https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r193755 | tilghman | 2009-05-11 17:48:20 -0500 (Mon, 11 May 2009) | 18 lines
Move 300 bytes around on the stack, to make more room for an extension buffer.
This allows more concurrent extensions to be copied for a single voicemail,
without creating a possibility of upsetting existing users, where a dialplan
could run out of stack space where it had run fine before. Alternatively,
we could have allocated off the heap, but that is a larger change and would
have increased the chance for instability introduced by this change.
This is really solved starting in 1.6.0.11, as the use of an ast_str buffer
allows an unlimited number of extensions (up to available memory). We
additionally create a new warning message when the buffer length is exceeded,
permitting administrators to see an issue after the fact, whereas previously
the list was silently truncated.
(closes issue #14739)
Reported by: p_lindheimer
Patches:
20090417__bug14739.diff.txt uploaded by tilghman (license 14)
Tested by: p_lindheimer
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193756 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In res_timer_timerfd, handle the case that set_rate gets called while a timer
is still in continuous mode. In this case, we want to remember the configured
rate, but not actually set it until continuous mode has been disabled.
Thanks to dvossel for finding and helping to debug the problem.
(closes issue #15080)
Reported by: dvossel
Tested by: dvossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193718 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r193613 | rmudgett | 2009-05-11 14:09:00 -0500 (Mon, 11 May 2009) | 12 lines
Sent wrong message to clear a call we started if the other end has not responed yet.
In the state MISDN_CALLING (i.e. SETUP was sent but no answer has arrived yet),
it is not allowed to clear the call with RELEASE_COMPLETE. It must be
cleared with DISCONNECT. A RELEASE_COMPLETE is only allowed as an answer
to a SETUP. (See Q.931 ch. 5.3.2, 5.3.2.a, 5.3.2.b)
Patches:
chan-misdn-ccstate7.patch uploaded by customer.
JIRA ABE-1862
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193614 65c4cc65-6c06-0410-ace0-fbb531ad65f3
find_peer() does not find a valid peer when using pvt->recv as the sockaddr_in argument. Because of the way TCP works, the port number in pvt->recv is not what we're looking for at all. There is currently only one place that find_peer searches for a peer using the sockaddr_in argument. If the peer is not found after using pvt->recv (works for UDP since the port number will be correct), a temp sockaddr_in struct is made using the Contact header in the sip_request. This has the correct port number in it.
Review: http://reviewboard.digium.com/r/236/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193387 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This helps to prevent odd scenarios where a queue will claim to have
taken 0 calls, but the members appear to have taken a non-zero amount.
(closes issue #15068)
Reported by: sum
Patches:
patchreset.patch uploaded by sum (license 766)
Tested by: sum
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193349 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r193193 | kpfleming | 2009-05-08 09:03:28 -0500 (Fri, 08 May 2009) | 7 lines
Make absolute paths for logger channels work properly
(Note: This is not a new feature, it was previously undocumented and broken.)
The Asterisk logger has a feature to support absolute pathnames for logger channels, but the code implementing the feature was broken. This has been fixed, and the absolute path feature is now documented in the sample logger.conf.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193194 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r193119 | tilghman | 2009-05-07 18:41:11 -0500 (Thu, 07 May 2009) | 19 lines
Fix Background within a Macro for FreePBX.
If the single digit DTMF is an extension in the specified context, then
go there and signal no DTMF. Otherwise, we should exit with that DTMF.
If we're in Macro, we'll exit and seek that DTMF as the beginning of an
extension in the Macro's calling context. If we're not in Macro, then
we'll simply seek that extension in the calling context. Previously,
someone complained about the behavior as it related to the interior of a
Gosub routine, and the fix (#14011) inadvertently broke FreePBX
(#14940). This change should fix both of these situations, but with the
possible incompatibility that if a single digit extension does not exist
(but a longer extension COULD have matched), it would have previously
gone immediately to the "i" extension, but will now need to wait for a
timeout.
(closes issue #14940)
Reported by: p_lindheimer
Patches:
20090420__bug14940.diff.txt uploaded by tilghman (license 14)
Tested by: p_lindheimer
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r193050 | rmudgett | 2009-05-07 17:17:06 -0500 (Thu, 07 May 2009) | 5 lines
Give a more helpful message when an incoming call's dialed extension does not match.
Added the dialed extension and context to the chan_misdn messages warning
that the dialed number cannot be matched in the dialplan.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193077 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r192932 | tilghman | 2009-05-07 11:29:08 -0500 (Thu, 07 May 2009) | 10 lines
Eliminate repetition of fullcontact during reconstruction.
If the fullcontact field appears in both the sippeers and the
sipregs table, then during reconstruction of the field, it will
otherwise be doubled.
(closes issue #14754)
Reported by: Alexei Gradinari
Patches:
20090506__bug14754.diff.txt uploaded by tilghman (license 14)
Tested by: lmadsen
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192933 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r192858 | jpeeler | 2009-05-06 17:15:19 -0500 (Wed, 06 May 2009) | 10 lines
Make ParkedCall application stop execution of the dialplan after hang up
Just changed park_exec to always return non-zero. I really wasn't entirely sure
at first if this was a bug. Decided it was since it would be surprising when
not using ParkedCall in the dialplan to hang up and have dialplan execution
continue.
