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${ noResults }
23181 Commits (a37fb2e8c89332f4cd9facf1884d982813614f0c)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Richard Mudgett
|
1420522c6e |
Fixed __ao2_ref() validating user_data twice.
(closes issue ASTERISK-19755)
Reported by: Gunther Kelleter
Patches:
ao2_ref.patch (license #6372) patch uploaded by Gunther Kelleter
........
Merged revisions 364902 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 364903 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364910 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
14 years ago |
Mark Michelson
|
b5f0647fc8 |
Fix Coverity-reported ARRAY_VS_SINGLETON error.
As it turned out, this wasn't a huge deal. We were calling ast_app_parse_options() for a set of options of which none took arguments. The proper thing to do for this case is to pass NULL for the "args" parameter here. We were instead passing a seemingly-randomly chosen char * from the function. While this would never get written to, you can rest assured things would have gotten bad had new options (which took arguments) been added to func_volume. (closes issue ASTERISK-19656) ........ Merged revisions 364899 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364900 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364901 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Richard Mudgett
|
2b7e1affac |
* Fix error path resouce leak in local_request().
* Restructure local_request() to reduce indentation. ........ Merged revisions 364840 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364845 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364846 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Jason Parker
|
885fbf6b04 |
Prevent a potential crash when using manager hooks.
Found by me while poking at DPMA-127. ........ Merged revisions 364841 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364842 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364844 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Kinsey Moore
|
a965f18695 |
Play conf-placeintoconf message to the correct channel
Correct the code in app_confbridge to play the conf-placeintoconf message to the marked user entering the bridge instead of to the conference while the marked user hears silence. (closes issue ASTERISK-19641) Reported-by: Mark A Walters ........ Merged revisions 364786 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364787 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364788 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Jonathan Rose
|
cefff2e52c |
Fix bad check in voicemail functions for ast_inboxcount2_func
Check looks for ast_inboxcount_func instead of ast_inboxcount2_func on ast_inboxcount2_func calls. (closes issue ASTERISK-19718) Reported by: Corey Farrell Patches: ast_app_inboxcount2-null-refcheck.patch uploaded by Corey Farrell (license 5909) ........ Merged revisions 364769 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364777 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364785 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Mark Michelson
|
6eb1ea3b79 |
Revert revision 360862.
Revision 360862 was intended to improve identities sent in dialog-info NOTIFY requests. Some users reported that hint became broken once this was done. It's not clear exactly what part of the patch has caused this regression, but broken hints are bad. For now, this revision is being reverted so that the next releases of Asterisk do not have bad behavior in them. The original reported issue will have to be fixed differently in the next version of Asterisk. (issue ASTERISK-16735) ........ Merged revisions 364706 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364707 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364708 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Mark Murawki
|
f3cde589cd |
Merged revisions 364635 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r364635 | markm | 2012-04-30 11:51:12 -0400 (Mon, 30 Apr 2012) | 10 lines Sanatize result from bfd_find_nearest_line (BETTER_BACKTRACES) bfd_find_nearest_line can possibly set file to null resulting in a crash when strrchr(file) runs (closes issue ASTERISK-19815) Reported by Mark Murawski Tested by Mark Murawski ........ ........ Merged revisions 364650 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364654 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Alexandr Anikin
|
6d87267542 |
Fix use freed pointer in return value from call thread
(issue ASTERISK-19663) Reported by: Matt Jordan Patches: ASTERISK-19663-ooh323.patch (License #5415) ........ Merged revisions 364649 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364651 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364652 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Matthew Jordan
|
60d9ca687c |
Fix error that caused truncate operations to fail
Another very inappropriate placement of a ')' (again introduced in r362151) caused the various truncate operations to attempt to truncate the sound file at a position of '0'. (issue ASTERISK-19655) Reported by: Matt Jordan (issue ASTERISK-19810) Reported by: colbec ........ Merged revisions 364578 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364579 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364580 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Michael L. Young
|
2cbcbc7f6b |
Fix configuring custom sound_leader_has_left in confbridge.conf
The configuration option to specify a custom sound_leader_has_left file for a conference bridge was not being parsed. This patch fixes it so that a custom sound file will now be used. (closes issue ASTERISK-19771) Reported by: Pawel Kuzak Tested by: Pawel Kuzak, Michael L. Young Patches: leaderhasleft_sound.dpatch uploaded by Pawel Kuzak (license 6380) Review: https://reviewboard.asterisk.org/r/1884/ ........ Merged revisions 364536 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364537 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Joshua Colp
|
ae1502be33 |
Add support for lightweight NAT keepalive.
If enabled using the keepalive option in sip.conf a small packet will be sent at a regular interval to keep the NAT mapping open. This is lightweight as the remote side does not need to parse and handle a SIP message. (closes issue AST-783) Review: https://reviewboard.asterisk.org/r/1756/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364500 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Russell Bryant
|
19097a4b62 |
md5: supress some compiler warnings.
md5.c: In function ‘MD5Final’: md5.c:154:2: error: dereferencing type-punned pointer will break strict-aliasing rules [-Werror=strict-aliasing] md5.c:155:2: error: dereferencing type-punned pointer will break strict-aliasing rules [-Werror=strict-aliasing] There is an md5 unit test and it still passes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364462 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Russell Bryant
|
eebdf35159 |
res_corosync: Fix build against corosync 2.0.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364444 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Russell Bryant
|
a498ef2aa0 |
app_minivm: Fix a couple compiler warnings.
The warnings were about argv[0] being used uninitialized, which is correct. Just remove setting username to this value, since username is set again before it actually gets used. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364438 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Russell Bryant
|
386c2c6edf |
features: Add FEATURE() and FEATUREMAP() functions.
Add two new dialplan functions: FEATURE() and FEATUREMAP(). FEATURE() lets you set some of the configuration options from the [general] section of features.conf on a per-channel basis. FEATUREMAP() lets you customize the key sequence used to activate built-in features, such as blindxfer, and automon. See the built-in documentation for details. Review: https://reviewboard.asterisk.org/r/1871/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364437 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Richard Mudgett
|
e8a6e0ef0e |
PreDial - Ability to run dialplan on callee and caller channels before Dial.
Thanks to Mark Murawski for the initial patch and feature definition. (closes issue ASTERISK-19548) Reported by: Mark Murawski Review: https://reviewboard.asterisk.org/r/1878/ Review: https://reviewboard.asterisk.org/r/1229/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364436 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Terry Wilson
|
f7c174ff84 |
Multiple revisions 364365,364369
........ r364365 | twilson | 2012-04-27 17:31:01 -0500 (Fri, 27 Apr 2012) | 11 lines Fix ast_parse_arg numeric type range checking and add tests ast_parse_arg wasn't checking for strto* parse errors or limiting the results by the actual range of the numeric types. This patch fixes that and adds unit tests as well. Review: https://reviewboard.asterisk.org/r/1879/ ........ Merged revisions 364340 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r364369 | twilson | 2012-04-27 17:33:10 -0500 (Fri, 27 Apr 2012) | 2 lines Add missing test_config.c ........ Merged revisions 364365,364369 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364397 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Mark Michelson
|
1a58b3b775 |
Don't attempt to make use of the dynamic_exclude_static ACL if DNS lookup fails.
