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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@76755 f38db490-d61c-443f-a65b-d21fe96a405b
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place. (issue #8378 reported by jcovert)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@72354 f38db490-d61c-443f-a65b-d21fe96a405b
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and stop during message playback.
(closes issue 9474, reported and patched by jaroth with modifications by me)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@72329 f38db490-d61c-443f-a65b-d21fe96a405b
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using IMAP storage.
This feature may be turned on by adding imapgreetings=yes to the general section of voicemail.conf
voicemail.conf.sample has details on the options added.
As a result, IMAP storage now has RETRIEVE and DISPOSE macros defined.
In addition to the IMAP greeting changes, I also have added an enum for the voicemail folders
and so now the code should be easier to understand and maintain when it comes to this area.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@72232 f38db490-d61c-443f-a65b-d21fe96a405b
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address and port that RTP (be it audio/video/text) is going to.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@71988 f38db490-d61c-443f-a65b-d21fe96a405b
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allowing users to declare local variables.
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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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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@64870 f38db490-d61c-443f-a65b-d21fe96a405b
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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)
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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)
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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)
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entries in the queue log.
(issue #7561, reported and originally patched by fkasumovic, patch slightly
modified and updated to trunk by me)
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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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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: http://svn.digium.com/svn/asterisk/trunk@62267 f38db490-d61c-443f-a65b-d21fe96a405b
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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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of the content.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@61283 f38db490-d61c-443f-a65b-d21fe96a405b
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the event. This replaces
"ChannelDriver" and "Channel", previously used to indicate channel driver. ChannelType is more
in line with "core show channeltypes"
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@60987 f38db490-d61c-443f-a65b-d21fe96a405b
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configured in realtime.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@59414 f38db490-d61c-443f-a65b-d21fe96a405b
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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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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: http://svn.digium.com/svn/asterisk/trunk@58304 f38db490-d61c-443f-a65b-d21fe96a405b
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use directly using the ZT_GETVERSION ioctl. (issue #9094 reported by tootai)
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parked calls. (issue #9082 reported by francesco_r)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@54889 f38db490-d61c-443f-a65b-d21fe96a405b
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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.
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from the channel, and the 'X' option which allows the user to exit to a valid single digit extension. (issue #8137 reported by mnicholson)
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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
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mr_mehul_shah)
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ExternalIVR. (issue #8165 reported by mnicholson)
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RFC so we might want to support it and please don't hurt me Marko ... (issue #7812 reported by drorlb)
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