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${ noResults }
23898 Commits (060ced4b54367f9822daaffe456da4460c04947c)
Author | SHA1 | Message | Date |
---|---|---|---|
|
174c84f6a2 |
main/file: Move test event to emit PLAYBACK event more consistently
This is being done in advance of the test for ASTERISK-23953 ........ Merged revisions 421059 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@421060 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
|
a63e70c46d |
logger: Don't store verbose-magic in the log files.
In r399267, the verbose2magic stuff was edited. This time it results in magic characters in the log files for multiline messages. In trunk (and 13) this was fixed by the "stripping" of those characters from multiline messages (in r414798). This is a backport of that fix to 11. That fix is altered to actually strip the characters and not replace them with blanks. Review: https://reviewboard.asterisk.org/r/3901/ Review: https://reviewboard.asterisk.org/r/3902/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@420897 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
|
9aba91e5a6 |
general: Fix memory Corruption in __ast_string_field_ptr_build_va.
If the space left in a stringfield is between 0 and (alignof(ast_string_field_allocation)-1) adding new data would cause memory corruption, because we would assume enough space (unsigned underrun). Thanks Arnd Schmitter for reporting and finding out the cause! ASTERISK-23508 #close Reported by: Arnd Schmitter Tested by: Arnd Schmitter, JoshE Review: https://reviewboard.asterisk.org/r/3898/ ........ Merged revisions 420680 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@420715 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
|
d43da8846c |
tcptls: Avoid compiler warning on non-dev-mode.
........ Merged revisions 420654 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@420655 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
|
d779523c8c |
chan_sip: Replace sip_tls_read() and resolve the large SDP poll issue.
Replace sip_tls_read() and sip_tcp_read() with a single function and resolve the poll/wait issue with large SDP payloads. ASTERISK-18345 #close Reported by: Stephane Chazelas Patches: tcptls_pollv4.diff (license #5835) patch uploaded by Elazar Broad Review: https://reviewboard.asterisk.org/r/3882/ ........ Merged revisions 420434 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@420435 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
|
0430c887f3 |
pbx_lua: fix regression with global sym export and context clash by pbx_config.
ASTERISK-23818 (lua contexts being overwritten by contexts of the same name in pbx_config) surfaced because pbx_lua, having the AST_MODFLAG_GLOBAL_SYMBOLS set, was always force loaded before pbx_config. Since I couldn't find any reason for pbx_lua to export it's symbols to the rest of Asterisk, I simply changed the flag to AST_MODFLAG_DEFAULT. Problem solved. What I didn't realize was that the symbols need to be exported not because Asterisk needs them but because any external Lua modules like luasql.mysql need the base Lua language APIs exported (ASTERISK-17279). Back to ASTERISK-23818... It looks like there's an issue in pbx.c where context_merge was only merging includes, switches and ignore patterns if the context was already existing AND has extensions, or if the context was brand new. If pbx_lua is loaded before pbx_config, the context will exist BUT pbx_lua, being implemented as a switch, will never place extensions in it, just the switch statement. The result is that when pbx_config loads, it never merges the switch statement created by pbx_lua into the final context. This patch sets pbx_lua's modflag back to AST_MODFLAG_GLOBAL_SYMBOLS and adds an "else if" in context_merge that catches the case where an existing context has includes, switchs or ingore patterns but no actual extensions. ASTERISK-23818 #close Reported by: Dennis Guse Reported by: Timo Teräs Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3891/ ........ Merged revisions 420146 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@420147 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
|
9707876002 |
format.c: Add reason comments for the format_list ordering.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@420054 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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af592bbc94 |
Manager - Improve documentation for manager commands Getvar and Setvar.
The documentation for these commands did not make it clear that they could accept expressions and functions. Modified to make this clear, but tried not to be overly explicit. ASTERISK-21178 #close Reported by: Rusty Newton Tested by: Rusty Newton Review: https://reviewboard.asterisk.org/r/3854 ........ Merged revisions 419942 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@419943 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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c2e464699f |
datastores: Audit ast_channel_datastore_remove usage.
Audit of v1.8 usage of ast_channel_datastore_remove() for datastore memory leaks. * Fixed leaks in app_speech_utils and func_frame_trace. * Fixed app_speech_utils not locking the channel when accessing the channel datastore list. Review: https://reviewboard.asterisk.org/r/3859/ Audit of v11 usage of ast_channel_datastore_remove() for datastore memory leaks. * Fixed leak in func_jitterbuffer. Review: https://reviewboard.asterisk.org/r/3860/ ........ Merged revisions 419684 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@419685 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
|
4c47641f0c |
features.c: Allow appliationmap to use Gosub.
Using DYNAMIC_FEATURES with a Gosub application as the mapped application does not work. It does not work because Gosub just pushes the current dialplan context, exten, and priority onto a stack and sets the specified Gosub location. Gosub does not have a dialplan execution loop to run dialplan like Macro. * Made the DYNAMIC_FEATURES application mapping feature call ast_app_exec_macro() and ast_app_exec_sub() for the Macro and Gosub applications respectively. * Backported ast_app_exec_macro() and ast_app_exec_sub() from v11 to execute dialplan routines from the DYNAMIC_FEATURES application mapping feature. NOTE: This issue does not affect v12+ because it already does what this patch implements. AST-1391 #close Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3844/ ........ Merged revisions 419630 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@419631 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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aaa44d6ea5 |
chan_sip: sip_subscribe_mwi_destroy should not call sip_destroy
sip_subscribe_mwi_destroy calls sip_destroy on the reference counted mwi->call. This results in the fields of mwi->call being freed, but mwi->call itself it leaked. If other code is still using mwi->call it can cause problems. This change uses dialog_unref instead, to balance the ref provided by sip_alloc(). ASTERISK-24087 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3834/ ........ Merged revisions 419440 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@419441 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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9219a97997 |
Don't cause Asterisk to exit if ooh323.conf not found.
(closes issue ASTERISK-23814) ........ Merged revisions 419374 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@419375 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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59795008d5 |
app_voicemail: use a consistent generator string
When updating voicemail.conf when a user changes their pin, change the generator string to be the same as the module name when reading so that the same config_hook will be called. Review: https://reviewboard.asterisk.org/r/3837/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@419284 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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22b9d0ddff |
Fix more dev-mode build issues
........ Merged revisions 419129 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@419162 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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bfc6904871 |
manager: Return ActionID on nominal responses to PresenceState action
When the PresenceState action is executed, the nominal path fails to include the ActionID in the successful response. This patch adds a call to astman_start_ack, which guarantees that an ActionID (if provided) will be sent back to the AMI client. Review: https://reviewboard.asterisk.org/r/3776/ ASTERISK-23985 #close git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@418713 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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3d419e8796 |
func_uri: URIENCODE/URIDECODE - allow empty strings as argument
Previously these two dialplan functions would issue warnings and return failure when an empty string is used as the argument. Now they will not issue a warning and will successfully return an empty string. ASTERISK-23911 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3745/ ........ Merged revisions 418641 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@418649 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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148b8d128e |
astobj2: work around REF_DEBUG race which causes out of order log entries
* Update refcounter.py to use delta's to track the current reference count. * Use result from internal_ao2_ref to write old_refcount to refs_log. Review: https://reviewboard.asterisk.org/r/3756/ ........ Merged revisions 418504 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@418505 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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7a914e14d0 |
Fix minor reference leaks in app_skel and TEST_FRAMEWORK
* Cleanup games object in app_skel. * Cleanup stasis subscription to TEST_FRAMEWORK in manager.c (12+). Review: https://reviewboard.asterisk.org/r/3757/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@418465 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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a100307eaa |
config: inform config hook of change when writing file
When updated configuration is written back to the conf file - for example when a user changes their voicemail pin, make sure that any config hook that wants to know of changes is informed. Review: https://reviewboard.asterisk.org/r/3708/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@418366 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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dfdf47a752 |
include/asterisk/xmpp.h: Convert indentation to tabs
This is a whitespace only change. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@418323 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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073356fa18 |
chan_dahdi/sig_pri: Fix type mismatch in the idledial feature's channel creation.
Square pegs in round holes don't work very well. ........ Merged revisions 418261 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@418262 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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2f1d2a9ef6 |
chan_dahdi: Add inband_on_setup_ack compatibility option.
The new inband_on_setup_ack option causes Asterisk to assume inband audio may be present when a SETUP_ACKNOWLEDGE message is received. Q.931 Section 5.1.3 says that in scenarios with overlap dialing, when a dialtone is sent from the network side, progress indicator 8 "Inband info now available" MAY be sent to the CPE if no digits were received with the SETUP. It is thus implied that the ie is mandatory if digits came with the SETUP and dialtone is needed. This option should be enabled, when the network sends dialtone and you want to hear it, but the network doesn't send the progress indicator when needed. NOTE: For Q.SIG setups this option should be enabled when outgoing overlap dialing is also enabled because Q.SIG does not send the progress indicator with the SETUP ACK. The commit -r413714 (AST-1338) which causes this issue was dealing with a SIP-to-ISDN interoperability issue. This commit is a merge of the two patches indicated below. ASTERISK-23897 #close Reported by: Pavel Troller Patches: pri-4.diff (license #6302) patch uploaded by Pavel Troller jira_asterisk_23897_v11.patch (license #5621) patch uploaded by rmudgett Review: https://reviewboard.asterisk.org/r/3633/ ........ Merged revisions 417956 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@417957 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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8f5461d3dc |
main/untils: Prevent potential infinite loop in ast_careful_fwrite
A loop in ast_careful_fwrite exists that will continually attempt to write to a file stream, even in the presence of EAGAIN/EINTR errors. However, if a connection that uses ast_careful_fwrite closes suddenly, ast_careful_fwrite's call to fflush may return EAGAIN/EINTER along with EOF. A subsequent call to fflush will return EOF but not clear errno, resulting in an infinite loop. This patch clears errno after it is detected and handled the loop, such that any subsequent call to fflush will not get erroneously stuck. Review: https://reviewboard.asterisk.org/r/3704 #ASTERISK-23984 #close Reported by: Steve Davies patches: fflush_loop_fix uploaded by one47 (License 5012) ........ Merged revisions 417797 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@417798 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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915de454f8 |
res_rtp_asterisk: Add SHA-256 support for DTLS and perform DTLS negotiation on RTCP.
This change fixes up DTLS support in res_rtp_asterisk so it can accept and provide a SHA-256 fingerprint, so it occurs on RTCP, and so it occurs after ICE negotiation completes. Configuration options to chan_sip have also been added to allow behavior to be tweaked (such as forcing the AVP type media transports in SDP). ASTERISK-22961 #close Reported by: Jay Jideliov Review: https://reviewboard.asterisk.org/r/3679/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@417677 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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1644cd874d |
chan_sip: be more tolerant of whitespace between attributes in SDP fmtp line
This patch is essentially a backport of a small portion of r397526 from ASTERISK-21981. In that patch, pass through support and format attribute negotiation was added for Opus. Part of that included being more tolerant to whitespace in the fmtp line of an SDP; that part of the patch is being applied here. As the author of the backport pointed out, in SDP, the fmtp line is allowed to include whitespace between attributes. RFC 3267 chapter 8.3 (from 2001) includes an example for this. This was not removed in the updated RFC 4867 in 2007. Review: https://reviewboard.asterisk.org/r/3658 ASTERISK-23916 #close Reported by: Alexander Traud patches: sdpFMTPspace_Asterisk11.patch uploaded by Alexander Traud (License 6520) ........ Merged revisions 417587 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@417588 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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23d1b904b0 |
Ensure REF_DEBUG records entrys for attempts to ao2_ref an invalid object
This change ensures that __ao2_ref_debug writes to ref_log when given a non-NULL pointer to an invalid ao2 object. This is to ensure that we record any attempt manipulate references of already freed objects. ASTERISK-23948 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3677/ ........ Merged revisions 417500 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@417505 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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fe70d27731 |
refcounter.py: prevent use of excessive RAM with large refs logs
When processing a 212MB refs file, refcounter.py used over 3GB of RAM. This change greatly reduces memory usage in two ways: * Saving object history in whole lines instead of separated values. * Not saving normal/skewed/leaked object lists unless they are requested. ASTERISK-23921 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3668/ ........ Merged revisions 417480 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@417481 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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5d3078bd48 |
res_http_websocket: Export symbol for ast_websocket_set_timeout
Thanks to Sean Bright for pointing out that this was missed in #asterisk-dev. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@417419 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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0b9601052b |
udptl: Correct FEC to not consider negative sequence numbers as missing
When using FEC, with span=3 and entries=4 Asterisk will attempt to repair the packet with sequence number 5, as it will see that packet -4 is missing. The result is Asterisk sending garbage packets that can kill a fax. This patch adds a check to see if the sequence number is valid before checking if the packet is missing. Review: https://reviewboard.asterisk.org/r/3657/ #ASTERISK-23908 #close Reported by: Torrey Searle patches: udptl_fec.patch uploaded by Torrey Searle (License 5334) ........ Merged revisions 417318 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@417320 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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32d684ee25 |
res_http_websocket: Close websocket correctly and use careful fwrite
When a client takes a long time to process information received from Asterisk, a write operation using fwrite may fail to write all information. This causes the underlying file stream to be in an unknown state, such that the socket must be disconnected. Unfortunately, there are two problems with this in Asterisk's existing websocket code: 1. Periodically, during the read loop, Asterisk must write to the connected websocket to respond to pings. As such, Asterisk maintains a reference to the session during the loop. When ast_http_websocket_write fails, it may cause the session to decrement its ref count, but this in and of itself does not break the read loop. The read loop's write, on the other hand, does not break the loop if it fails. This causes the socket to get in a 'stuck' state, preventing the client from reconnecting to the server. 2. More importantly, however, is that the fwrite in ast_http_websocket_write fails with a large volume of data when the client takes awhile to process the information. When it does fail, it fails writing only a portion of the bytes. With some debugging, it was shown that this was failing in a similar fashion to ASTERISK-12767. Switching this over to ast_careful_fwrite with a long enough timeout solved the problem. ASTERISK-23917 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3624/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@417310 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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2875174e97 |
chan_sip: Fix handling of "From" headers longer than 256 characters
From headers were processed using a 256 character buffer on the stack. This change replaces that with a heap allocation by ast_strdup. ASTERISK-23790 #close Reported by: uniken1 Tested by: uniken1 Review: https://reviewboard.asterisk.org/r/3669/ Patches: chan_sip-large-from-header-1.8-r3.patch uploaded by wdoekes (license 5674) ........ Merged revisions 417248 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@417249 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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3e642e66d1 |
res_rtp_asterisk: Return the length of data written when sending via ICE instead of 0.
ASTERISK-23834 #close Reported by: Richard Kenner git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@417141 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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cd6a931be5 |
main/features - documentation - reformat examples and options in features.conf.sample to show clearly which options apply in which section
The features.conf sample can be a bit confusing about what parking options can be set only in the general context, or both in the general context (for the default parking lot) and in other parking lot contexts. A bug was filed due to confusion and a little googling will show lots of other confused users. Despite some comments on the individual options, it still reads in a confusing way. In this patch I separate out those options with some headings in to attempt a better layout. I went ahead and modified other headings in the file, or added them to facilitate better visual scanning. ASTERISK-23667 Review: https://reviewboard.asterisk.org/r/3622/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@417077 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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c7e9233a81 |
build: Turn FORTIFY_SOURCE off if DONT_OPTIMIZE is set.
AST_FORTIFY_SOURCE is automatically set in ./Makefile even if DONT_OPTIMIZE is set in menuselect. This causes gcc to complain that _FORTIFY_SOURCE requires optimization and the build will fail. You can specify "make AST_FORTIFY_SOURCE=''" but I always forget. This patch moves the set of AST_FORTIFY_SOURCE to Makefile.rules and only sets it if DONT_OPTIMIZE is "no". The move is necessary because the top-level Makefile doesn't include menuselect.makeopts. This doesn't solve the entire problem however because res_config_mysql seems to force _FORTIFY_SOURCE so res_config_mysql has to be disabled for now if DONT_OPTIMIZE is set. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3664/ ........ Merged revisions 417016 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@417017 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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2b4423d2b3 |
build: Allow autoconf/ast_ext_tool_check to handle cross-compiling better.
ast_ext_tool_check.m4 isn't handling cases where a path to a package is provided (E.G. --with-mysqlclient=/some/sysroot) and the package has a config tool (E.G. mysql_config) and the package has its own subdirectories in include or lib. For example, mysql's libraries are in ${MYSQLCLIENT_DIR}/usr/lib/mysql but ast_ext_tool_check sets MYSQLCLIENT_LIB to ${MYSQLCLIENT_DIR}/usr/lib. libxml2 has the same problem with its includes. They're in ${LIBXML2_DIR}/usr/include/libxml2 not directly in ${LIBXML2_DIR}/usr/include. Both cause configure to fail and there are others in the same boat. The problem is caused by logic in ast_ext_tool_check that overrides the result of the config tool's --cflags and --libs options if package_DIR is set. This patch prepends package_DIR (if specified) to the -L and -I results from the package's config tool instead of overriding them. A regenerated ./configure and include/asterisk/autoconfig.h.in are included but can be regenerated by running ./bootstrap.sh at any time. Tested by: George Joseph Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3550/ ........ Merged revisions 416929 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@416930 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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a6ba838fce |
build: Allow autoconf/ast_ext_tool_check to handle cross-compiling better.
ast_ext_tool_check.m4 isn't handling cases where a path to a package is provided (E.G. --with-mysqlclient=/some/sysroot) and the package has a config tool (E.G. mysql_config) and the package has its own subdirectories in include or lib. For example, mysql's libraries are in ${MYSQLCLIENT_DIR}/usr/lib/mysql but ast_ext_tool_check sets MYSQLCLIENT_LIB to ${MYSQLCLIENT_DIR}/usr/lib. libxml2 has the same problem with its includes. They're in ${LIBXML2_DIR}/usr/include/libxml2 not directly in ${LIBXML2_DIR}/usr/include. Both cause configure to fail and there are others in the same boat. The problem is caused by logic in ast_ext_tool_check that overrides the result of the config tool's --cflags and --libs options if package_DIR is set. This patch prepends package_DIR (if specified) to the -L and -I results from the package's config tool instead of overriding them. Tested by: George Joseph Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3550/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@416870 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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b583f708cb |
Fix build warnings with TEST_FRAMEWORK enabled
........ Merged revisions 416732 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@416733 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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874be8b530 |
Remove the problematic and unneeded AST_MODFLAG_GLOBAL_SYMBOLS from pbx_lua.c
AST_MODFLAG_GLOBAL_SYMBOLS was causing the module to be incorrectly loaded before pbx_config. pbx_config was therefore blowing away contexts that were created by pbx_lua. With AST_MODFLAG_DEFAULT the load order is now correct and contexs are being properly merged. AST_MODFLAG_GLOBAL_SYMBOLS was not needed anyway since no other modules needed its global symbols that early. ASTERISK-23818 #close Reported by: Dennis Guse Tested by: Dennis Guse Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3629/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@416668 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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b960b9c2d5 |
Update extensions.lua.sample with naming conflict guidance.
The sample extensions.lua was causing pbx_lua to fail to load when parsing 'app.goto("default", "s", 1)' because in Lua 5.2, 'goto' is now a reserved word. This patch adds guidance to extensions.lua.sample and changed 'app.goto("default", "s", 1)' to 'app.['goto']("default", "s", 1)'. ASTERISK-23844 #close Reported by: rnewton Tested by: gtjoseph Review: https://reviewboard.asterisk.org/r/3627/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@416581 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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cc7bc40c2a |
Allow the PUSH and UNSHIFT functions to set inheritable channel variables.
........ Merged revisions 416500 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@416501 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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15d2f541b4 |
MoH: Don't restart stream on repeated start calls
Currently, music on hold will stop and then start again from the beginning if ast_moh_start() is called multiple times. This can happen if a call is put on hold repeatedly (the channel receives multiple HOLD control frames) and can be triggered from ARI by starting MoH on a channel multiple times. This is fairly jarring/annoying to users. This change prevents MoH from being restarted if the requested music class is the same as the one currently playing. This includes an extra check to prevent the errors previously experienced in the testsuite and has 100+ test runs behind it. Review: https://reviewboard.asterisk.org/r/3615/ ........ Merged revisions 416439 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@416440 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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07a0838e2a |
We have faced situation when using CDR and CEL by sqlite3 modules. With system having high load (~100 concurrent calls created by sipp) we found many cdr and cel records missed. There is special finction in sqlite3, that make able to fix this situation - sqlite3_wait_timeout, that also can replace awful code cdr_sqlite3 ad cel_sqlite3 modules. Also this function can be used for aastdb and res_config_sqlite3 to avoid missed writes to sqlite db.
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11 years ago |
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8d848c048d |
MoH: Undo commit r416150 (1.8)
This patch reverts r416150. When the comparison between mohclass->name and state->class->name is made, you are not guaranteed that (a) state->class is non-NULL or that state or state->class are in a safe state. Crashes caught by the bridges/transfer_capabilities test. ........ Merged revisions 416251 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@416252 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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71c9c7612e |
MoH: Don't restart stream on repeated start calls
Currently, music on hold will stop and then start again from the beginning if ast_moh_start() is called multiple times. This can happen if a call is put on hold repeatedly (the channel receives multiple HOLD control frames) and can be triggered from ARI by starting MoH on a channel multiple times. This is fairly jarring/annoying to users. This change prevents MoH from being restarted if the requested music class is the same as the one currently playing. Review: https://reviewboard.asterisk.org/r/3615/ ........ Merged revisions 416150 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@416151 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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11553fd489 |
AST-2014-007: Fix of fix to allow AMI and SIP TCP to send messages.
ASTERISK-23673 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/3617/ ........ Merged revisions 416066 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@416067 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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7e2ed2e032 |
main/pbx - documentation - enhance 'core show hints' and 'core show hint' help text
Adds descriptive help text to 'core show hints' and 'core show hint'. The text describes the various columns for the sake of clarity. ASTERISK-23764 Review: https://reviewboard.asterisk.org/r/3610/ ........ Merged revisions 415998 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@415999 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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c7df3bd093 |
chan_sip: DEBUG messages in sdp_crypto.c display despite a DEBUG level of zero
Change debug level for messages in sdp_crypto.c from zero to one. This ensures the messages are not displayed when debugging is disabled. Change does not apply to 12+ as it was already fixed in those versions. ASTERISK-23246 #close Reported by: Rusty Newton Review: https://reviewboard.asterisk.org/r/3605/ ........ Merged revisions 415908 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@415915 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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df686c50d8 |
AST-2014-007: Fix DOS by consuming the number of allowed HTTP connections.
Simply establishing a TCP connection and never sending anything to the configured HTTP port in http.conf will tie up a HTTP connection. Since there is a maximum number of open HTTP sessions allowed at a time you can block legitimate connections. A similar problem exists if a HTTP request is started but never finished. * Added http.conf session_inactivity timer option to close HTTP connections that aren't doing anything. Defaults to 30000 ms. * Removed the undocumented manager.conf block-sockets option. It interferes with TCP/TLS inactivity timeouts. * AMI and SIP TLS connections now have better authentication timeout protection. Though I didn't remove the bizzare TLS timeout polling code from chan_sip. * chan_sip can now handle SSL certificate renegotiations in the middle of a session. It couldn't do that before because the socket was non-blocking and the SSL calls were not restarted as documented by the OpenSSL documentation. * Fixed an off nominal leak of the ssl struct in handle_tcptls_connection() if the FILE stream failed to open and the SSL certificate negotiations failed. The patch creates a custom FILE stream handler to give the created FILE streams inactivity timeout and timeout after a specific moment in time capability. This approach eliminates the need for code using the FILE stream to be redesigned to deal with the timeouts. This patch indirectly fixes most of ASTERISK-18345 by fixing the usage of the SSL_read/SSL_write operations. ASTERISK-23673 #close Reported by: Richard Mudgett ........ Merged revisions 415841 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@415854 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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0df802b473 |
Correct UPGRADE.txt notes in r415825
The change was marked against the wrong version of Asterisk. My apologies. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@415837 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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a5f39fc2ba |
app_queue: delayed state can cause early leavewhenempty ringing
In app_queue, device state changes arrive in event messages and update the queue member status value. That value is checked in get_member_status() to decide that the caller should leave when there are no available members. Although event messages can be delayed by other activity, there is no adverse affect by lagged status except in one specific case: there is only one available member, it was just rung, and leavewhenempty is enabled set for ringing members. This change adds a direct check of the device state only under this condition where the caller may be dropped incorrectly, resolving this issue without affecting performance of app_queue normally. AST-1248 #close Review: https://reviewboard.asterisk.org/r/3595/ Reported by: Thomas Arimont ........ Merged revisions 415833 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@415835 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |