, 06 Apr 2007) | 13 lines
To be able to achieve the things that we would like to achieve with the
Asterisk GUI project, we need a fully functional HTTP interface with access
to the Asterisk manager interface. One of the things that was intended to be
a part of this system, but was never actually implemented, was the ability for
the GUI to be able to upload files to Asterisk. So, this commit adds this in
the most minimally invasive way that we could come up with.
A lot of work on minimime was done by Steve Murphy. He fixed a lot of bugs in
the parser, and updated it to be thread-safe. The ability to check
permissions of active manager sessions was added by Dwayne Hubbard. Then,
hacking this all together and do doing the modifications necessary to the HTTP
interface was done by me.
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r60323 | murf | 2007-04-05 16:35:11 -0600 (Thu, 05 Apr 2007) | 1 line
Added some clarification to the example configs for CDRs, on how to select a backend. Also, made cdr-csv the default if you 'make samples', and no other changes.
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r59452 | murf | 2007-03-29 18:56:36 -0600 (Thu, 29 Mar 2007) | 1 line
A small clarification to keep bugs from being filed, and confusion from rising, if clearglobalvars is set, and globals are set in the AEL file. (9419)
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r59040 | tilghman | 2007-03-19 10:42:26 -0500 (Mon, 19 Mar 2007) | 2 lines
Fix unescaped semicolon (reported via -dev list)
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r58957 | russell | 2007-03-15 20:42:37 -0500 (Thu, 15 Mar 2007) | 1 line
fix a couple SLA documentation references
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r58894 | russell | 2007-03-14 11:33:01 -0500 (Wed, 14 Mar 2007) | 8 lines
By default, don't attempt to do any CallerID handling at all with SLA because
it is known to not work properly in some situations. However, add an option to
enable it for those that would like to use it anyway.
The short story behind this is that to properly handle CallerID with SLA, we
need the ability to change the CallerID on an existing call, and we are not
ready to handle that.
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r58870 | russell | 2007-03-13 18:11:08 -0500 (Tue, 13 Mar 2007) | 1 line
fix the reference to the SLA documentation
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* Add new module, cdr_sqlite3_custom which allows logging custom CDRs into a
SQLite3 database. (issue #7149, alerios)
* Add new module, res_config_sqlite, which adds realtime database configuration
support for SQLite version 2. I decided that this was ok since we didn't have
any realtime support for version 3. If someone ports this to version 3, then
version 2 support can be removed or marked deprecated.
(issue #7790, rbarun_proformatique)
* Mark cdr_sqlite as deprecated in favor of cdr_sqlite3_custom.
Also, note that there were other modules on the bug tracker that did not make
the cut because they provided some duplicated functionality. Those are:
* cdr_sqlite3 (issue #6754, moy)
* cdr_sqlite3 (issue #8694, bsd)
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r58779 | file | 2007-03-11 20:51:16 -0400 (Sun, 11 Mar 2007) | 2 lines
Add matchexterniplocally setting which only substitutes your externip/externhost setting if it matches the localnet setting. I know of at least two people who need opposite settings, so I made it an option! (issue #8821 reported by kokoskarokoska)
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This can be done using a global variable or a dialplan function. Using the
SHELL() function will allow you to use an external script to determine what the
weight in the response should be. This can be very useful in load balancing
applications.
(inspired by discussions with blitzrage and jsmith in #asterisk-bugs)
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r58119 | russell | 2007-03-06 17:00:57 -0600 (Tue, 06 Mar 2007) | 3 lines
Clarify the documentation of the dialout and sendvoicemail options.
(issue #9000, caio1982 and serge-v)
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#9193 reported by junky)
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Thanks to Lacy Moore on the asterisk-users list for pointing out that this
was missing!
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r57364 | russell | 2007-03-01 17:42:53 -0600 (Thu, 01 Mar 2007) | 16 lines
Merge changes from svn/asterisk/team/russell/sla_updates
* Originally, I put in the documentation that only Zap interfaces would be
supported on the trunk side. However, after a discussion with Qwell, we came
up with a way to make IP trunks work as well, using some things already in
Asterisk. So, here it is, this now officially supports IP trunks.
* Update the SLA documentation to reflect how to setup IP trunks.
* Add a section in sla.txt that describes how to set up an SLA system with
voicemail.
* Simplify the way DTMF passthrough is handled in MeetMe.
* Fix a bug that exposed itself when using a Local channel on the trunk side
in SLA. The station's channel needs to be passed to the dial API when
dialing the trunk.
* Change a WARNING message to DEBUG in channel.h. This message is of no use
to users.
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r57207 | russell | 2007-02-28 17:01:52 -0600 (Wed, 28 Feb 2007) | 2 lines
minor tweaks to the sla docs
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r57203 | russell | 2007-02-28 16:07:05 -0600 (Wed, 28 Feb 2007) | 7 lines
Merge more changes from svn/asterisk/team/russell/sla_updates
* Add support for private hold. By setting "hold=private" for a trunk, only
the station that put the call on hold will be able to retrieve it from hold.
Also, by setting "hold=private" for a station, any call that station puts
on hold can only be retrieved by that station.
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r57144 | russell | 2007-02-28 13:56:20 -0600 (Wed, 28 Feb 2007) | 6 lines
Merge changes from svn/asterisk/team/russell/sla_updates
* Add support for the "barge=no" option for trunks. If this option is set,
then stations will not be able to join in on a call that is on progress
on this trunk.
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