(closes issue ASTERISK-18321) Reported by Dan Lukes Patches: ASTERISK-18321.patch by Mark Michelson (license #5049) ........ Merged revisions 364341 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364342 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364343 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Matthew Jordan
|
50c959580c |
Prevent overflow in calculation in ast_tvdiff_ms on 32-bit machines
The method ast_tvdiff_ms attempts to calculate the difference, in milliseconds, between two timeval structs, and return the difference in a 64-bit integer. Unfortunately, it assumes that the long tv_sec/tv_usec members in the timeval struct are large enough to hold the calculated values before it returns. On 64-bit machines, this might be the case, as a long may be 64-bits. On 32-bit machines, however, a long may be less (32-bits), in which case, the calculation can overflow. This overflow caused significant problems in MixMonitor, which uses the method to determine if an audio factory, which has not presented audio to an audiohook, is merely late in providing said audio or will never provide audio. In an overflow situation, the audiohook would incorrectly determine that an audio factory that will never provide audio is merely late instead. This led to situations where a MixMonitor never recorded any audio. Note that this happened most frequently when that MixMonitor was started by the ConfBridge application itself, or when the MixMonitor was attached to a Local channel. (issue ASTERISK-19497) Reported by: Ben Klang Tested by: Ben Klang Patches: 32-bit-time-overflow-10-2012-04-26.diff (license #6283) by mjordan (closes issue ASTERISK-19727) Reported by: Mark Murawski Tested by: Michael L. Young Patches: 32-bit-time-overflow-2012-04-27.diff (license #6283) by mjordan) (closes issue ASTERISK-19471) Reported by: feyfre Tested by: feyfre (issue ASTERISK-19426) Reported by: Johan Wilfer Review: https://reviewboard.asterisk.org/r/1889/ ........ Merged revisions 364277 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364285 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364287 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Kinsey Moore
|
83cf78deda |
Allow SIP pvts involved in Replaces transfers to fall out of reference sooner
Unref the SIP pvt stored in the refer structure as soon as it is no longer needed so that the pvt and associated file descriptors can be freed sooner. This change makes a reference decrement unnecessary in code that handles SIP BYE/Also transfers which should not touch the reference anyway. (Closes issue ASTERISK-19579) Reported by: Maciej Krajewski Tested by: Maciej Krajewski ........ Merged revisions 364258 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364259 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364260 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Matthew Jordan
|
103031330a |
Allow for reloading SRTP crypto keys within the same SIP dialog
As a continuation of the patch in r356604, which allowed for the reloading of SRTP keys in re-INVITE transfer scenarios, this patch addresses the more common case where a new key is requested within the context of a current SIP dialog. This can occur, for example, when certain phones request a SIP hold. Previously, once a dialog was associated with an SRTP object, any subsequent attempt to process crypto keys in any SDP offer - either the current one or a new offer in a new SIP request - were ignored. This patch changes this behavior to only ignore subsequent crypto keys within the current SDP offer, but allows future SDP offers to change the keys. (issue ASTERISK-19253) Reported by: Thomas Arimont Tested by: Thomas Arimont Review: https://reviewboard.asteriskorg/r/1885/ ........ Merged revisions 364203 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364204 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364205 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Stefan Schmidt
|
14b52ff9da |
fix a wrong behavior of alarm timezones in caldav and icalendar when an alarm doesnt use utc. This change uses the same timezone from the start time.
........ Merged revisions 364163 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364164 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Richard Mudgett
|
238640dc1b |
Update Pickup application documentation. (With feeling this time.)
........ Merged revisions 364108 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364109 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364110 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Richard Mudgett
|
b22874415e |
Fix DTMF atxfer running h exten after the wrong bridge ends.
When party B does an attended transfer of party A to party C, the
attending bridge between party B and C should not be running an h exten
when the bridge ends. Running an h exten now sets a softhangup flag to
ensure that an AGI will run in dead AGI mode.
* Set the AST_FLAG_BRIDGE_HANGUP_DONT on the party B channel for the
attending bridge between party B and C.
(closes issue AST-870)
(closes issue ASTERISK-19717)
Reported by: Mario
(closes issue ASTERISK-19633)
Reported by: Andrey Solovyev
Patches:
jira_asterisk_19633_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett, Andrey Solovyev, Mario
........
Merged revisions 364060 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 364065 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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14 years ago |
|
Terry Wilson
|
49a49a51ef |
Add more constness to the end_buf pointer in the netconsole
issue ASTERISK-18308 Review: https://reviewboard.asterisk.org/r/1876/ ........ Merged revisions 364046 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 364047 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364048 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Olle Johansson
|
e5c20ccb76 |
Code formatting fixes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363989 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Kinsey Moore
|
7bf6a01cfa |
Fix reference leaks involving SIP Replaces transfers
The reference held for SIP blind transfers using the Replaces header in an INVITE was never freed on success and also failed to be freed in some error conditions. This caused a file descriptor leak since the RTP structures in use at the time of the transfer were never freed. This reference leak and another relating to subscriptions in the same code path have now been corrected. (closes issue ASTERISK-19579) ........ Merged revisions 363986 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363987 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363988 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Alec L Davis
|
5746e0d2ac |
chan_sip: [general] maxforwards, not checked for a value greater than 255
The peer maxforwards is checked for both '< 1' and '> 255', but the default 'maxforwards' in the [general] section is only checked for '< 1' alecdavis (license 585) Reported by: alecdavis Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1888/ ........ Merged revisions 363934 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363935 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363936 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Richard Mudgett
|
9d655bd0e8 |
Update Pickup application documentation. (Even better)
........ Merged revisions 363875 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363876 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363877 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Richard Mudgett
|
e736a4fed3 |
* Put more information in pickup_exec() LOG_NOTICE.
* Delay duplicating a string on the stack in pickup_exec(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363839 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Richard Mudgett
|
0986873128 |
Update Pickup application documentation.
........ Merged revisions 363788 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363789 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363790 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Richard Mudgett
|
af39a4374e |
Make DAHDISendCallreroutingFacility wait 5 seconds for a reply before disconnecting the call.
Some switches may not handle the call-deflection/call-rerouting message if
the call is disconnected too soon after being sent. Asteisk was not
waiting for any reply before disconnecting the call.
* Added a 5 second delay before disconnecting the call to wait for a
potential response if the peer does not disconnect first.
(closes issue ASTERISK-19708)
Reported by: mehdi Shirazi
Patches:
jira_asterisk_19708_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett
........
Merged revisions 363730 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 363734 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363740 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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14 years ago |
|
Richard Mudgett
|
d2ac624b87 |
Clear ISDN channel resetting state if the peer continues to use it.
Some ISDN switches occasionally fail to send a RESTART ACKNOWLEDGE in
response to a RESTART request.
* Made the second SETUP received after sending a RESTART request clear the
channel resetting state as if the peer had sent the expected RESTART
ACKNOWLEDGE before continuing to process the SETUP. The peer may not be
sending the expected RESTART ACKNOWLEDGE.
(issue ASTERISK-19608)
(issue AST-844)
(issue AST-815)
Patches:
jira_ast_815_v1.8.patch (license #5621) patch uploaded by rmudgett (modified)
........
Merged revisions 363687 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 363688 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363689 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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14 years ago |
|
Olle Johansson
|
04ddb5714f |
Add documentation
Thanks Tilghman! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363637 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Olle Johansson
|
f102aecf12 |
Formatting changes only
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363599 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Olle Johansson
|
a8e755700e |
Use the DEFINED value for musicclass length.
For some reason, features.c has it's own definition. Should propably be fixed too. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363595 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Olle Johansson
|
7aa0c3c64b |
Make it possible to change the minimum DTMF duration in asterisk.conf
Asterisk has a setting for the minimum allowed DTMF. If we get shorter DTMF tones, these will be changed to the minimum on the outbound call leg. (closes issue ASTERISK-19772) Review: https://reviewboard.asterisk.org/r/1882/ Reported by: oej Tested by: oej Patches by: oej Thanks to the reviewers. 1.8 branch for this patch: agave-dtmf-duration-asterisk-conf-1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363558 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Olle Johansson
|
228ce5fd74 |
Formatting fixes
Developer guidelines are important. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363517 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Olle Johansson
|
db2b162e8c |
Formatting fixes
Found a small amount of curly brackets in my hotel room here in Denmark. I hereby donate them to the Asterisk project. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363480 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Richard Mudgett
|
7f0dce3bd1 |
Fix recalled party B feature flags for a failed DTMF atxfer.
1) B calls A with Dial option T 2) B DTMF atxfer to C 3) B hangs up 4) C does not answer 5) B is called back 6) B answers 7) B cannot initiate transfers anymore * Add dial features datastore to recalled party B channel that is a copy of the original party B channel's dial features datastore. * Extracted add_features_datastore() from add_features_datastores(). * Renamed struct ast_dial_features features_caller and features_callee members to my_features and peer_features respectively. These better names eliminate the need for some explanatory comments. * Simplified code accessing the struct ast_dial_features datastore. (closes issue ASTERISK-19383) Reported by: lgfsantos ........ Merged revisions 363428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363429 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363430 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Richard Mudgett
|
56d10c5677 |
Hangup affected channel in error paths of bridge_call_thread().
........ Merged revisions 363375 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363376 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363377 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Terry Wilson
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18045c9a07 |
OpenBSD doesn't have rawmemchr, use strchr
(closes issue ASTERISK-19758) Reported by: Barry Miller Tested by: Terry Wilson Patches: 362758-diff uploaded by Barry Miller (license 5434) ........ Merged revisions 362868 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362869 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363335 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Richard Mudgett
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f663924517 |
Make app_dial and app_queue use new macro and gosub calls.
* Simplify some code in app_dial and app_queue by calling ast_app_exec_macro() and ast_app_exec_sub(). * Fix minor locking issue in app_dial for post-answer macro/gosub MACRO/GOSUB_RESULT=GOTO: handling. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363269 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Tilghman Lesher
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f03d56a84d |
On some platforms, O_RDONLY is not a flag to be checked, but merely the absence of O_RDWR and O_WRONLY.
The POSIX specification does not mandate how these 3 flags must be specified, only that one of the three must be specified in every call. ........ Merged revisions 363209 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363212 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363215 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Jonathan Rose
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ceefcf8839 |
AST-2012-004: Fix an error that allows AMI users to run shell commands sans authorization.
As detailed in the advisory, AMI users without write authorization for SYSTEM class AMI actions were able to run system commands by going through other AMI commands which did not require that authorization. Specifically, GetVar and Status allowed users to do this by setting their variable/s options to the SHELL or EVAL functions. Also, within 1.8, 10, and trunk there was a similar flaw with the Originate action that allowed users with originate permission to run MixMonitor and supply a shell command in the Data argument. That flaw is fixed in those versions of this patch. (closes issue ASTERISK-17465) Reported By: David Woolley Patches: 162_ami_readfunc_security_r2.diff uploaded by jrose (license 6182) 18_ami_readfunc_security_r2.diff uploaded by jrose (license 6182) 10_ami_readfunc_security_r2.diff uploaded by jrose (license 6182) ........ Merged revisions 363117 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ Merged revisions 363141 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363156 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363159 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Matthew Jordan
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e8e12afc6a |
AST-2012-006: Fix crash in UPDATE handling when no channel owner exists
If Asterisk receives a SIP UPDATE request after a call has been terminated and the channel has been destroyed but before the SIP dialog has been destroyed, a condition exists where a connected line update would be attempted on a non-existing channel. This would cause Asterisk to crash. The patch resolves this by first ensuring that the SIP dialog has an owning channel before attempting a connected line update. If an UPDATE request is received and no channel is associated with the dialog, a 481 response is sent. (closes issue ASTERISK-19770) Reported by: Thomas Arimont Tested by: Matt Jordan Patches: ASTERISK-19278-2012-04-16.diff uploaded by Matt Jordan (license 6283) ........ Merged revisions 363106 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363107 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363108 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Matthew Jordan
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c37c7b4a2c |
AST-2012-005: Fix remotely exploitable heap overflow in keypad button handling
When handling a keypad button message event, the received digit is placed into a fixed length buffer that acts as a queue. When a new message event is received, the length of that buffer is not checked before placing the new digit on the end of the queue. The situation exists where sufficient keypad button message events would occur that would cause the buffer to be overrun. This patch explicitly checks that there is sufficient room in the buffer before appending a new digit. (closes issue ASTERISK-19592) Reported by: Russell Bryant ........ Merged revisions 363100 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ Merged revisions 363102 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363103 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363105 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Russell Bryant
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eb0a8df41c |
res_corosync: Recover if corosync gets restarted.
If corosync gets restarted while Asterisk is running, automatically recover. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363046 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Russell Bryant
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41826d203c |
res_corosync: reimplement "corosync show members" command.
Reimplement the "corosync show members" CLI command using a CPG iterator instead of the cpg_membership_get API call. This will also show all CPG members, including those in groups other than 'asterisk', which may be useful at some point for debugging purposes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363045 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |