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After 5 years in mantis and over a year on reviewboard, SRTP support is finally
being comitted. This includes generic CHANNEL dialplan functions that work for
getting the status of whether a call has secure media or signaling as defined
by the underlying channel technology and for setting whether or not a new
channel being bridged to a calling channel should have secure signaling or
media. See doc/tex/secure-calls.tex for examples.
Original patch by mikma, updated for trunk and revised by me.
(closes issue #5413)
Reported by: mikma
Tested by: twilson, notthematrix, hemanshurpatel
Review: https://reviewboard.asterisk.org/r/191/
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Also, optimize the allocation of mailboxes to avoid additional memory structures.
(closes issue #16320)
Reported by: Marquis
Patches:
20100525__issue16320.diff.txt uploaded by tilghman (license 14)
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Extract the SS7 specific code out of chan_dahdi like what was done to
ISDN/PRI and analog signaling. The new SS7 structures were modeled on
sig_pri.
The changes to sig_pri are an enhancement and a bug fix made possible
because SS7 was extracted.
1) The sig_pri TRANSFERCAPABILITY channel variable should have been set
unconditionally in sig_pri_new_ast_channel().
2) SS7/PRI transfer capability interaction in dahdi_new() fixed because of
SS7 extraction.
3) Module ref count error in dahdi_new() if startpbx failed to start the
PBX for some reason.
Review: https://reviewboard.asterisk.org/r/661/
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is expressed as a double, the last value would not also need to be a double.
(closes issue #15807)
Reported by: klaus3000
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(issue #17234)
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(closes issue #15685)
Reported by: david_s5
Patches:
dsp-silence-threshold-init.diff uploaded by dant (license 670)
issue15685.patch.v5 uploaded by pabelanger (license 224)
Tested by: danti
Review: https://reviewboard.asterisk.org/r/670/
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(closes issue #17237)
Reported by: pabelanger
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Suppress the warning about unexpected control subclass frames for
AST_CONTROL_CONNECTED_LINE, AST_CONTROL_REDIRECTING, and AST_CONTROL_AOC
in file.c:waitstream_core().
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the correct location for files in init.d.
(closes issue #16979)
Reported by: jw-asterisk
(issue #15691)
Reported by: itamarjp
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What I did not originally see in my previous commit was that even though the
next digit could be detected before the previous was considered ended, the
detection of the next digit effectively ends the detection of the previous.
Therefore, the length moves in lockstep with the digit, and no separate counter
is needed for the length alone.
(closes issue #17371)
Reported by: alecdavis
(closes issue #17474)
Reported by: kenner
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(closes issue #17234)
Reported by: mav3rick
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r268320 | tilghman | 2010-06-04 21:49:52 -0500 (Fri, 04 Jun 2010) | 3 lines
Rest In Peace
http://www.outandaboutnewspaper.com/article/4061
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option tweak
Changes.
1. RFC 3261 states in section 17.1.2.2 and 17.1.1.2 that retransmission
timers should initially be set to timer T1. T1 by default is 500ms.
Asterisk was starting the retransmission timers at T1*2 which shouldn't
cause any problems, but is not RFC compliant.
2. RFC 3261 states in section 17.1.2.2 that for a non-INVITE client transaction,
if the retransmit timer fires while in the proceeding state that
the request must be retransmitted. Asterisk currently ack's
requests for both INVITE and non-INVITE transactions when a
1XX response is received, this patch changes this for non-INVITE requests.
3. The 'registerattempts' option in sip.conf is supposed to set
how many registry attempts will be made before giving up. When
this option is set to 1, I would expect only one registry attempt
to be made before stopping because of a failure, but instead two are
made. In my opinion this is not expected behavior. This option does
not indicate that these are re-attempts. The logic behind this option
has been changed to only attempt registers the exact number of times
this option is set to. If this option is 0, it still continues to
re-attempt the registration forever.
Review: https://reviewboard.asterisk.org/r/687/
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r268126 | tilghman | 2010-06-04 15:41:24 -0500 (Fri, 04 Jun 2010) | 2 lines
AC_CONFIG_SUBDIRS has a bad side-effect on cross-compiles.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r268050 | tilghman | 2010-06-04 14:38:57 -0500 (Fri, 04 Jun 2010) | 6 lines
Build menuselect with the build environment's compiler, not the host (target)'s compiler.
(closes issue #17464)
Reported by: pprindeville
Tested by: tilghman
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r267971 | tilghman | 2010-06-04 11:27:02 -0500 (Fri, 04 Jun 2010) | 2 lines
As-fixiate the build process
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The problem would manifest itself if your dialplan matching could accept
more digits to match than were actually dialed. The time out waiting for
overlap digits disconnected the call instead of matching any accumulated
digits to the dialplan.
Accidental conversion of a break out of loop as a break out of switch.
(closes issue #17401)
Reported by: avalentin
Patches:
issue17401_digit_timeout.patch uploaded by rmudgett (license 664)
Tested by: avalentin, rmudgett
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require the values to be aligned in memory.
