asterisk

Commit Graph

Author	SHA1	Message	Date
Scott Griepentrog	ceab8bf73c	rtp_engine: fix rtp payloads copy and improve argument names In function ast_rtp_instance_early _bridge_make_compatible the use of instance 0/1 as arguments doesn't clearly communicate a direction that the copying of payloads from the source channel to the destination channel will occur, making it more probable to have the arguments to ast_rtp_codecs_payloads_copy() put in the reverse order. This patch renames the arguments with _dst and _src suffixes and corrects the copy direction. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@402000 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	08f1768dc3	pbx.c: fix confused match caller id that deleted exten still in hash This fixes a bug where a zero length callerid match adjacent to a no match callerid extension entry would be deleted together, which then resulted in hashtable references to free'd memory. A third state of the matchcid value has been added to indicate match to any extension which allows enforcing comparison of matchcid on/off without errors. (closes issue AST-1235) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/2930/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@401959 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	1ee20a668c	Put clicompat-r2.patch back in We've figured out how to resolve the problems this was causing in 12/trunk, so this can go back in now. (issue ASTERISK-22467) Reported by: Corey Farrell Patches: clicompat-r2.patch uploaded by coreyfarrell (license 5909) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@401914 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	eb6293eeaa	revert clicompat-r2.patch from r401704 Patch caused the following build errors against testsuite https://bamboo.asterisk.org/bamboo/browse/AST-ATRUNKBUILD4-244 (issue ASTERISK-22467) Reported by: Corey Farrell git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@401895 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	f313081314	utils: Fix memory leaks and missed unregistration of CLI commands on shutdown Final set of patches in a series of memory leak/cleanup patches by Corey Farrell (closes issue ASTERISK-22467) Reported by: Corey Farrell Patches: main-utils-1.8.patch uploaded by coreyfarrell (license 5909) main-utils-11.patch uploaded by coreyfarrell (license 5909) main-utils-12up.patch uploaded by coreyfarrell (license 5909) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@401829 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	240f7e9e73	test_linkedlists: Fix memory leak (issue ASTERISK-22467) Reported by: Corey Farrell Patches: test_linkedlists-1.8.patch uploaded by coreyfarrell (license 5909) test_linkedlists-11up.patch uploaded by coreyfarrell (license 5909) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@401790 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	7d03892ea8	jitterbuf: Fix memory leak on jitter buffer reset (issue ASTERISK-22467) Reported by: Corey Farrell Patches: jitterbuf-jb_reset-leak-1.8.patch jitterbuf-jb_reset-leak-11up.patch git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@401786 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	49716f1f8c	astobj2: Unregister debug CLI commands at exit (issue ASTERISK-22467) Reported by: Corey Farrell Patches: astobj2-clean-debug-cli-1.8-11.patch uploaded by coreyfarrell (license 5909) astobj2-clean-debug-cli-12up.patch uploaded by coreyfarrell (license 5909) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@401781 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	76428d4e12	app_voicemail: Memory Leaks against tests (issue ASTERISK-22467) Reported by: Corey Farrell Patches: app_voicemail-1.8.patch uploaded by coreyfarrell (license 5909) app_voicemail-11up.patch uploaded by coreyfarrell (license 5909) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@401743 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	d74eabf012	memory leaks: Memory leak cleanup patch by Corey Farrell (second set) Also covers ast_app_parse_timelen-fail-zero-length.patch, but the patch was replaced with one of my own. (issue ASTERISK-22467) Reported by: Corey Farrell Patches: chan_dahdi-cleanup_push.patch uploaded by coreyfarrell (license 5909) clicompat-r2.patch uploaded by coreyfarrell (license 5909) codecs-ilbc-doCPLC.patch uploaded by coreyfarrell (license 5909) data-cleanup-test-registration.patch uploaded by coreyfarrell (license 5909) main-asterisk-kill-listener.patch uploaded by coreyfarrell (license 5909) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@401704 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	9ec4719027	memory leaks: Memory leak cleanup patch by Corey Farrell (first set) (issue ASTERSIK-22467) Reported by: Corey Farrell Patches: chan_sip-parse_contact_header_test-free-contacts.patch uploaded by coreyfarrell (license 5909) cli-filename-completion-leak.patch uploaded by coreyfarrell (license 5909) func_math.patch uploaded by corefarrell (license 5909) main-test-cleanup.patch uploaded by coreyfarrell (license 5909) test_dlinklists.patch uploaded by coreyfarrell (license 5909) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@401660 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	8c07e036b2	res_rtp_asterisk: Address jittery DTMF events in RTP streams (closes issue ASTERISK-21170) Reported by: NITESH BANSAL Patches: dtmf-timestamp.patch uploaded by NITESH BANSAL (license 6418) Review: https://reviewboard.asterisk.org/r/2938/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@401619 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	bd2eb4ee0e	cdr_adaptive_odbc: Also apply a filter when the CDR value is empty. Extra CDR records are written if a filtered CDR value is empty because the filter is not checked. (closes issue ASTERISK-22272) Reported by: Jordi Llull Chavarria git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@401577 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	3a5f6aeeb9	chan_mgcp: Properly handle malformed media lines This corrects a situation in which a media line was not parsed properly and resulted in a crash. (closes issue ASTERISK-21190) Reported by: adomjan Patches: chan_mgcp.c-sscnaf_fix uploaded by adomjan (License 5448) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@401537 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	9d5e2cb9cf	chan_sip: Fix an issue where an incompatible audio format may be added to SDP. If preferred codecs included any non-audio format the code would mistakenly add the audio format, even if it was not a joint capability with the remote side. (closes issue ASTERISK-21131) Reported by: nbougues Patches: patch_unsupported_codec_1.8.patch uploaded by nbougues (license 6470) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@401497 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	58542a7c27	res_rtp_asterisk: Fix crash when RTCP is not available during SSRC change In r400089, a patch was put in to correct erroneous RTCP statistic resets. Unfortunately, ast_rtp_read can be called on an RTP instance that does not have RTCP information. This patch prevents that crash by only resetting the statistics if we do actually have an RTCP instance. (issue AST-1174) (closes issue ASTERISK-22667) Reported by: John Bigelow git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@401445 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	623d0656cc	chan_dahdi: Fix unable to get index warning when transferring an analog call. Transferring an analog call using flashhooks generated an unable to get index WARNING message when the transfer is completed. * Removed unnecessary analog subchannel shell games when transferring a call using flashhooks. Thanks to Tzafrir Cohen for mentioning this in a comment on issue ASTERISK-22720. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@401378 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	f5c2aa7a62	Segfault in LIBEDIT_INTERNAL after tgetstr(), when libncurses5-dev isn't installed Include the appropriate declarations when not using termcap, but term+curses and [n]curses do not exist. (closes issue ASTERISK-22351) Reported by: A. Iglesias Patches: issueA22351_libedit_internal_without_ncurses_dev.patch uploaded by wdoekes (license 5674) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@401325 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Walter Doekes	40e42ff3a4	Properly copy/remove the device state cache flag over a masquerade. In r378303 the AST_FLAG_DISABLE_DEVSTATE_CACHE flag was added that tells the devstate system to not cache states for non-real devices. However, when optimizing away channels (ast_do_masquerade), that flag wasn't copied. In my case, using Local devices as queue members created a situation where the endpoint was considered in use, but the state change of the device being available again was ignored (not cached). The endpoint channel was optimized into the (previously) Local channel, but kept the do-not-cache flag. The end result being that the queue member apparently stayed in use forever. (closes issue ASTERISK-22718) Reported by: Walter Doekes Review: https://reviewboard.asterisk.org/r/2925/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@401178 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	cd1d5ea15e	Reduce log level of a non-pubsub error message Drop an error log message to debug level 1 since distributed device state functions correctly when receiving this message and it spams the logs. (closes issue ASTERISK-22410) Reported by: abelbeck Patches: asterisk-1.8-res_jabber-log-nonpubsub-error-to-debug.patch uploaded by abelbeck (License 5903) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@401119 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Walter Doekes	f8a65f1c7f	Don't check all realtime queues when doing "queue show some_queue". When using realtime queues, queues have to be fetched from the database every now and then to see if any info has been changed or to see if the queue has been removed. When fetching info for an individual queue, the pruning of other queues is unnecessarily costly. Review: https://reviewboard.asterisk.org/r/2907/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@401049 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	3f0b77e77e	Prevent chan_sip from sending duplicate BYEs. When a 200 OK for an initial INVITE is received, we were doing the right thing by ACKing and sending an immediate BYE. However, we also were doing the wrong thing and queuing an answer frame, thus causing the call to be answered. This would cause the call to be hung up by the channel thread, thus resulting in a second BYE being sent out. In this fix, I also have set the hangupcause to be correct since the initial BYE being sent by Asterisk had an unknown hangup cause. I have changed to using "Bearer capabilty not available" since the call was hung up due to an SDP offer/answer error. (closes issue ASTERISK-22621) reported by Kinsey Moore git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@400970 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	9735b75ed2	chan_dahdi: Reflect the set software gain in the CLI "dahdi show channel" output. * Remember the swgain setting from CLI "dahdi set swgain" command so the CLI "dahdi show channel" output will reflect the current setting. * Updated CLI "dahdi set hwgain" and "dahdi set swgain" documentation. (issue ASTERISK-22429) Reported by: Jaco Kroon Patches: jira_asterisk_22429_v1.8_v2.patch (license #5621) patch uploaded by rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@400907 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	27f6318afb	chan_sip: Do not increment the SDP version between 183 and 200 responses. Bumping the SDP version number can cause interoperability problems since receivers of the responses will expect that a 200 SDP will be identical to a previous 183 SDP. (closes issue ASTERISK-21204) reported by NITESH BANSAL Patches: dont-increment-session-version-in-2xx-after-183.patch uploaded by NITESH BANSAL (License #6418) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@400906 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	02379cf262	Add warning when compiling with iODBC support When running configure, libiodbc2 development headers will fulfill the requirement for ODBC development headers, but will not function properly. This adds a warning when libiodbc2 development headers are detected instead of unixodbc development headers. (closes issue ASTERISK-22459) Reported by: Patrick Maille Tested by: Walter Doekes Patches: issueA22459_warn_when_using_iodbc.patch uploaded by Walter Doekes (License 5674) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@400767 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	2a74036b8a	Fix func_config list entry allocation The AST_CONFIG dialplan function defined in func_config.c allocates its config file list entries using ast_malloc. List entry allocations destined for use with Asterisk's linked list API must be ast_calloc()d or otherwise initialized so that list pointers are set to NULL. These uses of ast_malloc have been replaced by ast_calloc to prevent dereferencing of uninitialized pointer values when traversing the list. (closes issue ASTERISK-22483) Reported by: Brian Scott git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@400694 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	9fb45ae052	app_queue: Fix Queuelog EXITWITHKEY only logging two of four fields Commit r62462 added two extra fields for logging "the original position the caller entered the queue at, and the amount of time the caller was waiting in the queue." But when r75969 was merged from 1.4 into trunk (r75977), these two fields disappeared. Those two extra fields were not logged in 1.4 and when the patch was merged, those fields went away. Therefore, this is a regression and was caught by the reporter because he was reading the awesome "Asterisk: The Definitive Guide" book. (closes issue ASTERISK-22197) Reported by: Dalius M. Tested by: Dalius M. Patches: asterisk-22197-q-log-exitwithkey.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2901/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@400622 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	c2affa0f72	chan_sip: Don't ignore expires value in contact header if it lacks semicolon (closes issue ASTERISK-22574) Reported by: Filip Jenicek Patches: chan_sip_expires.patch uploaded by Filip Jenicek (license 6277) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@400469 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	4393ef861c	res_rtp_multicast: Ensure SSRC is set properly This fixes a bug where the SSRC field on multicast RTP can be stuck at 0 which can cause problems for endpoints trying to make sense of incoming streams. (closes issue ASTERISK-22567) Reported by: Simone Camporeale Patches: 22567_res_mulitcast_ssrc.patch uploaded by Simone Camporeale (License 6536) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@400393 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	ecc1bd0c34	Cast Integer Argument To Unsigned Char The member reg in the peercnt structure is an unsigned char and peercnt_modify() is expecting an unsigned char argument which gets assigned to peercnt->reg. This patch fixes that by casting the integer argument being passed to peercnt_modify to unsigned char. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@400314 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	6c77f74463	chan_sip: Allow Asterisk to retry after 403 on register This adds a global option in chan_sip to allow it to continue attempting registration if a 403 is received, clearing the cached nonce and treating it as a non-fatal response. Normally, this would cause registration attempts to that endpoint to stop. (closes issue ASTERISK-17138) Review: https://reviewboard.asterisk.org/r/2874/ Reported by: Rudi git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@400137 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	068e1928cb	res_rtp_asterisk: Correct erroneous lost packet information in RTCP reports RTCP's calculation of the number of lost packets in an RTP stream is based on that stream's sequence number count, the number of received packets, and how many packets we expect to receive. When the SSRC for an RTP stream changes, there can - and almost always will be - a large jump in the next packet's timestamp and sequence number. If we don't reset the number of received packets, sequence number count, and other metrics used by RTCP, the next RR/SR report will use the previous SSRC's values to calculate the lost packet count for the new SSRC - resulting in a very large number of lost packets. This patch modifies res_rtp_asterisk such that, if it detects a SSRC change, it will reset the various values used by the RTCP calculations. From the perspective of RTCP, this appears as a new media stream - which is what it is. Review: https://reviewboard.asterisk.org/r/2886/ (closes issue AST-1174) Reported by: Thomas Arimont git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@400089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	299ef1b3dc	Add check for openSUSE when detecting bfd library In ASTERISK-17842, some additional library checks were added to the configure script so that the bfd library could be found on CentOS and Fedora systems. As it turns out, openSUSE requires an additional library. This patch adds another check to the configure script for openSUSE that will add that library. Review: https://reviewboard.asterisk.org/r/2885/ (closes issue AST-1169) Reported by: Guenther Kelleter git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@400073 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	e51430e17e	chan_sip: Increase some scratch buffer sizes dealing with caller id. * Eliminated an unnecessary initialization in check_user_full(). (closes issue ASTERISK-22477) Reported by: Michael Shepelev git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@400013 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	cae74964d7	chan_sip: Reject calls on 200 OKs if no SDP has been received When Asterisk receives a 200 OK in response to an invite, that peer should have sent an SDP at some point by then. If the channel has never received an SDP, media won't have been set and the remote address won't be known. Endpoints in general should not be doing this. This patch makes it so that Asterisk will simply hang up a call if it sends a 200 OK at this point. So far this odd behavior for endpoints has only been observed in tests which involved manually created SIP transactions in SIPp. (closes issue ASTERISK-22424) Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/2827/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@399939 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	d371f21552	chan_dahdi: CLI "core stop gracefully" has needless delay for PRI and SS7. The PRI and SS7 link control threads are not stopped correctly when the chan_dahdi.so module is unloaded. The link control threads pri_dchannel() and ss7_linkset() are not awakened from a poll() to cancel the thread. * Added a SIGURG signal after requesting the thread cancel to break the link control thread poll() immediately. For SS7 it was slightly worse, the link poll() timeout would always be whatever was the last libss7 scheduled event time used. If no libss7 scheduled event was pending, the thread could run more often than necessary. * Set nextms to 60 seconds for the ss7_linkset() poll() if there is no other libss7 scheduled event. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@399818 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	bd46e6f482	chan_sip: Fix Realtime Peer Update Problem When Un-registering And Expires Header In 200ok 1st Issue When a realtime peer sends an un-REGISTER request, Asterisk un-registers the peer but the database table record still has regseconds and fullcontact for the peer. This results in calls attempting to be routed to the peer which is no longer registered. The expected behavior is to get busy/congested when attempting to call an un-registered peer through the dialplan. What was discovered is that we are clearing out the peer's registration in the database in parse_register_contact() when calling expire_register() but then upon returning from parse_register_contact(), update_peer() is run which stores back in the database table regseconds and fullcontact. 2nd Issue The reporter pointed out that the 200 ok being returned by Asterisk after un-registering a peer contains a Contact header with ;expires= and the Expires header is not set to 0. This is actually a regression. Tests were created for this second issue (ASTERISK-22548). The tests have been reviewed and a Ship It! was received on those tests. This patch does the following: * Do not ignore the Expires header value even when it is set to 0. The patch sets the pvt->expiry earlier on in the function so that it is set properly and used. * If pvt->expiry is 0, do not call update_peer since that means the peer has already been un-registered and there is no need to update the database record again since nothing has changed. (closes issue ASTERISK-22428) Reported by: Ben Smithurst Tested by: Ben Smithurst, Michael L. Young Patches: asterisk-22428-rt-peer-update-and-expires-header.diff by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2869/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@399794 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	de772e22b2	chan_iax2: Prevent some needless breaking of the native IAX2 bridge. * Clean up some twisted code in the iax2_bridge() loop. * Add AST_CONTROL_VIDUPDATE and AST_CONTROL_SRCCHANGE to a list of frames to prevent the native bridge loop from breaking. * Passing the AST_CONTROL_T38_PARAMETERS frame should also allow FAX over a native IAX2 bridge. (issue ABE-2912) Review: https://reviewboard.asterisk.org/r/2870/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@399697 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	2a2edececa	chan_sip: Make direct media reinvites for T38 put Asterisk in the media path Prior to this patch, Asterisk would incorrectly use the previous endpoint addresses in SDP in spite of providing its own port. T38 is never meant to be done through directmedia and Asterisk should always be in the media path for these streams. (closes issue ASTERISK-17273) Reported by: Kevin Stewart (closes issue ASTERISK-18706) Reported by: Jeremy Kister Review: https://reviewboard.asterisk.org/r/2853/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@399456 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	4626075eaf	Fix jitter buffer log file creation This adjusts '/'-to-'#' replacement to replace all instances of '/' instead of just the first to ensure that the jitter buffer log file gets the correct name as per Richard Kenner's suggestion. (closes issue ASTERISK-21036) Reported by: Richard Kenner git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@399402 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	dc23d0a19e	Update prep_tarball with new documentation files on the Asterisk wiki This will now pull both a command reference for the version being prepared, as well as an Admin Guide that applies to all versions of Asterisk. (issue ASTERISK-22439) Reported by: Olle Johansson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@399351 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	da09847884	Fix Segfault When Syntax Of A Line Under [applicationmap] Is Invalid When processing the lines under the [applicationmap] context in features.conf, a segfault occurs from attempting to process a line with an invalid syntax (basically missing most of the arguments). Example: [applicationmap] automon=6 This patch moves the checking for empty arguments to before they are accessed. * Also, checked the "todo" comment and removed it. Some applications do not require arguments. (closes issue ASTERISK-22416) Reported by: CGI.NET Tested by: CGI.NET Patches: asterisk-22416-check-syntax-first_v2.diff by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2803 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@399304 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	d847fec0cc	chan_iax2: Fix saving the wrong expiry time in astdb. When a new IAX2 client registers, the astdb database is updated with the value of minregexpire defined in iax.conf instead of using the expiry time that is provided by the client. The provided expiry time of the client is updated after inserting the astdb entry. As a consequence, restarting or reloading asterisk creates clients whose registration may expire before they reregister. The clients are therefore unavailable after minregexpire seconds until they reregister. * Move updating of the expiry time to before inserting into the astdb. (closes issue ASTERISK-22504) Reported by: Stefan Wachtler Patches: chan_iax2.c.patch (license #6533) patch uploaded by Stefan Wachtler git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@399158 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	d867a3f062	Don't write to /tmp/refs when REF_DEBUG is not defined. If MALLOC_DEBUG is enabled, then the debug destructor for the container is used, which would erroneously write to /tmp/refs. This patch only uses the debug destructor if ref_debug is used. (closes issue ASTERISK-22536) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@399098 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	c7b56581a9	Fix several crashes in MeetMeAdmin This change ensures that MeetMeAdmin commands requiring a user actually get a user and fixes another issue where an extra dereference could occur for a last-entered user being ejected if a user identifier was also provided. (closes issue ASTERISK-21907) Reported by: Alex Epshteyn Review: https://reviewboard.asterisk.org/r/2844/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@399033 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	89a99670fc	chan_sip: Revert r398835 due to failing tests involving originate (issue ASTERISK-22424) Reported by: Jonathan Rose git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@398977 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	be50098c82	res_musiconhold: Fix reference leaks caused when reloading with REF_DEBUG set Due to a faulty function for debugging reference decrementing, it was possible to reduce the refcount on the wrong object if two moh classes of the same name were in the moh class container. (closes issue ASTERISK-22252) Reported by: Walter Doekes Patches: 18_moh_debug_ref_patch.diff Uploaded by Jonathan Rose (license 6182) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@398937 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	75f0bf2d99	'queue add member' help text correction You are adding dial strings to the queue, not channels. An aribitrary string could be used, but you are typically referencing a channel. Correcting the command help text. (issue ASTERISK-22263) (closes issue ASTERISK-22263) Reported By: Rusty Newton git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@398884 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	14344fb803	Documentation fix - waitfordialtone is not boolean, it's time in milliseconds Changing text in chan_dahdi.conf sample to be accurate. (issue ASTERISK-22308) (closes issue ASTERISK-22308) Reported By: Malcolm Davenport git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@398880 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	79e1ec6fc1	chan_sip: Reject calls without prior SDP on 200 OK If we receive a 200 OK without SDP, we will now check to see if the remote address has been established for that channel's RTP session and if the to tag for that channel has changed from the most recent to tag in a response less than 200. If either a change has been made since the last to-tag was received or the remote address is unset, then we will drop the call. (closes issue ASTERISK-22424) Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/2827/diff/#index_header git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@398835 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago

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