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press DTMF digits to switch between spying modes. Pressing 4 activates spy mode,
pressing 5 activates whisper mode, and pressing 6 activates barge mode. Use of
this feature overrides the normal operation of DTMF numbers.
This feature is courtesy of Switchvox.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116522 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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in text stream
Work sponsored by Omnitor AB, Stockholm, Sweden (http://www.omnitor.se)
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This is to enable video and text in originated calls. Development sponsored
by Omnitor AB, Sweden. (http://www.omnitor.se)
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marked "urgent" are considered to be higher priority than other messages
and so they will be played before any other messages in a user's mailbox.
There are two ways to leave an urgent message.
1. send the 'U' option to VoiceMail().
2. Set review=yes in voicemail.conf. This will give instructions for
a caller to mark a message as urgent after the message has been recorded.
I have tested that this works correctly with file and ODBC storage, and James
Rothenberger (who wrote initial support for this feature) has tested its use
with IMAP storage.
(closes issue #11817)
Reported by: jaroth
Based on branch http://svn.digium.com/svn/asterisk/team/jrothenberger/asterisk-urgent
Tested by: putnopvut, jaroth
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(closes issue #12090)
Reported by: jaroth
Patches:
vm-check-newpassword.diff.txt uploaded by mvanbaak (license 7)
20080509__bug12090.diff.txt uploaded by Corydon76 (license 14)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(Closes issue AST-34)
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will show the version of each library respectively.
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(closes issue #10540)
Reported by: spendergrass
Patches:
20080417__bug10540.diff.txt uploaded by Corydon76 (license 14)
Tested by: blitzrage
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This commit adds some new features to the SMDI_MSG_RETRIEVE() dialplan function.
Previously, this function only allowed searching by the forwarding station.
I have added some options to allow you to also search for messages in the queue
by the message desk terminal ID, as well as the message desk number.
This originally came up as a suggestion on the asterisk-dev mailing list.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115021 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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and denoising to a channel, AGC() and DENOISE(). Also included, is a change
to the audiohook API to add a new function (ast_audiohook_remove) that can
remove an audiohook from a channel before it is detached.
This code is based on a contribution from Switchvox.
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basis in the register line. This comes from a Switchvox patch. (issue AST-24)
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to announce-position, "limit" and "more," as well as a new option,
announce-position-limit. For more information on the use of these options,
see CHANGES or configs/queues.conf.sample.
(closes issue #10991)
Reported by: slavon
Patches:
app_q.diff uploaded by slavon (license 288)
Tested by: slavon, putnopvut
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the spied-on
party to be spoken instead of the channel name or number.
This was accomplished by adding a new function pointer to point to a function in app_voicemail
which retrieves the name file and plays it. This makes for an easy way that applications may play
a user's name should it be necessary. app_directory, in particular, can be simplified greatly by
this change.
This change comes as a suggestion from Switchvox, which already has this feature. AST-23
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barge on the call. It is like the existing whisper option, except that
it allows the spy to talk to both sides of the conversation on which
he is spying.
This feature has existed in Switchvox, and this merges the functionality
into Asterisk.
(AST-32)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114678 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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supplying
a custom client name. Using the channel name is still the default. This was done
at the request of Jared Smith.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114533 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Reported by: atis
Tested by: murf
This upgrade adds the ~~ (concatenation) string operator to expr2.
While not needed in normal runtime pbx operation, it is needed when
raw exprs are being syntax checked. This plays into future syntax-
unification plans. By permission of atis, this addition in trunk
and the reason of why things are as they are will suffice to close
this bug.
I also added a short note about the previous addition of "sip show sched"
to the CLI in CHANGES, which I discovered I forgot in a previous commit.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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a Meetme conference.
(closes issue #11691)
Reported by: junky
Patches:
meetme_info.patch uploaded by jpeeler (license 325)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114261 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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saw your e-mail.
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astobj2-ification of chan_sip. I've tested a number of scenarios like crazy. It used to have 4x the call setup/teardown performance of trunk, but now it's roughly at parity. I will attempt to find the bottlenecks and get it back to the 4x mark. The changes made were somewhat invasive, but the value to the community of these upgrades outweighs waiting further for more testing. Every change being made to chan_sip was lousing this code up when we tried to merge. Peers, Users, Dialogs, are all now astobj2 objects, indexed via hashtables. Refcounting is used to track objects and free them at the bitter end of their lives. Please file issues on bugs.digium.com, and PLEASE, please, please be patient. One natural advantage to all the hash-table work is that loading large sip.conf files full of thousands of peers now goes much faster. One more please: PLEASE help thrash this code and test it.
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eating up lots of cpu cycles. See CHANGES. From the team/murf/bug11210 branch.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114182 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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feature, and various documentation updates in astobj2.h, and the addition of standalone utility, refcounter, that will filter the trace output for unbalanced, unfreed objects. This comes from the team/murf/bug11210 branch.
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channel that is behind it when queried. This is useful for transfer scenarios as the actual channel will be transferred, not the Local channel. If you have been using Local channels as queue members and having issues when the agent did a blind transfer this option may solve the issue.
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for a given domain, the hostname and port listed in the SRV record will be used. If no SRV record exists or a SRV lookup is not attempted, the DNS lookup on the specified domain will be performed as normal. Chan_sip has been modified to take advantage of the new SRV support.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r110880 | kpfleming | 2008-03-26 09:42:35 -0700 (Wed, 26 Mar 2008) | 10 lines
Merged revisions 110869 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r110869 | kpfleming | 2008-03-26 08:53:46 -0700 (Wed, 26 Mar 2008) | 2 lines
due to licensing restrictions, we cannot distribute the source code for iLBC encoding and decoding... so remove it, and add instructions on how the user can obtain it themselves
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to be played upon completion of an attended transfer.
(closes issue #9239)
Reported by: sunder
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is subject to change while we work out the remaining issues.
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peer entry. It looks like there is support for DNS manager for realtime peers as well, however it is not implemented correctly. The improper usage occurs when ast_dnsmgr_lookup is called with one of the arguments being an address from the stack to be continually updated. The variable from the stack will go out of scope and dnsmgr will continue to try and update the memory there, causing possible stack corruption. This problem will be worked on next as well as adding DNS manager support for peer entries.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110087 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #7767)
Reported by: Corydon76
Patches:
20070314__simple_hint_lookup.diff.txt uploaded by Corydon76
pbx-trunk-98436.diff uploaded by plack (license 365)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109970 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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allow the list of periodic announcments specified to be played in a random
order instead of being played sequentially.
(closes issue #6681)
Reported by: alt_phil
Tested by: putnopvut
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109621 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #11959)
Reported by: mostyn
Patches:
peerstatus3.patch uploaded by mostyn (license 398)
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file parsing
(closes issue #9503)
Reported by: tzafrir
Patches:
fix_cleanups uploaded by tzafrir (license 46)
zapata_sections uploaded by tzafrir (license 46)
skipchannel_options uploaded by tzafrir (license 46)
conf_sample uploaded by tzafrir (license 46)
patches updated by me to better conform to coding guidelines and fix some problems
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108286 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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a variable from an Asterisk configuration file in the dialplan, or anywhere
else where dialplan functions can be used.
(Inspired by a discussion with Tilghman and Pari)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107787 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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an attended transfer over AMI
(closes issue #10585)
Reported by: ornati
Patches:
atxfer-trunk-r90428.diff uploaded by ornati (license 210)
(with modifications from me)
Tested by: putnopvut
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106236 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #11236)
Reported by: philipps
Patches:
20080218__bug11236.diff.txt uploaded by Corydon76 (license 14)
Tested by: philipps
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do some additional code cleanup and improvement in passing.
(closes issue #12106)
Reported by: nizon
Patches:
devstate-patch.txt uploaded by nizon (license 415)
-- Updated to trunk, and tab completion added by me
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(closes issue #11170)
Reported by: kratzers
Patches:
ResetCDR.1.diff uploaded by kratzers (license 307)
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changes are backward-compatible, so no changes to UPGRADE.txt are
necessary.
(closes issue #9279)
Reported by: rottenroddy
Patches:
20080125__bug9279.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76
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privilege to call out to a subshell.
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