Kinsey Moore: User Summary
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kmoore
Kinsey Moore
Last 10 Builds
| Build | Completed | Code Changes | Tests |
|---|---|---|---|
| TESTING › ASTERISKTRUNK › #277 | 1 day ago | No tests found | |
| ASTTEAM › ASTERISKTRUNKDIGIUMPHONES › #333 | 1 day ago |
Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE)
This is the starting point for the Asterisk 11: Who Hung Up work and provides a framework which will allow channel drivers to report the types of hangup cause information available in SIP_CAUSE without incurring the overhead of the MASTER_CHANNEL dialplan function. The initial implementation only includes cause generation for chan_sip and does not include cause code translation utilities. This change deprecates SIP_CAUSE and replaces its method of reporting cause codes with the new framework. This change also deprecates the 'storesipcause' option in sip.conf. Review: https://reviewboard.asterisk.org/r/1822/ (Closes issue SWP-4221) |
No tests found |
| TESTING › ASTERISKTRUNK › #270 | 4 days ago |
Resolve FORWARD_NULL static analysis warnings
This resolves core findings from ASTERISK-19650 numbers 0-2, 6, 7, 9-11, 14-20, 22-24, 28, 30-32, 34-36, 42-56, 82-84, 87, 89-90, 93-102, 104, 105, 109-111, and 115. Finding numbers 26, 33, and 29 were already resolved. Those skipped were either extended/deprecated or in areas of code that shouldn't be disturbed. (Closes issue ASTERISK-19650) ........ Merged revisions 366167 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366168 from http://svn.asterisk.org/svn/asterisk/branches/10 |
1 of 232 failed |
| ASTTEAM › ASTERISKTRUNKDIGIUMPHONES › #326 | 5 days ago |
Resolve FORWARD_NULL static analysis warnings
This resolves core findings from ASTERISK-19650 numbers 0-2, 6, 7, 9-11, 14-20, 22-24, 28, 30-32, 34-36, 42-56, 82-84, 87, 89-90, 93-102, 104, 105, 109-111, and 115. Finding numbers 26, 33, and 29 were already resolved. Those skipped were either extended/deprecated or in areas of code that shouldn't be disturbed. (Closes issue ASTERISK-19650) ........ Merged revisions 366167 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366168 from http://svn.asterisk.org/svn/asterisk/branches/10 |
232 passed |
| TESTING › ASTERISK10BRANCH › #201 | 5 days ago |
Resolve FORWARD_NULL static analysis warnings
This resolves core findings from ASTERISK-19650 numbers 0-2, 6, 7, 9-11, 14-20, 22-24, 28, 30-32, 34-36, 42-56, 82-84, 87, 89-90, 93-102, 104, 105, 109-111, and 115. Finding numbers 26, 33, and 29 were already resolved. Those skipped were either extended/deprecated or in areas of code that shouldn't be disturbed. (Closes issue ASTERISK-19650) ........ Merged revisions 366167 from http://svn.asterisk.org/svn/asterisk/branches/1.8 |
210 passed |
| TESTING › ASTERISK18BRANCH › #198 | 5 days ago |
Resolve FORWARD_NULL static analysis warnings
This resolves core findings from ASTERISK-19650 numbers 0-2, 6, 7, 9-11, 14-20, 22-24, 28, 30-32, 34-36, 42-56, 82-84, 87, 89-90, 93-102, 104, 105, 109-111, and 115. Finding numbers 26, 33, and 29 were already resolved. Those skipped were either extended/deprecated or in areas of code that shouldn't be disturbed. (Closes issue ASTERISK-19650) |
178 passed |
| TESTING › ASTERISKTRUNK › #261 | 6 days ago |
Fix many issues from the NULL_RETURNS Coverity report
Most of the changes here are trivial NULL checks. There are a couple optimizations to remove the need to check for NULL and outboundproxy parsing in chan_sip.c was rewritten to avoid use of strtok. Additionally, a bug was found and fixed with the parsing of outboundproxy when "outboundproxy=," was set. (Closes issue ASTERISK-19654) ........ Merged revisions 365398 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365399 from http://svn.asterisk.org/svn/asterisk/branches/10 |
232 passed |
| TESTING › ASTERISK10BRANCH › #194 | 6 days ago |
Fix many issues from the NULL_RETURNS Coverity report
Most of the changes here are trivial NULL checks. There are a couple optimizations to remove the need to check for NULL and outboundproxy parsing in chan_sip.c was rewritten to avoid use of strtok. Additionally, a bug was found and fixed with the parsing of outboundproxy when "outboundproxy=," was set. (Closes issue ASTERISK-19654) ........ Merged revisions 365398 from http://svn.asterisk.org/svn/asterisk/branches/1.8 |
210 passed |
| TESTING › ASTERISK18BRANCH › #190 | 6 days ago |
Fix many issues from the NULL_RETURNS Coverity report
Most of the changes here are trivial NULL checks. There are a couple optimizations to remove the need to check for NULL and outboundproxy parsing in chan_sip.c was rewritten to avoid use of strtok. Additionally, a bug was found and fixed with the parsing of outboundproxy when "outboundproxy=," was set. (Closes issue ASTERISK-19654) |
178 passed |
| TESTING › ASTERISKTRUNK › #245 | 2 weeks ago |
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 |
232 passed |
Last 10 Broken
| Build | Completed | Code Changes | Tests |
|---|---|---|---|
| TESTING › ASTERISKTRUNK › #270 | 4 days ago |
Resolve FORWARD_NULL static analysis warnings
This resolves core findings from ASTERISK-19650 numbers 0-2, 6, 7, 9-11, 14-20, 22-24, 28, 30-32, 34-36, 42-56, 82-84, 87, 89-90, 93-102, 104, 105, 109-111, and 115. Finding numbers 26, 33, and 29 were already resolved. Those skipped were either extended/deprecated or in areas of code that shouldn't be disturbed. (Closes issue ASTERISK-19650) ........ Merged revisions 366167 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366168 from http://svn.asterisk.org/svn/asterisk/branches/10 |
1 of 232 failed |
| TESTING › ASTERISKTRUNK › #230 | 2 weeks ago |
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. (related to issue ASTERISK-19579) ........ 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 |
1 of 230 failed |
| TESTING › ASTERISK18BRANCH › #104 | 4 weeks ago |
Simplify build system architecture optimization
This change to the build system rips out any usage of PROC along with architecture-specific optimizations in favor of using -march=native where it is supported. This fixes broken builds on 64bit Intel systems and results in better optimized code on systems running GCC 4.2+. Review: https://reviewboard.asterisk.org/r/1852/ (closes issue ASTERISK-19462) |
1 of 176 failed |
| TESTING › ASTERISKTRUNK › #28 | 2 months ago |
Prevent outbound SIP NOTIFY packets from displaying a port of 0
In the change from 1.6.2 to 1.8, ast_sockaddr was introduced which changed the behavior of ast_find_ourip such that port number was wiped out. This caused the port in internip (which is used for Contact and Call-ID on NOTIFYs) to be 0. This change causes ast_find_ourip to be port-preserving again. (closes issue ASTERISK-19430) ........ Merged revisions 357665 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 357667 from http://svn.asterisk.org/svn/asterisk/branches/10 |
79 passed |
| TESTING › ASTERISK18BRANCH › #19 | 2 months ago |
Prevent outbound SIP NOTIFY packets from displaying a port of 0
In the change from 1.6.2 to 1.8, ast_sockaddr was introduced which changed the behavior of ast_find_ourip such that port number was wiped out. This caused the port in internip (which is used for Contact and Call-ID on NOTIFYs) to be 0. This change causes ast_find_ourip to be port-preserving again. (closes issue ASTERISK-19430) |
1 of 227 failed |
| PRI › 14 › #53 | 3 months ago |
Make PRI_DEBUG_Q921_RAW work independantly of PRI_DEBUG_Q921_DUMP
Ensure that the DUMP and RAW flags work independently in q921_dump(). Closes mantis issue: 18528 Patch-by: wimpy |
No tests found |
| ASTTRUNK › LUCID › #1331 | 4 months ago |
Prevent SLA settings from getting wiped out on reload
If SLA was reloaded without the config file being changed, current settings got wiped out before the SLA reload code decided it wasn't going to reload the file since nothing was changed. Moving the settings reset later in the reload process fixes this. (closes issue AST-744) ........ Merged revisions 350023 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350024 from http://svn.asterisk.org/svn/asterisk/branches/10 |
1 of 198 failed |
| ASTTRUNK › SNOWLEOPARD › #428 | 4 months ago |
Fix missing doc tags found while fixing ASTERISK-18689
Add missing <variable></variable> tags in app_dial documentation. ........ Merged revisions 348992 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348993 from http://svn.asterisk.org/svn/asterisk/branches/10 |
No tests found |
| ASTTRUNK › SNOWLEOPARD › #424 | 4 months ago |
Fix res_jabber resource leaks
This should fix almost all resource leaks in res_jabber that involve ASTOBJ_CONTAINER_FIND and resolves an ambiguous situation where ast_aji_get_client would sometimes bump an object's refcount and sometimes not. Review: https://reviewboard.asterisk.org/r/1553 ........ Merged revisions 346086 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346087 from http://svn.asterisk.org/svn/asterisk/branches/10 Fix chan_jingle/gtalk load regression introduced in r346087
Add missing symbol exports for ast_aji_client_destroy and ast_aji_buddy_destroy for usage outside res_jabber. Testing of these changes focused on res_jabber itself, so this problem was missed. Reported-by: Michael Spiceland ........ Merged revisions 346951 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346952 from http://svn.asterisk.org/svn/asterisk/branches/10 |
No tests found |
| AST10 › LUCID › #345 | 6 months ago |
Ensure that a null vmexten does not cause a segfault
When sip_send_mwi_to_peer was modified recently to avoid deadlocks, vmexten was not expected to be null. This change handles that situation to avoid a segfault. ........ Merged revisions 345063 from http://svn.asterisk.org/svn/asterisk/branches/1.8 |
1 of 179 failed |
Last 10 Fixed
| Build | Completed | Code Changes | Tests |
|---|---|---|---|
| TESTING › ASTERISKTRUNK › #261 | 6 days ago |
Fix many issues from the NULL_RETURNS Coverity report
Most of the changes here are trivial NULL checks. There are a couple optimizations to remove the need to check for NULL and outboundproxy parsing in chan_sip.c was rewritten to avoid use of strtok. Additionally, a bug was found and fixed with the parsing of outboundproxy when "outboundproxy=," was set. (Closes issue ASTERISK-19654) ........ Merged revisions 365398 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365399 from http://svn.asterisk.org/svn/asterisk/branches/10 |
232 passed |
| TESTING › ASTERISK10BRANCH › #172 | 2 weeks ago |
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. (related to issue ASTERISK-19579) ........ Merged revisions 364258 from http://svn.asterisk.org/svn/asterisk/branches/1.8 |
208 passed |
| TESTING › ASTERISKTRUNK › #126 | 1 month ago |
Stop sending out RTCP if RTP is inactive
This change prevents Asterisk from sending RTCP receiver reports during a remote bridge since it is no longer receiving media and should not be reporting anything. (related to ASTERISK-19366) ........ Merged revisions 360987 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360993 from http://svn.asterisk.org/svn/asterisk/branches/10 |
455 passed |
| TESTING › ASTERISK18BRANCH › #86 | 1 month ago |
Stop sending out RTCP if RTP is inactive
This change prevents Asterisk from sending RTCP receiver reports during a remote bridge since it is no longer receiving media and should not be reporting anything. (related to ASTERISK-19366) |
349 passed |
| TESTING › ASTERISK10BRANCH › #24 | 2 months ago |
Fix case-sensitivity for device-specific event subscriptions and CCSS
This change fixes case-sensitivity for device-specific subscriptions such that the technology identifier is case-insensitive while the remainder of the device string is still case-sensitive. This should also preserve the original case of the device string as passed in to the event system. CCSS is the only feature affected as it is the only consumer of device-specific event subscriptions. The second part of this patch addresses similar case-sensitivity issues within CCSS itself that prevented it from functioning correctly after the fix to the events system. This adds a unit test to verify that the event system works as expected. (closes issue ASTERISK-19422) Review: https://reviewboard.asterisk.org/r/1780/ ........ Merged revisions 357940 from http://svn.asterisk.org/svn/asterisk/branches/1.8 |
204 passed |
| ASTTRUNK › LUCID › #1315 | 4 months ago |
Update autosupport script and man page
Added information collection from the output of the utilities: top, free, uptime, ifconfig Added information collection from the output of the Asterisk command 'dahdi show status' Added option / flag '-n, --non-interactive' Updated man page to reflect new option / flag '-n, --non-interactive' Patch-by: John Bigelow (itzanger) (closes issue AST-749) ........ Merged revisions 349504 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 349505 from http://svn.asterisk.org/svn/asterisk/branches/10 |
194 passed |
| AST10 › LUCID › #418 | 4 months ago |
Fix missing doc tags found while fixing ASTERISK-18689
Add missing <variable></variable> tags in app_dial documentation. ........ Merged revisions 348992 from http://svn.asterisk.org/svn/asterisk/branches/1.8 |
189 passed |
| AST10 › LUCID › #380 | 5 months ago |
Fix chan_jingle/gtalk load regression introduced in r346087
Add missing symbol exports for ast_aji_client_destroy and ast_aji_buddy_destroy for usage outside res_jabber. Testing of these changes focused on res_jabber itself, so this problem was missed. Reported-by: Michael Spiceland ........ Merged revisions 346951 from http://svn.asterisk.org/svn/asterisk/branches/1.8 |
181 passed |
| ASTTRUNK › LUCID › #1243 | 5 months ago |
Fix res_jabber resource leaks
This should fix almost all resource leaks in res_jabber that involve ASTOBJ_CONTAINER_FIND and resolves an ambiguous situation where ast_aji_get_client would sometimes bump an object's refcount and sometimes not. Review: https://reviewboard.asterisk.org/r/1553 ........ Merged revisions 346086 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346087 from http://svn.asterisk.org/svn/asterisk/branches/10 |
178 passed |
| AST10 › LUCID › #368 | 5 months ago |
Fix res_jabber resource leaks
This should fix almost all resource leaks in res_jabber that involve ASTOBJ_CONTAINER_FIND and resolves an ambiguous situation where ast_aji_get_client would sometimes bump an object's refcount and sometimes not. Review: https://reviewboard.asterisk.org/r/1553 ........ Merged revisions 346086 from http://svn.asterisk.org/svn/asterisk/branches/1.8 |
178 passed |
This is the starting point for the Asterisk 11: Who Hung Up work and provides
a framework which will allow channel drivers to report the types of hangup
cause information available in SIP_CAUSE without incurring the overhead of the
MASTER_CHANNEL dialplan function. The initial implementation only includes
cause generation for chan_sip and does not include cause code translation
utilities.
This change deprecates SIP_CAUSE and replaces its method of reporting cause
codes with the new framework. This change also deprecates the 'storesipcause'
option in sip.conf.
Review: https://reviewboard.asterisk.org/r/1822/
(Closes issue SWP-4221)