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r312286 | tilghman | 2011-04-01 05:44:33 -0500 (Fri, 01 Apr 2011) | 2 lines
Reload must react correctly against a possibly changed table, so dropping the conditional reload flag.
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r312288 | tilghman | 2011-04-01 05:58:45 -0500 (Fri, 01 Apr 2011) | 21 lines
Merged revisions 312287 via svnmerge from
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r312287 | tilghman | 2011-04-01 05:51:24 -0500 (Fri, 01 Apr 2011) | 14 lines
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r312285 | tilghman | 2011-04-01 05:36:42 -0500 (Fri, 01 Apr 2011) | 7 lines
Found some leaking file descriptors while looking at ast_FD_SETSIZE dead code.
(issue #18969)
Reported by: oej
Patches:
20110315__issue18969__14.diff.txt uploaded by tilghman (license 14)
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r312211 | alecdavis | 2011-04-01 22:03:11 +1300 (Fri, 01 Apr 2011) | 36 lines
Merged revisions 312210 via svnmerge from
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r312210 | alecdavis | 2011-04-01 21:47:29 +1300 (Fri, 01 Apr 2011) | 29 lines
Merged revisions 312174 via svnmerge from
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r312174 | alecdavis | 2011-04-01 21:29:49 +1300 (Fri, 01 Apr 2011) | 23 lines
voicemail: get real last_message_index and count_messages, ODBC resequence
change last_message_index to read the max msgnum stored in the database
change count_messages to actually count the number of messages.
last_message_index change:
This fixed overwriting of the last message if msgnum=0 was missing.
Previously every incoming message would overwrite msgnum=1.
count_messages change:
allows us to detect when requencing is required in opneA_mailbox.
resequence enabled for ODBC storage:
Assists with fixing up corrupt databases with gaps, but only when
a user actively opens there mailboxes.
(closes issue #18692,#18582,#19032)
Reported by: elguero
Patches:
based on odbc_resequence_mailbox2.1.diff uploaded by elguero (license 37)
Tested by: elguero, nivek, alecdavis
Review: https://reviewboard.asterisk.org/r/1153/
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r312117 | alecdavis | 2011-04-01 20:32:12 +1300 (Fri, 01 Apr 2011) | 29 lines
Merged revisions 312103 via svnmerge from
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r312103 | alecdavis | 2011-04-01 20:25:54 +1300 (Fri, 01 Apr 2011) | 22 lines
Merged revisions 312070 via svnmerge from
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r312070 | alecdavis | 2011-04-01 19:46:56 +1300 (Fri, 01 Apr 2011) | 16 lines
app_voicemail: close_mailbox needs to respect additional messages while mailbox is open.
close_mailbox leave gaps in message sequence if messages are deleted and new messages
arrive during this time, this is because the shuffle down to slot 0, only shuffles
the number of pre-existing messages when mailbox is opened, ignoring new arrivals.
Fix: in close_mailbox re-evaluate number of messages before the shuffle, this then includes new arrivals.
Happens on filebased or ODBC storage.
(issues #19032,#18582,#18692,#18998)
Reported by: alecdavis,tootai,afosorio
Review: https://reviewboard.asterisk.org/r/1153/
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r312022 | rmudgett | 2011-03-31 15:11:40 -0500 (Thu, 31 Mar 2011) | 14 lines
chan_misdn segfaults when DEBUG_THREADS is enabled.
The segfault happens because jb->mutexjb is uninitialized from the
ast_malloc(). The internals of ast_mutex_init() were assuming a nonzero
value meant mutex tracking initialization had already happened. Recent
changes to mutex tracking code to reduce excessive memory consumption
exposed this uninitialized value.
Converted misdn_jb_init() to use ast_calloc() instead of ast_malloc().
Also eliminated redundant zero initialization code in the routine.
(closes issue #18975)
Reported by: irroot
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r311930 | tilghman | 2011-03-31 01:43:18 -0500 (Thu, 31 Mar 2011) | 6 lines
Incorrect default example; the field is actually internally named "clid", not "callerid".
(closes issue #19040)
Reported by: wcselby
Tested by: tilghman
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r311874 | rmudgett | 2011-03-29 20:56:05 -0500 (Tue, 29 Mar 2011) | 1 line
Update some setup_dahdi_int() comments.
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r311799 | tilghman | 2011-03-29 02:08:39 -0500 (Tue, 29 Mar 2011) | 7 lines
Remove extraneous check from integer-type fields.
(closes issue #19027)
Reported by: mlehner
Review: https://reviewboard.asterisk.org/r/1149/
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r311751 | russell | 2011-03-28 17:00:01 -0500 (Mon, 28 Mar 2011) | 2 lines
Cross-reference VoiceMail() and VoiceMailMain() in the xml docs.
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r311687 | may | 2011-03-28 01:47:13 +0400 (Mon, 28 Mar 2011) | 2 lines
correct return values in ooh323_indicate for AST_CONTROL_T38_PARAMETERS
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r311615 | bbryant | 2011-03-23 17:54:11 -0400 (Wed, 23 Mar 2011) | 8 lines
This patch fixes a bug with MeetMe behavior where the 'P' option for always
prompting for a pin is ignored for the first caller.
(closes issue #18070)
Reported by: mav3rick
Review: https://reviewboard.asterisk.org/r/1132/
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r311612 | bbryant | 2011-03-23 17:45:46 -0400 (Wed, 23 Mar 2011) | 9 lines
Fix a possible crash in sip/reqresp_parser.c that is caused by a possible null
value.
(closes issue #18821)
Reported by: cmaj
Patches:
patch-reqresp_parser_sip_uri_domain_cmp_c_locale-crash-1.8.3-rc2.diff.tx
uploaded by cmaj (license 830)
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r311558 | twilson | 2011-03-22 19:24:53 -0700 (Tue, 22 Mar 2011) | 5 lines
Don't use static declared buf in parse_name_andor_addr
This function isn't used anywhere yet, but we definitely don't want
to keep the same value for buf between calls to the function.
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r311497 | dvossel | 2011-03-22 10:25:24 -0500 (Tue, 22 Mar 2011) | 9 lines
Merged revisions 311496 via svnmerge from
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r311496 | dvossel | 2011-03-22 10:24:45 -0500 (Tue, 22 Mar 2011) | 2 lines
Fixes memory leak in MeetMe AMI action
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solve. Still, due to the nature of FollowMe, it makes sense to have this option since it keeps apps bound to channels that would otherwise go away from being lost.
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r311352 | jrose | 2011-03-18 11:19:05 -0500 (Fri, 18 Mar 2011) | 10 lines
Changes some print statements/events to use a blank string in place of NULL if the string in question is NULL.
This is supposed to improve Solaris compatibility since Solaris goes berserk when trying to output NULL strings.
(closes issue #18759)
Reported by: bklang
Patches:
null-strings.patch uploaded by bklang (license 919)
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r311342 | mnicholson | 2011-03-18 11:02:50 -0500 (Fri, 18 Mar 2011) | 2 lines
Properly populate the LOCALSTATIONID channel variable.
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r311297 | rmudgett | 2011-03-17 21:59:05 -0500 (Thu, 17 Mar 2011) | 12 lines
Race condition when ISDN CallRerouting/CallDeflection invoked.
The queued AST_CONTROL_BUSY could sometimes be processed before the
call_forward dial string is recognized.
* Moved setting the call_forwarding dial string after sending a response
to the initiator and just queue an empty frame to wake up the media thread
instead of an AST_CONTROL_BUSY.
* Added check for empty rerouting/deflection number and respond with an
error.
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r311295 | rmudgett | 2011-03-17 21:22:07 -0500 (Thu, 17 Mar 2011) | 35 lines
Merged revision 310986 from
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r310986 | rmudgett | 2011-03-16 13:56:28 -0500 (Wed, 16 Mar 2011) | 28 lines
Dial() o option broke when connected line feature added.
The patch restores the o option behavior and adds the ability to specify
the CallerID. The Dial o and f options are complementary to each other.
The o option stores the CallerID on the outgoing channel as the channel's
CallerID. The f option forces the CallerID sent by the outgoing channel.
o(x) - The argument 'x' is optional. If not present, then specify that
the CallerID that was present on the *calling* channel be stored as the
CallerID on the *called* channel. This was the behavior of Asterisk 1.0
and earlier. If present, then specify the CallerID stored on the *called*
channel. Note that o(${CALLERID(all)}) is similar to option o without
parameters.
f(x) - The argument 'x' is optional and its presence changes the behavior
of this option. If not present, then force the outgoing CallerID on a
call-forward or deflection to the dialplan extension for this Dial() using
a dialplan 'hint'. For example, some PSTNs do not allow CallerID to be
set to anything other than the numbers assigned to you. If present, then
force the outgoing CallerID to 'x'.
Patches:
jira_abe_2752_dial_fo_options.patch uploaded by rmudgett (license 664)
Tested by: rmudgett
JIRA ABE-2752
JIRA SWP-3096
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r311197 | jrose | 2011-03-17 14:03:34 -0500 (Thu, 17 Mar 2011) | 11 lines
This fixes a nasty chanspy bug which was causing a channel leak every time a spied on channel made a call.
In addition to the above, it makes certain channel destruction occurs so that applications don't get stuck waiting for datastore destruction while monitored by chanspy.
(closes issue #18742)
Reported by: jkister
Tested by: jkister, jcovert, jrose
Review: http://reviewboard.digium.internal/r/106/
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r311141 | mnicholson | 2011-03-17 10:00:33 -0500 (Thu, 17 Mar 2011) | 11 lines
Merged revisions 311140 via svnmerge from
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r311140 | mnicholson | 2011-03-17 09:58:52 -0500 (Thu, 17 Mar 2011) | 4 lines
Don't write items to the manager socket twice.
AST-2011-003
(closes issue 0018987)
Reported by: ks-steven
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r311050 | alecdavis | 2011-03-17 23:49:41 +1300 (Thu, 17 Mar 2011) | 24 lines
Merged revisions 311049 via svnmerge from
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r311049 | alecdavis | 2011-03-17 23:45:47 +1300 (Thu, 17 Mar 2011) | 17 lines
Merged revisions 311048 via svnmerge from
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r311048 | alecdavis | 2011-03-17 23:43:35 +1300 (Thu, 17 Mar 2011) | 12 lines
Remove extra quote in indications.conf
Picking low hanging fruit.
(closes issue #18971)
Reported by: IgorG
Patches:
based on indications.conf.sample.diff uploaded by IgorG (license 20)
Tested by: IgorG
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r310999 | twilson | 2011-03-16 14:47:59 -0500 (Wed, 16 Mar 2011) | 18 lines
Merged revisions 310998 via svnmerge from
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r310998 | twilson | 2011-03-16 14:46:36 -0500 (Wed, 16 Mar 2011) | 11 lines
Fix crash on fdopen failure
See security advisory AST-2011-004
(closes issue #18845)
Reported by: cmaj
Patches:
patch-main-tcptls-1.8.3-rc2-open-session-crash-take2.diff.txt uploaded by cmaj (license 830)
patch-main-tcptls-1.8.3-rc2-open-session-crash-take3.diff.txt uploaded by cmaj (license 830)
Tested by: cmaj, twilson
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r310993 | twilson | 2011-03-16 14:26:57 -0500 (Wed, 16 Mar 2011) | 11 lines
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r310992 | twilson | 2011-03-16 14:23:03 -0500 (Wed, 16 Mar 2011) | 4 lines
Don't keep trying to write to a closed connection
See security advisory AST-2011-003.
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r310902 | twilson | 2011-03-16 12:19:57 -0500 (Wed, 16 Mar 2011) | 43 lines
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r310889 | twilson | 2011-03-16 12:03:27 -0500 (Wed, 16 Mar 2011) | 36 lines
Merged revisions 310888 via svnmerge from
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r310888 | twilson | 2011-03-16 11:58:42 -0500 (Wed, 16 Mar 2011) | 29 lines
Don't delay DTMF in core bridge while listening for DTMF features
This patch is mostly the work of Olle Johansson. I did some cleanup and
added the silence generating code if transmit_silence is set.
When a channel listens for DTMF in the core bridge, the outbound DTMF is not
sent until we have received DTMF_END. For a long DTMF, this is a disaster. We
send 4 seconds of DTMF to Asterisk, which sends no audio for those 4 seconds.
Some products see this delay and the time skew on RTP packets that results and
start ignoring the audio that is sent afterward.
With this change, the DTMF_BEGIN frame is inspected and checked. If it matches
a feature code, we wait for DTMF_END and activate the feature as before. If
transmit_silence=yes in asterisk.conf, silence is sent if we paritally match a
multi-digit feature. If it doesn't match a feature, the frame is forwarded
along with the DTMF_END without delay. By doing it this way, DTMF is not delayed.
(closes issue #15642)
Reported by: jasonshugart
Patches:
issue_15652_dtmf_ast-1.4.patch.txt uploaded by twilson (license 396)
Tested by: globalnetinc, jde
(closes issue #16625)
Reported by: sharvanek
Review: https://reviewboard.asterisk.org/r/1092/
Review: https://reviewboard.asterisk.org/r/1125/
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r310834 | tilghman | 2011-03-14 20:48:25 -0500 (Mon, 14 Mar 2011) | 2 lines
Fix branch compile.
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r310781 | alecdavis | 2011-03-15 14:00:55 +1300 (Tue, 15 Mar 2011) | 10 lines
core show locks: display ThreadID in hexadecimal
Allow easier cross referencing of thread ID's with GDB backtraces
(closes issue #18968)
Reported by: alecdavis
Patches:
bug18968.diff.txt uploaded by alecdavis (license 585)
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(closes issue #18693)
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r310734 | may | 2011-03-15 00:45:53 +0300 (Tue, 15 Mar 2011) | 12 lines
Introduce t.38 parameters control functionality not full but enough for
Send/RcvFax support
Introduce t.38 controls between asterisk core and channel/proto layers.
Not all parameters are transferred from proto layers but *Fax apps
tested and work ok.
(issue #18693)
Reported by: benngard2
Patches:
issue-18693.patch uploaded by may213 (license 454)
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r310636 | rmudgett | 2011-03-14 11:50:59 -0500 (Mon, 14 Mar 2011) | 39 lines
Merged revisions 310635 via svnmerge from
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r310635 | rmudgett | 2011-03-14 11:47:54 -0500 (Mon, 14 Mar 2011) | 32 lines
Merged revisions 310633 via svnmerge from
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r310633 | rmudgett | 2011-03-14 11:38:24 -0500 (Mon, 14 Mar 2011) | 25 lines
"Caller*ID failed checksum" on Wildcard TDM2400P and TDM410
The last character in the caller id message is getting a framing error.
The checksum is the last character in the message. A framing error in the
checksum could be because:
1) The sender did not send a full stop bit.
2) The sender cut off the FSK carrier too soon.
3) The sender opted to send zero of the specified zero to 10 trailing mark
bits and round-off errors in the code resulted in the code not being where
it thought it was in the demodulated bit stream.
Bit 8 of 'b' is set when parity error.
Bit 9 of 'b' is set when framing error.
Made ignore the framing and parity error bits if the errored character is
the checksum. We can tolerate a framing/parity error there. The checksum
character validates the message.
(closes issue #18474)
Reported by: nivek
Patches:
callerid.c.1.patch uploaded by nivek (license 636) (with modifications)
Tested by: nivek
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r310587 | jrose | 2011-03-14 10:27:57 -0500 (Mon, 14 Mar 2011) | 15 lines
Merged revisions 310585 via svnmerge from
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r310585 | jrose | 2011-03-14 08:56:22 -0500 (Mon, 14 Mar 2011) | 8 lines
Adds 'p' as an option to func_volume. When it is on, the old behavior with DTMF controlling volume adjustment will be enforced.
When it is off, DTMF will not be processed by the function.
Programmed by Jonathan Rose
Reviewed by David Vossel, Leif Madsen, and Russell Bryant
http://reviewboard.digium.internal/r/93/
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various OS distributions. Thanks David.
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r310462 | tilghman | 2011-03-12 14:27:54 -0600 (Sat, 12 Mar 2011) | 45 lines
Merged revisions 310448 via svnmerge from
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r310448 | tilghman | 2011-03-12 14:24:54 -0600 (Sat, 12 Mar 2011) | 38 lines
Recorded merge of revisions 310435 via svnmerge from
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r310435 | tilghman | 2011-03-12 14:22:07 -0600 (Sat, 12 Mar 2011) | 31 lines
Add AELSub, which provides a stable entry point into AEL subroutines.
This commit needs some explanation, given that we're adding a new application
into an existing release branch. This is generally a violation of our release
policy, except in very limited circumstances, and I believe this is one of
those circumstances.
The problem that this solves is one of the sanity of using multiple dialplan
languages to define a dialplan. In the case of the reporter, he or she is
using AEL is define subroutines, while using Realtime extensions to invoke
those subroutines. While you can do this, it's based upon the reality of AEL
using actual dialplan extensions; however, there is no guarantee that the
details of _how_ AEL is compiled into extensions will remain stable. In fact,
at the time of this commit, it has already changed twice, once in a
fundamental way.
Now normally, a new application would only be added to trunk. However, this
application is explicitly to create a stable user-level API between versions,
and adding it to trunk only will not solve the user's problem of switching
between 1.6.2 and 1.8, nor will it help anybody switching from 1.8 to 1.10.
Therefore, it needs to go into existing release branches. For the sake of
consistency, and also because one of the changes was between 1.4 and 1.6.x,
I am also electing to commit this to 1.4.
(closes issue #18910)
Reported by: alexandrekeller
Patches:
20110304__issue18919__1.6.2.diff.txt uploaded by tilghman (license 14)
20110304__issue18919__1.4.diff.txt uploaded by tilghman (license 14)
Tested by: alexandrekeller
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r310415 | tilghman | 2011-03-12 14:05:46 -0600 (Sat, 12 Mar 2011) | 14 lines
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r310414 | tilghman | 2011-03-12 13:51:23 -0600 (Sat, 12 Mar 2011) | 7 lines
Transactional handles should be used for the insertbuf, if available.
Also, fix a possible resource leak.
(closes issue #18943)
Reported by: irroot
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Recent versions of GCC have a tuning option value of 'native', which causes
the compiler to optimize the build for the CPU the compile is performed on.
Since most people are building Asterisk on the machine they plan to run it on,
the configure script and build system will now use this value unless a different
value is specified by the user in CFLAGS when the configure script is executed.
In addition, this value will be used for building the GSM and LPC10 codecs as
well, in preference to the logic that has been in their Makefiles forever to
optimize for certain types of CPUs.
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r310287 | alecdavis | 2011-03-11 19:47:44 +1300 (Fri, 11 Mar 2011) | 17 lines
remote_bridge_loop: prevent segfault when after transfer of IAX2 of DAHDI call
If the channel condition is one of the following after breaking out of the loop, don't try to update_peer
(where x = 0/1)
1). ZOMBIE
2). cx->tech_pvt != pvtx
3). gluex != ast_rtp_instance_get_glue(cx->tech->type))
(closes issue #18781)
Reported by: alecdavis
Patches:
bug18781.diff3.txt uploaded by alecdavis (license 585)
Tested by: alecdavis, ZX81
Review: https://reviewboard.asterisk.org/r/1128/
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r310240 | twilson | 2011-03-10 10:05:45 -0600 (Thu, 10 Mar 2011) | 13 lines
Add \r\n to remaining http headers passed to ast_http_send
r309204 changed the behavior of ast_http_send. It now requires headers
to be passed with trailing \r\n. This change updates the remaining
instances in the code that did not pass the \r\n.
(closes issue #18186)
Reported by: nivaldomjunior
Patches:
res_phoneprov.c.diff uploaded by lathama (license 1028)
manager.diff.txt uploaded by twilson (license 396)
Tested by: lathama
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r310231 | mmichelson | 2011-03-10 09:17:04 -0600 (Thu, 10 Mar 2011) | 9 lines
Be more tolerant of what URI we accept for call completion PUBLISH requests.
(closes issue #18946)
Reported by: GeorgeKonopacki
Patches:
18946.patch uploaded by mmichelson (license 60)
Tested by: GeorgeKonopacki
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r310142 | tilghman | 2011-03-09 23:53:29 -0600 (Wed, 09 Mar 2011) | 19 lines
Merged revisions 310141 via svnmerge from
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r310141 | tilghman | 2011-03-09 23:51:37 -0600 (Wed, 09 Mar 2011) | 12 lines
Merged revisions 310140 via svnmerge from
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r310140 | tilghman | 2011-03-09 23:38:44 -0600 (Wed, 09 Mar 2011) | 5 lines
Initialize column size to 0 to deal with a potential UnixODBC bug on 64-bit systems.
(closes issue #18295)
Reported by: pruiz
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r310088 | jrose | 2011-03-08 14:19:32 -0600 (Tue, 08 Mar 2011) | 9 lines
Returns with an error notice if CHANNEL function of SIP channel is read without arguments.
(Closes issue #18653)
Reported by: wuwu
Patches:
diff.patch uploaded by jrose (license 1225)
Tested by: jrose
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r310039 | twilson | 2011-03-08 10:10:50 -0800 (Tue, 08 Mar 2011) | 11 lines
Spelling fix in "calendar show calendar"
s/Cartegories/Catagories/
(closes issue #18931)
Reported by: pdugas
Patches:
res_calendar.c.patch uploaded by pdugas (license 1222)
Review: [full review board URL with trailing slash]
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r309994 | rmudgett | 2011-03-08 10:37:02 -0600 (Tue, 08 Mar 2011) | 1 line
Make pri parameter description consistent.
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r309858 | jrose | 2011-03-07 16:07:25 -0600 (Mon, 07 Mar 2011) | 22 lines
Merged revisions 309857 via svnmerge from
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r309857 | jrose | 2011-03-07 16:04:44 -0600 (Mon, 07 Mar 2011) | 15 lines
Merged revisions 309856 via svnmerge from
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r309856 | jrose | 2011-03-07 16:02:12 -0600 (Mon, 07 Mar 2011) | 8 lines
Bug fix for MixMonitor involving filenames with '.' not in the extension
Closes issue #18391)
Reported by: pabelanger
Patches:
bugfix.patch uploaded by jrose (license 1225)
Tested by: jrose
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r309808 | tilghman | 2011-03-06 18:54:42 -0600 (Sun, 06 Mar 2011) | 14 lines
Merged revisions 309251 via svnmerge from
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r309251 | tilghman | 2011-03-01 19:06:02 -0600 (Tue, 01 Mar 2011) | 7 lines
Revert previous 2 commits, and instead conditionally redefine the same macro used in flex 2.5.35 that clashed with our workaround.
Not surprisingly, the workaround was exactly the same code as was provided by
the Flex maintainers, albeit in two different places, in different macros.
This should fix the FreeBSD builds, which have an older version of Flex.
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r309765 | mmichelson | 2011-03-06 18:13:36 -0600 (Sun, 06 Mar 2011) | 3 lines
Indicate that Asterisk uses the Allow header to determine if MESSAGE requests should be sent.
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r309720 | moy | 2011-03-05 12:44:30 -0500 (Sat, 05 Mar 2011) | 6 lines
Fix caller id passed to openr2_chan_make_call
(closes issue #18894)
Reported by: malufrj
Tested by: moy
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r309678 | tilghman | 2011-03-05 04:29:30 -0600 (Sat, 05 Mar 2011) | 14 lines
Merged revisions 309677 via svnmerge from
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r309677 | tilghman | 2011-03-05 04:28:24 -0600 (Sat, 05 Mar 2011) | 7 lines
Missed part of the conversion when we started passing ppid to astcanary.
(closes issue #18850)
Reported by: viraptor
Patches:
canary_ppid.patch uploaded by viraptor (license 543)
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Adding the setvar option with variable substitution on the value allows things
like setting the outbound caller id name to the summary of a calendar event,
etc. Values could be chained together as they are appended in order to do some
scripting if necessary.
Review: https://reviewboard.asterisk.org/r/1134/
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r309585 | mnicholson | 2011-03-04 13:38:25 -0600 (Fri, 04 Mar 2011) | 9 lines
Merged revisions 309584 via svnmerge from
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r309584 | mnicholson | 2011-03-04 13:37:13 -0600 (Fri, 04 Mar 2011) | 2 lines
Restore mysterious lua_pushvalue() call removed in r309494. The mystery has been solved.
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