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

I had some work to do to port these changes to trunk; the 
check_expr stuff hasn't been updated here for quite some
time, it appears. I added some more tests to the check_expr2
suite. I had to play around with the makefile a bit, etc.

I added STANDALONE2 #ifdefs to ast_expr2.y so as not to
conflict structure with aelparse.

........
  r179807 | murf | 2009-03-03 11:11:34 -0700 (Tue, 03 Mar 2009) | 19 lines
  
  These changes allow AEL to better check ${} constructs within $[...], that are concatenated with text.
  
  I modified and added rules in ast_expr2.fl to better handle
  the concatenations.
  
  I added some default routines to ast_expr2.y so the standalone would
  compile. It also looks like I haven't run this thru bison since 2.1, so
  it's good to get this updated.
  
  The Makefile has comments added now for check_expr2 and check_expr to
  explain what they are for, and how to run them. 
  
  The testexpr2s stuff has been removed, in favor of check_expr2.
  
  expr2.testinput has been updated to include the two expressions
  that inspired these changes (from mcnobody on #asterisk this morning)
  The regression has been run and all looks well.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179973 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When app_meetme finds a realtime conference, it doesn't get the filename and fileformat correctly when 'r' is set.  Now app_meetme first checks to see if fileformat and filename are declared in the db, if they're not it checks the .conf file, if its not declared there either it then uses defaults. 

(closes issue #14545)
Reported by: dalbaech
Patches:
	app_meetme-realtime5.patch uploaded by dvossel (license 671)
	Realtime_Conference_Record_workaround.txt uploaded by dalbaech (license 705)
Tested by: dvossel, dalbaech
Review: http://reviewboard.digium.com/r/180/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179972 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This document specifies the timing modules available in Asterisk beginning
with Asterisk 1.6.1. The document goes into detail about the differences
between each and gives a general overview of what timing is used for in
Asterisk. There is also a section which can be used to help customize
your setup or to troubleshoot timing issues you may have.

I also added messages to the DAHDI timing test used in res_timing_dahdi.c
that points to this new documentation if people experience problems.

Big thanks to all who contributed comments on this.

(closes issue #14490)
Reported by: mmichelson
Patches:
      timing.txt uploaded by mmichelson (license 60)

Review: http://reviewboard.digium.com/r/164/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179937 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179903 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r179840 | file | 2009-03-03 14:27:09 -0400 (Tue, 03 Mar 2009) | 9 lines
  
  Do not assume that the bridge_cdr is still attached to the channel when the 'h' exten is finished executing.
  
  It is possible for a masquerade operation to occur when the 'h' exten is operating. This operation moves
  the CDR records around causing the bridge_cdr to no longer exist on the channel where it is expected to.
  We can not safely modify it afterwards because of this, so don't even try.
  
  (closes issue #14564)
  Reported by: meric
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179841 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In tests run after making this conversion, I noticed an approximate 85% 
reduction in memory usage for call file processing.

Review: http://reviewboard.digium.com/r/168/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179745 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
r179741 | russell | 2009-03-03 10:45:46 -0600 (Tue, 03 Mar 2009) | 6 lines

Ensure chan->fdno always gets reset to -1 after handling a channel fd event.

Since setting fdno to -1 had to be moved, a couple of other code paths that
do process an fd event return early and do not pass through the code path
where it was moved to.  So, set it to -1 in a few other places, too.

........


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179675 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r179671 | file | 2009-03-03 10:38:09 -0400 (Tue, 03 Mar 2009) | 3 lines
  
  Move where fdno is set to the default value to *after* the read callback of the channel driver is called.
  We have to do this as the underlying channel driver may need the fdno value to determine what to read.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179672 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
r179608 | russell | 2009-03-03 07:53:52 -0600 (Tue, 03 Mar 2009) | 9 lines

Make it easier to detect an improper call to ast_read().

When you call ast_waitfor() on a channel, the index into the channel fds array
that holds the file descriptor that poll() determines has input available is
stored in fdno.  This patch clears out this value after a call to ast_read()
and also reports errors if ast_read() is called without an fdno set.

From a discussion on the asterisk-dev list.

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179609 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r179536 | jpeeler | 2009-03-02 17:54:39 -0600 (Mon, 02 Mar 2009) | 15 lines
  
  Fix bridging regression from commit 176701
  
  This fixes a bad regression where the bridge would exit after an attended
  transfer was made. The problem was due to nexteventts getting set after the
  masquerade which caused the bridge to return AST_BRIDGE_COMPLETE.
  
  (closes issue #14315)
  Reported by: tim_ringenbach
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179537 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
r179532 | russell | 2009-03-02 17:34:13 -0600 (Mon, 02 Mar 2009) | 40 lines

Move ast_waitfor() down to avoid the results of the API call becoming stale.

This call to ast_waitfor() was being done way too soon in this section of code.
Specifically, there was code in between the call to waitfor and the code that
uses the result that puts the channel in autoservice.  By putting the channel
in autoservice, the previous results of ast_waitfor() become meaningless,
as the autoservice thread will do it's own ast_waitfor() and ast_read()
on the channel.

So, when we came back out of autoservice and eventually hit the block of code
that calls ast_read() on the channel, there may not actually be any input on
the channel available.  Even though the previous call to ast_waitfor() in
app_meetme said there was input, the autoservice thread has since serviced
the channel for some period of time.

This bug manifested itself while dvossel was doing some testing of MeetMe in
Asterisk trunk.  He was using the timerfd timing module.  When the code hit
ast_read() erroneously, it determined that it must have been called because of
input on the timer fd, as chan->fdno was set to AST_TIMING_FD, since that was 
the cause of the last legitimate call to ast_read() done by autoservice.  

In this test, an IAX2 channel was calling into the MeetMe conference.  It was
_much_ more likely to be seen with an IAX2 channel because of the way audio
is handled.  Every audio frame that comes in results in a call to
ast_queue_frame(), which then uses ast_timer_enable_continuous() to notify
the channel thread that a frame is waiting to be handled.  So, the chances
of ast_waitfor() indicating that a channel needs servicing due to a timer
event on an IAX2 event is very high.

Finally, it is interesting to note that if a different timing interface was
being used, this bug would probably not be noticed.  When ast_read() is called
and erroneously thinks that there is a timer event to handle, it calls the
ast_timer_ack() function.  The pthread and dahdi timing modules handle the
ack() function being called when there is no event by simply ignoring it.
In the case of the timerfd module, it results in a read() on the timer fd
that will block forever, as there is no data to read.  This caused Asterisk
to lock up very quickly.

Thanks to dvossel and mmichelson for the fun debugging session.  :-)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179533 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r179468 | tilghman | 2009-03-02 17:09:01 -0600 (Mon, 02 Mar 2009) | 10 lines
  
  When ending a recording with silence detection, remember to reduce the duration.
  The end of the recording is correspondingly trimmed, but the duration was not
  trimmed by the number of seconds trimmed, so the saved duration was necessarily
  longer than the actual soundfile duration.
  (closes issue #14406)
   Reported by: sasargen
   Patches: 
         20090226__bug14406.diff.txt uploaded by tilghman (license 14)
   Tested by: sasargen
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179469 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(found during a debugging session with dvossel and mmichelson.)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179465 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
r179461 | russell | 2009-03-02 16:58:18 -0600 (Mon, 02 Mar 2009) | 8 lines

Ensure that only one thread is calling ast_settimeout() on a channel at a time.

For example, with an IAX2 channel, you can have both the channel thread and the
chan_iax2 processing threads calling this function, and doing so twice at the
same time is a bad thing.

(Found in a debugging session with dvossel and mmichelson)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179462 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r179395 | qwell | 2009-03-02 14:14:57 -0600 (Mon, 02 Mar 2009) | 1 line
  
  Remove several silly warnings in editline.  One about a broken preprocessor directive, and another about strlcpy/strlcat.

  (closes issue #14264)
  Reported by: dimas
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179396 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179361 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Do not try to remove a registration scheduled item if the scheduler context has already been destroyed.

(closes issue #14580)
Reported by: alecdavis


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179323 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #14574)
Reported by: KNK
Patches:
      audiohook_volume_fix.diff uploaded by KNK (license 545)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179291 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This was pointed out by ScribbleJ in #asterisk-dev. Thanks very much, ScribbleJ!



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179254 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Previously, only the "data" field of the sip_pkt created during __sip_reliable_xmit 
was freed when XMIT_ERROR was returned by __sip_xmit. When retrans_pkt was called,
this inevitably resulted in the reading and writing of freed memory.

XMIT_ERROR is a condition meaning that we don't want to attempt resending the packet
at all. The proper action to take is to remove the scheduler entry we just created,
free the packet's data as well as the packet itself, and unlink it from the list of
packets on the sip_pvt structure.

(closes issue #14455)
Reported by: Nick_Lewis
Patches:
      14455.patch uploaded by mmichelson (license 60)
Tested by: Nick_Lewis



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179219 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179164 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(Closes issue #14563)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179161 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179154 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Special thanks goes to DEA who had to redo this patch twice
because we first put unload/load support in and later redid the way
we configure devices and lines.

(closes issue #10297)
Reported by: DEA
Patches:
      skinny-reload-trunkv2.diff uploaded by wedhorn (license 30)
      skinny-reload-trunk-v4.txt uploaded by DEA (license 3)
	  With mods by me based on feedback from wedhorn and Russell and seanbright
Tested by: DEA, mvanbaak, pj

Review: http://reviewboard.digium.com/r/130/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179122 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #14566)
Reported by: klaus3000
Patches:
      ANSWEREDTIME-1.4-patch.txt uploaded by klaus3000 (license 65)
      ANSWEREDTIME-trunk-patch.txt uploaded by klaus3000 (license 65)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179057 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In passing, also fix downloading SLIN16 extra sound packages.

(closes issue #14565)
Reported by: jtodd


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179021 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

In this case, it's just a matter of reducing the default timeouts from 2000
to 1000 msec, as the max def feature digit timeout is no longer halved.

........
  r178956 | murf | 2009-02-26 14:27:32 -0700 (Thu, 26 Feb 2009) | 18 lines
  
  This change moves the default feature digit timeout to 1000 ms from the previous default of 500.
  
  As per bug 14515, a dev discussion arrived at a "mediated concensus" 
  of a default feature digit timeout of 1.0 sec. Some voted for 1300;
  ctooley thought 1500 for distracted phone users in phone booths; 
  kpfleming put his foot down at 1.0 sec. 
  
  Users who found the previous default max delay of 250 msec perfect,
  are welcome to override the new default. Notice that I said that
  250 msec was the default; wait a minute, you might say, the config
  file said it was 500 msec!; well, because of the bug fix for 14515,
  we found that 500 msec was actually enforcing a max of 250. The bug
  fix would restore 500 msec, but we felt even that was a bit tight
  for most users... 2000 msec was pushed earlier by mmichelson, so
  that reduces to 1000 msec after the bug fix. Enjoy!
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178986 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #13826)
 Reported by: azielke
 Patches: 
       pickupsound2-trunk.patch uploaded by azielke (license 548)
       __20081124_bug_13826_updated.patch uploaded by lmadsen (license 10)
 Tested by: lmadsen


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178919 65c4cc65-6c06-0410-ace0-fbb531ad65f3

A return statement was missing which caused unexpected cli output.

issue #14479


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178871 65c4cc65-6c06-0410-ace0-fbb531ad65f3

These small fixes prevent compiler warnings with ubuntu 8.10's gcc-4.3.2, which tend to break my dev-mode build. Not a problem in 1.6.x.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178870 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........
  r178838 | dvossel | 2009-02-26 11:24:02 -0600 (Thu, 26 Feb 2009) | 9 lines
  
  IAX2 prune realtime fix
  
  Now prune_users() and prune_peers() are called instead of reload_config() to prune all users/peers that are realtime.  These functions remove all users/peers with the rtfriend and delme flags set. iax2_prune_realtime() also lacked the code to properly delete a single friend.  For example. if iax2 prune realtime <friend> was called, only the peer instance would be removed. The user would still remain.
  
  (closes issue #14479)
  Reported by: mousepad99
  Review: http://reviewboard.digium.com/r/176/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178848 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r178804 | murf | 2009-02-26 10:09:03 -0700 (Thu, 26 Feb 2009) | 28 lines
  
  This patch prevents the feature detection timeout from being cut in half.
  
  Because the ast_channel_bridge() call will return 0 and pass
  a frame pointer for both DTMF_BEGIN and DTMF_END, the feature_timer
  field in hte config struct is getting decremented twice, which 
  effectively cuts the digittimeout in half. I added conditions
  to the if statement to only let DTMF_END frames to flow thru,
  which solved the problem. Also, when the frame pointer is null,
  let control flow thru-- this usually happens on timeouts. I added
  a comment to the code to explain what's going on and why.
  
  Many thanks to sodom for reporting this problem. Personnally, it always seemed
  like something was wrong with the featuredigittimeout, but I never
  could quite decide what... and was too busy to investigate.
  This bug forced the issue, and now we know.
  
  Sodom had other issues in 14515, but I couldn't reproduce them. If
  he still has problems, and wants to get them solved, he is welcome
  to reopen 14515.
  
  
  (closes issue #14515)
  Reported by: sodom
  Patches:
        14515.patch uploaded by murf (license 17)
  Tested by: murf, sodom
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178828 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #14541)
Reported by: grant


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178801 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Iax2 prune realtime had issues.  If "iax2 prune realtime all" was called, it would appear like the command was successful, but in reality nothing happened.  This is because the reload that was supposed to take place checks the config files, sees no changes, and does nothing.  If there had been a change in the the config file, the realtime users would have been marked for deletion and everything would have been fine.  Now prune_users() and prune_peers() are called instead of reload_config() to prune all users/peers that are realtime.  These functions remove all users/peers with the rtfriend and delme flags set. iax2_prune_realtime() also lacked the code to properly delete a single friend.  For example. if iax2 prune realtime <friend> was called, only the peer instance would be removed. The user would still remain.  

(closes issue #14479)
Reported by: mousepad99
Review: http://reviewboard.digium.com/r/176/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178767 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #14558)
Reported by: alecdavis


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178764 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178733 65c4cc65-6c06-0410-ace0-fbb531ad65f3

another minor commit to test post-commit script changes (now testing post-revprop-change as well, third try)




git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178704 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178703 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........
  r178640 | murf | 2009-02-25 14:00:50 -0700 (Wed, 25 Feb 2009) | 17 lines
  
  This patch completes the fixes nec. to make 1.4 asterisk dialplan expressions ($[...]) 8-bit transparent
  
  While I was updating ast_expr2.fl, I missed one rule that would allow 8-bit chars to be caught
  in tokens; and in so doing, it absorbs the ${ sequence and messes up the
  checking of raw exprs by AEL.
  
  Trunk already has these changes.
  
  
  
  (closes issue #14543)
  Reported by: klaus3000
  Patches:
        patch.14543 uploaded by murf (license 17)
  Tested by: murf
........


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