mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
22603 Commits (c6142cf2cc7312122abcf74d6ab1daf0ad1d72bf)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Kinsey Moore
|
c6142cf2cc |
Fix coverity UNUSED_VALUE findings in core support level files
Most of these were just saving returned values without using them and in some cases the variable being saved to could be removed as well. (issue ASTERISK-19672) ........ Merged revisions 368738 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368739 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368751 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Kinsey Moore
|
566ea22e18 |
Recorded merge of revisions 368721 from http://svn.asterisk.org/svn/asterisk/branches/10
........ Fix compilation in dev-mode Backport a compilation fix in md5.c from trunk that only showed up in dev-mode under certain compiler versions. ........ Merged revisions 368719 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368722 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Richard Mudgett
|
745484e1b3 |
Fix error paths in action_hangup() for AMI Hangup action.
* Check allocation function return values for failure. Crashing is bad. * Tweak ast_regex_string_to_regex_pattern() parameters for proper ast_str usage. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368714 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Richard Mudgett
|
8b2412db28 |
Tweak ast_channel_softhangup_withcause_locked() to take a typed parameter.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368712 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Igor Goncharovskiy
|
4ca35e0907 |
Fix MWI update so LED display correct voicemail state after phone usage. Also fixes few warnings.
|
13 years ago |
Damien Wedhorn
|
d979399071 |
Skinny cleanup (mwi_event_cb).
Original was testing for d->session, setting and testing again (all nested). Removed duplicate testing and restructured function to test/return and then the main code. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368681 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Damien Wedhorn
|
0271734f2e |
Skinny cleanup.
Removed d->registered which was mirroring d->session. Changed relevant references to use d->session instead. Moved setting and unsetting of l->device from session register to device configuration. As such, l->device will always be valid unless it is has not been configured to a device. Revised various test where checking if a device is registered to use l->device->session. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368680 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Richard Mudgett
|
3f59ad990c |
Fix app_queue debug message use of args.options after the string has been parsed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368675 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Richard Mudgett
|
9ecd6c9ab4 |
Fix inverted test in app_queue for ringinuse.
Regression from -r367080 ringinuse commit. (issue ASTERISK-19536) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368674 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Terry Wilson
|
9f704b5d59 |
Fix reloading an unchanged file with the Config Options API
Adding multiple file support broke reloading an unchanged file. This adds an enum for return values for the aco_process_* functions and ensures that the config is not applied if res is not ACO_PROCESS_OK. Review: https://reviewboard.asterisk.org/r/1979/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368673 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Tzafrir Cohen
|
ac6ec71fd2 |
Fix a typo in format_ogg_vorbis.c: suport
Review: https://reviewboard.asterisk.org/r/1970/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368668 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Terry Wilson
|
aeeff8cfa2 |
Add default handler documentation and standardize acl handler
Added documentation describing what flags and arguments to pass to aco_option_register for default option types. Also changed the ACL handler to use the flags parameter to differentiate between "permit" and "deny" instead of adding an additional vararg parameter. Review: https://reviewboard.asterisk.org/r/1969/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368663 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Richard Mudgett
|
0f71b29e2f |
Fix POTS flash hook to orignate a second call deadlock.
A deadlock can occur when a POTS phone tries to flash hook to originate a second call for 3-way or transfer. If another process is scanning the channels container when the POTS line flash hooks then a deadlock will occur. * Release the channel and private locks when creating a new channel as a result of a flash hook. (closes issue ASTERISK-19842) Reported by: rmudgett Tested by: rmudgett ........ Merged revisions 368644 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368645 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368646 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Mark Michelson
|
ea8cf8b5f3 |
Fix a specific scenario where ACKs are not matched.
