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Previously only configurable globally. A unit test has also been written to
provide protection against parse failures for supported mailbox options.
(closes issue #16864)
Reported by: kobaz
Patches:
voicemail2.patch uploaded by kobaz (license 834)
Review: https://reviewboard.asterisk.org/r/555/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@254321 f38db490-d61c-443f-a65b-d21fe96a405b
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users expect them to work.
'core set debug' and 'core set verbose' can optionally change the
level for a specific filename; however, this is actually for a
specific source file name, not the module that source file is included
in. With examples like chan_sip, chan_iax2, chan_misdn and others
consisting of multiple source files, this will not lead to the
behavior that users expect. If they want to set the debug level for
chan_sip, they want it set for all of chan_sip, and not to have to
also set it for reqresp_parser and other files that comprise the
chan_sip module.
This patch changes this functionality to be module-name based instead
of file-name based.
To make this work, some Makefile modifications were required to ensure
that the AST_MODULE definition is present in each object file produced
for each module as well.
Review: https://reviewboard.asterisk.org/r/574/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@253917 f38db490-d61c-443f-a65b-d21fe96a405b
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The PITCH_SHIFT function can be used on a channel to independently
modify the pitch of both rx and tx audio streams. Now you can
improve your conference calls by assigning a random pitch effect
to everyone entering a meetme room, or just make your day more
interesting by making your co-workers sound funny. These are just
some of the numerious practical uses for this function. Enjoy!
https://reviewboard.asterisk.org/r/526/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@251038 f38db490-d61c-443f-a65b-d21fe96a405b
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New config parameter "reportalarms" added in chan_dahdi.conf which supports the
following possible values:
"channels": report each channel alarms (current behavior, default for backward compatibility)
"spans": report an "SpanAlarm" event when the span of any configured channel is alarmed
"all": report channel and span alarms (aggregated behavior)
"none": do not report any alarms
(closes issue #16709)
Reported by: nahuelgreco
Patches:
chan_dahdi.c.reportalarms.patch uploaded by nahuelgreco (license 162)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250392 f38db490-d61c-443f-a65b-d21fe96a405b
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Also fixed MODULEINFO depends and conflicts for app_fax, res_fax, and res_fax_spandsp.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250302 f38db490-d61c-443f-a65b-d21fe96a405b
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The 'p' option allows the PickupChan app to pickup
a ringing phone by looking for the first match to a
partial channel name rather than requiring a full match.
(closes issue #16613)
Reported by: syspert
Patches:
pickipbycallid.patch uploaded by syspert (license 938)
pickupbycallerid_v2.patch uploaded by dvossel (license 671)
Tested by: dvossel, syspert
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250141 f38db490-d61c-443f-a65b-d21fe96a405b
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VMSayName that will play the recorded name of the voicemail user if it exists,
otherwise will play the mailbox number. A unit test has been written to verify
correct functionality called test_voicemail_vmsayname.
(closes issue #14973)
Reported by: ghjm
Review: https://reviewboard.asterisk.org/r/530/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249889 f38db490-d61c-443f-a65b-d21fe96a405b
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The new JabberStatus event gives a concise view of the status change to the AMI
clients. Thanks fiddur!
(closes issue #16760)
Reported by: fiddur
Patches:
244498.2.diff uploaded by fiddur (license 678)
Tested by: fiddur, phsultan
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@247500 f38db490-d61c-443f-a65b-d21fe96a405b
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Current support for regex matching was previously only available on the group.
Also, error reporting for regex failures has been added. In addition to this
feature enhancement a unit test has been written to check the regular expression
logic to ensure the count operation is working as expected.
(closes issue #16642)
Reported by: kobaz
Patches:
groupmatch2.patch uploaded by kobaz (license 834)
Review: https://reviewboard.asterisk.org/r/503/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@247295 f38db490-d61c-443f-a65b-d21fe96a405b
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This feature allows for parkinglots to be created dynamically within
the dialplan. Thanks to all who were involved with getting this patch
written and tested!
