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12.0.0
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certified/1.8.11-cert1
certified/1.8.11-cert10
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certified/1.8.11-cert3-rc1
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certified/1.8.11-cert4
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certified/1.8.11-cert5-rc1
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certified/1.8.11-cert6
certified/1.8.11-cert7
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certified/1.8.11-cert9-rc1
certified/1.8.15-cert1
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certified/1.8.15-cert2
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certified/1.8.15-cert4
certified/1.8.15-cert5
certified/1.8.15-cert6
certified/1.8.15-cert7
certified/1.8.28-cert1
certified/1.8.28-cert1-rc1
certified/1.8.28-cert2
certified/1.8.28-cert3
certified/1.8.28-cert4
certified/1.8.28-cert5
certified/1.8.6-cert1
certified/11.2-cert1
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certified/11.2-cert1-rc2
certified/11.2-cert2
certified/11.2-cert3
certified/11.6-cert1
certified/11.6-cert1-rc1
certified/11.6-cert1-rc2
certified/11.6-cert10
certified/11.6-cert11
certified/11.6-cert12
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certified/11.6-cert3
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certified/11.6-cert5
certified/11.6-cert6
certified/11.6-cert7
certified/11.6-cert8
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certified/13.1-cert1
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certified/13.1-cert1-rc2
certified/13.1-cert1-rc3
certified/13.1-cert2
certified/13.1-cert3
${ noResults }
23245 Commits (ad582f9aef40a56fc13f9eea40adb3ea51fe37a8)
Author | SHA1 | Message | Date |
---|---|---|---|
|
ad582f9aef |
Add missing test event
This test event was missing from channel.c causing the dial_LS_options test to fail intermittently because of a race condition where most code paths emitted the test event but this one did not. The dial_LS_options test should stop bouncing now. ........ Merged revisions 378455 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378459 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378460 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
da7c2e3ffe |
chan_agent: Misc code cleanup.
* Fix off-nominal path resource cleanup in agent_request(). * Create agent_pvt_destroy() to eliminate inlined versions in many places. * Pull invariant code out of loop in add_agent(). * Remove redundant module user references in login_exec(). * Remove unused struct agent_pvt logincallerid[] member. ........ Merged revisions 378456 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378457 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378458 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
11571714fe |
chan_agent: Fix agent_indicate() locking.
Avoid deadlock potential with local channels and simplify the locking. ........ Merged revisions 378427 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378428 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378429 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
7739ed9989 |
Add aliases to the Directory.
This is an interesting feature that allows additional strings to be used to search the Directory, primarily intended to be used with nicknames, but could be used with affiliations and the like. Because the name field is used in more than one place (such as email notifications), it is important that these additional strings not be placed in the name field, but be specified separately. Review: https://reviewboard.asterisk.org/r/2244/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378414 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
099cd07887 |
Prevent exhaustion of system resources through exploitation of event cache
This patch changes res_xmpp to no longer cache events under certain circumstances. (issue ASTERISK-20175) Reported by: Russell Bryant, Leif Madsen, Joshua Colp Tested by: kmoore ........ Merged revisions 378411 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378412 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
8bf1f1745b |
Prevent crashes in res_xmpp when receiving large messages
Similar to r378287, res_xmpp was marshaling data read from an external source onto the stack. For a sufficiently large message, this could cause a stack overflow. This patch modifies res_xmpp in a similar fashion to res_jabber by removing the stack allocation, as it was unnecessary. (issue ASTERISK-20658) Reported by: wdoekes ........ Merged revisions 378409 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378410 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
473ec0b1c3 |
Clean up app_mysql's application entry points to properly parse arguments
When parsing arguments, application entry points should not attempt to directly modify the parameters to the function. This patch properly duplicates the passed in parameters before attempting to parse them. (issue ASTERISK-20658) Reported by: wdoekes patches: issueA20658_sanitize_app_mysql.patch uploaded by wdoekes (license 5674) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378384 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
89f9e077d7 |
Prevent crashes from occurring when reading from data sources with large values
When reading configuration data from an Asterisk .conf file or when pulling data from an Asterisk RealTime backend, Asterisk was copying the data on the stack for manipulation. Unfortunately, it is possible to read configuration data or realtime data from some data source that provides a large blob of characters. This could potentially cause a crash via a stack overflow. This patch prevents large sets of data from being read from an ARA backend or from an Asterisk conf file. (issue ASTERISK-20658) Reported by: wdoekes Tested by: wdoekes, mmichelson patches: * issueA20658_dont_process_overlong_config_lines.patch uploaded by wdoekes (license 5674) * issueA20658_func_realtime_limit.patch uploaded by wdoekes (license 5674) ........ Merged revisions 378375 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378376 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378377 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
5601f3be43 |
Fix AMI redirect action with two channels failing to redirect both channels.
The AMI redirect action can fail to redirect two channels that are bridged together. There is a race between the AMI thread redirecting the two channels and the bridge thread noticing that a channel is hungup from the redirects. * Made the bridge wait for both channels to be redirected before exiting. * Made the AMI redirect check that all required headers are present before proceeding with the redirection. * Made the AMI redirect require that any supplied ExtraChannel exist before proceeding. Previously the code fell back to a single channel redirect operation. (closes issue ASTERISK-18975) Reported by: Ben Klang (closes issue ASTERISK-19948) Reported by: Brent Dalgleish Patches: jira_asterisk_19948_v11.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett, Thomas Sevestre, Deepak Lohani, Kayode Review: https://reviewboard.asterisk.org/r/2243/ ........ Merged revisions 378356 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378358 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378374 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
8fb5bdce9a |
Prevent exhaustion of system resources through exploitation of event cache
Asterisk maintains an internal cache for devices in the event subsystem. The device state cache holds the state of each device known to Asterisk, such that consumers of device state information can query for the last known state for a particular device, even if it is not part of an active call. The concept of a device in Asterisk can include entities that do not have a physical representation. One way that this occurred was when anonymous calls are allowed in Asterisk. A device was automatically created and stored in the cache for each anonymous call that occurred; this was possible in the SIP and IAX2 channel drivers and through channel drivers that utilized the res_jabber/res_xmpp resource modules (Gtalk, Jingle, and Motif). These devices are never removed from the system, allowing anonymous calls to potentially exhaust a system's resources. This patch changes the event cache subsystem and device state management to no longer cache devices that are not associated with a physical entity. (issue ASTERISK-20175) Reported by: Russell Bryant, Leif Madsen, Joshua Colp Tested by: kmoore patches: event-cachability-3.diff uploaded by jcolp (license 5000) ........ Merged revisions 378303 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378320 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 378321 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378322 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
1fb06fde95 |
Resolve crashes due to large stack allocations when using TCP
Asterisk had several places where messages received over various network transports may be copied in a single stack allocation. In the case of TCP, since multiple packets in a stream may be concatenated together, this can lead to large allocations that overflow the stack. This patch modifies those portions of Asterisk using TCP to either favor heap allocations or use an upper bound to ensure that the stack will not overflow: * For SIP, the allocation now has an upper limit * For HTTP, the allocation is now a heap allocation instead of a stack allocation * For XMPP (in res_jabber), the allocation has been eliminated since it was unnecesary. Note that the HTTP portion of this issue was independently found by Brandon Edwards of Exodus Intelligence. (issue ASTERISK-20658) Reported by: wdoekes, Brandon Edwards Tested by: mmichelson, wdoekes patches: ASTERISK-20658_res_jabber.c.patch uploaded by mmichelson (license 5049) issueA20658_http_postvars_use_malloc2.patch uploaded by wdoekes (license 5674) issueA20658_limit_sip_packet_size3.patch uploaded by wdoekes (license 5674) ........ Merged revisions 378269 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378286 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 378287 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378288 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
2db81d7fc2 |
Add UUID packages now required to configure
In ASTERISK-20726 UUID was added to Asterisk. This commit is to add the dependancies to the install script git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378259 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
cdf8498784 |
Revert 378248. I changed the logic of this function unitentionally, pointed out by file.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378249 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
9c20603dfc |
Bail out early when building an ast_trans_pvt and the translator doesn't supply a 'newpvt'
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378248 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
32472eca70 |
Ensure chan_sip rejects encrypted streams without crypto info
This ensures that Asterisk rejects encrypted media streams (RTP/SAVP audio and video) that are missing cryptographic keys and ensures that the incoming SDP is consistent with RFC4568 as far as having a crypto attribute present for any SAVP streams. Review: https://reviewboard.asterisk.org/r/2204/ ........ Merged revisions 378217 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378218 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 378219 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378220 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
255d82aae3 |
Give the causes[] a struct name.
........ Merged revisions 378164 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378165 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378166 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
dec9ee2f04 |
Add test events for time limit-related hangups
This patch adds hangup-related test events in order to support testing of time-limited bridges. This aids in testing the S() and L() bridge options. (issue SWP-4713) ........ Merged revisions 378119 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378120 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 378121 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378122 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
5e2c52a546 |
Fix potential double free when unloading a module.
........ Merged revisions 378092 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378093 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 378094 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378095 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
23b94b9211 |
Make chan_local module references tied to local_pvt lifetime.
The chan_local module references were manually tied to the existence of the ;1 and ;2 channel links. * Made chan_local module references tied to the existence of the local_pvt structure as well as automatically take care of the module references. * Tweaked the wording of the local_fixup() failure warning message to make sense. Review: https://reviewboard.asterisk.org/r/2181/ ........ Merged revisions 378088 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378089 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 378090 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378091 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
0494456ae6 |
chan_local: Parse dial string consistently.
* Fix local_alloc() unexpected limitation of exten and context length from a combined length of 80 characters to a normal 80 characters each. * Made local_alloc() and local_devicestate() parse the same way. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378081 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
7ecf0aa94b |
Make libasteriskssl.so symlink use a relative path.
This was causing issues when using DESTDIR, since the path to which the link pointed is not likely to exist (and not useful to exist) on the target system. (issue ASTNOW-284) ........ Merged revisions 378073 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378074 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
87cb8e94cd |
chan_local: Misc lock and ref tweaks.
* awesome_locking() does not need to thrash the pvt lock as much. * local_setoption() does not need to check for NULL pvt on cleanup since it will never be NULL. * Made ref the pvt before locking for consistency. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378072 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
de026cf92f |
chan_agent: Remove some duplicated code.
No need to check for an agent twice. Santa does that. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378064 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
ae655031b9 |
Features: BRIDGE_FEATURES variable automixmonitor support and use proper party
BRIDGE_FEATURES did not previously support the automixmonitor feature. Now it does. In addition, the BRIDGE_FEATURES variable would not apply features to the proper party based on whether the feature option letter was in caps or in lowercase (both ways would apply it to the caller). Now uppercase applies to the caller while lowercase applies to the callee (like with the dial option) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378063 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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0f54b3ee37 |
app_queue: Revert bad ringinuse=no patch.
With the option ringinuse=no set, the patch committed for ASTERISK-16115 causes non-SIP queue members to never be called because the device state is checked after a channel is created to determine if the member is busy. These queue members always get the "Member %s is busy, cannot dial" message. Most channel drivers other than chan_sip use the default device state handling. The default device-state state is considered in use or unknown if the channel exists or not respectively. (closes issue ASTERISK-20801) Reported by: rmudgett Patches: jira_asterisk_16115_revert_r370418_v1.8.patch (license #5621) patch uploaded by rmudgett ........ Merged revisions 378036 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378037 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 378038 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378039 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
1c2f27c4a9 |
app_queue: Make update_status() not return anything.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378029 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
cb6e00b408 |
Fix skinny to recognise vmexten in general section of conf
Fixup the vmexten so if globally set in general section will be honored by chan_skinny. Also get rid of the 'global_' part of variable name to match regexten. (closes issue ASTERISK-20790) Reported by: snuffy Tested by: snuffy, myself Patches: skinny-vm.diff uploaded by snuffy (license 5024) ........ Merged revisions 378010 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378011 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
b514659d1c |
Add g722 codec support to skinny
(closes issue ASTERISK-20788) Reported by: snuffy Tested by: snuffy, myself Patches: skinny-g722.diff uploaded by snuffy (license 5024) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378006 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
b17f7cab95 |
confbridge: Fix MOH on simultaneous user entry to a new conference.
When two users entered a new conference simultaneously, one of the callers hears MOH. This happened if two unmarked users entered simultaneously and also if a waitmarked and a marked user entered simultaneously. * Created a confbridge internal MOH API to eliminate the inlined MOH handling code. Note that the conference mixing bridge needs to be locked when actually starting/stopping MOH because there is a small window between the conference join unsuspend MOH and actually joining the mixing bridge. * Created the concept of suspended MOH so it can be interrupted while conference join announcements to the user and DTMF features can operate. * Suspend any MOH until the user is about to actually join the mixing bridge of the conference. This way any pre-join file playback does not need to worry about MOH. * Made post-join actions only play deferred entry announcement files. Changing the user/conference state during that time is not protected or controlled by the state machine. (closes issue ASTERISK-20606) Reported by: Eugenia Belova Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/2232/ ........ Merged revisions 377992 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 377993 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378002 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
5cf8a1f2e5 |
Minor fixes for chan_skinny
Whitespace, change SUBSTATE_ONHOOK to correct SKINNY_ONHOOK and correct len of 2 strcmp in skinny_setdebug(). (see opticron's review on https://reviewboard.asterisk.org/r/2240/) ........ Merged revisions 377991 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378001 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
1e51b9eaa1 |
Make generate_exchange_uuid() always return the passed ast_str pointer.
I changed this code earlier to return NULL if it wasn't able to generate a UUID, whereas the earlier code would always return the ast_str that was passed in. Switch back to returning the ast_str, only set it to the empty string instead if UUID generation fails. We still do a validity check later which will catch this and blow up if necessary. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378000 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
0f62fa6e47 |
Fixed svn merge property breakage from r377986
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377994 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
758cad0984 |
Fix skinny debug tab completion
Review the syntax of the 'skinny debug' command to show more than just 'show' for options to 'skinny debug' command. (closes issue ASTERISK-20789) Reported by: snuffy Tested by: snuffy, myself Patches: skinny-debug.diff uploaded by snuffy (license 5024) ........ Merged revisions 377985 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377986 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
24b0d2365b |
Bail configure if it can't find libuuid.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377981 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
fad2637297 |
Remove compile time check HAVE_DEV_URANDOM.
The code was doing a runtime check, anyway. The compile time check isn't always valid (cross-compiling, packages). Review: https://reviewboard.asterisk.org/r/2245/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377977 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
71b1e411b0 |
Re-add taskprocessor cleanup code that was removed by the UUID merge.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377975 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
d33e46a47f |
Use the UUID API to generate and validate UUIDs for res_calendar_exchange.
Currently the res_calendar_exchange module uses its own method of generating UUIDs using ast_random(). Now that we have a UUID API we should use that instead. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377974 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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609ffb429a |
The UUID commit removed changes made in res_clialiases.c
This puts back in the changes that are designed to work around a memory leak fix in the CLI code. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377973 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
251cbe3c51 |
Fixed configure.ac to look for proper uuid.h file
Introduced in r377846, the configure script was looking for uuid.h instead of uuid/uuid.h. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377972 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
ab894d5af9 |
This change adds a SIP peer configuration feature to allow the peer's
configured codecs to take precedence on an outgoing call. This change introduces a new peer configuration property named 'ignore_requested_pref' that causes the requested codec to be ignored when determining the preferred codec for an outgoing call leg. The consequence is that Asterisk's usual efforts to prefer avoiding transcoding can be overridden on a peer-by-peer basis where appropriate. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377971 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
4f6064584d |
Ensure Min-SE is included in outbound INVITEs
Asterisk now includes Min-SE in outbound INVITEs when the value is not 90 (the default) and session timers are not disabled. This has the effect of Asterisk following RFC4028 more closely with regard to 422 responses and preventing situations in which Asterisk would be forced to temporarily accept a call to tear it down based on a Session-Expires below the locally configured Min-SE. (issue SWP-5051) Review: https://reviewboard.asterisk.org/r/2222/ Reported-by: Kinsey Moore Patch-by: Kinsey Moore ........ Merged revisions 377946 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 377947 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 377948 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377966 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
98646b1ea0 |
Incremented EXTRA_SOUNDS_VERSION in sounds/Makefile to 1.4.12 for new Extra Sounds releases
See CHANGES-* files in English extra 1.4.12 tarballs for new sound prompts added. (closes ASTERISK-20328) Reported by: Matt Jordan (closes AST-755) Reported by: John Bigelow ........ Merged revisions 377922 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 377923 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 377924 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377925 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
ff2414ce86 |
Convert Dynamic Features Buffer To Use ast_str
Currently, the buffer for the dynamic features list is set to a fixed size of 128. If the list is bigger than that, it results in the dynamic feature(s) not being recognized. This patch changes the buffer from a fixed size to a dynamic one. (closes issue ASTERISK-20680) Reported by: Clod Patry Tested by: Michael L. Young Patches: asterisk-20680-dynamic-features-v2.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2221/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377915 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
607a5d898c |
Fix a potential deadlock in chan_sip during transfers.
The issue comes from the fact that transfers may perform a redirecting update on a channel. The issue is that lock inversion between the channel and its tech_pvt occurs since the channel lock is released during the transfer process. The fix is to move when the redirecting update occurs to a place where neither the tech_pvt or the channel is locked so that the two can be locked in the proper order. (closes issue ASTERISK-20708) reported by Mark Michelson patches: ASTERISK-20708-3.patch uploaded by Mark Michelson (License #5049) Tested by: Tim Ringenbach at Asteria Solutions Group ........ Merged revisions 377910 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377911 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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9c74a60ba5 |
Add test events necessary for bridging tests to be able to properly run.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377906 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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d7c59c19a8 |
Cleanup CLI commands on exit for several files.
(issue ASTERISK-20649) Reported by: Corey Farrell Patches: unregister-cli-multiple-all.patch (license #5909) patch uploaded by Corey Farrell ........ Merged revisions 377881 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 377882 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 377883 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377884 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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4ba38ce38b |
And remove svnmerge-integrated property.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377880 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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09cd2ebe9c |
Remove automerge properties.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377878 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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b059bd4c28 |
Cleanup udptl on exit.
* Cleanup CLI commands on exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches: udptl-shutdown-1_8-10.patch (license #5909) patch uploaded by Corey Farrell udptl-shutdown-11-trunk.patch (license #5909) patch uploaded by Corey Farrell Modified ........ Merged revisions 377847 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 377848 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 377849 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377867 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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8cb156bfc1 |
Add UUID support to Asterisk.
This provides a common API for dealing with unique identifiers. The API provides methods to create, parse, copy, and stringify UUIDs. An accompanying unit test is provided that tests all operations. (closes issue ASTERISK-20726) reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2217 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377846 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |