This change builds upon the recent change to menuselect to add 'touch_on_change'
as an attribute of both categories and members. This should allow only the most
invasive defines to cause a complete rebuild, while defines which only affect a
subset of modules will only cause a rebuild of that smaller set.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246789 65c4cc65-6c06-0410-ace0-fbb531ad65f3
After further discussion with Steve Underwood, we should not (yet) be offering
to receive MMR or JBIG transcoded streams from T.38 endpoints. A future spandsp
release will support those features, and then they can be enabled during
negotiation
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245680 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@244393 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Add file information to data element of T event so
the file information is sent to the client when it is
interrupted. Previously only notification of pending
files that were dropped was sent
(closes issue #16147)
Reported by: thedavidfactor
Tested by: thedavidfactor
Review: https://reviewboard.asterisk.org/r/449/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240969 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r240414 | tilghman | 2010-01-15 14:52:27 -0600 (Fri, 15 Jan 2010) | 15 lines
Disallow leaving more than maxmsg voicemails.
This is a possibility because our previous method assumed that no messages are
left in parallel, which is not a safe assumption. Due to the vmu structure
duplication, it was necessary to track in-process messages via a separate
structure. If at some point, we switch vmu to an ao2-reference-counted
structure, which would eliminate the prior noted duplication of structures,
then we could incorporate this new in-process structure directly into vmu.
(closes issue #16271)
Reported by: sohosys
Patches:
20100108__issue16271.diff.txt uploaded by tilghman (license 14)
20100108__issue16271__trunk.diff.txt uploaded by tilghman (license 14)
20100108__issue16271__1.6.0.diff.txt uploaded by tilghman (license 14)
Tested by: jsutton
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240415 65c4cc65-6c06-0410-ace0-fbb531ad65f3
asterisk.conf's 'transmit_silence' option existed before
this patch, but was limited to only generating silence
while recording and sending DTMF. Now enabling the
transmit_silence option generates silence during wait
times as well.
To achieve this, ast_safe_sleep has been modified to
generate silence anytime no other generators are present
and transmit_silence is enabled. Wait apps not using
ast_safe_sleep now generate silence when transmit_silence
is enabled as well.
(closes issue #16524)
Reported by: kobaz
(closes issue #16523)
Reported by: kobaz
Tested by: dvossel
Review: https://reviewboard.asterisk.org/r/456/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239712 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When reporting hold time, the number of seconds should be mod 60.
Otherwise audio playback could be something like "2 minutes 123 seconds"
rather than "2 minutes 3 seconds".
Also, the "minute" sound file is missing, so for the moment until
that file can be created the "minutes" file is used instead.
(closes issue #16168)
Reported by: nickilo
Patches:
patch-unified-trunk-rev-222176 uploaded by nickilo (license )
Tested by: nickilo, wonderg
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237920 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@237803 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Also add an XXX comment that I'm baffled nobody has ever complained about. We
say "first message", and then we go into language-specific stuff where we
proceed to say..."first message".
(closes issue #15053)
Reported by: dinhtrung
Patches:
vietnamese.ods uploaded by dinhtrung (license 776)
app_voicemail.c.diff uploaded by dinhtrung (license 776)
(closes issue #15626)
Reported by: dinhtrung
Patches:
say.c.diff uploaded by dinhtrung (license 776)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237050 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r236509 | seanbright | 2009-12-28 07:43:36 -0500 (Mon, 28 Dec 2009) | 12 lines
Avoid a crash with large numbers of MeetMe conferences.
Similar to changes made to Queue(), when we have large numbers of conferences in
meetme.conf (1000s) and we use alloca()/strdupa(), we can blow out the stack and
crash, so instead just use a single fixed buffer.
(closes issue #16509)
Reported by: Kashif Raza
Patches:
20091223_16509.patch uploaded by seanbright (license 71)
Tested by: seanbright
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236510 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The QUEUE_MEMBER dialplan function can return total members,
logged-in members and "free" members count. A member is counted
as "free" immediately after his call ends, even though its wrap-up
time, if specified in queues.conf, has not yet expired, and the
queue will not actually route a call to it.
This Patch introduces a new "ready" option that only counts
free agents no longer in the wrap up time period.
(closes issue #16240)
Reported by: kkm
Patches:
appqueue-memberfun-readyoption-trunk.diff uploaded by kkm (license 888)
Tested by: kkm, dvossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236308 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The 'R' argument stops moh and indicates ringing once the agent is
ringing. This allows the person in the queue to know their call
is potentially about to be answered.
(closes issue #16384)
Reported by: haakon
Patches:
new_app_queue.c.patch uploaded by haakon (license 880)
Tested by: haakon, loloski, dvossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236304 65c4cc65-6c06-0410-ace0-fbb531ad65f3
so we should generate T38MaxBitRate of 14400 (even though that doesn't really
affect the FAX transmission much at all)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ReadExten already supported playing a tonezone from indications.conf.
It now has the ability to use a tonelist like 440+480/2000|0/4000
(closes issue #15185)
Reported by: jcovert
Patches:
app_readexten.c.patch uploaded by jcovert (license 551)
Tested by: qwell
Patch modified by me, to maintain backwards compatibility.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234776 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r234379 | jpeeler | 2009-12-11 16:37:21 -0600 (Fri, 11 Dec 2009) | 11 lines
Fix talking detection status after conference user is muted.
This patch ensures that when a conference user is muted that the accompanying
AMI Meetme talking off event is sent. Also, the meetme list output is updated
to show the muted user as unmonitored.
(closes issue #16247)
Reported by: dimas
Patches:
v3-16247.patch uploaded by dimas (license 88)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234380 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@234173 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Fix a couple of very minor bugs that prevent the socket client from working. The wrong set of properties were used in one place and the size of the address variable isn't set if the host name is an ip address. Also includes a fix for a bug that was introduced previously.
(closes issue #16121)
Reported by: thedavidfactor
Tested by: thedavidfactor
Review: https://reviewboard.asterisk.org/r/439/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233545 65c4cc65-6c06-0410-ace0-fbb531ad65f3