(closes issue #16912)
Reported by: michaelevdokimov
Patches:
asterisk.patch uploaded by michaelevdokimov (license 997)
Tested by: michaelevdokimov
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From issue ABE-2247. RFC 3261 compliance for sections 13.2.24 and 17.1.1.2.
Review: https://reviewboard.asterisk.org/r/692/
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require the values to be aligned in memory.
(closes issue #16912)
Reported by: michaelevdokimov
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r267759 | tilghman | 2010-06-03 20:16:26 -0500 (Thu, 03 Jun 2010) | 7 lines
Make the default install path appear to be /usr on Linux, instead of /usr/local.
Also, reorganize the options, so that they're more alphabetical.
(closes issue #17013)
Reported by: klaus3000
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This came up when using the sample configs, and just indicates expected behavior.
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(closes issue #17084)
Reported by: falves11
Patches:
issue17084_162_A.diff uploaded by falves11 (license 374)
Tested by: falves11
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Updated the UPGRADE.txt and CHANGES file stating that CDR records will not be explicity
written unless cdr.conf exists and is configured.
(closes issue #17373)
Reported by: wdoekes
Tested by: pabelanger
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The logic for handling generic PLC is now handled in ast_write in
channel.c instead of in translation code.
Review: https://reviewboard.asterisk.org/r/683/
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Add the ability to report waiting messages to ISDN endpoints (phones).
Relevant specification: EN 300 650 and EN 300 745
Review: https://reviewboard.asterisk.org/r/599/
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After some debugging, the random chan_h323 build failures appear to be due
to complications introduced by some chan_h323 specific build stuff getting
triggered during a clean. Simplify this by moving the h323 clean commands
down into channels/makefile.
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Add the ability to report malicious callers as an AMI event in the call
event class.
Relevant specification: EN 300 180
Review: https://reviewboard.asterisk.org/r/576/
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specified.
When ASTCFLAGS was specified with the make command, Makefile.rules was using
the specified value from the command line and not the one here, making it so this
flag would go missing.
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Add the ability to announce a call to an endpoint when there are no B
channels available. A call waiting call is a SETUP message with no B
channel selected.
Relevant specification: EN 300 056, EN 300 057, EN 300 058
For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
"no_media_path" option.
* Returns "0" if there is a B channel associated with the call.
* Returns "1" if no B channel is associated with the call. The call is
either on hold or is a call waiting call.
If you are going to allow incoming call waiting calls then you need to use
CHANNEL(no_media_path) do determine if you must drop a call to accept the
new call.
Review: https://reviewboard.asterisk.org/r/568/
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Review: https://reviewboard.asterisk.org/r/684/
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Two struct sockaddr_ins are created when applying directmedia
host access rules. The addresses of these are passed to the RTP
engine to be filled in. However, the RTP engine inspects the fields
of the structs before actually taking action. This inspection caused
valgrind to be a bit unhappy.
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Asterisk Generic AOC Representation
- Generic AOC encode/decode routines.
(Generic AOC must be encoded to be passed on the wire in the AST_CONTROL_AOC frame)
- AST_CONTROL_AOC frame type to represent generic encoded AOC data
- Manager events for AOC-S, AOC-D, and AOC-E messages
Asterisk App Support
- app_dial AOC-S pass-through support on call setup
- app_queue AOC-S pass-through support on call setup
AOC Unit Tests
- AOC Unit Tests for encode/decode routines
- AOC Unit Test for manager event representation.
SIP AOC Support
- Pass-through of generic AOC-D and AOC-E messages to snom phones via the
snom AOC specification.
- Creation of chan_sip page3 flags for the addition of the new
'snom_aoc_enabled' sip.conf option.
IAX AOC Support
- Natively supports AOC pass-through through the use of the new
AST_CONTROL_AOC frame type
DAHDI AOC Support
- ETSI PRI full AOC Pass-through support
- 'aoc_enable' chan_dahdi.conf option for independently enabling
pass-through of AOC-S, AOC-D, AOC-E.
- 'aoce_delayhangup' option for retrieving AOC-E on disconnect.
- DAHDI A() dial string option for requesting AOC services.
example usage:
;requests AOC-S, AOC-D, and AOC-E on call setup
exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e))
Review: https://reviewboard.asterisk.org/r/552/
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This changes the sample slinear frame data to contain non-zero data so that
translation calculations for speex works when preprocessing and VAD is turned
on. The encoder expects samples to be returned, but when attempted with the
mentioned two options and silent sample frames everything was discarded.
(closes issue #17240)
Reported by: seandarcy
Review: https://reviewboard.asterisk.org/r/682/
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r267009 | pabelanger | 2010-06-02 13:14:37 -0400 (Wed, 02 Jun 2010) | 7 lines
Cleanup error/warning messages in AEL2 parser
(closes issue #16684)
Reported by: Silmaril
Patches:
patch_ael2_logmsg.diff uploaded by Silmaril (license 979)
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