If a dialog-starting INVITE contains a to-tag, then Asterisk will respond with a 481. In this case, the resulting incoming ACK would not be matched, so Asterisk would continue retransmitting the 481 until the transaction times out. There were two issues. Asterisk, upon creating a sip_pvt would generate a local tag. However, when the time came to transmit the 481, since there was a to-tag in the INVITE, Asterisk would place this original to-tag in the 481 response. When the ACK came in, Asterisk would attempt to match the to-tag in the ACK to the generated local tag. Unfortunately, Asterisk never actually transmitted a response with the generated local tag, so the to-tag in the ACK would not match. The other problem was that when the 481 was sent, nothing was set on the sip_pvt to indicate what CSeq is expected in the ACK. To fix the first problem, we zero out the to-tag seen in the incoming INVITE. This way, Asterisk, when time to send a response, will send its generated local tag instead. To fix the second problem, we set the sip_pvt's pendinginvite to the CSeq of the INVITE when we send a 481. (closes issue ASTERISK-19892) Reported by Mark Michelson ........ Merged revisions 368625 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368629 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368637 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Matthew Jordan
|
d197f69107 |
Add feature modifier to versions produced from branches
Certain branches, such as Certified Asterisk, may have a modifier added to them that specifies the features available in that branch. For branches, this modifier is expected to be reflected in the location of the branch in subversion. For example, a subversion of URL of /certified/branches/1.8.11 would have a feature modifier of 'certified'. This is slightly different then how features are determined for tags, where the feature is part of the actual tag name, e.g., "10.5.0-digiumphones". In keeping with the nomenclature used for tags, the feature specifier for branches is translated and placed after the revision numbers. For the example given previously, this would result in a branch version of "Asterisk SVN-branch-1.8.11-cert-rXXXXXX". ........ Merged revisions 368604 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368605 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368606 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Kinsey Moore
|
1492177b7b |
Ensure overlapping hold flags do not conflict
When changing between different modes of hold, the flags were not being cleared out properly causing a failure to change hold states. (closes issue ASTERISK-19919) Patch-by: Morten Tryfoss Reported-by: Morten Tryfoss ........ Merged revisions 368586 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368587 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368588 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Richard Mudgett
|
a2402dbe25 |
Fix parked call performing a DTMF blind transfer after being retrieved.
When a parked call was retrieved from the parking lot, it could not do a blind transfer because it caused the involved calls to be hung up unconditionally. * Made the ParkedCall application return the ast_bridge_call() return value. (closes issue ABE-2862) Reported by: Vlad Povorozniuc ........ Merged revisions 368567 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368568 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368569 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Richard Mudgett
|
faacb8ba52 |
Make builtin_blindtransfer() fully use ast_async_goto() abilities.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368566 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Jonathan Rose
|
37677a8cc2 |
Merge 'core' and 'core changes' sections in CHANGES file.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368550 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Kinsey Moore
|
f6b5fd5411 |
Recorded merge of revisions 368536 from http://svn.asterisk.org/svn/asterisk/branches/10
........ Resolve some build warnings My newly upgraded compiler caught these usages of uninitialized values. They weren't actually used. ........ Merged revisions 368533 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368537 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Kinsey Moore
|
bd958c037f |
Ensure that pages and emails are sent using RFC822-compliant date format
When localization was added to app_voicemail, these headers were altered when they should have remained in en_US format for RFC compliance. This reverts the changes to those two lines. (closes issue ASTERISK-19876) ........ Merged revisions 368520 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368524 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368529 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Kinsey Moore
|
571445ab9c |
Convert AST_FLAG_ANSWERED_ELSEWHERE usage to AST_CAUSE_ANSWERED_ELSEWHERE
This was essentially duplicated functionality where normal channels used AST_CAUSE_ANSWERED_ELSEWHERE while local channels and queues used AST_FLAG_ANSWERED_ELSEWHERE. This removes the flag and converts that usage into AST_CAUSE_ANSWERED_ELSEWHER usage. Review: https://reviewboard.asterisk.org/r/1944 (closes issue ASTERISK-19865) Patch-by: Birger Harzenetter git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368519 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Mark Michelson
|
d210685a20 |
Relay proper SIP responses on calling side.
Revision 351130 broke corect HANGUPCAUSE setting for the 404 case in chan_sip. Other cases were also potentially broken. This patch fixes the relaying of causes to be what they used to be. (closes issue ASTERISK-19914) Reported by Pavel Troller Tested by Walter Doekes (via a reviewboard test to be committed later) Patches: chan_sip.diff uploaded by Pavel Troller (license #6302) ........ Merged revisions 368498 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368499 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368500 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Richard Mudgett
|
cc69a0deaf |
Document BLINDTRANSFER behavior change.
(issue ASTERISK-19322) (closes issue ASTERISK-19875) Reported by: call ........ Merged revisions 368469 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368470 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368472 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Mark Michelson
|
f4218dc4e6 |
Also have vim syntax-highlight type=network.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368467 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Mark Michelson
|
005661bfdf |
Add vim syntax highlighting for type=line, type=phone, and type=application.
(closes issue ASTERISK-19800) Reported by: Billy Chia Patches: asterisk.vim.patch uploaded by Billy Chia (license #6381) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368466 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Mark Michelson
|
c6a2cbab19 |
Remove some extra debugging I forgot to remove in the merge of Digium phone support.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368455 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Mark Michelson
|
458f6c4bc0 |
Remove automerge properties.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368441 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Mark Michelson
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14a985560e |
Merge changes dealing with support for Digium phones.
Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Richard Mudgett
|
c1bbe79748 |
Fix potential deadlock between masquerade and chan_local.
* Restructure ast_do_masquerade() to not hold channel locks while it calls ast_indicate(). * Simplify many calls to ast_do_masquerade() since it will never return a failure now. If it does fail internally because a channel driver callback operation failed, the only thing ast_do_masquerade() can do is generate a warning message about strange things may happen and press on. * Fixed the call to ast_bridged_channel() in ast_do_masquerade(). This change fixes half of the deadlock reported in ASTERISK-19801 between masquerades and chan_iax. (closes issue ASTERISK-19537) Reported by: rmudgett Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1915/ ........ Merged revisions 368405 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368407 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368421 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Joshua Colp
|
380c7c5c39 |
Add res_http_websocket module which implements the WebSocket protocol according to RFC 6455.
Review: https://reviewboard.asterisk.org/r/1952/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368359 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Richard Mudgett
|
91a20ee2f9 |
Fix deadlock when Gosub used with alternate dialplan switches.
Attempting to remove a channel from autoservice with the channel lock held will result in deadlock. * Restructured gosub_exec() to not call ast_parseable_goto() and ast_exists_extension() with the channel lock held. (closes issue ASTERISK-19764) Reported by: rmudgett Tested by: rmudgett ........ Merged revisions 368308 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368310 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368311 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Kevin P. Fleming
|
dd02d976f5 |
Improve SDP offer/answer RFC compliance
Asterisk should not accept SDP offers that contain unknown RTP profiles (for audio/video streams) or unknown top-level media types. When it does, it answers with an SDP that does not match the offer properly, and this will nearly always result in a broken call. This patch causes such offers to be rejected. Review: https://reviewboard.asterisk.org/r/1811/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368269 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Kevin P. Fleming
|
66e5c30716 |
Improve SDP parsing warning messages
* 'Unsupported media type' is only reported when that is in fact the case, not when a supported media type is included in an 'm' line that has an invalid format. * All warning messages related to parsing 'm' lines now include the 'm' line contents. * (minor bugfix) newline added to port-number-zero warning messages. * Warning messages improved to use RFC-specified terminology for various items. * Warnings for offers that include more than one port for a single media type now include the media type. Review: https://reviewboard.asterisk.org/r/1811/ ........ Merged revisions 368218 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368267 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368268 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Terry Wilson
|
6016094db7 |
Add missing config for config API test
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368221 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Terry Wilson
|
d54717c39e |
Add new config-parsing framework
This framework adds a way to register the various options in a config file with Asterisk and to handle loading and reloading of that config in a consistent and atomic manner. Review: https://reviewboard.asterisk.org/r/1873/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368181 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Mark Michelson
|
463f9d729a |
Help mitigate potential reinvite glare scenarios.
