asterisk

Commit Graph

Author	SHA1	Message	Date
Kinsey Moore	2f0051982a	Make sure asterisk builds on OpenBSD OpenBSD defines SO_PEERCRED, but it returns a 'struct sockpeercred', not 'struct ucred', which causes compilation of main/asterisk.c to fail in read_credentials(). This allows configure to check for sockpeercred and asterisk to deal with it properly. (closes issue ASTERISK-18929) Reported-by: Barry Miller Patch-by: Barry Miller git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350730 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	f7876c1dfe	Set port to a default sane value if a bogus one is provided when parsing hostnames. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350679 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	26966a5cd8	Use compatible names for event extra data for various CEL backends. * Change eventextra to extra in cel_psql.c and cel_odbc.c. * Change EventExtra to Extra in cel_manager.c. (issue ASTERISK-17190) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350571 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	773074f3b7	Add missing CEL logging fields to various CEL backends. * Add missing eventextra to cel_psql.c and cel_odbc.c. * Add missing PeerAccount and EventExtra to cel_manager.c. * Add missing userdeftype support for cel_custom.conf.sample and cel_sqlite3_custom.conf.sample. (closes issue ASTERISK-17190) Reported by: Bryant Zimmerman git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350555 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	a364b1e068	Realtime queues failed to load queue information without queue member table Previously, realtime queues could be loaded without defining the queue member table. This allowed for queue members to be dynamic, while the realtime queue definitions could exist in some backing storage. Revision 342223 broke this when it changed the return value for realtime_multientry to return NULL when no results are returned. Previously, an empty ast_config object was expected. (closes issue ASTERISK-19170) Reported by: Rene Mendoza Tested by: Rene Mendoza Patches: rt_queue_member_patch.diff uploaded by Matt Jordan (license 6283) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350552 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	f22392dbdc	Adds peer to CEL report on CEL_BRIDGE_START and CEL_BRIDGE_END (closes issue ASTERISK-17940) Reporter: Nic Colledge Patches: features_18.patch uploaded by Nic Colledge (license 6245) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350501 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	929c8c3cb7	Remove extraneous BRIDGEPEER AMI VarSet event on a CEL dummy channel. (closes issue ASTERISK-19180) Reported by: Corey Farrell Patches: asterisk_cel_noevent_varset.diff (license #5909) patch uploaded by Corey Farrell git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350452 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	99868648e4	Make FollowMe optionally update connected line information when the accepting endpoint is bridged. Like Dial and Queue, FollowMe needs to deal with AST_CONTROL_CONNECTED_LINE information so when the parties are initially bridged, the connected line information will be correct. * Added the 'I' option just like the app_dial and app_queue 'I' option. (closes issue ASTERISK-18969) Reported by: rmudgett Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1656/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350364 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	ed61726748	Fix absolute/relative time mismatch in LOCK function. The time passed by the LOCK function to an internal function was relative time when the function expected absolute time. * Don't use C++ keywords in get_lock(). (closes issue ASTERISK-16868) Reported by: Andrey Solovyev Patches: 20101102__issue18207.diff.txt (license #5003) patch uploaded by Andrey Solovyev (modified) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350311 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	fb6b399324	Fix joinable thread terminating without joiner memory leak in chan_iax.c. The iax2_process_thread() can exit without anyone waiting to join the thread. If noone is waiting to join the thread then a large memory leak occurs. * Made iax2_process_thread() deatach itself if nobody is waiting to join the thread. (closes issue ASTERISK-17339) Reported by: Tzafrir Cohen Patches: asterisk-1.8.4.4-chan_iax2-detach-thread-on-non-stop-exit.patch (license #5617) patch uploaded by Alex Villacis Lasso (modified) (closes issue ASTERISK-17825) Reported by: wangjin git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350220 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	8726d89f58	live_ast: valgrind: run asterisk under valgrind Adds a new sub-command, "valgrind" to live_ast. It runs asterisk under valgrind. The extra command-line parameters are passed to Asterisk as usual, and parameters to valgrind are passed through LIVE_AST_VALGRIND_ARGS in live.conf . Review: https://reviewboard.asterisk.org/r/1109/ Merged revisions 326636 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350128 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	3609b8c19b	Update contrib script live_ast to invoke Asterisk with valgrind and suppression file. * Added valgrind_compare script to compare two valgrind log files for differences. (issue ASTERISK-17339) Reported by: Tzafrir Cohen Patches: valgrind_compare (license #5035) script uploaded by Tzafrir Cohen live_ast_valgrind.diff (license #5035) patch uploaded by Tzafrir Cohen live_ast_valgrind_v2.diff (license #5185) patch uploaded by Paul Belanger git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350127 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	c1f9b1277e	Make Asterisk -x command line parameter imply -r parameter presence. The Asterisk -x command line parameter is documented inconsistently. * Made the -x documentation and behavior consistent. * Since this is also a new year, updated the copyright notices while here. (closes issue ASTERISK-19094) Reported by: Eugene Patches: issueA19094_correct_asterisk_option_x.patch (license #5674) patch uploaded by Walter Doekes (modified) Tested by: Eugene git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350075 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	cf9df098bd	Prevent SLA settings from getting wiped out on reload If SLA was reloaded without the config file being changed, current settings got wiped out before the SLA reload code decided it wasn't going to reload the file since nothing was changed. Moving the settings reset later in the reload process fixes this. (closes issue AST-744) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350023 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	96770386a7	Don't leak CID in From header when presentation=unavailable When someone does Set(CALLERPRES()=unavailable) (or Set(CALLERID(pres)=unavailable)) when sendrpid=no, the From header shows "Anonymous" <anonymous@anonymous.invalid>. When sendrpid=yes/pai, the From header will still display the callerid info, even though we supply an rpid header with the anonymous info. It seems like we shouldn't leak that info in any case. Skimming http://tools.ietf.org/html/draft-ietf-sip-privacy-04 seems to indicate that one shouldn't send identifying info in the From in this case. This patch anonymizes the From header as well even when sendrpid=yes/pai. (closes issue ASTERISK-16538) Review: https://reviewboard.asterisk.org/r/1649/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@349968 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	d377bc31ce	Fix memory leaks in app_followme find_realtime(). (closes issue ASTERISK-19055) Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@349872 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	9376320c82	Make not assume that the cel_sqlite3_custom SQL table primary key is AcctId. If a table is created by some other application and the primary key is not named "AcctId", cel/cel_sqlite3_custom.c will always try to create the table and fail because it already exists. * Change the SQL table query to not require AcctId as the primary key. (closes issue ASTERISK-18963) Reported by: socketpair Patches: fix.patch (license #6337) patch uploaded by socketpair git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@349819 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	49a980a427	Allow playback of formats that don't support seeking ast_streamfile previously did unconditional seeking on files that broke playback of formats that don't support that functionality. This patch avoids the seek that was causing the problem. This regression was introduced in r158062. (closes issue ASTERISK-18994) Patch-by: Timo Teras git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@349731 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	1035b21f62	Fix an issue where dsp.c would interpret multiple dtmf events from a single key press. When receiving calls from a mobile phone into a DISA system on a connection with significant interference, the reporter's Asterisk system would interpret DTMF incorrectly and replicate digits received. This patch resolves that by increasing the number of frames a mismatch has to be detected before assuming the DTMF is over by 1 frame and adjusts dtmf_detect function to reset hits and misses only when an edge is detected. (closes issue ASTERISK-17493) Reported by: Alec Davis Patches: bug18904-refactor.diff.txt uploaded by Alec Davis (license 5546) Review: https://reviewboard.asterisk.org/r/1130/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@349728 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	879208db8e	Ensures Asterisk closes when receiving terminal signals in 'no fork' mode. When catching a signal, in no fork mode the console thread is identical to the thread responsible for catching the signal and closing Asterisk, which requires it to first dispense with the console thread. Prior to this patch, if these threads were identical, upon receiving a killing signal, the thread will send an URG signal to itself, which we also catch and then promptly do nothing with. Obviously this isn't useful behavior. (closes issue ASTERISK-19127) Reported By: Bryon Clark Patches: quit_on_signals.patch uploaded by Bryon Clark (license 6157) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@349672 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	6f7d995edc	Fix segfault in chan_dahdi for CHANNEL(dahdi_span) evaluation on hangup. * Added NULL private pointer checks in the following chan_dahdi channel callbacks: dahdi_func_read(), dahdi_func_write(), dahdi_setoption(), and dahdi_queryoption(). (closes issue ASTERISK-19142) Reported by: Diego Aguirre Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@349558 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	1414d5b804	Make debian init script conform to the LSB standard Previously, this init script would return 1 if Asterisk was already running. This is incorrect behavior according to the LSB standard and has been fixed by returning 0 instead. (closes issue ASTERISK-17958) Reported-by: johnc git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@349529 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	7369c261ea	Update autosupport script and man page Added information collection from the output of the utilities: top, free, uptime, ifconfig Added information collection from the output of the Asterisk command 'dahdi show status' Added option / flag '-n, --non-interactive' Updated man page to reflect new option / flag '-n, --non-interactive' Patch-by: John Bigelow (itzanger) (closes issue AST-749) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@349504 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	21fa14919d	Adds Subscription-State header to notify with call completion. per RFC3265 (Closes issue ASTERISK-17953) Reported by: George Konopacki Patches: 19400.patch uploaded by mmichelson (license 5049) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@349482 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	fc698edaad	Fix documentation for SayNumber to reflect the fact that language is changed in CHANNEL() (closes issue ASTERISK-18962) reported by: Nir Simionovich git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@349450 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	5bdcc834df	Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop Failing to handle AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop causes the loop to exit prematurely. This causes a variety of negative side effects, depending on when the loop exits. This patch handles the frame by essentially swallowing the frame in the local loop, as the current channel drivers expect the RTP bridge to handle the frame, and, in the case of the local bridge loop, no additional action is necessary. (issue ASTERISK-19040) (issue ASTERISK-19128) (issue ASTERISK-17725) (issue ASTERISK-18340) (closes issue ASTERISK-19095) Reported by: Stefan Schmidt Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1640/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@349339 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	a05723fd5f	Use ast_audiohook_write_list_empty to determine if our lists are empty instead of duplicating that logic. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@349289 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	0ee313f076	Fix timing source dependency issues with MOH Prior to this patch, res_musiconhold existed at the same module priority level as the timing sources that it depends on. This would cause a problem when music on hold was reloaded, as the timing source could be changed after res_musiconhold was processed. This patch adds a new module priority level, AST_MODPRI_TIMING, that the various timing modules are now loaded at. This now occurs before loading other resource modules, such that the timing source is guaranteed to be set prior to resolving the timing source dependencies. (closes issue ASTERISK-17474) Reporter: Luke H Tested by: Luke H, Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont Patches: asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-1.8.diff uploaded by elguero (License #5026) asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-10.diff uploaded by elguero (License #5026) asterisk-17474-dahdi_timing-infinite-wait-fix_v3.diff uploaded by elguero (License #5026) Review: https://reviewboard.asterisk.org/r/1578/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@349194 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	c74793e570	Once an audiohook is attached to a channel, we continue to transcode all of the frames, even after all of the hooks are detached. This patch short-cicuits us out before we transcode unnecessarily. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@349144 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	b9bfee7ee6	In ChanSpy, don't create audiohooks that will never be used. When ChanSpy is initialized it creates and attaches 3 audiohooks: 1) Read audio off of the channel that we are spying on 2) Write audio to the channel that we are spying on 3) Write audio to the channel that is bridged to the channel that we are spying on. The first is always necessary, but the others are used only when specific options are passed to the ChanSpy application (B, d, w, and W to be specific). When those flags are not passed, neither of those audiohooks are ever sent frames, but we still try to process the hooks for each voice frame that we recieve on the channel. So in short - only create and attach audiohooks that we actually need. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@349044 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	77fb12285d	Fix missing doc tags found while fixing ASTERISK-18689 Add missing <variable></variable> tags in app_dial documentation. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348992 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	9698383360	Fix extension state callback references in chan_sip. Chan_sip gives a dialog reference to the extension state callback and assumes that when ast_extension_state_del() returns, the callback cannot happen anymore. Chan_sip then reduces the dialog reference count associated with the callback. Recent changes (ASTERISK-17760) have resulted in the potential for the callback to happen after ast_extension_state_del() has returned. For chan_sip, this could be very bad because the dialog pointer could have already been destroyed. * Added ast_extension_state_add_destroy() so chan_sip can account for the sip_pvt reference given to the extension state callback when the extension state callback is deleted. * Fix pbx.c awkward statecbs handling in ast_extension_state_add_destroy() and handle_statechange() now that the struct ast_state_cb has a destructor to call. * Ensure that ast_extension_state_add_destroy() will never return -1 or 0 for a successful registration. * Fixed pbx.c statecbs_cmp() to compare the correct information. The passed in value to compare is a change_cb function pointer not an object pointer. * Make pbx.c ast_merge_contexts_and_delete() not perform callbacks with AST_EXTENSION_REMOVED with locks held. Chan_sip is notorious for deadlocking when those locks are held during the callback. * Removed unused lock declaration for the pbx.c store_hints list. (closes issue ASTERISK-18844) Reported by: rmudgett Review: https://reviewboard.asterisk.org/r/1635/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348940 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	78f0d8d50b	Fix for memory leaks / cleanup in cel_pgsql There were a number of issues in cel_pgsql's pgsql_log method: * If either sql or sql2 could not be allocated, the method would return while the pgsql_lock was still locked * If the execution of the log statement succeeded, the sql and sql2 structs were never free'd * Reconnection successes were logged as ERRORs. In general, the severity of several logging statements was reduced (closes issue ASTERISK-18879) Reported by: Niolas Bouliane Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1624/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348888 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	05078f24e1	Allow packetization vaules > 127 According to the RTP packetization documentation, and the maximum values listed in AST_FORMAT_LIST, we should support values > that the signed char array that ast_codec_pref makes available to store the value. All places in the code treat the framing field as though it were an int array instaead of a char array anyway, so this just fixes the type of the array. (closes issue ASTERISK-18876) Review: https://reviewboard.asterisk.org/r/1639/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348833 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	fc96f6eb9d	Fix chan_iax2 to not report an RDNIS number if it is blank. Some ISDN switches complain or block the call if the RDNIS number is empty. * Made chan_iax2 not save a RDNIS number into the ast_channel if the string is blank. This is what other channel drivers do. (closes issue ASTERISK-17152) Reported by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348735 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	4a5ed19cd7	Fix crashes on other platforms caused by interference from Darwin weak symbol support. Support weak symbols on a platform specific basis. The Mac OS X (Darwin) support must be isolated from the other platforms because it has caused other platforms to crash. Several other platforms including Linux have GCC versions that define the weak attribute. However, this attribute is only setup for use in the code by Darwin. (closes issue ASTERISK-18728) Reported by: Ben Klang Review: https://reviewboard.asterisk.org/r/1617/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348647 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	6dbb78d453	Correct two flaws in sip.conf.sample related to AST-2011-013. * The sample file listed two values for the 'nat' option as being the default. Only 'force_rport' is the default. * The warning about having differing 'nat' settings confusingly referred to both peers and users. ........ Merged revisions 348515 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348516 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	ec6b5be4b9	Clean-up on isle five for __ast_request_and_dial() and ast_call_forward(). * Add locking when a channel inherits variables and datastores in __ast_request_and_dial() and ast_call_forward(). Note: The involved channels are not active so there was minimal potential for problems. * Remove calls to ast_set_callerid() in __ast_request_and_dial() and ast_call_forward() because the set information is for the wrong direction. * Don't use C++ keywords for variable names in ast_call_forward(). * Run the redirecting interception macro if defined when forwarding a call in ast_call_forward(). Note: Currently will never execute because the only callers that supply a calling channel supply a hungup or zombie channel. * Make feature_request_and_dial() put the transferee into autoservice when it calls ast_call_forward() in case a redirection interception macro is run. Note: Currently will never happen because the caller channel (Party B) is always hungup at this time. * Make feature_request_and_dial() ignore the AST_CONTROL_PROCEEDING frame to silence a log message. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348464 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	d1c0c7c6c5	Fix cut and past error in ast_call_forward(). (issue ASTERISK-18836) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348401 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	74da7648bb	Fix crash during CDR update. The ast_cdr_setcid() and ast_cdr_update() were shown in ASTERISK-18836 to be called by different threads for the same channel. The channel driver thread and the PBX thread running dialplan. * Add lock protection around CDR API calls that access an ast_channel pointer. (closes issue ASTERISK-18836) Reported by: gpluser Review: https://reviewboard.asterisk.org/r/1628/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348362 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	6b17e5e23c	Fix ParkAndAnnounce to pass the CallerID to the announcing channel. ParkAndAnnounce tried to pass the CallerID to the announcing channel but the ID was wiped out by the channel masquerade done when parking the call. * Save the CallerID before parking the channel to pass it to the announcing channel. * Fixed a minor memory leak in ParkAndAnnounce. * Updated some ParkAndAnnounce log messages. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348310 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	5921a96450	Don't clear LOCALSTATIONID before sending or receiving. The user may set that variable. ASTERISK-18921 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348212 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	ea8c309ecf	Fix accidental use of tabs instead of spaces from previous features.conf.sample change git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348157 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	9b52c5f1c9	Document PARKINGSLOT variable in features.conf.sample (issue ASTERISK-16239) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348154 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	bf8ba13e66	Fix FollowMe CallerID on outgoing calls. The addition of the Connected Line support changed how CallerID is passed to outgoing calls. The FollowMe application was not updated to pass CallerID to the outgoing calls. * Fix FollowMe CallerID on outgoing calls. * Restructured findmeexec() to fix several memory leaks and eliminate some duplicated code. * Made check the return value of create_followme_number(). Putting a NULL into the numbers list is bad if create_followme_number() fails. * Fixed a couple uses of ast_strdupa() inside loops. * The changes to bridge_builtin_features.c fix a similar CallerID issue with the bridging API attended and blind transfers. (Not used at this time.) (closes issue ASTERISK-17557) Reported by: hamlet505a Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1612/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348101 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Stefan Schmidt	9d53f3352b	Fix possible misshandling of an incoming SIP response as a peer poke response. Also make sure peer has even qualify enabled when handle a peer poke response. (closes issue ASTERISK-18940) Reported by: Vitaliy Tested by: Vitaliy and UnixDev Review: https://reviewboard.asterisk.org/r/1620 Reviewed by: David Vossel git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348048 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	607398d450	Add a separate buffer for SRTCP packets The function ast_srtp_protect used a common buffer for both SRTP and SRTCP packets. Since this function can be called from multiple threads for the same SRTP session (scheduler for SRTCP and channel for SRTP) it was possible for the packets to become corrupted as the buffer was used by both threads simultaneously. This patch adds a separate buffer for SRTCP packets to avoid the problem. (closes issue ASTERISK-18889, Reported/patch by Daniel Collins) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@347995 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	47783a00a9	Fix some parsing issues in add_exten_to_pattern_tree(). * Simplify compare_char() and avoid potential sign extension issue. * Fix infinite loop in add_exten_to_pattern_tree() handling of character set escape handling. * Added buffer overflow checks in add_exten_to_pattern_tree() character set collection. * Made ignore empty character sets. * Added escape character handling to end-of-range character in character sets. This has a slight change in behavior if the end-of-range character is an escape character. You must now escape it. * Fix potential sign extension issue when expanding character set ranges. * Made remove duplicated characters from character sets. The duplicate characters lower extension matching priority and prevent duplicate extension detection. * Fix escape character handling when the escape character is trying to escape the end-of-string. We could have continued processing characters after the end of the exten string. We could have added the previous character to the pattern matching tree incorrectly. (closes issue ASTERISK-18909) Reported by: Luke-Jr git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@347811 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	3240b06b10	Fix regression when using tcpenable=no and tlsenable=yes. The tlsenable settings are tucked away in main/tcptls.c, so I missed them when resolving ASTERISK-18837. This should resolve the test suite breakage of the sip tls tests. Review: https://reviewboard.asterisk.org/r/1615 Reviewed by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@347718 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	0e9e42e044	Mark channel running the h exten with the soft-hangup flag. When a bridge is broken, ast_bridge_call() might execute the h exten on the calling channel. However, that channel may not have been the channel that broke the bridge by hanging up. The channel executing the h exten must be in a hung up state so things like AGI run in the correct mode. * Make sure ast_bridge_call() marks the channel it is executing the h exten on as hung up. (The AST_SOFTHANGUP_APPUNLOAD flag is used so as to match the pbx.c main dialplan execution loop when it executes the h exten.) (closes issue ASTERISK-18811) Reported by: David Hajek Patches: jira_asterisk_18811_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: David Hajek, rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@347595 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago

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