except it lets you operate on a channel by name instead of conference member
number. It is very useful in combination with the 'X' option to ChanSpy.
(issue #9671, patch by mnicholson, with some small modifications by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63319 65c4cc65-6c06-0410-ace0-fbb531ad65f3
created will now be stored. Then, every channel that joins the conference will
have the MEETMEUNIQUEID channel variable set with this ID. This can be used to
relate callers that come and go from long standing conferences.
(issue #7295, patch by softins)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62794 65c4cc65-6c06-0410-ace0-fbb531ad65f3
NOT_INUSE or INUSE when Local channels are in use as opposed to just UNKNOWN.
It will still return INVALID if the extension doesn't exist at all.
(issue #8048, patch from tim_ringenbach)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62673 65c4cc65-6c06-0410-ace0-fbb531ad65f3
entries in the queue log.
(issue #7561, reported and originally patched by fkasumovic, patch slightly
modified and updated to trunk by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62462 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62292 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This introduces two new dialplan functions: DUNDIQUERY and DUNDIRESULT.
DUNDIQUERY lets you intitiate a DUNDi query from the dialplan. Then,
DUNDIRESULT will let you find out how many results there are, and access each
one without having to the query again.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62267 65c4cc65-6c06-0410-ace0-fbb531ad65f3
minimum amount of time between queue announcements for use when the caller's queue
position changes frequently.
(issue #9604, patch by Matthew Roth)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62242 65c4cc65-6c06-0410-ace0-fbb531ad65f3
"ChannelDriver" and "Channel", previously used to indicate channel driver. ChannelType is more
in line with "core show channeltypes"
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60987 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* 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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58866 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58304 65c4cc65-6c06-0410-ace0-fbb531ad65f3
T.140/RFC 2793 is a live communication channel, originally
created for IP based text phones for hearing impaired.
Feels very much like the old Unix talk application.
This code is developed and disclaimed by John Martin of Aupix, UK.
Tested for interoperability by myself and Omnitor in Sweden,
the company that wrote most of the specifications.
A big thank you to everyone involved in this.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54838 65c4cc65-6c06-0410-ace0-fbb531ad65f3
pretty cool things.
First, you can get the device state of anything in the dialplan:
NoOp(SIP/mypeer has state ${DEVSTATE(SIP/mypeer)})
NoOp(The conference room 1234 has state ${DEVSTATE(MeetMe:1234)})
Most importantly, this allows you to create custom device states so you can
control phone lamps directly from the dialplan.
Set(DEVSTATE(Custom:mycustomlamp)=BUSY)
...
exten => mycustomlamp,hint,Custom:mycustomlamp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54261 65c4cc65-6c06-0410-ace0-fbb531ad65f3
this, implementing locking of this list to make it thread-safe.
- Add a "redirect" option to http.conf that allows redirecting one URI to
another. I was inspired to do this while playing with the Asterisk GUI. I
got tired of typing this URL to get to the GUI:
http://localhost:8088/asterisk/static/config/cfgadvanced.html
So, now I have the following line in http.conf:
redirect=/=/asterisk/static/config/cfgadvanced.html
Now, I can type the following into my browser and go to the GUI:
http://localhost:8088
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48930 65c4cc65-6c06-0410-ace0-fbb531ad65f3
see queues.conf.sample for details.
* Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
setqueueentryvar options for each queue, see queues.conf.sample for details.
(#8216, jmls reported and submitted)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46369 65c4cc65-6c06-0410-ace0-fbb531ad65f3
application by configuring them in voicemail.conf (issue #7415, patch by
fkasumovic, with some fixes and documentation updates by myself)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46360 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Russell Bryant
Olle Johansson
Joshua Colp
Steve Murphy
BJ Weschke