mirror of https://github.com/asterisk/asterisk
master
20
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certified/1.8.11-cert1
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certified/11.6-cert7
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certified/13.1-cert1-rc2
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certified/13.1-cert2
certified/13.1-cert3
${ noResults }
21408 Commits (6d49117675ecbb228016788045c27b728598a1ed)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
6d49117675 |
Fix potential memory allocation failure crashes in config.c.
* Added required checks to the returned memory allocation pointers to prevent crashes. * Made ast_include_rename() create a replacement ast_variable list node if the new filename is longer than the available space. Fixes potential crash and memory leak. * Factored out ast_variable_move() from ast_variable_update() so ast_include_rename() can also use it when creating a replacement ast_variable list node. * Made the filename stuffed at the end of the struct a minimum allocated size in ast_variable_new() in case ast_include_rename() changes the stored filename. * Constify struct char pointers pointing to strings stuffed at the end of the struct for: ast_variable, cache_file_mtime, and ast_config_map. * Factored out cfmtime_new() to remove inlined code and allow some struct pointers to become const. * Removed the list lock from struct cache_file_mtime that was never used. * Added doxygen comments to several structure elements and better documented what strings are stuffed at the struct end char array. * Reworked ast_config_text_file_save() and set_fn() to handle allocation failure of the include file scratch pad object tracking blank lines. * Made ast_config_text_file_save() fn[] declared with PATH_MAX to ensure it is long enough for any filename with path. Also reduced the number of container fileset buckets from a rediculus 180,000 to 1023. JIRA AST-618 Review: https://reviewboard.asterisk.org/r/1378/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@334296 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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b178214e07 |
Remove 1.6 compatibility documentation from 1.8, as it no longer applies.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@334234 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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7db9485aa8 |
Create a local alias for ast_odbc_clear_cache.
As a function pointer, the reference has to be resolved at load time irrespective of the RTLD_LAZY flag. Creating a local alias solves this problem, because the structure is initialized with that local function pointer, while the actual function can remain lazily linked until runtime. The reason why this is important is because we lazily load function references during the module loading process, in order to obtain priority values for each module, ensuring that modules are loaded in the correct order. Previous to this change, when this module was initially loaded, the module loader would emit a symbol resolution error, because of the above requirement. Closes ASTERISK-18399 (reported by Mikael Carlsson, fix suggested by Walter Doekes, patch by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@334229 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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dac29dd12a |
Disable T.38 when we get a invite with image media port set to 0
ASTERISK-17678 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@334156 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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37835270a4 |
No DAHDI channel available for conference, user introduction disabled.
The following error will consistently occur when trying to dial into a
MeetMe conference when the server does not have DAHDI hardware installed:
app_meetme.c: No DAHDI channel available for conference, user introduction
disabled (is chan_dahdi loaded?)
While chan_dahdi is loaded correctly during compilation and install of
Asterisk/Dahdi, including associated modules, etc., a chan_dahdi.conf
configuration file in /etc/asterisk is not created by FreePBX if hardware
does not exist, causing MeetMe to be unable to open a DAHDI pseudo
channel.
* Allow chan_dahdi to create a pseudo channel when there is no
chan_dahdi.conf file to load.
(closes issue ASTERISK-17398)
Reported by: Preston Edwards
Patches:
jira_asterisk_17398_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@334012 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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15 years ago |
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9e99b1819e |
Call pickup race leaves orphaned channels or crashes.
Multiple users attempting to pickup a call that has been forked to multiple extensions either crashes or fails a masquerade with a "bad things may happen" message. This is the scenario that is causing all the grief: 1) Pickup target is selected 2) target is marked as being picked up in ast_do_pickup() 3) target is unlocked by ast_do_pickup() 4) app dial or queue gets a chance to hang up losing calls and calls ast_hangup() on target 5) SINCE A MASQUERADE HAS NOT BEEN SETUP YET BY ast_do_pickup() with ast_channel_masquerade(), ast_hangup() completes successfully and the channel is no longer in the channels container. 6) ast_do_pickup() then calls ast_channel_masquerade() to schedule the masquerade on the dead channel. 7) ast_do_pickup() then calls ast_do_masquerade() on the dead channel 8) bad things happen while doing the masquerade and in the process ast_do_masquerade() puts the dead channel back into the channels container 9) The "orphaned" channel is visible in the channels list if a crash does not happen. This patch does the following: * Made ast_hangup() set AST_FLAG_ZOMBIE on a successfully hung-up channel and not release the channel lock until that has happened. * Made __ast_channel_masquerade() not setup a masquerade if either channel has AST_FLAG_ZOMBIE set. * Fix chan_agent misuse of AST_FLAG_ZOMBIE since it would no longer work. (closes issue ASTERISK-18222) Reported by: Alec Davis Tested by: rmudgett, Alec Davis, irroot, Karsten Wemheuer (closes issue ASTERISK-18273) Reported by: Karsten Wemheuer Tested by: rmudgett, Alec Davis, irroot, Karsten Wemheuer Review: https://reviewboard.asterisk.org/r/1400/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@334009 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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c2636419b4 |
Correct an AMI protocol violation with SIPshowpeer
The response of SIPshowpeer ends with "\r\n\r\n". Since other commands are ended by using \r\n this confuses any interfacing script. (closes issue ASTERISK-17486) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@334006 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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1ae6d66b92 |
cleanups in ACF/ARJ GK replies processing
fixed long (24 sec) pause if acf/arj proccessed before ast_cond_wait called to wait this git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@333947 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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13c15462d8 |
Refresh peer address if DNS unavailable at peer creation
If Asterisk starts and no DNS is available, outbound registrations will fail indefinitely. This patch copies the address from the sip_registry struct, which will be updated, to the peer->addr when necessary. If dnsmgr is enabled, the registration fails without the patch because even though the address on the registry is updated via dnsmgr, the address is just copied on the first try. Since we use ast_sockaddr_copy, dnsmgr can't update the address that is copied to the sip_pvt or peers. Closes issue ASTERISK-18000 Review: https://reviewboard.asterisk.org/r/1335/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@333836 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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b806cbbccb |
Add some do not hold locks notes to channel.h
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@333785 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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97959f3e9a |
Fix deadlock potential of chan_mobile.c:mbl_ast_hangup().
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@333784 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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92ad64998c |
Fixed improperly formatted TestEvent AMI message in app_voicemail
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@333630 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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3b2f8bc772 |
Accidental use of variable client->status instead of client->state in from ASTERISK-18078
(issue ASTERISK-18078) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@333569 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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1025557c72 |
chan_vpb: remove unused variables (gcc4.6)
GCC 4.6 detects variables that get assined to, but never used later. Also removes some remmed-out lines that become invalid. (closes issue ASTERISK-18336) Signed-off-by: Tzafrir Cohen (License #5035) <tzafrir.cohen@xorcom.com>, git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@333507 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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7f1ca01155 |
[patch] Buddies are always auto-registered when processing the roster
Reporter said autoregister flag was ignored for registering 'buddies' which
had a subscription to us. Verified that this was the case and observed how
the patch addressed this and made sure it didn't break anything.
(closes issue ASTERISK-14233)
Reported by: Simon Arlott
Patches:
asterisk-0015229.patch (license #5756) patch uploaded by Simon Arlott
Tested by: Jonathan Rose
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@333378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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15 years ago |
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3a29ee54db |
Fixed incorrect pointer copy to structure copy in revision 333339
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@333354 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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792c3a2d56 |
Bug fixes for voicemail user emailsubject / emailbody.
This code change fixes a few issues with the voicemail user override of emailbody and emailsubject, including escaping the strings, potential memory leaks, and not overriding the voicemail defaults. Revision 325877 fixed this for ASTERISK-16795, but did not fix it for ASTERISK-16781. A subsequent check-in prevented 325877 from being applied to 10. This check-in resolves both issues, and applies the changes to 1.8, 10, and trunk. (closes issue ASTERISK-16781) Reported by: Sebastien Couture Tested by: mjordan (closes issue ASTERISK-16795) Reported by: mdeneen Tested by: mjordan Review: https://reviewboard.asterisk.org/r/1374 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@333339 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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1ab513dbac |
Fix for DESTDIR spaces patch.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@333267 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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a17bc6dabd |
Segfault when publishing device states via XMPP and not connected
When using publishing device state with res_jabber, Asterisk will attempt
to send a device state using the unconnected client using iks_send_raw
and crash. This patch checks the validity of the connection before
attempting to send the device state.
(closes issue ASTERISK-18078)
Reported by: Michael L. Young
Patches:
res_jabber-segfault-pubsub-not-connected2.patch (license #5026) patch uploaded by Michael L. Young
Tested by: Jonathan Rose
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@333265 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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15 years ago |
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cb8c7da373 |
Fix installation into directories containing spaces.
This also vastly simplifies the logic in sounds/Makefile (Closes issue ASTERISK-18290) Reported by: Paul Belanger Review: https://reviewboard.asterisk.org/r/1379/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@333201 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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185f890c89 |
Memory Leak in app_queue
The patch that was committed in the 1.6.x versions of Asterisk for
ASTERISK-15862 actually fixed two issues. One was not applicable to 1.8
but the other is. queue_leak.patch fixes the portion applicable to 1.8.
(closes issue ASTERISK-18265)
Reported by: Fred Schroeder
Patches:
queue_leak.patch (license #5049) patch uploaded by mmichelson
Tested by: Thomas Arimont
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@333010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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15 years ago |
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c9325708c8 |
default 'sipstorecause' to no
We've decided to disable this feature by default in future 1.8 versions. This would be an unexpected behavior change for anyone depending on that SIP_CAUSE update in their dialplan. Please refer to the asterisk-dev mailing list more information: http://lists.digium.com/pipermail/asterisk-dev/2011-August/050626.html (issue AST-580) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@333009 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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b1f11e0df4 |
Revert previous commit. Not ready yet.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332945 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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c0ce03d77f |
Signed
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332943 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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b06b8acd20 |
Minor code optimizations.
* Simplify ast_category_browse() logic for easier understanding. * Remove dead code in ast_variable_delete() and simplify some of its logic. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332939 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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ad133138fa |
Revert previous commit
It seems google is still making changes to the protocol. (issue ASTERISK-18301) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332876 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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def9e8fe45 |
Reference leaks in app_queue.
* Fixed load_realtime_queue() leaking a queue reference when it overwrites q when processing a realtime queue. (issue ASTERISK-18265) * Make join_queue() unreference the queue returned by load_realtime_queue() when it is done with the pointer. The load_realtime_queue() returns a reference to the just loaded realtime queue. * Fixed queues container reference leak in queues_data_provider_get(). * queue_unref() should not return q that was just unreferenced. * Made logic in __queues_show() and queues_data_provider_get() when calling load_realtime_queue() easier to understand. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332874 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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56549c96ab |
Review: https://reviewboard.asterisk.org/r/1364/
This update adds a new AMI event, TestEvent, which is enabled when the TEST_FRAMEWORK compiler flag is defined. It also adds initial usage of this event to app_voicemail. The TestEvent AMI event is used extensively by the voicemail tests in the Asterisk Test Suite. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332817 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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e55b37bb5a |
Memory leaks in realtime_multi_xxx() when database access returns error.
* Fix realtime_multi_pgsql() configuration memory leak when the database access returns an error. * Fix realtime_multi_odbc() configuration category use after free when the database access returns an error. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332816 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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08b76290a2 |
Memory leak reading realtime database variable list.
Calling ast_load_realtime() can leak the last list node if the read list
only contains empty variable value items.
* Fixed list filter loop in ast_load_realtime() to delete the list node
immediately instead of the next time through the loop. The next time
through the loop may not happen if the node to delete is the last in the
list.
(issue ASTERISK-18277)
(issue ASTERISK-18265)
Patches:
jira_asterisk_18265_v1.8_config.patch (license #5621) patch uploaded by rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332759 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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15 years ago |
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a54ace8fc7 |
Fix outgoing calls in chan_gtalk
(closes issue ASTERISK-18301) Reported by: az1324 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332699 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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5dd7370c32 |
Fix possible error on stringification of IPv4-mapped addrs
The FreeBSD netsock2 test has been failing for a while. We were pasing sa->len to getnameinfo instead of sa_tmp->len. ASTERISK-18289 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332559 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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6eceaa5efb |
CRC4 in "dahdi show status" gives wrong impression to T1 users
Change CRC4 to CRC in the output of "dahdi show status" so that it can apply in more situations without confusing users, especially since T1 lines use CRC6 instead of CRC4. (closes issue AST-471) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332503 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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3893db6f57 |
Move BETTER_BACKTRACES out of development mode, as it's useful when DEBUG_THREADS is enabled.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332446 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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582aacb3d8 |
Re-add support for spaces in pathnames, including now spaces in DESTDIR.
This was initially added to 1.8 prior to release, primarily to support the standard paths on Mac OS X, but was partially reverted recently in Subversion, due to the lack of support for spaces in DESTDIR. This commit restores support for the standard paths on Mac OS X, and also includes support for spaces in DESTDIR. (closes issue ASTERISK-18290) Reported by: pabelanger Review: https://reviewboard.asterisk.org/r/1326/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332355 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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a9db588064 |
Don't read from a disarmed or invalid timerfd
Numerous isues have been reported for deadlocks that are caused by a blocking read in res_timing_timerfd on a file descriptor that will never be written to. This patch adds some checks to make sure that the timerfd is both valid and armed before calling read(). Should fix: ASTERISK-18142, ASTERISK-18197, ASTERISK-18166, AST-486 AST-495, AST-507 and possibly others. Review: https://reviewboard.asterisk.org/r/1361/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332320 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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9328590ddb |
Outgoing BRI calls fail when using Asterisk 1.8 with HA8, HB8, and B410P cards.
France Telecom brings layer 2 and layer 1 down on BRI lines when the line is idle. When layer 1 goes down Asterisk cannot make outgoing calls and the HA8 and HB8 cards also get IRQ misses. The inability to make outgoing calls is because the line is in red alarm and Asterisk will not make calls over a line it considers unavailable. The IRQ misses for the HA8 and HB8 card are because the hardware is switching clock sources from the line which just brought layer 1 down to internal timing. There is a DAHDI option for the B410P card to not tell Asterisk that layer 1 went down so Asterisk will allow outgoing calls: "modprobe wcb4xxp teignored=1". There is a similar DAHDI option for the HA8 and HB8 cards: "modprobe wctdm24xxp bri_teignored=1". Unfortunately that will not clear up the IRQ misses when the telco brings layer 1 down. * Add layer 2 persistence option to customize the layer 2 behavior on BRI PTMP lines. The new option has three settings: 1) Use libpri default layer 2 setting. 2) Keep layer 2 up. Bring layer 2 back up when the peer brings it down. 3) Leave layer 2 down when the peer brings it down. Layer 2 will be brought up as needed for outgoing calls. JIRA AST-598 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332264 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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8345854458 |
print a warning instructing the user to disable storesipcause if we process 100
or more scheduler entries at a time AST-580 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332234 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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1fc32a652e |
Flag test modules as 'core'
Review: https://reviewboard.asterisk.org/r/1369/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332176 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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a10e0544a5 |
ASTERISK-18067 ASTERISK-15479 - White Space affects mailbox value, multiple MWI subs
Before, having multiple subscriptions to mailboxes on a sip peer set via the mailbox setting in sip.conf would only result in updates being sent on whichever mailbox triggered the mwi event. Now all of them get counted regardless. Also fixes a bug involving parsing of the mailbox option in sip.conf so that trailing and leading spaces before/after commas are trimmed. (closes issue ASTERISK-18067) Reported by: aragon (closes issue ASTERISK-15479) Reported by: Ben Winslow Patches: chan_sip.c-mwi_multi_mailbox_fix-1.6.2.13.diff (License #5288) patch uploaded by Ben Winslow git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332118 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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422e191e03 |
Fix multiple parking issues.
JIRA ASTERISK-17183 Multi-parkinglot directs calls to wrong parkinglot. JIRA ASTERISK-17870 Cannot retrieve parked calls. JIRA ASTERISK-17430 ParkedCall() with no extension should pickup first available call and does not. JIRA AST-576 Issues with parking lots * Removed searching for parking lots by extension. Parking lots can only be found by the parking lot name since parking lot access extensions and spaces are not guaranteed to be unique. * Added parking_lot_name option to the Park and ParkedCall applications. Updated documentation for Park and ParkedCall applications. * Add parkext_exclusive configuration option to make parking entry extensions specify which parking lot they access. (closes issue ASTERISK-17183) Reported by: David Cabrejos Tested by: rmudgett, David Cabrejos (closes issue ASTERISK-17870) Reported by: Remi Quezada (closes issue ASTERISK-17430) Reported by: Philippe Lindheimer JIRA ASTERISK-17452 Parking_offset not used JIRA AST-624 'next' setting for findslot does nothing * Reimplemented since findslot feature option broken by -r114655. (closes issue ASTERISK-17452) Reported by: David Woolley Tested by: rmudgett JIRA ASTERISK-15792 Dialplan continues execution after transfer to park. This happens for DTMF attended transfer, DTMF blind transfer, and DTMF one-touch-parking if the party initiating these features also initiated the call. * Fixed the return code from the affected builtin features when parking a call. (closes issue ASTERISK-15792) Reported by: Mat Murdock Tested by: rmudgett, twilson JIRA AST-607 The courtesytone is not playing to the expected call when picking up a parked call. This is mostly a documentation problem. However, the option is not reset to the default when features.conf is reloaded. * Updated features.conf.sample documentation for courtesytone and parkedplay options. * Reset the parkedplay option to default when features.conf is reloaded. JIRA AST-615 AMI Park action followed by features reload results in orphaned channels in parking lot. * Reloading features.conf will not touch parking lots that have calls still parked in them. Reload again at a later time. Misc additional fixes: * Added unit test for parking lot dialplan usage checking. * Made update connected line when a parked call is retrieved from a parking lot. * Made retrieved parked call stop ringing or MOH depending upon how the call was waiting in the parking lot. * Made CLI "features show" indicate if the parking lot is enabled for use. * Added PARKINGDYNEXTEN channel variable to allow dynamic parking lots to specify the parking lot access extension. * Made AMI ParkedCalls action ParkedCall events have a Parkinglot header. * Made AMI ParkedCalls action ParkedCallsComplete event have a Total header. * Fixed potential deadlock from AMI Park action holding channel locks while calling masq_park_call(). * Fixed several places where ast_strdupa() were used inside of loops. (Mostly fixed by refactoring the loop body into its own function.) * Fixed copy_parkinglot() copying too much from the source parking lot. Extracted the parking lot configuration settings into struct parkinglot_cfg. * Refactored courtesytone playing code to put the channel not playing the tone in autoservice. * Fix when pbx-parkingfailed is played that the other channel is put in autoservice if it exists. * Fixed parkinglot reference leak in parked_call_exec() error paths. * Fixed parkinglot_unref() use of parkinglot after it was unreffed. * Made destroy the struct ast_parkinglot parkings lock when done. * Refactored the features.conf parking lot configuration code to eliminate redundancy. * Fixed feature reload to better protect parking lots. * Fixed parking lot container reference leak in handle_parkedcalls(). * Fixed the total count in handle_parkedcalls(). Review: https://reviewboard.asterisk.org/r/1358/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332100 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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3d709a2b55 |
use DEFAULT_STORE_SIP_CAUSE to set the default value for the 'storesipcause' option
AST-580 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332026 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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f01a484b48 |
Added the 'storesipcause' option to sip.conf to allow the user to disable the
setting of HASH(SIP_CAUSE,<chan name>) on the channel. Having chan_sip set HASH(SIP_CAUSE,<chan name>) on the channel carries a significant performance penalty because of the usage of the MASTER_CHANNEL() dialplan function. AST-580 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332021 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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acc2d27a47 |
Fix some minor chan_dahdi config load issues.
* Address chan_dahdi.conf dahdichan option todo item about needing line number. * Make ignore_failed_channels option also apply to dahdichan option. * Don't attempt to create a default pseudo channel if the chan_dahdi.conf channel/channels option is not allowed. * Add a similar check for dahdichan in normal chan_dahdi.conf sections as is done in users.conf. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@331955 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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0281f945c0 |
Fix noisy message when briding channels
(closes issue ASTERISK-18270) Reported by: Federico Alves git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@331886 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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53bc3bdbe6 |
Fixes locking inversion issues present in the handling of the sip REFER method.
(closes issue ASTERISK-18082) Reported by: James Van Vleet git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@331867 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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b92d5952e1 |
Unlock the channel before calling update_queue.
Holding the channel lock when calling update_queue which attempts to lock the queue lock can cause a deadlock. This deadlock involves the following chain: 1. hold chan lock -> wait queue lock 2. hold queue lock -> wait agent list lock 3. hold agent list lock -> wait chan list lock 4. hold chan list lock -> wait chan lock git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@331774 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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450ba7e060 |
Suppress warning message when using DAHDITransfer or DAHDIHangup.
* The fake event should only be processed by the channel that currently owns the private and not the associated call waiting or 3-way channel. JIRA AST-620 JIRA SWP-3616 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@331771 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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36c8e8ca15 |
AMI actions DAHDIHangup and DAHDITransfer have no effect.
The AMI actions DAHDIHangup and DAHDITransfer have no effect on a DAHDI channel. These two AMI actions are highly specialized to analog channels and appear to make the channel behave like a jack port for headsets. * Made the faked DAHDI event get processed before a normal media stream read in dahdi_read() instead of trying to trigger an exception read by setting the AST_FLAG_EXCEPTION flag. Apparently a change was made long ago that changed how AST_FLAG_EXCEPTION is processed in the core. Unfortunately, the faked DAHDI events no longer worked when that happened. * Updated the DAHDI AMI action documentation for the following actions: DAHDITransfer, DAHDIHangup, DAHDIDialOffhook, DAHDIDNDon, DAHDIDNDoff, DAHDIShowChannels, and DAHDIRestart. * Made use sscanf() instead of atoi() for better error checking of the DAHDIChannel header string. JIRA AST-620 JIRA SWP-3616 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@331714 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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be2099a976 |
Fix netsock2 multiple zero-expansion test
Remove erroneous single bracket. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@331658 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |