mirror of https://github.com/asterisk/asterisk
master
20
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${ noResults }
21454 Commits (7361deae1b0620095bc19e8dc3400b741b52883e)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
7361deae1b |
Check if a channel was created before using the pointer in sig_ss7_new_ast_channel().
Fixes the crash in ASTERISK-17955 gdb-11918.txt backtrace.
* Added some missing libss7 access lock protection.
* Prevent cancelling the ss7_linkset() thread at inoportune times just
like the pri_dchannel() thread.
(issue ASTERISK-17955)
Reported by: Ian M Sherman
Patches:
jira_asterisk_17955_v1.8.patch (license #5621) patch uploaded by rmudgett
(attached to related ASTERISK-17966)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@337007 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
14 years ago |
|
|
b48984e2fb |
Fix deadlock from not releasing SS7 linkset lock.
sig_ss7_hangup() failed to release the SS7 linkset lock if the call had
the alreadyhungup flag set.
* Made unlock the SS7 linkset lock in sig_ss7_hangup() if the
alreadyhungup flag is set.
* Made ss7_start_call() not hold any locks while creating the channel for
an incoming call to prevent deadlock.
* Made ss7_grab() a void function, since it could never fail, to simplify
calling code.
* Made obtain the channel lock to do softhangup in some places.
Patches:
jira_ast_668_v1.8.patch (license #5621) patch uploaded by rmudgett
JIRA AST-668
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@336977 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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14 years ago |
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df4d47dff4 |
Fix crashes in ast_rtcp_write().
This patch addresses crashes related to RTCP handling. The backtraces just show a crash in ast_rtcp_write() where it appears that the RTP instance is no longer valid. There is a race condition with scheduled RTCP transmissions and the destruction of the RTP instance. This patch utilizes the fact that ast_rtp_instance is a reference counted object and ensures that it will not get destroyed while a reference is still around due to scheduled RTCP transmissions. RTCP transmissions are scheduled and executed from the chan_sip scheduler context. This scheduler context is processed in the SIP monitor thread. The destruction of an RTP instance occurs when the associated sip_pvt gets destroyed (which happens when the sip_pvt reference count reaches 0). However, the SIP monitor thread is not the only thread that can cause a sip_pvt to get destroyed. The sip_hangup function, executed from a channel thread, also decrements the reference count on a sip_pvt and could cause it to get destroyed. While this is being changed anyway, the patch also removes calling ast_sched_del() from within the RTCP scheduler callback. It's not helpful. Simply returning 0 prevents the callback from being rescheduled. (closes issue ASTERISK-18570) Related issues that look like they are the same problem: (issue ASTERISK-17560) (issue ASTERISK-15406) (issue ASTERISK-15257) (issue ASTERISK-13334) (issue ASTERISK-9977) (issue ASTERISK-9716) Review: https://reviewboard.asterisk.org/r/1444/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@336877 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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0628cce193 |
Don't interfere with T.38 reinvites
This is an update to the fix for ASTERISK-18340 and ASTERISK-17725 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@336791 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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02795f190e |
Various changes to allow 1.8 to compile on Mac OS X Lion (10.7)
* Makefile workaround for 10.6 extended to work on 10.7 and later. * Now uses the 'weak' symbol for Lion systems, which no longer support 'weak_import' Closes ASTERISK-17612. Closes ASTERISK-18213. Tested by: tilghman, oej. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@336733 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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32c717b97c |
Document applications that play audio and do not answer unanswered calls.
This patch is part of an effort to document early media and its usage. If you are interested in contributing to this documentation effort, there are probably other applications worth documenting as well as an Asterisk wiki article at https://wiki.asterisk.org/wiki/display/AST/Early+Media+and+the+Progress+Application git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@336716 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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07a3a611a9 |
Made Dial d and H options no longer immediately auto-answer the calling leg.
The Dial d and H options break DTMF attended transfer atxferdropcall option. 1) Party A calls party B. 2) Party B does a DTMF attended transfer to Party C. If the dialplan uses the Dial d or H options to call Party C then the Dial application answers the call immediately before initiating the call leg to Party C. The premature answer causes the transfer code to not invoke the atxferdropcall=no behavior for a blonde transfer since Party C has "answered". The transfer code thinks that Party B has "consulted" with Party C when Party B hangs up and completes the transfer to Party A. Party A now hears ringback until Party C actually answers. ASTERISK-13294 Dial d option. ASTERISK-11067 Dial H option to disconnect before answer. The referenced issues made Dial answer with the d and H options because many SIP and ISDN phones cannot send DTMF before the call is connected. * Made require the dialplan to control when or if the call needs to be answered to use the Dial application d and H options. (The call is no longer surprise answered when using the Dial d or H options.) Review: https://reviewboard.asterisk.org/r/1381/ JIRA AST-623 JIRA AST-666 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@336658 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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a7b1c2eafb |
Remove weird mergeinfo props that make merges annoying sometimes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@336591 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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83e8f9b91c |
Update get_ilbc_source.sh script to work again.
Recently iLBC support in Asterisk has changed after the acquisition of GIPS by Google. More information about how this may affect you is available in a blog post at: http://blogs.asterisk.org/2011/09/19/ilbc-support-in-asterisk-after-googles-acquisition-of-gips/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@336572 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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9eb7ccef76 |
Rework sig_pri_hangup() to be simpler and clearer.
JIRA AST-675 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@336569 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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535817fe71 |
Add diversion header to a 302 redirect response if we have diversion data
(closes issue ASTERISK-18143) patch by oej git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@336501 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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aa50191685 |
A long time ago in a galaxy far far away a IPv6 update was made,
chan_h323 was not updated causeing all to flee to chan_ooh323. the brave Jedi [asterisk developers] pondered this miscarrige of justice and restored order to the force for the sake of closing out 2 old issues. (closes issue ASTERISK-17278) (closes issue ASTERISK-17500) Reported by: dread, sybasesql Tested by: irroot Reviewed by: IRC (russellb, kpfleming) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@336499 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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02a28f4afe |
Make sure manager_debug option is reset at reload
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@336440 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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309e3fe7fa |
Revert accidental change that fixes OS/X Lion support
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@336379 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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7a2e489631 |
Add missing unlock at MWI message sending time
(closes issue ASTERISK-18573) Patches: sip_mwi_lock.patch (license #5041) by Gregory Hinton Nietsky Thanks to irrot for the reminder, to Gregory for the patch! git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@336378 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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928de8c08a |
Whitespace fix
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@336314 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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19992c7246 |
Add missing frame types to func_frame_trace
Also casts control frames to the proper enum so that the compile will catch new additions. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@336312 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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21714a05b6 |
Fix bad RTP media bridges in directmedia calls on peers separated by multiple Asterisk nodes.
In a situation involving devices on separate Asterisk trunks, the remote RTP bridge would break when starting a call with directmedia. This patch queues a new type of control frame so that our RTP bridge loop can properly detect when these situations occur and check to see if peers need to be updated in order to send their media to the proper location. (Closes issue ASTERISK-18340) Reported by: Thomas Arimont (Closes issue ASTERISK-17725) Reported by: kwk Tested by: twilson, jrose git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@336294 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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ea573b112f |
Make a note that inotify won't work with an NFS mounted spooler directory.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@336234 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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bbc088b9fc |
The round robin routing routine in chan_misdn.c is broken.
it rotates between ports but never checks the channels in the ports.
i have extensivly tested it and verified it works on 1 upto 4 ports.
before the patch only 1 out of each port was used now all are used as
expected.
(closes issue ASTERISK-18413)
Reported by: irroot
Tested by: irroot
Reviewed by: irroot
Review: https://reviewboard.asterisk.org/r/1410/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@336166 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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14 years ago |
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f94fa3dba3 |
Locking order in app_queue.c causes deadlocks.
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14 years ago |
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46e2968917 |
lock the channel before calling ast_bridged_channel() to prevent a seg fault.
AMI agents list called on shutdown causes a segfault, introducing proper locking will prevent this. (closes issue ASTERISK-18092) Reported by: agustina Patches: chan_agent.patch (License #5041) patch uploaded by irroot git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335978 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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9910558356 |
Remove unnecessary libpri dependency checks in the configure script.
Using the --with-pri option with the configure script generated an error about not having PRI_L2_PERSISTENCE if you did not have the absolute latest libpri SVN checkout installed. The AST_EXT_LIB_SETUP_DEPENDENT macro in the configure.ac script seems to be for libraries that are dependent upon other libraries and not necessarily for optional/added features within a library. (closes issue ASTERISK-18535) Reported by: Michael Keuter git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335911 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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b695a91265 |
Fixed cut-n-paste regression using the wrong variable.
Fixes the missing DAHDI channels when using the newer chan_dahdi.conf
sections for channel configuration.
(closes issue ASTERISK-18496)
Reported by: Sean Darcy
Patches:
jira_asterisk_18496_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: Sean Darcy, rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335851 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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14 years ago |
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454969d783 |
The tech and data members of fast_originate_helper are not string fields.
ASTERISK-17709 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335790 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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5c5122d104 |
Remove obsolete todo comment about PICKUPRESULT.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335720 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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42840a2ef9 |
do parse defaultlanguage from asterisk.conf
Do parse the option "defaultlanguage" from the [options] section of asterisk.conf, as in the sample config file. Otherwise the build-time default language (normally "en") is always the default one. Review: https://reviewboard.asterisk.org/r/1342/ Signed-off-by: Tzafrir Cohen (License #5035) <tzafrir.cohen@xorcom.com> git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335716 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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28952b7ea5 |
Meetme should have 'core' support level
(closes issue ASTERISK-18542) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335714 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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28a4975127 |
Move mandatory checks closer to the beginning of the file.
If these are going to fail, they should fail as quickly as possible. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335655 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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ebb6110a13 |
Don't limit the size of appdata for manager originate actions.
ASTERISK-17709 Patch by: tilghman (with modifications) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335618 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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5f1882731a |
Fix a crash in res_ais.
This patch resolves a crash observed in a load testing environment that involved the use of the res_ais module. I observed some crashes where the event delivery callback would get called, but the length parameter incidcating how much data there was to read was 0. The code assumed (with good reason I would think) that if this callback got called, there was an event available to read. However, if the rare case that there's nothing there, catch it and return instead of blowing up. More specifically, the change always ensure that the size of the received event in the cluster is always big enough to be a real ast_event. Review: https://reviewboard.asterisk.org/r/1423/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335497 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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1aeb6f1242 |
Properly set caller_warning and callee_warning before we try to use them.
ASTERISK-18199 Patch by: elguero Testing by: rtang git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335433 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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828a733f58 |
Prevent a race condition when the bridge technology changes. This change was
ported from asterisk 10. ASTERISK-18155 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335431 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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263a410438 |
Ensure frames are not written to dialed channel if ringback is requested
When a single channel was dialed and there was media to be forwarded to the calling channel, the media was written without regard for ringback causing silence to be heard in some circumstances. This regression was introduced when the meaning of "single" changed to mean only the number of channels dialed. (closes issue ASTERISK-18083) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335341 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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b1b865d7b2 |
Prevent IAX2 from getting IPv6 addresses via DNS
IAX2 does not support IPv6 and getting such addresses from DNS can cause error messages on the remote end involving bad IPv4 address casts in the presence of IPv6/IPv4 tunnels. This patch ensures that IAX2 will not encounter IPv6 addresses via DNS queries. (closes issue ASTERISK-18090) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335320 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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c0ab1f3281 |
Lock the peer->mvipvt to avoid crashes with SIP history enabled
After the launch of 1.6 event-based MWI we have two threads handling the peer->mwipvt, which cause issues with SIP history additions in combination with the max limit for number of history entries. Review: https://reviewboard.asterisk.org/r/1373/ (closes issue ASTERISK-18288) Thanks to irrot for peer review. Work with this bug funded by IPvision AS git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335319 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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22b30eb82c |
build_peer doesnt unlink a peer object from peers_by_ip container which leads to a wrong refcounter value.
adding an ao2_unlink from the peers_by_ip container fix it. Review: https://reviewboard.asterisk.org/r/1428/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335259 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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7dc49195d8 |
Updated SIP 484 handling; added Incomplete control frame
When a SIP phone uses the dial application and receives a 484 Address Incomplete response, if overlapped dialing is enabled for SIP, then the 484 Address Incomplete is forwarded back to the SIP phone and the HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete application dialplan logic was automatically triggered; now, explicit dialplan usage of the application is required. Additionally, this patch adds a new AST_CONTOL_FRAME type called AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame, it is an indication that the dialplan expects more digits back from the device. If the device supports overlap dialing it should attempt to notify the device that the dialplan is waiting for more digits; otherwise, it can handle the frame in a manner appropriate to the channel driver. (closes issue ASTERISK-17288) Reported by: Mikael Carlsson Tested by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/1416/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335064 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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2fb25c3aea |
Fix crash with res_fax when MALLOC_DEBUG and "core stop gracefully" are used.
Asterisk crashes if MALLOC_DEBUG is enabled when res_fax tries to unregister its logger level. * Make ast_logger_unregister_level() use ast_free() instead of free(). When MALLOC_DEBUG is enabled, ast_free() does not degenerate into a call to free(). Therefore, if you allocated memory with a form of ast_malloc you must free it with ast_free. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@334953 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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f105f3e579 |
Cleanup chan_iax2.c log messages
Review: https://code.asterisk.org/code/cru/CR-AST-11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@334843 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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c6e7f17a68 |
Fix AMI action Park crash.
* Made AMI action Park not say anything to the parker channel (AMI header Channel2) since the AMI action is a third party parking the call. (This is a change in behavior that cannot be preserved without a lot of effort.) * Made not play pbx-parkingfailed if the Park 's' option is used. JIRA AST-660 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@334840 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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687e413fd8 |
Adding the Feature to sent a Reason Header in a SIP Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE before doing a masquerade in the pickup function.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@334682 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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74f9e66b41 |
peroid typo
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@334620 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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71f3dc01da |
Prevent segfault if call arrives before Asterisk is fully booted.
Prevent ast_pbx_start and ast_run_start from starting a new thread unless asterisk is fully booted. alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1407/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@334616 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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4b1398a82d |
Make SQL query in app_voicemail.c portable LIMIT is not portable.
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14 years ago |
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34919315ab |
MusicOnHold has extra unref which may lead to memory corruption and crash.
The problem happens when a call is disconnected and you had started a MOH
class that does not use the files mode. If you define REF_DEBUG and
recreate the problem, it will announce itself with the following warning:
Attempt to unref mohclass 0xb70722e0 (default) when only 1 ref remained,
and class is still in a container!
* Fixed moh_alloc() and moh_release() functions not handling the
state->class reference consistently.
(closes issue ASTERISK-18346)
Reported by: Mark Murawski
Patches:
jira_asterisk_18346_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett, Mark Murawski
Review: https://reviewboard.asterisk.org/r/1404/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@334355 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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15 years ago |
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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 |