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

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r175029 | phsultan | 2009-02-12 11:16:21 +0100 (Thu, 12 Feb 2009) | 12 lines

Set the initiator attribute to lowercase in our replies when receiving calls.

This attribute contains a JID that identifies the initiator of the GoogleTalk
voice session. The GoogleTalk client discards Asterisk's replies if the 
initiator attribute contains uppercase characters.

(closes issue #13984)
Reported by: jcovert
Patches:
      chan_gtalk.2.patch uploaded by jcovert (license 551)
Tested by: jcovert

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If someone has configured the queue to play an position or holdtime
announcement, then it is odd and potentially unexpected to hear a 
"Thank you for your patience" sound when no position or holdtime
was actually announced.

This fixes the announcement so that the "thanks" sound is only played
in the case that a position or holdtime was actually announced.

There is a way that the "thank you" sound can be played without a
position or holdtime, and that is to set announce-frequency to a value
but keep announce-position and announce-holdtime both turned off.

(closes issue #14227)
Reported by: caspy
Patches:
      14227_v3.patch uploaded by putnopvut (license 60)
Tested by: caspy


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174948 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The 'd' option would not work for channel types which use RTP to transport
DTMF digits. The only way to allow for this to work was to answer the channel
if we saw that this option was enabled.

I realized that this may cause issues with CDRs, specifically with giving false
dispositions and answer times. I therefore modified ast_answer to take another
parameter which would tell if the CDR should be marked answered.

I also extended this to the Answer application so that the channel may be answered
but not CDRified if desired.

I also modified app_dictate and app_waitforsilence to only answer the channel if it
is not already up, to help not allow for faulty CDR answer times.

All of these changes are going into Asterisk trunk. For 1.6.0 and 1.6.1, however, all
the changes except for the change to the Answer application will go in since we do
not introduce new features into stable branches

(closes issue #14164)
Reported by: DennisD
Patches:
      14164.patch uploaded by putnopvut (license 60)
Tested by: putnopvut

Review: http://reviewboard.digium.com/r/145



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  r174885 | tilghman | 2009-02-11 14:54:18 -0600 (Wed, 11 Feb 2009) | 13 lines
  
  Restore a behavior that was recently changed, when we fixed issue #13962 and
  issue #13363 (related to issue #6176).  When a hangup occurs during a Macro
  execution in earlier 1.4, the h extension would execute within the Macro
  context, whereas it was always supposed to execute only within the main context
  (where Macro was called).  So this fix checks for an "h" extension in the
  deepest macro context where a hangup occurred; if it exists, that "h" extension
  executes, otherwise the main context "h" is executed.
  (closes issue #14122)
   Reported by: wetwired
   Patches: 
         20090210__bug14122.diff.txt uploaded by Corydon76 (license 14)
   Tested by: andrew
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We need to do this because while we know that the freeing of the channel may cause something to become
not in use we do not know this for sure. There may be another channel that is still up which would cause
it to be in use.
(closes issue #13238)
Reported by: kowalma
Patches:
      20090121__bug13238.diff.txt uploaded by Corydon76 (license 14)
Tested by: alecdavis


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174844 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When using the 'g' or 'e' options, the stack allocations that
were used could cause a stack overflow if a spyer stayed on the
line long enough without actually successfully spying on anyone.

The problem has been corrected by using static buffers and copying
the contents of the appropriate strings into them instead of using
functions like alloca or ast_strdupa



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174805 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The explanation behind this fix is a bit complicated, and I've already
typed it up in the code as a huge comment inside of manager.c, so I'll
give the abridged version here.

We needed a way to separate action-specific data from session-specific data.
Unfortunately, the only way to maintain API compatibility and to not have to
change every single manager action was to rename the current mansession structure
and wrap it inside a new mansession structure which actually contains action-
specific data.

(closes issue #14364)
Reported by: awk
Patches:
      14364_better.patch uploaded by putnopvut (license 60)
Tested by: putnopvut

Review: http://reviewboard.digium.com/r/148/



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(issue #13238)
Reported by: kowalma


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174710 65c4cc65-6c06-0410-ace0-fbb531ad65f3

improve slinfactory API to remove implicit sample rate and require explicit sample rate selection by creator of the slinfactory



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  r174644 | file | 2009-02-10 14:50:50 -0400 (Tue, 10 Feb 2009) | 6 lines
  
  Go off hold when we get an empty reinvite telling us to.
  (closes issue #14448)
  Reported by: frawd
  Patches:
        hold_invite_nosdp.patch uploaded by frawd (license 610)
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  r174583 | mnicholson | 2009-02-10 11:52:42 -0600 (Tue, 10 Feb 2009) | 18 lines
  
  Improve behavior of jitterbuffer when maxjitterbuffer is set.
  