(closes issue #15135)
Reported by: IgorG
Patches:
features.dynamic_park.v3.diff uploaded by IgorG (license 20)
2009090400_dynamicpark.diff.txt uploaded by mvanbaak (license 7)
dynamic_parkinglot.diff uploaded by dvossel (license 671)
Tested by: eliel, IgorG, acunningham, mvanbaak, zktech
Review: https://reviewboard.asterisk.org/r/352/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@247248 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@246382 f38db490-d61c-443f-a65b-d21fe96a405b
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The options are: parkedcallparking, parkedcallhangup, parkedcallrecording, and
parkedcalltransfers. Previously these options were only available for the
default parking lot.
(closes issue #16641)
Reported by: bluecrow76
Patches:
asterisk-1.6.2.1-features.c.diff uploaded by bluecrow76 (license 270)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@244598 f38db490-d61c-443f-a65b-d21fe96a405b
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Previously, we would parse GOSUB_RESULT, but not actually do anything with it.
Also, allow GOSUB_RETVAL to be inherited back across a peer/master channel.
(closes issue #16687)
Reported by: bklang
Patches:
app_dial-preserve-gosub_retval.patch uploaded by bklang (license 919)
(with modifications)
(closes issue #16686)
Reported by: bklang
Patches:
app_dial-respect-gosub_result.patch uploaded by bklang (license 919)
(with modifications)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@244393 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@243652 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16358)
Reported by: raarts
Patches:
lockconfdir.diff uploaded by raarts (license 937)
modified by me
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@243551 f38db490-d61c-443f-a65b-d21fe96a405b
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AGI servers.
(closes issue #14775)
Reported by: _brent_
Patches:
20091215__issue14775.diff.txt uploaded by tilghman (license 14)
hagi-5.patch uploaded by brent (license 388)
Tested by: _brent_
Reviewboard: https://reviewboard.asterisk.org/r/378/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@241188 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@241012 f38db490-d61c-443f-a65b-d21fe96a405b
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In asterisk.conf, transmit_silence_during_record has been removed
in favor of using only the transmit_silence option. The
transmit_silence_during_record option remains a valid option in
asterisk.conf, but has been removed from the sample config and
noted in CHANGES.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@240971 f38db490-d61c-443f-a65b-d21fe96a405b
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Part of the work done for connected line was to add an optional
argument to the 'f' option to allow for the connected party information
of the outgoing channel to be set to the argument provided. This was
overlooked during the merge of the work to trunk and is being added
back now. The CHANGES file has also been updated to note this change.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@237803 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@237802 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@236312 f38db490-d61c-443f-a65b-d21fe96a405b
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to the feature patch
(issue #16384)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@236306 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@236028 f38db490-d61c-443f-a65b-d21fe96a405b
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indications.conf
(closes issue #14504)
Reported by: alecdavis
Tested by: alecdavis,jsmith
Patch
app_dial.play_ring_indications.diff7.txt uploaded by alecdavis (license 585)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@235740 f38db490-d61c-443f-a65b-d21fe96a405b
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The option is global and currently the acceptable values as noted in the sample
config are accept or deny.
(closes issue #15228)
Reported by: lp0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@235342 f38db490-d61c-443f-a65b-d21fe96a405b
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New parameters ExtraContext, ExtraExtension, and ExtraPriority have been added
to redirect the second channel to a different location. Previously, it was only
possible to redirect both channels to the same place.
(closes issue #15853)
Reported by: haakon
Patches:
trunk-manager.c.patch uploaded by haakon (license 880)
Tested by: jpeeler
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@235265 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #15132)
Reported by: floletarmo
Patches:
voicemail_changes.patch uploaded by floletarmo (license 784)
(with some additional changes by me)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@234820 f38db490-d61c-443f-a65b-d21fe96a405b
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As described in the CHANGES file:
* MeetMe has a new option 'G' to play an announcement before joining a
conference.
* Page has a new option 'A(x)' which will playback an announcement
simultaneously to all paged phones (and optionally excluding the caller's one
using the new option 'n') before the call is bridged.
To add the new option to meetme, the conference flag options had to be extended
to 64 bits.
(closes issue #14365)
Reported by: dferrer
Patches:
page_announce.patch uploaded by dferrer (license 525)
modified by me
Review: https://reviewboard.asterisk.org/r/188/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@234173 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@234055 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@234053 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@234051 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@234028 f38db490-d61c-443f-a65b-d21fe96a405b
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This feature was listed as a 1.6.2 feature, even though it's in all 1.6.X
versions. The description of the feature was also no longer accurate.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@234008 f38db490-d61c-443f-a65b-d21fe96a405b
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be).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@233967 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #14352)
Reported by: fiddur
Patches:
trunk-14352-2.diff uploaded by phsultan (license 73)
Tested by: fiddur
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@233468 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@233235 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@233198 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16263)
Reported by: andrew
Patches:
pagerdate.patch uploaded by andrew (license 240)
(with a slight modification by me)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@232916 f38db490-d61c-443f-a65b-d21fe96a405b
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Previously only possible per context, new option called imapfolder.
(closes issue #14298)
Reported by: jablko
Patches:
patch-200906202 uploaded by jablko (license 675)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@232700 f38db490-d61c-443f-a65b-d21fe96a405b
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and 1.6.2
(closes issue #16212)
Reported by: miki
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@232657 f38db490-d61c-443f-a65b-d21fe96a405b
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configuration files.
This option can be used to enable #exec support in the asterisk.conf configuration file.
(closes issue #16260)
Reported by: atis
Patches:
exec_includes.patch uploaded by atis (license 242)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@232510 f38db490-d61c-443f-a65b-d21fe96a405b
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terminate recording.
(closes issue #15436)
Reported by: Vince
Patches:
app_record.diff uploaded by Vince (license 823)
Tested by: dbrooks
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@232442 f38db490-d61c-443f-a65b-d21fe96a405b
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r230964
(related to issue #14155)
Reported by: junky
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@231025 f38db490-d61c-443f-a65b-d21fe96a405b
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the ast_get_encoded_* functions.
* Add REPLACE function, which searches a given variable for a set of
characters and replaces each with a given character.
* Add PASSTHRU function, which passes a literal string back, like a NoOp for
functions. Intent is to be able to specify a literal string to another
function that takes a variable name as an argument.
* Let the array manipulation functions work with dialplan functions, in
addition to variables. This allows the array manipulation functions to
modify ASTDB and ODBC backends, assuming the func_odbc configuration has
both read and write functions.
(closes issue #15223)
Reported by: ajohnson
Patches:
20091112__issue15223.diff.txt uploaded by tilghman (license 14)
Tested by: lmadsen, tilghman
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@230994 f38db490-d61c-443f-a65b-d21fe96a405b
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(Closes AST-33)
Reviewboard: https://reviewboard.asterisk.org/r/368/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@230111 f38db490-d61c-443f-a65b-d21fe96a405b
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ChanIsAvail in the
AVAILCAUSECODE dialplan variable instead of overwriting the device state in AVAILSTATUS.
(closes issue #14426)
Reported by: macli
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@229970 f38db490-d61c-443f-a65b-d21fe96a405b
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Review: https://reviewboard.asterisk.org/r/426/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@229819 f38db490-d61c-443f-a65b-d21fe96a405b
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Updating the CHANGES file after noticing an email on the asterisk-dev mailing
list from Russell.
(issue #15874)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@229431 f38db490-d61c-443f-a65b-d21fe96a405b
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calculating the time of announcments from the end of the previous announcment rather than from the beginning.
(closes issue #15260)
Reported by: tonils
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@228947 f38db490-d61c-443f-a65b-d21fe96a405b
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Dial(DAHDI/g1[/extension[/options]])
Current options:
K(<keypad_digits>)
R Reverse charging indication (Collect calls)
The earlier Dial(DAHDI/g1[/K<keypad_digits>][/extension] format was
variable and did not allow for the easy addition of more options.
The earlier 'C' prefix character for reverse charge indiation would
conflict with the a-d DTMF digits if ISDN uses them.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@228691 f38db490-d61c-443f-a65b-d21fe96a405b
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