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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


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268456 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #17234)
 Reported by: mav3rick


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https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  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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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

........
  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

........
  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

........
  r267971 | tilghman | 2010-06-04 11:27:02 -0500 (Fri, 04 Jun 2010) | 2 lines
  
  As-fixiate the build process
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267972 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267928 65c4cc65-6c06-0410-ace0-fbb531ad65f3

As signed linear audio data is accessed as 16-bit values, certain processors 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


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267877 65c4cc65-6c06-0410-ace0-fbb531ad65f3

From issue ABE-2247. RFC 3261 compliance for sections 13.2.24 and 17.1.1.2.

Review: https://reviewboard.asterisk.org/r/692/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267863 65c4cc65-6c06-0410-ace0-fbb531ad65f3

As signed linear audio data is accessed as 16-bit values, certain processors 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

........
  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
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267775 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This came up when using the sample configs, and just indicates expected behavior.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267714 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #17084)
 Reported by: falves11
 Patches: 
       issue17084_162_A.diff uploaded by falves11 (license 374)
 Tested by: falves11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267669 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267537 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267399 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267352 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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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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.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267303 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267138 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267097 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  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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This feature generates AMI events in the new aoc event class from the
events passed up by libpri.

Review:	https://reviewboard.asterisk.org/r/537/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267008 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Added ability to send and receive ETSI Explicit Call Transfer (ECT)
messages to eliminate tromboned calls.

Note: Asterisk already supported initiating the transfer of calls to
eliminate tromboned calls to libpri so there was nothing to do for the
asterisk portion.

Review:	https://reviewboard.asterisk.org/r/520/


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