(closes issue #14555)
Reported by: francesco_r
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192861 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This scenario crept up if chan_iax2 was loaded with no configuration file present.
It would create a timer and tell it to go at an interval but the thread that normally
acknowledges it would not be created because no configuration file was present. The timer
will now be closed if no configuration file is present.
(closes issue #15014)
Reported by: madkins
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192808 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r192633 | file | 2009-05-06 10:30:51 -0300 (Wed, 06 May 2009) | 7 lines
Update some old logic to stop both begin and end DTMF frames from reaching the core if rfc2833 is not enabled.
(closes issue #15036)
Reported by: dimas
Patches:
v1-15036.patch uploaded by dimas (license 88)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192634 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The local struct ast_party_connected_line connected_caller variable
was uninitialized when the copy function was called.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192590 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r192524 | seanbright | 2009-05-05 15:56:11 -0400 (Tue, 05 May 2009) | 11 lines
Fix Javascript error when using astman.js in Internet Explorer.
Internet Explorer (tested with 7.0) does not like trailing commas on constructs
like object initializers, so get rid of them to avoid some errors.
(closes issue #15026)
Reported by: rajnishgiri
Patches:
bug15026.patch uploaded by seanbright (license 71)
Tested by: seanbright
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r192454 | file | 2009-05-05 15:22:27 -0300 (Tue, 05 May 2009) | 8 lines
Fix an incorrect assumption that certain values on the channel will always exist when they may not.
The CDR code involved with bridges wrongly assumed that the currently executing application and data
values will always exist. It is possible for this to be false when call forwarding is involved.
(closes issue #14984)
Reported by: gincantalupo
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192462 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If an incoming call authenticated as a user or peer and t38pt_udptl was
not set to yes in general then no UDPTL session would be present and any
T38 related things would fail. This commit changes it so that if after
authenticating T38 is enabled but no UDPTL session is present one will be
created.
(issue AST-215)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192387 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit adds an API call that can be used to allocate a structure along with this stringfield storage in a single allocation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192362 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Under some conditions, the memory allocation for stringfields and ao2 objects would not have supplied valid file/function names for MALLOC_DEBUG tracking, so this commit corrects that.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192357 65c4cc65-6c06-0410-ace0-fbb531ad65f3
As in previous commits, when channels are allocated (with ast_channel_alloc) or datastores are allocated (with ast_datastore_alloc) properly account for the memory being owned by the caller, instead of the allocator function itself.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192318 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit modifies the stringfield pool allocator to remember the 'owner' of the stringfield manager the pool is being allocated for, and ensures that pools allocated in the future when fields are populated are owned by that file/function.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192279 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r192213 | dvossel | 2009-05-04 17:37:31 -0500 (Mon, 04 May 2009) | 11 lines
global mohinterpret setting is ignored
mohinterpret and mohsuggest global variables were not copied over during build_users and build_peers.
(closes issue #14728)
Reported by: dimas
Patches:
v1-14728.patch uploaded by dimas (license 88)
Tested by: dimas, dvossel
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192214 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit ensures that all astobj2 allocated objects are properly accounted for in MALLOC_DEBUG mode by passing down the file/function/line information from the module/function that actually called the astobj2 allocation function.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192059 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Correct command description in test_sched.c and include asterisk/cli.h in test_skel.c, since it's highly unlikely that a test module will *not* want to provide CLI commands to execute the tests
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191997 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This configuration file was changed to ensure that only one console channel driver
(chan_oss) is loaded by default, but the change would only work if chan_console
was not built. Now it will work as expected; if chan_alsa or chan_console are built
and installed, they will not be loaded unless explicity requested.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191955 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch add a new payload type for information elements, a set
of bit flags. The payload is transported as a 32-bit unsigned integer
but when matching is performed between events and subscribers,
the matching is done by using a bitwise AND instead of numeric value
comparison.
Review: http://reviewboard.asterisk.org/r/242/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191919 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- convert handling of IE PLTYPEs to switch statements
- add braces to various small blocks
- remove a bit of trailing whitespace
- remove a couple of unnecessary ast_strdupa() uses
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191848 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In discussions today at the Europe Asterisk Developer Meet-Up, we determined that
the event_log was used in only 9 places in the entire tree, and really was not needed
at all. The users have been converted to use LOG_NOTICE, or the messages have been
removed since other messages were already in place that provided the same information.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191785 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r191778 | mmichelson | 2009-05-02 13:48:20 -0500 (Sat, 02 May 2009) | 11 lines
Fix a bug which resulted from the Hebrew voicemail commit.
This fixes a case where a certain message could get played twice.
(closes issue #13155)
Reported by: greenfieldtech
Patches:
app_voicemail.c.multi-lang-patch uploaded by greenfieldtech (license 369)
Tested by: greenfieldtech
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191781 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This code was copy-and-pasted without properly changing references to event_rotate into queue_rotate, so under some conditions the log rotation would rotate queue_log even though it was not necessary.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191775 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Tilghman Lesher
Russell Bryant
Richard Mudgett
Leif Madsen
Joshua Colp
David Vossel
Mark Michelson
Sean Bright
Kevin P. Fleming
Jeff Peeler
Matthew Fredrickson
Eliel C. Sardanons