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  r172438 | tilghman | 2009-01-29 16:54:29 -0600 (Thu, 29 Jan 2009) | 9 lines
  
  Lose the CAP_NET_ADMIN at every fork, instead of at startup.  Otherwise, if
  Asterisk runs as a non-root user and the administrator does a 'restart now',
  Asterisk loses the ability to set QOS on packets.
  (closes issue #14004)
   Reported by: nemo
   Patches: 
         20090105__bug14004.diff.txt uploaded by Corydon76 (license 14)
   Tested by: Corydon76
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*  Added doxygen comments to the major dahdi structures.
*  Fixed PRI and SS7 using an incorrect string value if the extension
delimiter is not present in the Dial() function.
*  Fixed SS7 not checking if the dialed extension is at least as long
as the stripmsd option.
*  Fixed PRI not handling unknown TON/NPI prefix letters correctly.
*  Fixed some uninitialized string variables on FXS ports.

configs/chan_dahdi.conf.sample
*  Updated some documentation.


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Also, implement a private cause code (as suggested by Tilghman). This works with
chan_sip, but doesn't propagate through chan_local.



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closes issue #14339)
Reported by: fiddur
Patches:
      realtime_pgsql.sql.diff uploaded by fiddur (license 678)

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- Also, change a function in app.c to return a userful value instead of always returning 0.

Patch by fnordian, changed by Corydon76 and myself.

This does not close the bug report, as fnordian had an additional change we're still discussing.

(related to issue #14059)
Reported by: fnordian
Patches: 
      chan_sip_hfield.patch uploaded by fnordian (license 110)
      20090116__bug14059.diff.txt uploaded by Corydon76 (license 14)
Tested by: fnordian, Corydon76, oej



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(closes issue #14185)
Reported by: Nick_Lewis
Patches: 
      chan_sip.c-expiryrequest6.patch uploaded by Nick (license 657)
Tested by: Nick_Lewis


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r172169 | oej | 2009-01-29 09:48:18 +0100 (Tor, 29 Jan 2009) | 16 lines

Make sure that we always add the hangupcause headers. In some cases, the owner was disconnected before we checked for the cause.
This patch implements a temporary storage in the pvt and use that instead.

The code is based on ideas from code from Adomjan in issue #13385 (Add support for Reason: header)
Thanks to Klaus Darillion for testing!

(closes issue #14294)
related to issue #13385

Reported by: klaus3000 and adomjan
Patches: 
      bug14294b.diff uploaded by oej (license 306)
      Based on 20080829_chan_sip.c-q850reason_header.patch uploaded by adomjan (license 487)
Tested by: oej, klaus3000


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(closes issue #14205)
 Reported by: maxgo
 Patches: 
       20090128__bug14205__5.diff.txt uploaded by Corydon76 (license 14)
 Tested by: blitzrage


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my gcc (Ubuntu 4.3.2-1ubuntu11) 4.3.2 didn't like the \%ld and issued a warning, breaking my dev-mode build. This fixes it.

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  r172030 | murf | 2009-01-28 11:51:16 -0700 (Wed, 28 Jan 2009) | 46 lines
  
  This patch fixes h-exten running misbehavior in manager-redirected 
  situations.
  
  What it does:
  1. A new Flag value is defined in include/asterisk/channel.h,
   AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the
   bridge hangup exten code not to run the h-exten there (nor
   publish the bridge cdr there). It will done at the pbx-loop
   level instead.
  2. In the manager Redirect code, I set this flag on the channel
   if the channel has a non-null pbx pointer. I did the same for the
   second (chan2) channel, which gets run if name2 is set...
   and the first succeeds.
  3. I restored the ending of the cdr for the pbx loop h-exten
   running code. Don't know why it was removed in the first place.
  4. The first attempt at the fix for this bug was to place code
     directly in the async_goto routine, which was called from a
     large number of places, and could affect a large number of
     cases, so I tested that fix against a fair number of transfer
     scenarios, both with and without the patch. In the process,
     I saw that putting the fix in async_goto seemed not to affect
     any of the blind or attended scenarios, but still, I was
     was highly concerned that some other scenarios I had not tested
     might be negatively impacted, so I refined the patch to 
     its current scope, and jmls tested both. In the process, tho,
     I saw that blind xfers in one situation, when the one-touch
     blind-xfer feature is used by the peer, we got strange 
     h-exten behavior.  So, I  inserted code to swap CDRs and
     to set the HANGUP_DONT field, to get uniform behavior.
  5. I added code to the bridge to obey the HANGUP_DONT flag,
     skipping both publishing the bridge CDR, and running
     the h-exten; they will be done at the pbx-loop (higher)
     level instead.
  6. I removed all the debug logs from the patch before committing.
  7. I moved the AUTOLOOP set/reset in the h-exten code in res_features
     so it's only done if the h-exten is going to be run. A very
     minor performance improvement, but technically correct.
  
  
  (closes issue #14241)
  Reported by: jmls
  Patches:
        14241_redirect_no_bridgeCDR_or_h_exten_via_transfer uploaded by murf (license 17)
  Tested by: murf, jmls
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  r171963 | tilghman | 2009-01-28 11:25:18 -0600 (Wed, 28 Jan 2009) | 2 lines
  
  Clarify log message (suggested by manxpower on #asterisk-dev)
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Add final part of previously committed work for answered elsewhere in queue - the missing piece that started with app_dial() earlier on.

This is to avoid having the list and counter of missed calls being touched by queue calls. Add the C option to queue() and nothing 
will be logged on phones that support the Reason: header on SIP cancel, like the SNOM phones.



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r171837 | oej | 2009-01-28 14:07:27 +0100 (Ons, 28 Jan 2009) | 2 lines

Add a better explanation of the difference between the device namespace and the dialplan for newbies.

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(closes issue #14301)
Reported by: amorsen

review: http://reviewboard.digium.com/r/128/


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r171689 | mmichelson | 2009-01-27 15:55:08 -0600 (Tue, 27 Jan 2009) | 39 lines

Fix devicestate problems for "always-on" agent channels

A revision to chan_agent attempted to "inherit" the device
state of the underlying channel in order to report the
device state of an agent channel more accurately.

The problem with the logic here is that it makes no sense to
use this for always-on agents. If the agent is logged in, then
to the underlying channel, the agent will always appear to be
"in use," no matter if the agent is on a call or not. The reason
is that to the underlying channel, the channel is currently in use
on a call to the AgentLogin application.

The most common cause that I found for this issue to occur was for
a SIP channel to be the underlying channel type for an Agent channel.
If the SIP phone re-registers, then the registration will cause the
device state core to query the device state of the SIP channel. Since the
SIP channel is in use, the Agent channel would also inherit this status.
Once the agent channel was set to "in use" there was no way that the device
state could change on that channel unless the agent logged out.

The solution for this problem is a bit different in 1.4 than it is in the
other branches. In 1.4, there will be a one-line fix to make sure that only
callback agents will inherit device state from their underlying channel type.
For the other branches of Asterisk, since callback support has been removed, there
is also no need for device state inheritance in chan_agent, so I will simply be
removing it from the code.

In addition, the 1.4 source is getting a new comment to help the next person who
edits chan_agent.c. I'm adding a comment that a agent_pvt's loginchan field may be
used to determine if the agent is a callback agent or not.

(closes issue #14173)
Reported by: nathan
Patches:
      14173.patch uploaded by putnopvut (license 60)
Tested by: nathan, aramirez


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r171621 | mmichelson | 2009-01-27 14:06:01 -0600 (Tue, 27 Jan 2009) | 18 lines

Prevent a crash from occurring when a jitter buffer interpolated frame is
removed from a slinfactory

slinfactory used the "samples" field of an ast_frame in order to determine
the amount of data contained within the frame. In certain cases, such as
jitter buffer interpolated frames, the frame would have a non-zero value for
"samples" but have NULL "data"

This caused a problem when a memcpy call in ast_slinfactory_read would attempt
to access invalid memory. The solution in use here is to never feed frames into
the slinfactory if they have NULL "data"

(closes issue #13116)
Reported by: aragon
Patches:
      13116.diff uploaded by putnopvut (license 60)


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The data passed to the end_bridge_callback was assumed to be data which was
still stack'd. The problem was that with some call features, attended transfers
in particular, a new bridge thread is started once the feature completes, meaning
that when the end_bridge_callback is called, the end_bridge_callback_data was
invalid.

To fix this problem, there are two measures taken

1. Instead of pointing to stacked data, we now used heap-allocated data for
passing to the end_bridge_callback in app_queue
2. Since bridges can end multiple times on a single logical call, we wait until
the final bridge is broken to actually set any queue variables. This is accomplished
through reference-counting and the use of an end_bridge_callback_data_fixup function
in app_queue.c

(closes issue #14260)
Reported by: ccesario
Patches:
      14260.patch uploaded by putnopvut (license 60)
Tested by: ccesario



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(issue #14104)
Reported by: alecdavis
Tested by: dbailey



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Merged revisions 171527 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r171527 | oej | 2009-01-27 15:33:20 +0100 (Tis, 27 Jan 2009) | 13 lines

Use the same branch tag in CANCEL as in INVITE

Originally putnopvut implemented some changes in revision 142079 that according to the bug report seemed to have worked then, but somehow fails now.
I guess code, as humans, get old and forget stuff. Anyway, this bug caused CANCEL not to work with picky systems. 

Thanks Fredrik for pointing out where the bug in the SIP messaging was.

(closes issue #14346)
Reported by: oej
Patches: 
      bug14346.diff uploaded by oej (license 306)
Tested by: oej

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r171452 | russell | 2009-01-26 15:31:59 -0600 (Mon, 26 Jan 2009) | 13 lines

Resolve some synchronization issues in chan_iax2 scheduler handling.

The important changes here are related to the synchronization between threads
adding items into the scheduler and the scheduler handling thread.  By adjusting
the lock and condition handling, we ensure that the scheduler thread sleeps no
longer and no less than it is supposed to.  We also ensure that it does not
wake up more often than it has to.

There is no bug report associated with this.  It is just something that I found
while putting scheduler thread handling into a reusable form (review 129).

Review: http://reviewboard.digium.com/r/131/ 

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Continue to move variables into the sip_cfg structure to make them easier to handle in the future as a group of settings for a group of devices. 
At some point, I want one sip_cfg per domain handled, so we can have "group" settings.



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Default setting is set before we activate the channel or at reloads, not where we declare the variable.



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r171264 | oej | 2009-01-26 13:51:53 +0100 (MÃ¥n, 26 Jan 2009) | 9 lines

Don't retransmit 401 on REGISTER requests when alwaysauthreject=yes

(closes issue #14284)
Reported by: klaus3000
Patches: 
      patch_chan_sip_unreliable_1.4.23_14284.txt uploaded by klaus3000 (license 65)
Tested by: klaus3000


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  r171187 | tilghman | 2009-01-25 17:44:01 -0600 (Sun, 25 Jan 2009) | 6 lines
  
  Correctly track the hookstate
  (closes issue #13686)
   Reported by: itiliti
   Patches: 
         20081013__bug13686.diff.txt uploaded by Corydon76 (license 14)
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  r171122 | tilghman | 2009-01-25 14:40:44 -0600 (Sun, 25 Jan 2009) | 2 lines
  
  Err, yeah.
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  r171120 | tilghman | 2009-01-25 14:30:41 -0600 (Sun, 25 Jan 2009) | 8 lines
  
  Add thread to kill zombies, when child processes don't die immediately on
  SIGHUP.
  (closes issue #13968)
   Reported by: eldadran
   Patches: 
         20090114__bug13968.diff.txt uploaded by Corydon76 (license 14)
   Tested by: eldadran
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