git.asterisk.org Git - asterisk/asterisk.git/log

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r162663 | mmichelson | 2008-12-10 10:24:56 -0600 (Wed, 10 Dec 2008) | 11 lines

Revert fix for issue 13570. It has caused more problems than
it helped to fix.

(closes issue #13783)
Reported by: navkumar

(closes issue #14025)
Reported by: ffs

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162664 65c4cc65-6c06-0410-ace0-fbb531ad65f3

FreeBSD also needs libgen.h
(closes issue #14051)
Reported by: ys
Patches:
res_http_post.c.diff uploaded by ys (license 281)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162660 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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  r162653 | file | 2008-12-10 12:05:29 -0400 (Wed, 10 Dec 2008) | 6 lines

  Increment the sequence number on the end packets for RFC2833. After reading the RFC some more and doing some testing I agree with this change.
  (closes issue #12983)
  Reported by: vt
  Patches:
        dtmf_inc_seqnum_on_end_pkts.diff uploaded by vt (license 520)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162656 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When transmitting a register set the socket port to the local one for the transport being used, not the port for the remote server.
(closes issue #13633)
Reported by: performer

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162619 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Make res_snmp.so compile on OpenBSD.

OpenBSD uses an old version of gcc which throws an error
if you use a macro that's not #defined

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162583 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Finish conversion to using ARRAY_LEN and remove it as a janitor project.
(closes issue #14032)
Reported by: bkruse
Patches:
14032.patch uploaded by bkruse (license 132)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162542 65c4cc65-6c06-0410-ace0-fbb531ad65f3

it does help if the compiler attribute syntax is correct

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r162463 | tilghman | 2008-12-09 17:08:53 -0600 (Tue, 09 Dec 2008) | 2 lines

Oops, should be "tz", not "zonetag".
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Add some additional Asterisk project developer documentation.

After the nightly update of the documentation on asterisk.org, I'll post
an update to asterisk-dev with a pointer to the changes. This covers some
release branch and commit policy information. None of this should be a
surprise, since it's just documenting what we have already been doing.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162418 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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r162413 | russell | 2008-12-09 16:17:39 -0600 (Tue, 09 Dec 2008) | 8 lines

Remove the test_for_thread_safety() function completely.

The test is not valid. Besides, if we actually suspected that recursive
mutexes were not working, we would get a ton of LOG_ERROR messages when
DEBUG_THREADS is turned on.

(inspired by a discussion on the asterisk-dev list)

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  r162348 | tilghman | 2008-12-09 15:53:25 -0600 (Tue, 09 Dec 2008) | 4 lines

  We appear to have documented tz= in the [general] section of voicemail.conf,
  without actually having implemented it.  Oops.
  (Reported by Olivier on the -users list)
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  r162341 | file | 2008-12-09 17:14:29 -0400 (Tue, 09 Dec 2008) | 4 lines

  Add 'down' as a valid state for directed call pickup. This creeps up when we receive session progress when dialing a device and not ringing.
  (closes issue #14005)
  Reported by: ddl
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r162286 | russell | 2008-12-09 14:57:35 -0600 (Tue, 09 Dec 2008) | 9 lines

Fix an issue where callers on an incoming call on an SLA trunk would not hear ringback.

We need to make sure that we don't start writing audio to the trunk channel until we're
actually ready to answer it. Otherwise, the channel driver will treat it as inband
progress, even though all they are getting is silence.

(closes issue #12471)
Reported by: mthomasslo

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  r162273 | file | 2008-12-09 16:44:32 -0400 (Tue, 09 Dec 2008) | 4 lines

  Fix double declaration of 'x' on the PPC platform.
  (closes issue #14038)
  Reported by: ffloimair
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r162264 | murf | 2008-12-09 13:20:54 -0700 (Tue, 09 Dec 2008) | 1 line

In discussion with seanbright on #asterisk-dev, I have added a default rule, and an option to suppress the default rule from being generated in the flex output, for the sake of those OS's where they didn't tweak flex's ECHO macro, and the compiler doesn't like it. The regressions are OK with this.
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r162265 | mmichelson | 2008-12-09 14:28:44 -0600 (Tue, 09 Dec 2008) | 6 lines

If we fail to start a thread for the pbx to run in, we need to
be sure to decrease the number of active calls on the system.

This fix may relate to ABE-1713, but it is not certain yet.

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  r162204 | file | 2008-12-09 15:47:07 -0400 (Tue, 09 Dec 2008) | 7 lines

  Make sure that the timestamp for DTMF is not the same as the previous voice frame and do not send audio when transmitting DTMF as this confuses some equipment.
  (closes issue #13209)
  Reported by: ip-rob
  Patches:
        13209.diff uploaded by file (license 11)
  Tested by: ip-rob, bujones
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  r162071 | tilghman | 2008-12-09 11:07:50 -0600 (Tue, 09 Dec 2008) | 3 lines

  For some reason, after a distclean, gcc started returning
  'value computed is not used'.  Fixing (for --enable-dev-mode).
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  r162188 | file | 2008-12-09 15:06:14 -0400 (Tue, 09 Dec 2008) | 4 lines

  Take video into account when early bridging RTP.
  (closes issue #13535)
  Reported by: davidw
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r162136 | murf | 2008-12-09 11:13:39 -0700 (Tue, 09 Dec 2008) | 1 line

Previous fix used ast_malloc and ast_copy_string and messed up the standalone stuff. Fixed.
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r162013 | murf | 2008-12-09 09:31:55 -0700 (Tue, 09 Dec 2008) | 45 lines

(closes issue #14019)
Reported by: ckjohnsonme
Patches:
14019.diff uploaded by murf (license 17)
Tested by: ckjohnsonme, murf

This crash was the result of a few small errors that
would combine in 64-bit land to result in a crash.

32-bit land might have seen these combine to mysteriously
drop the args to an application call, in certain
circumstances.

Also, in trying to find this bug, I spotted
a situation in the flex input, where, in passing
back a 'word' to the parser, it would allocate
a buffer larger than necessary. I changed the
usage in such situations, so that strdup was
not used, but rather, an ast_malloc, followed
by ast_copy_string.

I removed a field from the pval struct, in
u2, that was never getting used, and set in
one spot in the code. I believe it was an
artifact of a previous fix to make switch
cases work invisibly with extens.

And, for goto's I removed a '!' from
before a strcmp, that has been there
since the initial merging of AEL2, that
might prevent the proper target of a
goto from being found. This was pretty
harmless on its own, as it would just
louse up a consistency check for users.

Many thanks to ckjohnsonme for providing
a simplified and complete set of information
about the bug, that helped considerably in
finding and fixing the problem.

Now, to get aelparse up and running again
in trunk, and out of its "horribly broken" state,
so I can run the regression suite!

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r162014 | russell | 2008-12-09 10:46:53 -0600 (Tue, 09 Dec 2008) | 5 lines

Allow DISA to handle extensions that start with #.

(closes issue #13330)
Reported by: jcovert

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r161948 | russell | 2008-12-09 08:52:25 -0600 (Tue, 09 Dec 2008) | 15 lines

Fix a problem with GROUP() settings on a masquerade.

The previous code carried over group settings from the old channel to the new
one.  However, it did nothing with the group settings that were already on the
new channel.  This patch removes all group settings that already existed on the
new channel.

I have a more complicated version of this patch which addresses only the most
blatant problem with this, which is that a channel can end up with multiple
group settings in the same category.  However, I could not think of a use case
for keeping any of the group settings from the old channel, so I went this route
for now.

(closes AST-152)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161951 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Avoid allocating memory for a thread that don't need it. Also, this memory was not being freed until the
main thread ends. (That is never).

(closes issue #14040)
Reported by: eliel
Patches:
func_odbc.c.patch uploaded by eliel (license 64)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161947 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Note that the recently changed waittime parameter is in milliseconds.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161911 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Add alw as a valid file extension for alaw and ulw as a valid file extension for ulaw.
(closes issue #14001)
Reported by: henrikw
Patches:
alw.diff uploaded by henrikw (license 627)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161869 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Update autosupport script with a few changes.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161830 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Allocate enough space initially for the message.
(closes issue #14027)
Reported by: junky
Patches:
M14027.diff uploaded by junky (license 177)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161790 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fix a regression introduced when the PBX timeouts were converted to milliseconds. collect_digits now gets milliseconds fed to it, not seconds.
(closes issue #14012)
Reported by: dveiga
Patches:
14012.patch uploaded by bkruse (license 132)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161787 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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  r161725 | file | 2008-12-08 13:52:10 -0400 (Mon, 08 Dec 2008) | 6 lines

  Make the usereqphone option work again.
  (closes issue #13474)
  Reported by: mmaguire
  Patches:
        20080912_bug13474.diff uploaded by mmaguire (license 571)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161726 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fix a crash that can occur on a transfer in chan_sip when attempting to collect
rtp stats.

(closes issue #13956)
Reported by: chris-mac
Tested by: chris-mac

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161721 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Add the ability to play a courtesy tone to the transfer target in a native SIP attended transfer by setting the variable ATTENEDED_TRANSFER_COMPLETE_SOUND.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161679 65c4cc65-6c06-0410-ace0-fbb531ad65f3

- Fix a leak while printing an argument description.
- Avoid printing the name of an argument in the [Arguments] tag if there is no description
for that argument.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161637 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Add voicemail related applications and functions XML documentation:
    applications:
      - VoiceMail()
      - VoiceMailMain()
      - MailboxExists()
      - VMAuthenticate()
    functions:
      - MAILBOX_EXISTS()

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161604 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Introduce SMS() application XML documentation.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161571 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Move Speech* applications and functions documentation to XML.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161536 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If the autoloop flag is set on a channel, then we need to
add 1 to the priority when checking if the extension exists. Otherwise,
gosubs will fail.

This was discovered when investigating an asterisk-users mailing list post
made by Gary Hawkins.

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  r161426 | seanbright | 2008-12-05 16:02:20 -0500 (Fri, 05 Dec 2008) | 15 lines

  Merged revisions 161421 via svnmerge from
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    r161421 | seanbright | 2008-12-05 15:50:23 -0500 (Fri, 05 Dec 2008) | 8 lines

    Fix build errors on FreeBSD (uint -> unsigned int).

    (closes issue #14006)
    Reported by: alphaque
    Patches:
          astobj2.h-patch uploaded by alphaque (license 259)
          (Slightly modified by seanbright)
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Use ast_free() instead of free(), pointed out by eliel on IRC.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161350 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When using IMAP_STORAGE, it's important to convert bare newlines (\n) in
emailbody and pagerbody to CR-LF so that the IMAP server doesn't spit out an
error. This was informally reported on #asterisk-dev a few weeks ago. Reviewed
by Mark M. on IRC.

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r161287 | russell | 2008-12-05 08:12:14 -0600 (Fri, 05 Dec 2008) | 2 lines

Fix a NULL format string warning found by buildbot.

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Resolve a compiler warning from buildbot about a NULL format string.

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Janitor, use ARRAY_LEN() when possible.

(closes issue #13990)
Reported by: eliel
Patches:
array_len.diff uploaded by eliel (license 64)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161218 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The first file should have a blank config filename in the structure, so that
when a save occurs to a different filename, everything goes to the alternate
filename, instead of appending to the original.  This is important for the
AMI command UpdateConfig.
(closes issue #13301)
Reported by: trevo
Patches:
       20081113__bug13301.diff.txt uploaded by Corydon76 (license 14)
       20081113__bug13301__1.6.0.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76, blitzrage

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Check the return value of fread/fwrite so the compiler doesn't complain. Only a
problem when IMAP_STORAGE is enabled.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161147 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If 'faxdetect=yes' in sip.conf, switch to a 'fax' extension (if it exists) after T38 is negotiated.
Terry Wilson created the original patch for this functionality, which I slightly modified and added
the faxdetect=yes|no configuration option. This patch is only for T38 fax detection and does not
do anything for G711 over SIP fax detection. By default, this option is disabled.

Reviewboard: http://reviewboard.digium.com/r/69/

This functionality is for issue AST-140.

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Fix minor coding guidelines introduced with CLI permissions.

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r161013 | jpeeler | 2008-12-04 12:30:41 -0600 (Thu, 04 Dec 2008) | 9 lines

(closes issue #13835)
Reported by: matt_b
Tested by: jpeeler

This mirrors a check that was present in ast_rtp_read to also be in ast_rtp_raw_write to not schedule sending the receiver report if the remote RTCP endpoint address isn't present in the RTCP structure.

Closes AST-142.

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r160943 | mmichelson | 2008-12-04 10:44:18 -0600 (Thu, 04 Dec 2008) | 15 lines

Fix a callerid parsing issue. If someone formatted callerid like the
following: "name <number>" (including the quotation marks), then the parts
would be parsed as

name: "name
number: number

This is because the closing quotation mark was not discovered since the number
and everything after was parsed out of the string earlier. Now, there is a check
to see if the closing quote occurs after the number, so that we can know if we
should strip off the opening quote on the name.

Closes AST-158

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Add debug flag so skinny debug will show information about packets.
We dont want to scare users with this, so we added a devmode compile flag

(closes issue #13952)
Reported by: wedhorn
Patches:
packetdebug3.diff uploaded by wedhorn (license 30)
Tested by: mvanbaak, wedhorn

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Added XML documentation for the following AGI commands:
        - get option
        - get variable
        - hangup
        - noop

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Jcolp pointed out that num will also match number

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* Found a couple more places where num/number needed to be done
so 1.4 upgraders will not have problems.
* Added curly braces and minor tweaks.

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r160770 | tilghman | 2008-12-03 15:54:07 -0600 (Wed, 03 Dec 2008) | 2 lines

Some compilers warn on null format strings; some don't (caught by buildbot)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160791 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Blocked revisions 160764 via svnmerge

........
  r160764 | qwell | 2008-12-03 15:38:07 -0600 (Wed, 03 Dec 2008) | 7 lines

  Only show this warning when we want to show it.

  (closes issue #13982)
  Reported by: coolmig
  Patches:
        chan_agent.c.patch uploaded by coolmig (license 621)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160765 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merged revisions 160703 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r160703 | murf | 2008-12-03 13:41:42 -0700 (Wed, 03 Dec 2008) | 11 lines

(closes issue #13597)
Reported by: john8675309
Patches:
patch.13597 uploaded by murf (license 17)
Tested by: murf, john8675309

This patch causes the setcid func to update the CDR
clid after setting the channel field.

I also notice that in trunk, the num/number of 1.4 is
left out; I decided to include the option to use
either in trunk, so as not to have 1.4 upgraders
not to have problems.

........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160760 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Another place this is missing

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160700 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fix typo when ListCategories returns none.

(closes issue #13994)
Reported by: mika
Patches:
ListCategoriesActionPatch.diff uploaded by mika (license 624)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160699 65c4cc65-6c06-0410-ace0-fbb531ad65f3

- iax2-provision was not freeing iax_templates structure when unloading the chan_iax2.so module.
- Move the code to start using the LIST macros.

Review: http://reviewboard.digium.com/r/72

(closes issue #13232)
Reported by: eliel
Patches:
iax2-provision.patch.txt uploaded by eliel (license 64)
(with minor changes pointed by Mark Michelson on review board)
Tested by: eliel

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160663 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Add some safety measures when using gosub, especially when using the options
for app_dial and app_queue to run a gosub when the call is answered.

* Check for the existence of the gosub target in gosub_exec. If it is nonexistent,
  then this will cause errors when we attempt to actually run the gosub, including
  a definite memory leak and potential crashes. Return an error in this situation
* Check the return value of pbx_exec in app_dial and app_queue before attempting
  to actually run the gosub routine. If there was an error, we should not attempt
  to run the gosub.
* Change a '|' to a ',' in app_queue.
* Add some extra curly braces where they had been missing previously.

(closes issue #13548)
Reported by: fiddur

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160626 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Blocked revisions 160570 via svnmerge

........
  r160570 | tilghman | 2008-12-03 11:55:12 -0600 (Wed, 03 Dec 2008) | 5 lines

  During bridge code, the channel bridge may return a retry code, if a transfer
  was initiated but not yet completed.  If the bridge is immediately retried,
  then we may send a storm of TXREQ packets, even though the first set is sent
  reliably (retransmitted).  Fixes AST-137.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160585 65c4cc65-6c06-0410-ace0-fbb531ad65f3

- Add <variable /> tags when naming a channel variable.
- Add <filename /> tags when naming a filename.
- Simplify the xml formatting putting some enters.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160562 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merged revisions 160558 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r160558 | tilghman | 2008-12-03 11:34:34 -0600 (Wed, 03 Dec 2008) | 7 lines

  If an entry is added to the directory during a scan when another entry expires,
  then that new entry will not be processed promptly, but must wait for either a
  future entry to start or a current entry's retry to occur.  If no other entries
  exist in the directory (other than the new entries) when a bunch expire, then
  the new entries must wait until another new entry is added to be processed.
  This was a rather weird race condition, really.  Fixes AST-147.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160559 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When investigating issue #13548, I found that gosub
handling in app_queue was just completely wrong, mostly
because the channel operations being performed were being
done on the incorrect channel.

With this set of changes, a gosub will correctly run on
the answering queue member's channel. There are still crash
issues which occur if there are dialplan syntax errors, so
I cannot yet close the referenced issue.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160555 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merged revisions 160551 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r160551 | tilghman | 2008-12-03 10:58:34 -0600 (Wed, 03 Dec 2008) | 4 lines

  Don't start scanning the directory until all modules are loaded, because some
  required modules (channels, apps, functions) may not yet be in memory yet.
  Fixes AST-149.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160552 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merged revisions 160480 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r160480 | tilghman | 2008-12-03 08:09:35 -0600 (Wed, 03 Dec 2008) | 7 lines

  Jon Bonilla (Manwe) pointed out on the -dev list:
  "I guess that having only ip-phones in mind is not a good approach. Since it is
  possible to have a sip proxy connected to asterisk we could receive a 407
  (unauthorized) or 483 (too many hops) as response and dialog ending would not be
  a good behavior."
  So modified.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160481 65c4cc65-6c06-0410-ace0-fbb531ad65f3

- Avoid setting .synopsis and .syntax if we are using XML documentation (or the
xml documentation wont be loaded).
- Use <variable></variable> to refer to a dialplan variable.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160447 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Info on LOCAL_PEEK function.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160346 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Add LOCAL_PEEK function, as requested by lmadsen.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160344 65c4cc65-6c06-0410-ace0-fbb531ad65f3

remove duplicate comment that I accidentally merged

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160333 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #13786)
Reported by: tzafrir

Readding DAHDI_CHECK_HOOKSTATE define that was removed in r134260 which fixes not being able to make outgoing calls on some FXO adapters:
http://lists.digium.com/pipermail/asterisk-users/2008-November/thread.html#221553

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160319 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merged revisions 160297 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r160297 | tilghman | 2008-12-02 11:42:09 -0600 (Tue, 02 Dec 2008) | 10 lines

  When the text does not match exactly (e.g. RTP/SAVP), then the %n conversion
  fails, and the resulting integer is garbage.  Thus, we must initialize the
  integer and check it afterwards for success.
  (closes issue #14000)
   Reported by: folke
   Patches:
         asterisk-sipbg-sscanf-1.4.22.diff uploaded by folke (license 626)
         asterisk-sipbg-sscanf-1.6.0.1.diff uploaded by folke (license 626)
         asterisk-sipbg-sscanf-trunk-r159896.diff uploaded by folke (license 626)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160308 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merged revisions 160207 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r160207 | tilghman | 2008-12-01 18:25:16 -0600 (Mon, 01 Dec 2008) | 3 lines

  Ensure that Asterisk builds with --enable-dev-mode, even on the latest gcc
  and glibc.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160208 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merged revisions 159976 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r159976 | mvanbaak | 2008-12-01 11:08:36 -0500 (Mon, 01 Dec 2008) | 3 lines

  Get rid of the useless format string and argument in the Bogus/ manager channelname.
  Noted by kpfleming and name Bogus/manager suggested by eliel
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160172 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Silence a build warning. (chan_phone.c:810: warning: value computed is not used)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160171 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Pay attention to the return value of system(), even if we basically ignore it.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160170 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Use AST_EXT_LIB_SETUP before using AST_EXT_LIB_CHECK or bad things happen.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160097 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Introduce CLI permissions.
Based on cli_permissions.conf configuration file, we are able to permit or deny
cli commands based on some patterns and the local user and group running rasterisk.

(Sorry if I missed some of the testers).

Reviewboard: http://reviewboard.digium.com/r/11/

(closes issue #11123)
Reported by: eliel
Tested by: eliel, IgorG, Laureano, otherwiseguy, mvanbaak

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160062 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merged revisions 160003 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r160003 | russell | 2008-12-01 11:27:30 -0600 (Mon, 01 Dec 2008) | 6 lines

Apply some logic used in iax2_indicate() to iax2_setoption(), as well, since they
both have the potential to send control frames in the middle of call setup. We
have to wait until we have received a message back from the remote end before
we try to send any more frames. Otherwise, the remote end will consider it
invalid, and we'll get stuck in an INVAL/VNAK storm.

........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160004 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merged revisions 159900 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r159900 | russell | 2008-12-01 08:52:56 -0600 (Mon, 01 Dec 2008) | 2 lines

Force a "make clean" to avoid a bizarre build issue ...

........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159911 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merged revisions 159897 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r159897 | mvanbaak | 2008-12-01 15:05:41 +0100 (Mon, 01 Dec 2008) | 4 lines

  make manager compile on OpenBSD.
  The last (10th) argument to ast_channel_alloc here should be a pointer
  and NULL is not really a pointer.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159898 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Allow the '#' sign to exist within an extension (inspired by issue #13330)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159853 65c4cc65-6c06-0410-ace0-fbb531ad65f3

incorporates r159808 from branches/1.4:
------------------------------------------------------------------------
r159808 | kpfleming | 2008-11-29 10:58:29 -0600 (Sat, 29 Nov 2008) | 7 lines

update dev-mode compiler flags to match the ones used by default on Ubuntu Intrepid, so all developers will see the same warnings and errors

since this branch already had some printf format attributes, enable checking for them and tag functions that didn't have them

format attributes in a consistent way

------------------------------------------------------------------------

in addition:

move some format attributes from main/utils.c to the header files they belong in, and fix up references to the relevant functions based on new compiler warnings

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159818 65c4cc65-6c06-0410-ace0-fbb531ad65f3

we can now build with -Wformat=2, which found a couple of real bugs

because SPRINTF() use non-literal format strings (which cannot be checked), move it into its own module so the rest of func_strings can benefit from format string checking

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159774 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Make res_config_ldap compile with the official OpenLDAP 2.3.X versions.
They removed the LDAP_DEPRECATED define from their source and since we are using a couple
of deprecated function calls we should define it with a CFLAG.

Tested by me on OpenBSD 4.4 and snuff-home on Linux to make sure everything keeps compiling.
It shouldn't break, we only define the LDAP_DEPRECATED with this which is what
all 2.2.X and older versions of OpenLDAP did in their own tree.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159734 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Removed duplicate code

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159701 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Make a formatting change to test a new post-commit hook for reviewboard.

http://reviewboard.digium.com/r/65/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159666 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Make a formatting change to test a new post-commit hook for reviewboard.

http://reviewboard.digium.com/r/65/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159665 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Make a formatting change to test a new post-commit hook for reviewboard.

http://reviewboard.digium.com/r/65/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159664 65c4cc65-6c06-0410-ace0-fbb531ad65f3

improve handling of API calls provided by loaded modules through use of some GCC features; this makes app_stack's usage of AGI APIs even cleaner, and will allow it to work 'as expected' either with or without res_agi being loaded

reviewed at http://reviewboard.digium.com/r/62

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159631 65c4cc65-6c06-0410-ace0-fbb531ad65f3

add support for event suppression for AMI-over-HTTP

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159629 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Blocked revisions 159571 via svnmerge

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r159571 | kpfleming | 2008-11-26 14:21:04 -0600 (Wed, 26 Nov 2008) | 2 lines

rename these files so as to avoid conflicts when users update their working copies and have unversioned files already in place
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159573 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Add some necessary hangup commands in the case that forwarding
a call fails

1) Hang up the original destination if the local channel cannot
   be requested.
2) Hang up the local channel (in addition to the original destination)
   if ast_call fails when calling the newly created local channel.

This prevents channels from sticking around forever in the
case of a botched call forward (e.g. to an extension which does not
exist).

(closes issue #13764)
Reported by: davidw
Patches:
      13764_v2.patch uploaded by putnopvut (license 60)
Tested by: putnopvut, davidw

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159554 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merged revisions 159476 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r159476 | kpfleming | 2008-11-26 12:36:24 -0600 (Wed, 26 Nov 2008) | 7 lines

  simplify (and slightly bug-fix) the recent developer-oriented COMPILE_DOUBLE mode

  ensure that 'make clean' removes dependency files for .i files that are created in COMPILE_DOUBLE mode
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159534 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If the config file does not exist, then the first use crashes Asterisk.
(closes issue #13848)
Reported by: klaus3000
Patches:
udptl.c.patch uploaded by eliel (license 64)
Tested by: blitzrage

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159475 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Don't allow for configuration options to overwrite options
set via channel variables on a reload.

(closes issue #13921)
Reported by: davidw
Patches:
13921.patch uploaded by putnopvut (license 60)
Tested by: davidw

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159437 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Always parse arguments in park_call_exec so that app_args is valid. This prevents a crash when executing Park from the dialplan with no arguments.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159402 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merged revisions 159316 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r159316 | murf | 2008-11-25 15:41:10 -0700 (Tue, 25 Nov 2008) | 15 lines

(closes issue #12694)
Reported by: yraber
Patches:
12694.2nd.diff uploaded by murf (license 17)
Tested by: murf, laurav

Thanks to file (Joshua Colp) for his IAX fix.

the change to cdr.c allows no-answer to percolate
up into CDR's, and feels like the right place to
locate this fix; if BUSY is done here, no-answer
should be, too.

........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159360 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Add an option, waitfordialtone, for UK analog lines which do not end a call
until the originating line hangs up.
(closes issue #12382)
Reported by: one47
Patches:
zap-waitfordialtone-trunk.080901.patch uploaded by one47 (license 23)
zap-waitfordialtone-bra-1.4.21.2.patch uploaded by fleed (license 463)
Tested by: fleed

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159317 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merged revisions 159269 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r159269 | tilghman | 2008-11-25 15:56:48 -0600 (Tue, 25 Nov 2008) | 7 lines

  Don't try to send a response on a NULL pvt.
  (closes issue #13919)
   Reported by: barthpbx
   Patches:
         chan_iax2.c.patch uploaded by eliel (license 64)
   Tested by: barthpbx
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159276 65c4cc65-6c06-0410-ace0-fbb531ad65f3