(Closes issue #14820)
Reported by: phsultan


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186297 65c4cc65-6c06-0410-ace0-fbb531ad65f3

A recent change made interactive vm_states no longer get
added to the list of vm_states and instead get stored in
thread-local storage.

In trunk and all the 1.6.X branches, the problem is that
when we search for messages in a voicemail box, we would
attempt to update the appropriate vm_state struct by directly
searching in the list of vm_states instead of using the
get_vm_state_by_imap_user function. This meant we could not
find the interactive vm_state that we wanted.

(closes issue #14685)
Reported by: BlargMaN
Patches:
      14685.patch uploaded by mmichelson (license 60)
Tested by: BlargMaN, qualleyiv, mmichelson



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186286 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
r186229 | russell | 2009-04-02 20:57:44 -0500 (Thu, 02 Apr 2009) | 21 lines

Fix a memory leak in cdr_radius.

I came across this while doing some testing of my ast_channel_ao2 branch.
After running a test overnight that generated over 5 million calls, Asterisk
had taken up about 1 GB of my system memory.  So, I re-ran the test with
MALLOC_DEBUG turned on.  However, it showed no leaks in Asterisk during the
test, even though Asterisk was still consuming it somehow.

Instead, I turned to valgrind, which when run with --leak-check=full, told
me exactly where the leak came from, which was from allocations inside the
radiusclient-ng library.  This explains why MALLOC_DEBUG did not report it.

After a bit of analysis, I found that we were leaking a little bit of memory
every time a CDR record was passed to cdr_radius.

I don't actually have a radius server set up to receive CDR records.  However,
I always have my development systems compile and install all modules.  In
addition to making sure there are not build errors across modules, always
loading modules helps find bugs like this, too, so it is strongly recommend for
all developers.

........


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........
  r186174 | mmichelson | 2009-04-02 16:55:34 -0500 (Thu, 02 Apr 2009) | 5 lines
  
  Fix instructions in one-step parking comment to make more sense.
  
  Changed a capital K to a lowercase k.
........


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........
  r186081 | kpfleming | 2009-04-02 12:21:29 -0500 (Thu, 02 Apr 2009) | 3 lines
  
  ensure that the buffer passed to DAHDI_SET_BUFINFO is fully initialized
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186101 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This API provides a generic way for multiple RTP stacks to be
integrated into Asterisk. Right now there is only one present, res_rtp_asterisk,
which is the existing Asterisk RTP stack. Functionality wise this commit
performs the same as previously. API documentation can be viewed in the
rtp_engine.h header file.

Review: http://reviewboard.digium.com/r/209/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186078 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

................
  r186059 | tilghman | 2009-04-02 12:09:13 -0500 (Thu, 02 Apr 2009) | 9 lines
  
  Merged revisions 186056 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.2
  
  ........
    r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02 Apr 2009) | 2 lines
    
    Fix for AST-2009-003
  ........
................


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........
  r186057 | tilghman | 2009-04-02 12:03:59 -0500 (Thu, 02 Apr 2009) | 2 lines
  
  Avoid multiple warning messages in SIP, due to this column not existing
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186058 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #14716)
 Reported by: sum
 Patches: 
       20090402__bug14716.diff.txt uploaded by tilghman (license 14)
 Tested by: sum


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186021 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r185952 | kpfleming | 2009-04-02 08:43:43 -0500 (Thu, 02 Apr 2009) | 5 lines
  
  the DAHDI_GETCONF, DAHDI_SETCONF and DAHDI_GET_PARAMS ioctls were recently corrected to show that they do, in fact, read data from userspace as part of their work. due to this fix, valgrind now reports a number of cases where chan_dahdi passed an uninitialized (or partially) buffer to these ioctls, which could lead to unexpected behavior.
  
  this patch corrects chan_dahdi to ensure that buffers passed to these ioctls are always fully initialized.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185953 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Included is a small bugfix to an ast_str helper, but most of these changes
are simply doxygen fixes.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185912 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r185845 | dvossel | 2009-04-01 14:02:00 -0500 (Wed, 01 Apr 2009) | 10 lines
  
  Fixes issue with dropped calles due to re-Invite glare and re-Invites never executing after a 491
  
  Acknowledgement for 491 responses were never being processed because it didn't match our pending invite's seqno.  Since the ACK was never processed, the 491 frame would continue to be retransmitted until eventually the call was dropped due to max retries.  Now during a pending invite, if we receive another invite, we send an 491 and hold on to that glare invite's seqno in the "glareinvite" variable for that sip_pvt struct.  When ACK's are received, we first check to see if it is in response to our pending invite, if not we check to see if it is in response to a glare invite.  In this case, it is in response to the glare invite and must be dealt with or the call is dropped.  I've changed the wait time for resending the re-Invite after receving a 491 response to comply with RFC 3261.  Before this patch the scheduled re-Invite would only change a flag indicating that the re-Invite should be sent out, now it actually sends it out as well. 
  
  (closes issue #12013)
  Reported by: alx
  
  Review: http://reviewboard.digium.com/r/213/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185846 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Use ast_debug and ast_softhangup_nolock.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185777 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
r185771 | russell | 2009-04-01 08:47:30 -0500 (Wed, 01 Apr 2009) | 6 lines

Fix a case where DTMF could bypass audiohooks.

This change fixes a situation where an audiohook that wants DTMF would not
actually get it.  This is in the code path where we end DTMF digit length
emulation while handling a NULL frame.

........


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185741 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #14695)
Reported by: mneuhauser
Patches:
      cause-for-hangup-manager-action.patch uploaded by mneuhauser (license 425)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185704 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185664 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185604 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r185599 | mmichelson | 2009-03-31 17:00:01 -0500 (Tue, 31 Mar 2009) | 6 lines
  
  Fix crash that would occur if an empty member was specified in queues.conf.
  
  (closes issue #14796)
  Reported by: pida
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185600 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch provides a number of optimizations to the stringfields API, focused around saving (not wasting) memory whenever possible. Thanks to Mark Michelson for inspiring this work and coming up with the first two optimizations that are represented here:

Changes:

- Cleanup of some code, fix incorrect doxygen comments

- When a field is emptied or replaced with a new allocation, decrease the amount of 'active' space in the pool it was held in; if that pool reaches zero active space, and is not the current pool, then free it as it is no longer in use

- When allocating a pool, try to allocate a size that will fit in a 'standard' malloc() allocation without wasting space

- When allocating space for a field, store the amount of space in the two bytes immediately preceding the field; this eliminates the need to call strlen() on the field when overwriting it, and more importantly it 'remembers' the amount of space the field has available, even if a shorter string has been stored in it since it was allocated

- Don't automatically double the size of each successive pool allocated; it's wasteful

http://reviewboard.digium.com/r/165/



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........
  r185531 | mmichelson | 2009-03-31 15:55:47 -0500 (Tue, 31 Mar 2009) | 3 lines
  
  Use AST_SCHED_DEL_SPINLOCK instead of manually using the logic.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185532 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r185468 | mmichelson | 2009-03-31 14:45:30 -0500 (Tue, 31 Mar 2009) | 8 lines
  
  Fix Russian voicemail intro to say the word "messages" properly.
  
  (closes issue #14736)
  Reported by: chappell
  Patches:
        voicemail_no_messages.diff uploaded by chappell (license 8)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185469 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In my tests that exercised full frame handling in chan_iax2, the version with
these changes took 30% to 40% of the CPU time compared to the same test of
Asterisk trunk before these modifications.

While doing some profiling for <http://reviewboard.digium.com/r/205/>,
one function that caught my eye was network_thread() in chan_iax2.c.
After the things that I was working on there, it was the next target
for analysis and optimization.  I used oprofile's source annotation
functionality and found that the loop traversing the frame queue in
network_thread() was to blame for the excessive CPU cycle consumption.

The frame_queue in chan_iax2 previously held all frames that either were
pending transmission or had been transmitted and are still pending
acknowledgment.

In network_thread(), the previous code would go back through the main
for loop after reading a single incoming frame or after being signaled
because a frame had been queued up for initial transmission.  In each
iteration of the loop, it traverses the entire frame queue looking for
frames that need to be transmitted.  On a busy server, this could easily
be quite a few entries.

This patch is actually quite simple.  The frame_queue has become only a list
of frames pending acknowledgment.  Frames that need to be transmitted are
queued up to a dedicated transmit thread via the taskprocessor API.

As a result, the code in network_thread() becomes much simpler, as its only
job is to read incoming frames.

In addition to the previously described changes, this patch includes some
additional changes to the frame_queue.  Instead of one big frame_queue, now
there is a list per call number to further reduce wasted list traversals.
The biggest impact of this change is in socket_process().

For additional details on testing and test results, see the review request.

Review: http://reviewboard.digium.com/r/212/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185432 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r185362 | dbrooks | 2009-03-31 11:37:12 -0500 (Tue, 31 Mar 2009) | 35 lines
  
  Fix incorrect parsing in chan_gtalk when xmpp contains extra whitespaces
  
  To drill into the xmpp to find the capabilities between channels, chan_gtalk 
  calls iks_child() and iks_next(). iks_child() and iks_next() are functions in 
  the iksemel xml parsing library that traverse xml nodes. The bug here is that 
  both iks_child() and iks_next() will return the next iks_struct node 
  *regardless* of type. chan_gtalk expects the next node to be of type IKS_TAG, 
  which in most cases, it is, but in this case (a call being made from the 
  Empathy IM client), there exists iks_struct nodes which are not IKS_TAG data 
  (they are extraneous whitespaces), and chan_gtalk doesn't handle that case, 
  so capabilities don't match, and a call cannot be made.
  
  iks_first_tag() and iks_next_tag(), on the other hand, will not return the 
  very next iks_struct, but will check to see if the next iks_struct is of 
  type IKS_TAG. If it isn't, it will be skipped, and the next struct of type 
  IKS_TAG it finds will be returned. This assures that chan_gtalk will find 
  the iks_struct it is looking for.
  
  This fix simply changes all calls to iks_child() and iks_next() to become 
  calls to iks_first_tag() and iks_next_tag(), which resolves the capability 
  matching.
  
  The following is a payload listing from Empathy, which, due to the extraneous 
  whitespace, will not be parsed correctly by iksemel:
  
  <iq from='dbrooksjab@235-22-24-10/Telepathy' to='astjab@235-22-24-10/asterisk' type='set' id='542757715704'> <session xmlns='http://www.google.com/session' initiator='dbrooksjab@235-22-24-10/Telepathy' type='initiate' id='1837267342'> <description xmlns='http://www.google.com/session/phone'> <payload-type clockrate='16000' name='speex' id='96'/>
   <payload-type clockrate='8000' name='PCMA' id='8'/>
   <payload-type clockrate='8000' name='PCMU' id='0'/>
   <payload-type clockrate='90000' name='MPA' id='97'/>
   <payload-type clockrate='16000' name='SIREN' id='98'/>
   <payload-type clockrate='8000' name='telephone-event' id='99'/>
  </description>
  </session>
  </iq>

Review: http://reviewboard.digium.com/r/181/
........


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........
  r185298 | mmichelson | 2009-03-31 10:34:05 -0500 (Tue, 31 Mar 2009) | 10 lines
  
  Fix some state_interface stuff that was in trunk but not in the backport to 1.4.
  
  Issue #14359 was fixed between the time that I posted the review of the backport
  of the state interface change for 1.4. This merges the changes from that issue
  back into 1.4.
  
  (closes issue #14359)
  Reported by: francesco_r
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185299 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #14672)
Reported by: makoto


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185261 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r185196 | file | 2009-03-31 11:06:39 -0300 (Tue, 31 Mar 2009) | 8 lines
  
  Fix crash when moving audiohooks between channels.
  
  Handle the scenario where we are called to move audiohooks between channels
  and the source channel does not actually have any on it.
  
  (closes issue #14734)
  Reported by: corruptor
........


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  r185121 | rmudgett | 2009-03-30 15:40:11 -0500 (Mon, 30 Mar 2009) | 1 line
  
  Update the channel allocation method documentation.
........


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........
  r185120 | rmudgett | 2009-03-30 15:38:11 -0500 (Mon, 30 Mar 2009) | 19 lines
  
  Make chan_misdn BRI TE side normally defer channel selection to the NT side.
  
  Channel allocation collisions are not handled by chan_misdn very well.
  This patch simply avoids the problem for BRI only.
  
  For PRI, allocation collisions are still possible but less likely since
  there are simply more channels available and each end could use a different
  allocation strategy.
  
  misdn.conf options available:
  te_choose_channel - Use to force the TE side to allocate channels.
  method - Specify the channel allocation strategy.
  
  (closes issue #13488)
  Reported by: Christian_Pinedo
  Patches:
        isdn_lib.patch.txt uploaded by crich
  Tested by: crich, siepkes, festr
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185122 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r185031 | mmichelson | 2009-03-30 11:17:35 -0500 (Mon, 30 Mar 2009) | 39 lines
  
  Fix queue weight behavior so that calls in low-weight queues are not inappropriately blocked.
  
  (This is copied and pasted from the review request I made for this patch)
  
  Asterisk has some odd behavior when queue weights are used. The current logic used when
  potentially calling a queue member is:
  
  If the member we are going to call is part of another queue and _that other queue has any 
  callers in it_ and has a higher weight than the queue we are calling from, then don't try 
  to contact that member. The issue here is what I have marked with underscores. If the 
  higher-weighted queue has any callers in it at all, then the queue member will be unreachable 
  from the lower-weighted queue. This has the potential to be really really bad if using a 
  queue strategy, such as leastrecent or fewestcalls, with the potential to call the same 
  member repeatedly.
  
  The fix proposed by garychen on issue 13220 is very simple and, as far as I can see, works 
  well for this situation. With this set of changes, the logic used becomes:
  
  If the member we are going to call is part of another queue, the other queue has a higher 
  weight than the queue we are calling from, and the higher weight queue has at least as many 
  callers as available members, then do not try to contact the queue member. If the higher 
  weighted queue has fewer callers than available members, then there is no reason to deny 
  the call to this member since the other queue can afford to spare a member.
  
  Since the fix involved writing a generic function for determining the number of available 
  members in the queue, I also modified the is_our_turn function to make use of the new 
  num_available_members function to determine if it is our turn to try calling a member. There 
  is one small behavior change. Before writing this patch, if you had autofill disabled, then 
  if you were the head caller in a queue, you would automatically be told that it was your 
  turn to try calling a member. This did not take into account whether there were actually any 
  queue members available to take the call. Now we actually make sure there is at least one 
  member available to take the call if autofill is disabled.
  
  (closes issue #13220)
  Reported by: garychen
  
  Review: http://reviewboard.digium.com/r/202/
........


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........
  r184980 | mmichelson | 2009-03-30 10:23:59 -0500 (Mon, 30 Mar 2009) | 22 lines
  
  Backport state interface changes to app_queue from trunk.
  
  After several issues raised on the Asterisk bugtracker against
  the 1.4 branch were determined to be fixable with the state interface
  change available in the 1.6.X series, it finally came time to just
  suck it up and backport the change.
  
  For a detailed explanation of what this change entails, the original
  trunk commit for this feature may be found here:
  
  http://svn.digium.com/view/asterisk?view=revision&revision=97203
  
  In addition, the details for the use of this change to fix the problems
  stated in issue #12970 may be found in the review request I made for
  this change. It is linked below.
  
  (closes issue #12970)
  Reported by: edugs15
  
  Review: http://reviewboard.digium.com/r/116
........


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........
  r184947 | file | 2009-03-30 11:35:47 -0300 (Mon, 30 Mar 2009) | 14 lines
  
  Improve our handling of T38 in the initial INVITE from a device.
  
  We now answer with matching media streams to what is requested. If an INVITE
  is received with both a T38 and RTP media stream this means we answer with both.
  For any outgoing calls created as a result of this inbound one no T38 is requested
  in the initial INVITE. Instead if we start receiving udptl packets we trigger a
  reinvite on the outbound side.
  
  (closes issue #12437)
  Reported by: marsosa
  Tested by: pinga-fogo, okrief, file, afu
  
  Review: http://reviewboard.digium.com/r/208/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184948 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(reported by cai1982 in #asterisk-dev)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184910 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
r184842 | russell | 2009-03-29 00:51:55 -0500 (Sun, 29 Mar 2009) | 5 lines

Ensure targs variable is fully initialized.

(closes issue #14758)
Reported by: tim_ringenbach

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184843 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #14715)
Reported by: jthurman
Patches:
      h323-makefile-1.6.2.0-beta1.patch uploaded by jthurman (license 614)
Tested by: tzafrir, russell


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184838 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #14765)
Reported by: timeshell
Patches:
      app_ices.svn-1.6.0.diff uploaded by timeshell (license 399)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184801 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The doxygen documentation has now been updated to state explicitly that I want
punctuation atthe end of the first sentence in a commit message. :).


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184798 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The ability to load/unload timing interfaces is nice, but it means that when a timer is allocated, it may come from provider A, but later provider B becomes the 'preferred' provider. If this happens, all timer API calls on the timer that was provided by provider A will actually be handed to provider B, which will say WTF and return an error.

This patch changes the timer API to include a pointer to the provider of the timer handle so that future operations on the timer will be forwarded to the proper provider.

(closes issue #14697)
Reported by: moy

Review: http://reviewboard.digium.com/r/211/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184762 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184726 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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