When Asterisk servers are set up back-to-back, and direct media is to be used betweeen endpoints, it is fairly common for the two Asterisk servers to send direct media reinvites to each other simultaneously. This results in 491s and ACKs being exchanged between the servers. While the media eventually gets set up properly, the problem is that there can be a noticeable delay for the streams to stabilize. This patch adds a new directmedia option called "outgoing". With this set, an immediate direct media reinvite will only be sent if the call direction is outgoing. For incoming dialogs, an immediate direct media reinvite will not be sent, but further "reactionary" direct media reinvites may be sent. Review: https://reviewboard.asterisk.org/r/1954 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368143 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Michael L. Young
|
a7a3050de9 |
Add documentation to function CHANNEL for options echocan_mode and buffers
The ability to set "echocan_mode" and "buffers" through the dialplan was added to chan_dahdi some time ago. This patch adds some documentation to func_channel. (Closes issue ASTERISK-19911) Reported by: Dale Noll Tested by: Michael L. Young Patches: asterisk-19911-branch18.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/1949/ ........ Merged revisions 368092 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368093 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368094 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Richard Mudgett
|
dd2427c141 |
Coverity Report: Fix issues for error type REVERSE_INULL (core modules)
* Fixes findings: 0-2,5,7-15,24-26,28-31 (issue ASTERISK-19648) Reported by: Matt Jordan ........ Merged revisions 368039 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368042 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368052 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Richard Mudgett
|
fdb002a43a |
Use the DEADLOCK_AVOIDANCE() macro instead.
(issue ASTERISK-19854) ........ Merged revisions 367980 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 367981 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367982 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Richard Mudgett
|
e65ad34770 |
Fix deadlock when executing CLI "pri show channels" and "ss7 show channels" commands.
* Fix sig_pri_lock_owner() to avoid deadlock properly.
* Code pri_grab() better.
* Fix sig_ss7_lock_owner() to avoid deadlock properly.
* Code ss7_grab() better.
(closes issue ASTERISK-19854)
Reported by: Jaxon
Patches:
jira_asterisk_19854_v1.8.6.patch (license #5621) patch uploaded by rmudgett (Modified to do the same thing to sig_ss7)
Tested by: Jaxon
........
Merged revisions 367976 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 367978 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367979 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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13 years ago |
|
Richard Mudgett
|
77f5e86e4d |
Coverity Report: Fix issues for error type REVERSE_INULL (deprecated modules)
* Fix only issue pointed out by deprecated_REVERSE_INULL.txt for app_meetme.c in find_user(). * Change use of %i to %d in sscanf() in find_user(). The use of %i gives unexpected parsing because it can accept hex, octal, and decimal integer formats. * Changed other uses of %i in app_meetme() to use %d for consistency. (issue ASTERISK-19648) Reported by: Matt Jordan ........ Merged revisions 367906 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 367907 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367908 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Matthew Jordan
|
94187aafc0 |
AST-2012-008: Fix remote crash vulnerability in chan_skinny
When a skinny session is unregistered, the corresponding device pointer is set to NULL in the channel private data. If the client was not in the on-hook state at the time the connection was closed, the device pointer can later be dereferened if a message or channel event attempts to use a line's pointer to said device. The patches prevent this from occurring by checking the line's pointer in message handlers and channel callbacks that can fire after an unregistration attempt. (closes issue ASTERISK-19905) Reported by: Christoph Hebeisen Tested by: mjordan, Damien Wedhorn Patches: AST-2012-008-1.8.diff uploaded by mjordan (license 6283) AST-2012-008-10.diff uploaded by mjordan (licesen 6283) ........ Merged revisions 367844 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367845 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Richard Mudgett
|
2d418b596c |
AST-2012-007: Fix IAX receiving HOLD without suggested MOH class crash.
* Made schedule_delivery() set the received frame f->data.ptr to NULL if
the datalen is zero.
* Fix queue_signalling() memcpy() size error.
* Made queue_signalling() not use C++ keyword variable names.
(closes issue ASTERISK-19597)
Reported by: mgrobecker
Patches:
jira_asterisk_19597_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett, Michael L. Young
........
Merged revisions 367781 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 367782 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367783 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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13 years ago |
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Michael L. Young
|
2eff35bafa |
Fix pvt_sip for inbound call to use peer's allowtransfer setting
The pvt_sip allowtransfer was not being set to that of the peer's setting.
Therefore, the global allowtransfer setting was being used instead which would
lead to calls not being transfered if the global setting was set to 'no' despite
the setting on the peer being 'yes' and vice versa, calls would be allowed to
transfer even if the peer's setting was 'no' but the global setting was 'yes'.
(Closes issue ASTERISK-19856)
Reported by: Jacek
Tested by: Michael L. Young, Jacek
Patches:
issue-asterisk-19856-branch10-v3.diff uploaded by
Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/1923/
........
Merged revisions 367730 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 367731 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367732 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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13 years ago |
|
Richard Mudgett
|
e518536773 |
Fix Dial I option ignored if dial forked and one fork redirects.
The Dial and Queue I option is intended to block connected line updates and redirecting updates. However, it is a feature that when a call is locally redirected, the I option is disabled if the redirected call runs as a local channel so the administrator can have an opportunity to setup new connected line information. Unfortunately, the Dial and Queue I option is disabled for *all* forked calls if one of those calls is redirected. * Make the Dial and Queue I option apply to each outgoing call leg independently. Now if one outgoing call leg is locally redirected, the other outgoing calls are not affected. * Made Dial not pass any redirecting updates when forking calls. Redirecting updates do not make sense for this scenario. * Made Queue not pass any redirecting updates when using the ringall strategy. Redirecting updates do not make sense for this scenario. * Fixed deadlock potential with chan_local when Dial and Queue send redirecting updates for a local redirect. * Converted the Queue stillgoing flag to a boolean bitfield. (closes issue ASTERISK-19511) Reported by: rmudgett Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1920/ ........ Merged revisions 367678 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 367679 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367693 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Jonathan Rose
|
bdaecbb66b |
chan_sip: fix problem directmediapermit/deny uses the wrong address
When remotely bridging calls with directmedia, Asterisk would check the address of the peers/users holding directmedia ACLs (set via directmediapermit/directmediadeny) instead of the bridged peer. This is similar to r366547, but trunk specific and involves changes to the rtpengine instead of just chan_sip. (closes issue AST-876) review: https://reviewboard.asterisk.org/r/1924/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367640 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Jonathan Rose
|
d0ed332750 |
Blocked revisions 366591
........ chan_sip: Check the right channel's host address for directmediapermit/deny Prior to this patch, when checking the addresses for directmediapermit and directmediadeny, Asterisk would check the host address of the channel permit/deny was specified, which differs from the expectations of both our users and the development team. Instead, directmediapermit/deny now checks against the address of the channel that the peer with the ACL is connected to. (issue AST-876) Review: https://reviewboard.asterisk.org/r/1899/ ........ Merged revisions 366547 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367603 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Matthew Jordan
|
66754b3f3d |
Fix crash in ConfBridge when user announcement is played for more than 2 users
A patch introduced in r354938 made it so that ConfBridge would not attempt to play sound files if those files did not exist. Unfortunately, ConfBridge uses the same underlying function, play_sound_helper, to playback both sound files and numbers to callers. When a number is being played back, the name of the sound file is expected to be NULL. This NULL value was passed into a function that tested for the existance of a sound file and is not tolerant to NULL file names, causing a crash. This patch fixes the behavior, such that if a sound file does not exist we do not attempt to play it, but we only attempt that check if the a sound file was specified in the first place. If a sound file was not specified, we use the 'play number' logic in the helper function. (closes issue ASTERISK-19899) Reported by: Florian Gilcher Tested by: Florian Gilcher patches: asterisk-19899.diff uploaded by mjordan (license 6283) ........ Merged revisions 367562 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367563 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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Richard Mudgett
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8de31699d8 |
Made use IAX frame cache only for cacheable frame types.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367520 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |