Commit Graph

16927 Commits

Author SHA1 Message Date
Joshua Colp
75d4f7e1c2 Per kpfleming add a note describing why you must never change the first element of peer_finding_info.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-29 14:52:22 +00:00
Eliel C. Sardanons
f2a3836106 Blocked revisions 166823 via svnmerge
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  r166823 | eliel | 2008-12-28 10:36:25 -0500 (Sun, 28 Dec 2008) | 3 lines
  
  Fix a typo in the XML documentation of the AUDIOHOOK_INHERIT dialplan function.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166831 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-28 15:41:44 +00:00
Russell Bryant
941a82916a Merged revisions 166773 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r166773 | russell | 2008-12-28 09:15:14 -0600 (Sun, 28 Dec 2008) | 12 lines

Merged revisions 166772 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r166772 | russell | 2008-12-28 09:13:48 -0600 (Sun, 28 Dec 2008) | 4 lines

Use strncat() instead of an sprintf() in which source and target buffers overlap

http://lists.digium.com/pipermail/asterisk-dev/2008-December/035919.html

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166775 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-28 15:16:11 +00:00
Terry Wilson
869388718f Blocked revisions 166731 via svnmerge
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  r166731 | twilson | 2008-12-24 09:10:42 -0600 (Wed, 24 Dec 2008) | 10 lines
  
  There is no section 22.2.2 in rfc 3261.  I believe 26.2.2 is what was meant:
  
        Note that in the SIPS URI scheme, transport is independent of TLS,
        and thus "sips:alice@atlanta.com;transport=tcp" and
        "sips:alice@atlanta.com;transport=sctp" are both valid (although
        note that UDP is not a valid transport for SIPS).  The use of
        "transport=tls" has consequently been deprecated, partly because
        it was specific to a single hop of the request.  This is a change
        since RFC 2543.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166735 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-24 15:19:33 +00:00
Steve Murphy
a7aeaf341b Merged revisions 166665 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

This merged from trunk with no conflicts. I tested
mostly the 'tired' cases, and for the most part
ignored the tests for reconnecting and dialing in
to fetch a parked call, after the first case.

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  r166665 | murf | 2008-12-23 11:13:49 -0700 (Tue, 23 Dec 2008) | 153 lines
  
  Merged revisions 166093 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  In order to merge this 1.4 patch into trunk,
  I had to resolve some conflicts and wait for
  Russell to make some changes to res_agi.
  I re-ran all the tests; 39 calls in all, and
  made fairly careful notes and comparisons: I
  don't want this to blow up some aspect of 
  asterisk; I completely removed the KEEPALIVE
  from the pbx.h decls. The first 3 scenarios
  involving feature park; feature xfer to 700;
  hookflash park to Park() app call all behave
  the same, don't appear to leave hung channels,
  and no crashes.
  
  ........
    r166093 | murf | 2008-12-19 15:30:32 -0700 (Fri, 19 Dec 2008) | 131 lines
    
    This merges the masqpark branch into 1.4
    
    These changes eliminate the need for (and use of)
    the KEEPALIVE return code in res_features.c;
    There are other places that use this result code
    for similar purposes at a higher level, these appear
    to be left alone in 1.4, but attacked in trunk.
    
    The reason these changes are being made in 1.4, is
    that parking ends a channel's life, in some situations,
    and the code in the bridge (and some other places),
    was not checking the result code properly, and dereferencing
    the channel pointer, which could lead to memory corruption
    and crashes.
    
    Calling the masq_park function eliminates this danger 
    in higher levels.
    
    A series of previous commits have replaced some parking calls
    with masq_park, but this patch puts them ALL to rest,
    (except one, purposely left alone because a masquerade
    is done anyway), and gets rid of the code that tests
    the KEEPALIVE result, and the NOHANGUP_PEER result codes.
    
    While bug 13820 inspired this work, this patch does
    not solve all the problems mentioned there.
    
    I have tested this patch (again) to make sure I have
    not introduced regressions. 
    
    Crashes that occurred when a parked party hung up
    while the parking party was listening to the numbers
    of the parking stall being assigned, is eliminated.
    
    These are the cases where parking code may be activated:
    
    1. Feature one touch (eg. *3)
    2. Feature blind xfer to parking lot (eg ##700)
    3. Run Park() app from dialplan (eg sip xfer to 700)
       (eg. dahdi hookflash xfer to 700)
    4. Run Park via manager.
    
    The interesting testing cases for parking are:
    I. A calls B, A parks B
        a. B hangs up while A is getting the numbers announced.
        b. B hangs up after A gets the announcement, but 
           before the parking time expires
        c. B waits, time expires, A is redialed,
           A answers, B and A are connected, after
           which, B hangs up.
        d. C picks up B while still in parking lot.
    
    II. A calls B, B parks A
        a. A hangs up while B is getting the numbers announced.
        b. A hangs up after B gets the announcement, but 
           before the parking time expires
        c. A waits, time expires, B is redialed,
           B answers, A and B are connected, after
           which, A hangs up.
        d. C picks up A while still in parking lot.
    
    Testing this throroughly involves acting all the permutations
    of I and II, in situations 1,2,3, and 4.
    
    Since I added a few more changes (ALL references to KEEPALIVE in the bridge
    code eliimated (I missed one earlier), I retested
    most of the above cases, and no crashes.
    
    H-extension weirdness.
    
    Current h-extension execution is not completely
    correct for several of the cases.
    
    For the case where A calls B, and A parks B, the
    'h' exten is run on A's channel as soon as the park
    is accomplished. This is expected behavior.
    
    But when A calls B, and B parks A, this will be
    current behavior:
    
    After B parks A, B is hung up by the system, and
    the 'h' (hangup) exten gets run, but the channel
    mentioned will be a derivative of A's...
    
    Thus, if A is DAHDI/1, and B is DAHDI/2,
    the h-extension will be run on channel
    Parked/DAHDI/1-1<ZOMBIE>, and the 
    start/answer/end info will be those 
    relating to Channel A.
    
    And, in the case where A is reconnected to
    B after the park time expires, when both parties
    hang up after the joyful reunion, no h-exten
    will be run at all.
    
    In the case where C picks up A from the 
    parking lot, when either A or C hang up,
    the h-exten will be run for the C channel.
    
    CDR's are a separate issue, and not addressed
    here.
    
    As to WHY this strange behavior occurs, 
    the answer lies in the procedure followed
    to accomplish handing over the channel
    to the parking manager thread. This procedure
    is called masquerading. In the process,
    a duplicate copy of the channel is created,
    and most of the active data is given to the
    new copy. The original channel gets its name
    changed to XXX<ZOMBIE> and keeps the PBX
    information for the sake of the original
    thread (preserving its role as a call 
    originator, if it had this role to begin
    with), while the new channel is without
    this info and becomes a call target (a
    "peer").
    
    In this case, the parking lot manager
    thread is handed the new (masqueraded)
    channel. It will not run an h-exten
    on the channel if it hangs up while
    in the parking lot. The h exten will
    be run on the original channel instead,
    in the original thread, after the bridge
    completes.
    
    See bug 13820 for our intentions as
    to how to clean up the h exten behavior.
  
  Review: http://reviewboard.digium.com/r/29/
  
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166730 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-24 01:15:43 +00:00
Tilghman Lesher
827f655dac Merged revisions 166696 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r166696 | tilghman | 2008-12-23 14:47:08 -0600 (Tue, 23 Dec 2008) | 7 lines
  
  Allow semicolons and extended characters in user-specified SIP headers.
  (closes issue #14110)
   Reported by: gork
   Patches: 
         20081222__bug14110__2.diff.txt uploaded by Corydon76 (license 14)
   Tested by: gork, putnopvut
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166698 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-23 20:56:10 +00:00
Mark Michelson
63f2d52453 Merged revisions 166569 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r166569 | mmichelson | 2008-12-23 09:17:54 -0600 (Tue, 23 Dec 2008) | 20 lines

Merged revisions 166568 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r166568 | mmichelson | 2008-12-23 09:16:26 -0600 (Tue, 23 Dec 2008) | 12 lines

Fix a crash resulting from a datastore with inheritance but no duplicate callback

The fix for this is to simply set the newly created datastore's data pointer
to NULL if it is inherited but has no duplicate callback.

(closes issue #14113)
Reported by: francesco_r
Patches:
      14113.patch uploaded by putnopvut (license 60)
Tested by: francesco_r


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166571 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-23 15:20:15 +00:00
Tilghman Lesher
9981739dd1 Merged revisions 166533 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r166533 | tilghman | 2008-12-22 22:32:15 -0600 (Mon, 22 Dec 2008) | 11 lines
  
  Merged revisions 166509 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r166509 | tilghman | 2008-12-22 22:05:25 -0600 (Mon, 22 Dec 2008) | 4 lines
    
    Use the integer form of condition for integer comparisons.
    (closes issue #14127)
     Reported by: andrew
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166535 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-23 04:34:13 +00:00
Mark Michelson
2ff6f18737 Merged revisions 166470 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r166470 | mmichelson | 2008-12-22 17:25:34 -0600 (Mon, 22 Dec 2008) | 11 lines

Always use the value of the AGISIGHUP when running an AGI.

Prior to this patch, the value of AGISIGUP was not always
honored when set on a channel.

(closes issue #13711)
Reported by: fmueller
Patches:
      13711.patch uploaded by putnopvut (license 60)


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166472 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-22 23:27:37 +00:00
Mark Michelson
25342b814f Merged revisions 166382 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r166382 | mmichelson | 2008-12-22 15:08:03 -0600 (Mon, 22 Dec 2008) | 44 lines

Merged revisions 166380 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r166380 | mmichelson | 2008-12-22 14:56:29 -0600 (Mon, 22 Dec 2008) | 36 lines

Fix a deadlock relating to channel locks and autoservice

It has been discovered that if a channel is locked prior
to a call to ast_autoservice_stop, then it is likely that
a deadlock will occur. The reason is that the call to 
ast_autoservice_stop has a check built into it to be sure
that the thread running autoservice is not currently trying
to manipulate the channel we are about to pull out of 
autoservice.

The autoservice thread, however, cannot advance beyond where
it currently is, though, because it is trying to acquire
the lock of the channel for which autoservice is attempting
to be stopped.

The gist of all this is that a channel MUST NOT be locked
when attempting to stop autoservice on the channel.

In this particular case, the channel was locked by a call
to ast_read. A call to ast_exists_extension led to autoservice
being started and stopped due to the existence of dialplan
switches.

It may be that there are future commits which handle the same
symptoms but in a different location, but based on my looks through
the code, it is very rare to see a construct such as this one.

(closes issue #14057)
Reported by: rtrauntvein
Patches:
      14057v3.patch uploaded by putnopvut (license 60)
Tested by: rtrauntvein

Review: http://reviewboard.digium.com/r/107/


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166440 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-22 21:56:36 +00:00
Russell Bryant
80159b17cc Merged revisions 166436 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r166436 | russell | 2008-12-22 15:45:28 -0600 (Mon, 22 Dec 2008) | 2 lines

Cosmetic change - don't mix struct initializer styles.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166438 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-22 21:46:53 +00:00
Russell Bryant
4df667e276 Merged revisions 166377 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r166377 | russell | 2008-12-22 14:26:48 -0600 (Mon, 22 Dec 2008) | 2 lines

Fix a bad typo.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166379 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-22 20:28:12 +00:00
Russell Bryant
511692bc3d Merged revisions 166342 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r166342 | russell | 2008-12-22 11:44:23 -0600 (Mon, 22 Dec 2008) | 2 lines

Remove some error messages.  This is the default handler that is valid to use.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166343 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-22 17:46:05 +00:00
Russell Bryant
3d6899f584 Merged revisions 166317 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r166317 | russell | 2008-12-22 11:29:10 -0600 (Mon, 22 Dec 2008) | 10 lines

Merged revisions 166297 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r166297 | russell | 2008-12-22 11:22:56 -0600 (Mon, 22 Dec 2008) | 2 lines

Fix up timeout handling in ast_carefulwrite().

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166326 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-22 17:32:14 +00:00
Russell Bryant
3db74e5bd7 Merged revisions 166282 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r166282 | russell | 2008-12-22 11:09:36 -0600 (Mon, 22 Dec 2008) | 12 lines

Introduce ast_careful_fwrite() and use in AMI to prevent partial writes.

This patch introduces a function to do careful writes on a file stream which
will handle timeouts and partial writes.  It is currently used in AMI to
address the issue that has been reported.  However, there are probably a few
other places where this could be used.

(closes issue #13546)
Reported by: srt
Tested by: russell
http://reviewboard.digium.com/r/104/

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-22 17:19:05 +00:00
Russell Bryant
0bfcfd0848 Merged revisions 166273 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r166273 | russell | 2008-12-22 10:10:40 -0600 (Mon, 22 Dec 2008) | 7 lines

Re-work ref count handling of MoH classes using astobj2 to resolve crashes.

(closes issue #13566)
Reported by: igorcarneiro
Tested by: russell
Review: http://reviewboard.digium.com/r/106/

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166277 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-22 16:21:07 +00:00
Mark Michelson
198eedad1f Merged revisions 166267 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r166267 | mmichelson | 2008-12-22 10:07:59 -0600 (Mon, 22 Dec 2008) | 17 lines

Fix a file playback crash and explicitly initialize values in func_timeout.c

A crash was brought up on the bugtracker. The first run through valgrind
was full of legitimate complaints of uninitialized values in func_timeout when
setting a response timeout. These were fixed but the crash persisted.

A second run through showed the real problem. The reference counting used
for filestreams was incorrect because there were some missing increments
when a frame was read from a format module.

(closes issue #14118)
Reported by: blitzrage
Patches:
      14118v2.patch uploaded by putnopvut (license 60)
Tested by: blitzrage


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166275 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-22 16:17:17 +00:00
Joshua Colp
69c484879f Merged revisions 166268 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r166268 | file | 2008-12-22 12:08:13 -0400 (Mon, 22 Dec 2008) | 7 lines
  
  Record the previous port in the temporary address structure so that the comparison does not treat the host as having changed even if it did not. This would have been uninitialized before and would have led to a baddddd port.
  (closes issue #13628)
  Reported by: pananix
  Patches:
        bug13628.patch uploaded by jpeeler (license 325)
  Tested by: file, blitzrage
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-22 16:10:22 +00:00
Russell Bryant
e8952a420c Merged revisions 166258 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r166258 | russell | 2008-12-22 08:16:54 -0600 (Mon, 22 Dec 2008) | 26 lines

Remove AST_PBX_KEEPALIVE usage from res_agi.

This patch removes the usage of AST_PBX_KEEPALIVE from res_agi.  The only usage
was for the AGI command, "asyncagi break".  This patch removes this feature.
Normally, a feature would not be removed like this.  However, this code is
broken and usage of it will result in a memory leak.

Usage of this feature will make the AGI code return a result of 
AST_PBX_KEEPALIVE.  The PBX handler assumes that another thread has assumed
ownership of the channel.  The channel thread will exit without destroying the
channel.  Unfortunately, _no_ thread has ownership of the channel at this
point.  There are a couple of serious problems here:

1) The only way to recover the caller is to issue a channel redirect.  This
   will work, but this will be done with a masquerade, and the old ast_channel
   structure will be lost.

2) Until the channel redirect happens, there is no code servicing the channel.
   That means nothing is reading audio or handling events coming from the
   channel.  This is very bad.

The recommended way to get this same "break" functionality is to issue the
redirect while the channel is still being handled by the AGI code.  That way,
there will be no memory leak, and there will be no period of time that the
channel is not being serviced.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166260 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-22 14:19:06 +00:00
Mark Michelson
732fd97b6b Merged revisions 166162 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r166162 | mmichelson | 2008-12-19 17:45:00 -0600 (Fri, 19 Dec 2008) | 6 lines

Get rid of an extra space.

I don't know how this crept back in when I had already
fixed it earlier


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166164 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-19 23:45:59 +00:00
Mark Michelson
4bb7e35704 Switch documentation formats for func_audiohookinherit.c
1.6.1 does not have xml documentation, so I reverted
to the old way here.



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166161 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-19 23:43:35 +00:00
Mark Michelson
c665735e2e Blocked revisions 166158 via svnmerge
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r166158 | mmichelson | 2008-12-19 17:36:22 -0600 (Fri, 19 Dec 2008) | 16 lines

Blocked revisions 166157 via svnmerge

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r166157 | mmichelson | 2008-12-19 17:34:57 -0600 (Fri, 19 Dec 2008) | 9 lines

Backport of AUDIOHOOK_INHERIT for Asterisk 1.4

(closes issue #13538)
Reported by: mbit
Patches:
      13538.patch uploaded by putnopvut (license 60)
Tested by: putnopvut


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166159 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-19 23:36:47 +00:00
Mark Michelson
7ed70a7a3f Merged revisions 166092,166095 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r166092 | mmichelson | 2008-12-19 16:26:16 -0600 (Fri, 19 Dec 2008) | 28 lines

Adding a new dialplan function AUDIOHOOK_INHERIT

This function is being added as a method to allow for
an audiohook to move to a new channel during a channel
masquerade. The most obvious use for such a facility is
for MixMonitor when a transfer is performed. Prior to
the addition of this functionality, if a channel 
running MixMonitor was transferred by another party, then
the recording would stop once the transfer had completed.
By using AUDIOHOOK_INHERIT, you can make MixMonitor 
continue recording the call even after the transfer
has completed.

It has also been determined that since this is seen
by most as a bug fix and is not an invasive change,
this functionality will also be backported to 1.4 and
merged into the 1.6.0 branches, even though they are
feature-frozen.

(closes issue #13538)
Reported by: mbit
Patches:
      13538.patch uploaded by putnopvut (license 60)
	  Tested by: putnopvut

Review: http://reviewboard.digium.com/r/102/


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r166095 | mmichelson | 2008-12-19 16:40:57 -0600 (Fri, 19 Dec 2008) | 5 lines

Remove the verbatim tag from the author line

I could have sworn I already did that before, though...


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166098 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-19 23:07:33 +00:00
Eliel C. Sardanons
8e78d8ebd4 Blocked revisions 165954 via svnmerge
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  r165954 | eliel | 2008-12-19 16:20:46 -0200 (Fri, 19 Dec 2008) | 4 lines
  
  Fix the XML documentation for Record().
  <value> tags inside <variable> elements must have CDATA and no
  another XML node.
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2008-12-19 18:22:37 +00:00
Russell Bryant
e3f0577bda Merged revisions 165890 via svnmerge from
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r165890 | russell | 2008-12-19 09:05:09 -0600 (Fri, 19 Dec 2008) | 17 lines

Merged revisions 165889 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r165889 | russell | 2008-12-19 09:03:02 -0600 (Fri, 19 Dec 2008) | 9 lines

Ensure that the chanspy datastore is fully initialized.

This patch resolved some random crash issues observed by a user on a BSD system

(closes issue #14111)
Reported by: ys
Patches:
      app_chanspy.c.diff uploaded by ys (license 281)

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2008-12-19 15:08:59 +00:00
Russell Bryant
6562e1d88d Blocked revisions 165886 via svnmerge
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r165886 | russell | 2008-12-19 08:48:35 -0600 (Fri, 19 Dec 2008) | 2 lines

Disable some automatic links generated by doxygen.

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2008-12-19 14:49:00 +00:00
Russell Bryant
4b91befbd7 Blocked revisions 165883 via svnmerge
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r165883 | russell | 2008-12-19 08:42:51 -0600 (Fri, 19 Dec 2008) | 8 lines

Introduce commit message formatting guidelines.

This documents the recommended outline to use for commit message.  It also
covers information on special tags that can be used in commit messages, as well
as the template functionality that is available on bugs.digium.com.

Review: http://reviewboard.digium.com/r/96/

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2008-12-19 14:43:46 +00:00
Tilghman Lesher
a3825a040a Merged revisions 165797 via svnmerge from
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  r165797 | tilghman | 2008-12-18 15:41:02 -0600 (Thu, 18 Dec 2008) | 15 lines
  
  Merged revisions 165767 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
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    r165767 | tilghman | 2008-12-18 15:14:47 -0600 (Thu, 18 Dec 2008) | 8 lines
    
    Add mutexes around accesses to the IMAP library interface.  This prevents
    certain crashes, especially when shared mailboxes are used.
    (closes issue #13653)
     Reported by: howardwilkinson
     Patches: 
           asterisk-1.4.21.2-appvoicemail-sharedimap-lock.patch uploaded by howardwilkinson (license 590)
     Tested by: jpeeler
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2008-12-18 21:57:14 +00:00
Russell Bryant
c871ef2aeb Merged revisions 165801 via svnmerge from
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r165801 | russell | 2008-12-18 15:44:47 -0600 (Thu, 18 Dec 2008) | 19 lines

Merged revisions 165796 via svnmerge from 
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r165796 | russell | 2008-12-18 15:39:25 -0600 (Thu, 18 Dec 2008) | 11 lines

Make ast_carefulwrite() be more careful.

This patch handles some additional cases that could result in partial writes
to the file description.  This was done to address complaints about partial
writes on AMI.

(issue #13546) (more changes needed to address potential problems in 1.6)
Reported by: srt
Tested by: russell
Review: http://reviewboard.digium.com/r/99/

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2008-12-18 21:47:07 +00:00
Jeff Peeler
48a29209e6 Blocked revisions 165798 via svnmerge
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r165798 | jpeeler | 2008-12-18 15:43:17 -0600 (Thu, 18 Dec 2008) | 6 lines

(closes issue #13993)
Reported by: mika

Add ActionID response to ping if sent with request.


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2008-12-18 21:44:20 +00:00
Joshua Colp
4142e300f7 Merged revisions 165792 via svnmerge from
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  r165792 | file | 2008-12-18 17:21:44 -0400 (Thu, 18 Dec 2008) | 6 lines
  
  Numerous documentation updates.
  (closes issue #13970)
  Reported by: pkempgen
  Patches:
        __20081217_cli_usage_fixes.patch.txt uploaded by blitzrage (license 10)
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2008-12-18 21:24:00 +00:00
Russell Bryant
52f5e8fcfb Merged revisions 165723 via svnmerge from
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r165723 | russell | 2008-12-18 13:33:42 -0600 (Thu, 18 Dec 2008) | 14 lines

Remove the need for AST_PBX_KEEPALIVE with the GoSub option from Dial.

This is part of an effort to completely remove AST_PBX_KEEPALIVE and other
similar return codes from the source.  While this usage was perfectly safe,
there are others that are problematic.  Since we know ahead of time that
we do not want to PBX to destroy the channel, the PBX API has been changed
so that information can be provided as an argument, instead, thus removing
the need for the KEEPALIVE return value.

Further changes to get rid of KEEPALIVE and related code is being done by
murf.  There is a patch up for that on review 29.

Review: http://reviewboard.digium.com/r/98/

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2008-12-18 19:45:11 +00:00
Mark Michelson
0101f564a4 Merged revisions 165724 via svnmerge from
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r165724 | mmichelson | 2008-12-18 13:34:33 -0600 (Thu, 18 Dec 2008) | 8 lines

Fix crashes in res_odbc.

The variable "class" was being set NULL just prior to
being dereferenced in an ao2_link call. I have moved
the setting of the variable to NULL until after the
ao2_link call.


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2008-12-18 19:36:23 +00:00
Russell Bryant
03d73a0522 Merged revisions 165662 via svnmerge from
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r165662 | russell | 2008-12-18 12:54:47 -0600 (Thu, 18 Dec 2008) | 15 lines

Merged revisions 165661 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r165661 | russell | 2008-12-18 12:52:18 -0600 (Thu, 18 Dec 2008) | 7 lines

Set the process group ID on the MOH process so that all children will get killed

(closes issue #14099)
Reported by: caspy
Patches:
      res_musiconhold.c.patch.killpg.try2 uploaded by caspy (license 645)

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2008-12-18 18:58:05 +00:00
Tilghman Lesher
fcb9c545f4 Merged revisions 165658 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r165658 | tilghman | 2008-12-18 12:36:48 -0600 (Thu, 18 Dec 2008) | 2 lines
  
  Fix 2 resource leaks and fix another pipe-to-comma conversion
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2008-12-18 18:47:08 +00:00
Joshua Colp
1f96800e79 Merge in changes to return chan_sip to matching based on how it was previously done and how it is done in trunk. It will do name based for users and friends and IP based for peers.
(closes issue #14107)
Reported by: jsmith


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2008-12-18 17:59:37 +00:00
Joshua Colp
e5a8f92ab1 Merged revisions 165599 via svnmerge from
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  r165599 | file | 2008-12-18 13:13:32 -0400 (Thu, 18 Dec 2008) | 11 lines
  
  Merged revisions 165591 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
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    r165591 | file | 2008-12-18 13:11:42 -0400 (Thu, 18 Dec 2008) | 4 lines
    
    Only care about a compatible codec for early bridging if we are actually bridging to another channel. If we are not we actually want to bring the audio back to us.
    (closes issue #13545)
    Reported by: davidw
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2008-12-18 17:15:55 +00:00
Tilghman Lesher
51bcaa624f Merged revisions 165541 via svnmerge from
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  r165541 | tilghman | 2008-12-18 10:36:48 -0600 (Thu, 18 Dec 2008) | 2 lines
  
  Fix reference counts of the class and add an assertion to the end.
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2008-12-18 16:48:13 +00:00
Joshua Colp
e0480df461 Blocked revisions 165538 via svnmerge
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  r165538 | file | 2008-12-18 12:11:08 -0400 (Thu, 18 Dec 2008) | 12 lines
  
  Blocked revisions 165537 via svnmerge
  
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    r165537 | file | 2008-12-18 12:07:37 -0400 (Thu, 18 Dec 2008) | 6 lines
    
    Do not crash if we are not passed in a followme id.
    (closes issue #14106)
    Reported by: ys
    Patches:
          app_followme.c.2.diff uploaded by ys (license 281)
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2008-12-18 16:12:28 +00:00
Eliel C. Sardanons
4745521710 Blocked revisions 165502 via svnmerge
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  r165502 | eliel | 2008-12-18 13:25:15 -0200 (Thu, 18 Dec 2008) | 9 lines
  
  Remove duplicate code from the ast_str API. We now use __AST_STR_* to
  access 'struct ast_str' members, but this must only be used inside the API implementation.
  
  (closes issue #14098)
  Reported by: eliel
  Patches:
        ast_str.patch uploaded by eliel (license 64)
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2008-12-18 15:27:04 +00:00
Russell Bryant
3b0d24cede Blocked revisions 165433 via svnmerge
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r165433 | russell | 2008-12-18 07:33:34 -0600 (Thu, 18 Dec 2008) | 9 lines

Add a new application, Originate.

(closes issue #14075)
Reported by: rcasas
Patches:
      app_originate.c uploaded by rcasas (license 641), heavily modified by me
Tested by: russell
Review: http://reviewboard.digium.com/r/95/

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2008-12-18 13:34:26 +00:00
Tilghman Lesher
6b559884cf Blocked revisions 165397 via svnmerge
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  r165397 | tilghman | 2008-12-17 17:39:48 -0600 (Wed, 17 Dec 2008) | 3 lines
  
  Add RECORD_STATUS variable, as requested on the -users list.
  Patch by me (license 14)
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2008-12-17 23:40:32 +00:00
Mark Michelson
e76e724b3b Merged revisions 165330 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r165330 | mmichelson | 2008-12-17 15:46:19 -0600 (Wed, 17 Dec 2008) | 3 lines

Fix a refcount leak in res_odbc


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Tilghman Lesher
d70704c277 Merged revisions 165325 via svnmerge from
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  r165325 | tilghman | 2008-12-17 15:28:51 -0600 (Wed, 17 Dec 2008) | 2 lines
  
  Oops, broke trunk
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2008-12-17 21:31:15 +00:00
Mark Michelson
0d8c8393e1 Blocked revisions 165326 via svnmerge
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r165326 | mmichelson | 2008-12-17 15:29:30 -0600 (Wed, 17 Dec 2008) | 3 lines

Fix the build


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2008-12-17 21:30:12 +00:00
Mark Michelson
04b50b8b8c Merged revisions 165318 via svnmerge from
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r165318 | mmichelson | 2008-12-17 15:17:20 -0600 (Wed, 17 Dec 2008) | 15 lines

Merged revisions 165255 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r165255 | mmichelson | 2008-12-17 14:51:38 -0600 (Wed, 17 Dec 2008) | 7 lines

Fix some memory leaks found while looking at how realtime
configs are handled.

Also cleaned up some coding guidelines violations in app_realtime.c,
mostly related to spacing


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2008-12-17 21:25:32 +00:00
Tilghman Lesher
c383e518fd Merged revisions 165319 via svnmerge from
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  r165319 | tilghman | 2008-12-17 15:18:57 -0600 (Wed, 17 Dec 2008) | 11 lines
  
  Merged revisions 165317 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
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    r165317 | tilghman | 2008-12-17 15:14:37 -0600 (Wed, 17 Dec 2008) | 4 lines
    
    Reverse the fix from issue #6176 and add proper handling for that issue.
    (Closes issue #13962, closes issue #13363)
    Fixed by myself (license 14)
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2008-12-17 21:22:38 +00:00
Steve Murphy
77014eebbe Merged revisions 165254 via svnmerge from
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r165254 | murf | 2008-12-17 13:50:19 -0700 (Wed, 17 Dec 2008) | 5 lines

This patch is here committed to satisfy the
buildbot, who has a problem with the const.



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2008-12-17 21:02:45 +00:00
Terry Wilson
fe4f241693 Merged revisions 165219 via svnmerge from
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  r165219 | twilson | 2008-12-17 13:55:10 -0600 (Wed, 17 Dec 2008) | 2 lines
  
  Polycom phones close the connection after reading a little bit of the firmware files, we should stop sending in that case. Also, make that case print out a debug statement instead of a scary WARNING.
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2008-12-17 20:02:14 +00:00
Joshua Colp
f9973f81a3 Merged revisions 165216 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r165216 | file | 2008-12-17 15:52:40 -0400 (Wed, 17 Dec 2008) | 4 lines
  
  Call proxy_update so that the IP address gets populated. Sending stuff to 0.0.0.0 is silly!
  (closes issue #14055)
  Reported by: chris-mac
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