  This change improves the way the jitterbuffer handles maxjitterbuffer and
  dramatically reduces the number of frames dropped when maxjitterbuffer is
  exceeded.  In the previous jitterbuffer, when maxjitterbuffer was exceeded, all
  new frames were dropped until the jitterbuffer is empty.  This change modifies
  the code to only drop frames until maxjitterbuffer is no longer exceeded.
  
  Also, previously when maxjitterbuffer was exceeded, dropped frames were not
  tracked causing stats for dropped frames to be incorrect, this change also
  addresses that problem.
  
  (closes issue #14044)
  Patches:
        bug14044-1.diff uploaded by mnicholson (license 96)
  Tested by: mnicholson
  Review: http://reviewboard.digium.com/r/144/
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(closes issue #14447)
Reported by: triccyx


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Make the logic for inuse and inringing manipluation match that of 1.4. The old broken logic would reset the values back to 0 during certain scenarios causing the wrong state to be reported.
(closes issue #14399)
Reported by: caspy
(issue #13238)
Reported by: kowalma


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from app_rpt.c.


(closes issue #14435)
Reported by: D_McNaul



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

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  r174369 | murf | 2009-02-09 19:27:40 -0700 (Mon, 09 Feb 2009) | 5 lines
  
  This patch solves some compiler complaints
  in both 32 and 64-bit environments.
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The ignore_hangup, run_dead, and noanswer flags were never initilized to zero causing hangups to never be issued.  If the external script expects to be notified of a hangup and never receives one, it runs indefinitely. 

(closes issue #14251)
Reported by: chris-mac
Tested by: dvossel



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

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r174282 | mmichelson | 2009-02-09 11:11:05 -0600 (Mon, 09 Feb 2009) | 12 lines

Don't do an SRV lookup if a port is specified

RFC 3263 says to do A record lookups on a hostname
if a port has been specified, so that's what we're
going to do. See section 4.2.

(closes issue #14419)
Reported by: klaus3000
Patches:
      patch_chan_sip_nosrvifport_1.4.23.txt uploaded by klaus3000 (license 65)


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  r174218 | file | 2009-02-09 10:48:21 -0400 (Mon, 09 Feb 2009) | 4 lines
  
  Don't overwrite our pointer to the music class when music on hold stops. We will use this if it starts again to see if we can resume the music where it left off.
  (closes issue #14407)
  Reported by: mostyn
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r174148 | russell | 2009-02-07 10:15:07 -0600 (Sat, 07 Feb 2009) | 2 lines

Fix a race condition that could cause a crash.

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r174082 | dhubbard | 2009-02-06 17:36:03 -0600 (Fri, 06 Feb 2009) | 5 lines

check ast_strlen_zero() before calling ast_strdupa() in sip_uri_headers_cmp()
and sip_uri_params_cmp()

The reporter didn't actually upload a properly-formed patch, instead a 
modified chan_sip.c file was uploaded.  I created a patch to determine the
changes, then modified the suggested changes to create a proper fix.  The
summary above is a complete description of the changes.

(closes issue #13547)
Reported by: tecnoxarxa
Patches:
      chan_sip.c.gz uploaded by tecnoxarxa (license 258)
Tested by: tecnoxarxa

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This is very similar to the DAHDI immediate=yes option.  When the phone is picked up, instead of giving a dialtone it connects directly to the "s" extension.  Changes where implemented in chan_iax2.c to directly connect to the "s" extension in the appropriate context when this option is enabled.  Examples explaining its use are added to iax2.conf.sample.  CHANGES has been updated as well. 

(closes issue #14266)
Reported by: jcovert
Patches:
      chan_iax2.c.patch-trunk uploaded by jcovert (license 551)
      iax.conf.sample.patch uploaded by jcovert (license 551)
Tested by: jcovert, dvossel
Review: http://reviewboard.digium.com/r/143/


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Don't subscribe to a mailbox on pseudo channels. It is futile. This solves an issue where duplicated pseudo channels would cause a crash because the first one would unsubscribe and the next one would also try to unsubscribe the same subscription.
(closes issue #14322)
Reported by: amessina


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  r173967 | file | 2009-02-06 13:14:15 -0400 (Fri, 06 Feb 2009) | 4 lines
  
  Some clients do not put the call-id for replaces at the beginning, so support it being anywhere in the string.
  (closes issue #14350)
  Reported by: fhackenberger
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  r173968 | file | 2009-02-06 13:15:01 -0400 (Fri, 06 Feb 2009) | 2 lines
  
  Remove a debug message I put in by accident.
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  r173917 | mnicholson | 2009-02-06 10:20:23 -0600 (Fri, 06 Feb 2009) | 7 lines
  
  Limit the addition of the Contact header in SIP responses according to various
  SIP RFCs.
  
  (closes issue #13602)
  Reported by: hjourdain
  Tested by: mnicholson
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Always detach and destroy the whisper and barge audiohooks. Additionally also allow an audiohook to be detached if it has not been attached.
(closes issue #14414)
Reported by: bluecrow76


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  r173900 | tilghman | 2009-02-06 09:43:32 -0600 (Fri, 06 Feb 2009) | 3 lines
  
  Backport OS X fix from trunk
  (AGAIN, closes issue #14360)
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This commit expands the Asterisk scheduler API to include a common implementation
of a scheduler context being processed by a dedicated thread.  chan_iax2 has been
updated to use this new code.  Also, as a result, this resolves some race
conditions related to the previous chan_iax2 scheduler handling.

Related to rev 171452 which resolved the same issues in 1.4.

Code from team/russell/sched_thread2

Review: http://reviewboard.digium.com/r/129/


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In trunk, the only necessary change pointed out was that the call
to ChanIsAvail uses an option that has been removed.

For the 1.6.1 branch, however, it appears that the sample file is
badly in need of updating since there are |'s used all over the place
there. My tentative plan is just to copy trunk's sample config file
to those branches since the info there is most up-to-date and should
be correct for use in 1.6.1

Thanks to macli in #asterisk-dev for bringing this up



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Properly set "seen" and "unseen" flags when moving messages from the new to the old folder when using IMAP for voicemail storage

(closes issue #13905)
Reported by: jaroth
Patches:
      foldermove_v2.patch uploaded by jaroth (license 50)


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r173770 | mmichelson | 2009-02-05 17:19:16 -0600 (Thu, 05 Feb 2009) | 20 lines

Fix logic regarding when to perform an SRV lookup for outgoing REGISTER requests

With this fix, we only will perform an SRV lookup at the following times:

* The first time we register with a remote registrar
* If we send a REGISTER but do not receive a response
* If the sendto() function returns an error

While I wrote the patch that fixes this issue, a huge amount of credit is due
to Brett Bryant, who wrote the initial patch on which I based this one.

(closes issue #12312)
Reported by: jrast
Patches:
      12312.patch uploaded by putnopvut (license 60)
Tested by: blitzrage

Review: http://reviewboard.digium.com/r/132/


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  r173696 | jpeeler | 2009-02-05 14:47:51 -0600 (Thu, 05 Feb 2009) | 12 lines
  
  Add new configuration option to make shared IMAP mailboxes function as expected.
  
  The new option is "imapvmshareid" which is an ID to tag multiple mailboxes
  using the same IMAP storage location to function as one mailbox. This allows
  all messages to be retrieved for any user in the group. The patch alters the
  'X-Asterisk-VM-Extension' header that is responsible for matching voicemails
  for a given user.
  
  (closes issue #13673)
  Reported by: howardwilkinson
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r173692 | mmichelson | 2009-02-05 14:29:09 -0600 (Thu, 05 Feb 2009) | 12 lines

Fix situations where queue members could be autopaused unexpectedly

Specifically, this patch prevents us from autopausing members when
we receive a busy or congestion frame from them.

(closes issue #14376)
Reported by: fiddur
Patches:
      14376.patch uploaded by putnopvut (license 60)
Tested by: fiddur


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r173592 | mmichelson | 2009-02-05 12:47:24 -0600 (Thu, 05 Feb 2009) | 3 lines

Add some missing cleanup to app_mixmonitor


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