asterisk

Commit Graph

Author	SHA1	Message	Date
Matthew Jordan	8235a16c2b	Fix remotely exploitable stack overrun in Milliwatt Milliwatt is vulnerable to a remotely exploitable stack overrun when using the 'o' option. This occurs due to the milliwatt_generate function not accounting for AST_FRIENDLY_OFFSET when calculating the maximum number of samples it can put in the output buffer. This patch resolves this issue by taking into account AST_FRIENDLY_OFFSET when determining the maximum number of samples allowed. Note that at no point is remote code execution possible. The data that is written into the buffer is the pre-defined Milliwatt data, and not custom data. (issue ASTERISK-19541) Reported by: Russell Bryant Tested by: Matt Jordan Patches: milliwatt_stack_overrun.rev1.txt by Russell Bryant (license 6283) Note that this patch was written by Russell, even though Matt uploaded it git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@359615 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Leif Madsen	14fdd09490	Update svn:externals to use menuselect from 1.4.42 and not later. This change is required because when making security releases, if you pull from menuselect/trunk you'll get changes meant for later versions of Asterisk. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@347657 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	656d2e043a	Default to nat=yes; warn when nat in general and peer differ It is possible to enumerate SIP usernames when the general and user/peer nat settings differ in whether to respond to the port a request is sent from or the port listed for responses in the Via header. In 1.4 and 1.6.2, this would mean if one setting was nat=yes or nat=route and the other was either nat=no or nat=never. In 1.8 and 10, this would mean when one was nat=force_rport and the other was nat=no. In order to address this problem, it was decided to switch the default behavior to nat=yes/force_rport as it is the most commonly used option and to strongly discourage setting nat per-peer/user when at all possible. For more discussion of the issue, please see: http://lists.digium.com/pipermail/asterisk-dev/2011-November/052191.html (closes issue ASTERISK-18862) Review: https://reviewboard.asterisk.org/r/1591/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@345776 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	36b9cc1d49	Add .o files to svn:ignore property, since it's only ignored if locally configured to do so. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@327251 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	1bca16589d	Don't leak SIP username information git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@325275 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	9bc8ac021e	Addresses AST-2011-010, remote crash in IAX2 driver Thanks to twilson for identifying the issue and providing the patches. AST-2011-010 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@324627 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	dcddfdc148	Fix DYNAMIC_FEATURES DYNAMIC_FEATURES were broken by a recent DTMF change. This patch makes sure that dynamic features are also checked when deciding whether or not to pass DTMF through or store it for interpreting. (closes issue ASTERISK-17914) Reported by: vrban git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@323732 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	224b82262f	Resolve a segfault/bus error when we try to map memory that falls on a page boundary. The fix for ASTERISK-15359 was incorrect in that it added 1 to the length of the mmap'd region. The problem with this is that reading/writing to that extra byte outside of the bounds of the underlying fd causes a bus error. The real issue is that we are working with both a FILE * and the raw fd underneath it and not synchronizing between them. The code that was removed in ASTERISK-15359 was correct, but we weren't flushing the FILE * before mapping the fd. Looking at the manager code in 1.4 reveals that the FILE * in 'struct mansession' is never used except to create a temporary file that we immediately fdopen. This means we just need to write a 0 byte to the fd and everything will just work. The other branches require a call to fflush() which, while not a guaranteed fix, should reduce the likelihood of a crash. This all makes sense in my head. (closes issue ASTERISK-16460) Reported by: Ravelomanantsoa Hoby (hoby) Patches: issue17747_1.4_svn_markII.patch uploaded by Sean Bright (license #5060) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@323559 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	847e704100	unlock pvt when we drop voice frames received in early media when in t.38 mode git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@322698 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	f6d89f46b1	whitespace git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@322657 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	9ed0888605	don't drop any voice frames when checking for T.38 during early media (closes issue ASTERISK-17705) Review: https://reviewboard.asterisk.org/r/1186/ patch by oej reported by oej git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@322646 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	b92393f6c6	Solaris compatibility fixes git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@320393 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	82506ffa01	The meetme CLI command completion leaves conferences mutex locked. When issuing a meetme kick CLI command and an invalid (non-existent) conference number is specified, pressing Tab leaves the conferences mutex locked and, therefore, all conferences deadlock. Add missing unlock. (closes issue #19336) Reported by: zvision Patches: app_meetme.diff uploaded by zvision (license 798) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@320235 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	a93fce4278	chan_sip: Destroy variables on a sip_pvt before copying vars from the sip_peer. (closes issue #19202) Reported by: wdoekes Patches: issue19202_destroy_challenged_invite_chanvars.patch uploaded by wdoekes (license 717) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@320055 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	a5819437ae	Make sure everyone gets an unhold when a transfer succeeds Some phones, like the Snom phones, send a hold to the transfer target after before sending the REFER. We need to make sure that we unhold the parties that are being connected after the masquerade. If Local channels with the /nm option are used when dialing the parties, hold music would still be playing on the transfer target, even after being connected with the transferee. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@319652 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	5a1bd8ffbc	Fix app_dial ring groups Revert part of r315643. We need to remove the datastore here as well. The code in bridging code will catch anything that app_dial might miss. (closes issue #19311) Reported by: mspuhler Patches: issue_19311_no_answer.diff uploaded by elguero (license 37) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@319527 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	2f024e4544	Merged revisions 318671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 * The applicable fixes for v1.4 are the SIP deadlock and the in progress masquerade check for multiple parties trying to pickup the same call. issue18654_v1.4.patch uploaded by rmudgett (license 664) * Backported to v1.6.2. issue18654_v1.6.2.patch uploaded by rmudgett (license 664) ........ r318671 \| alecdavis \| 2011-05-13 10:52:08 +1200 (Fri, 13 May 2011) \| 30 lines Fix directed group pickup feature code *8 with pickupsounds enabled Since 1.6.2, the new pickupsound and pickupfailsound in features.conf cause many issues. 1). chan_sip:handle_request_invite() shouldn't be playing out the fail/success audio, as it has 'netlock' locked. 2). dialplan applications for directed_pickups shouldn't beep. 3). feature code for directed pickup should beep on success/failure if configured. Created a sip_pickup() thread to handle the pickup and playout the audio, spawned from handle_request_invite. Moved app_directed:pickup_do() to features:ast_do_pickup(). Functions below, all now use the new ast_do_pickup() app_directed_pickup.c: pickup_by_channel() pickup_by_exten() pickup_by_mark() pickup_by_part() features.c: ast_pickup_call() (closes issue #18654) Reported by: Docent Patches: ast_do_pickup_1.8_trunk.diff.txt uploaded by alecdavis (license 585) Tested by: lmadsen, francesco_r, amilcar, isis242, alecdavis, irroot, rymkus, loloski, rmudgett Review: https://reviewboard.asterisk.org/r/1185/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@318734 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	bdf971df60	Regression after r297603 (Improve handling of REGISTER requests with multiple contact headers.) Uninitialized variable. (issue #18640) (closes issue #18785) Reported by: pnlarsson Patches: issue18785_enegaard.patch uploaded by enegaard (license 1197) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@317719 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	c322f12666	Re-fix queue round-robin This part of the change for r315596 was incorrect. No bridge occurs when doing a roundrobin dial and no one answers, so this code shouldn't have been removed. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@317574 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	b2ad693b90	chan_sip: fix broken realtime peer count, fix memory leak This patch addresses two bugs in chan_sip: 1) The count of realtime peers and users was off. The increment checked the value of the caching option, while the decrement did not. 2) Add a missing regfree() for a regex. (closes issue #19108) Reported by: vrban Patches: missing_regfree.patch uploaded by vrban (license 756) sip_object_counter.patch uploaded by vrban (license 756) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@317211 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	73e8f5d57e	Disable console colourization inside safe_asterisk checks. (closes issue #19213) Reported by: lefoyer Patches: issue19213_strip_color_in_safe_asterisk-svn.patch uploaded by wdoekes (license 717) Tested by: wdoekes, lefoyer git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@317102 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	a864430b0b	If sox fails when processing a voicemail, don't delete the original file. (closes issue #18111) Reported by: sysreq Patches: issue18111_trunk.patch uploaded by seanbright (license 71) Tested by: seanbright git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@316707 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	806fef4ef5	Fixes chan_local crashs in local_fixup() Thanks OEJ for tracking down the issue and submitting the patch. (closes issue #19053) Reported by: oej Tested by: oej Review: https://reviewboard.asterisk.org/r/1158/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@316328 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	4731bae48d	Breakage from slightly before the outage; would have fixed sooner but for the outage. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@316089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	268318bf88	Backport the use of curl from 1.6.2 to make the 1.4 target work on Bamboo. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@316006 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	17ca06cdb8	Partial revert of r315671 which removed a logging statement and not a manager event. Reported by ibercom in #asterisk-bugs. (issue #16033) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@315989 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	acfd6d4bad	Fix our compliance with RFC 3261 section 18.2.2. This change optimizes the free_via() function and removes some redundant null checking. It also fixes compliance with RFC 3261 section 18.2.2 by always using the port specified in the Via header for routing responses (even when maddr is not set). Also the htons() function is now used when setting the port. Additional documentation comments have been added in various places to make the logic in the code clearer. (closes issue #18951) Reported by: jmls Patches: issue18951_set_proper_port_from_via.patch uploaded by wdoekes (license 717) (modified) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@315891 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	a851a4834f	Make sure unregistering a peer unlinks it from the peer container Instead of mostly copying the code from expire_register, just use the function that "does the right thing". (closes issue #16033) Reported by: kkm Patches: 016033-tilgman-fixed-refcount.diff uploaded by kkm (license 888) Tested by: kkm, tilghman, twilson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@315671 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	77acf5b626	Allow transfer loops without allowing forwarding loops We try to avoid the situation where two phones may be forwarded to each other causing an infinite loop by storing each dialed interface in a channel datastore and checking the list before dialing out. This works, but currently breaks situations like A calls B, A transfers B to C, B transfers C to A, and A transfers C to B. Since human interaction is happening here and not an automated forwarding loop, it should be allowed. This patch removes the dialed_interfaces datastore when a call is bridged (a suggestion from the brilliant mmichelson). If a call is being bridged, it should be safe to assume that we aren't stuck in a loop. Since we are now handling this is the bridge code, the previous attempts at handling it in app_dial and app_queue are removed. Review: https://reviewboard.asterisk.org/r/1195/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@315596 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	dac18ad759	Fix the bounds-checking code. The code that set the bit within the select bitfield was correct, but the bounds-checking code was not. The change to that line uses the new _bitsize macro for clarity. Also, FD_ZERO macro did not zero-out anything but the first word of the bitfield, so this could have caused problems with modules using that macro with the expanded bitfield. (closes issue #18773) Reported by: jamicque Patches: 20110423__issue18773.diff.txt uploaded by tilghman (license 14) Tested by: chris-mac git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@315501 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	112e8ef678	Be more flexible with unknown chunks in wav files. This patch makes format_wav ignore unknown chunks instead of erroring out on them. (closes issue #18306) Reported by: jhirsch Patches: wav_skip_unknown_blocks.diff uploaded by jhirsch (license 1156) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@315257 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	94fa98557d	Reverted part of r314607, as it can introduce a regression. Specifically, the security check for the "system" privilege was removed. If a user had the "call" privilege but not the "system" privilege, they would loose the ability to execute the system app and dialplan functions that run commands in a shell. This branch never used the "system" privilege for that purpose and did not need to be patched. AST-2011-006 (related to issue 0018787) Reported by: kobaz git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@315147 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	b3d58b6260	chan_local:check_bridge() misplaced misplaced ast_mutex_unlock if !p->chan->_bridge->_softhangup path isn't followed, brigde remains locked. (closes issue #19176) Reported by: alecdavis Patches: bug19176.diff.txt uploaded by alecdavis (license 585) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@315051 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	1cfde1088c	Prevent the login thread and the app threads from using the asterisk channel at the same time. ABE-2756 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@314908 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	5a3d415ea4	Initialize buffers in getvar and getvarfull. Initialize the buffers used to hold the result from GET VARIABLE or GET VARIABLE FULL. The bug report shows func_read returning garbage in the result. It assumed that the buffer passed in was initialized, like many other functions do. In the more common code path (through the dialplan), it is initialized, so just initialize it here too. (closes issue #19050) Reported by: johnz git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@314822 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	9bd88702b7	Added limits to the number of unauthenticated sessions TCP based protocols are allowed to have open simultaneously. Also added timeouts for unauthenticated sessions where it made sense to do so. Unrelated, the manager interface now properly checks if the user has the "system" privilege before executing shell commands via the Originate action. AST-2011-005 AST-2011-006 (closes issue #18787) Reported by: kobaz (related to issue #18996) Reported by: tzafrir git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@314607 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	05ea303ca4	app_voicemail: Fix ODBC Storage compile regression caused by me, from mantis bug #19032 / commit r312070 (closes issue #19142) Reported by: vrban Patches: app_voicemail_fix_for_312070.patch uploaded by vrban (license 756) Tested by: vrban, alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@314300 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	fa67a98ccc	Asterisk does not hangup a channel after endpoint hangs up. If the call that the dialplan started an AGI script for is hungup while the AGI script is in the middle of a command then the AGI script is not notified of the hangup. There are many AGI Exec commands that this can happen with. The reported applications have been: Background, Wait, Read, and Dial. Also the AGI Get Data command. * Don't wait on the Asterisk channel after it has hung up. The channel is likely to never need servicing again. * Restored the AGI script's ability to return the AGI_RESULT_HANGUP value in run_agi(). It previously only could return AGI_RESULT_SUCCESS or AGI_RESULT_FAILURE after the DeadAGI and AGI applications were merged. (closes issue #17954) Reported by: mn3250 Patches: issue17954_v1.8.patch uploaded by rmudgett (license 664) issue17954_v1.6.2.patch uploaded by rmudgett (license 664) issue17954_v1.4.patch uploaded by rmudgett (license 664) Tested by: rmudgett JIRA SWP-2171 (closes issue #18492) Reported by: devmod Tested by: rmudgett JIRA SWP-2761 (closes issue #18935) Reported by: nvitaly Tested by: astmiv, rmudgett JIRA SWP-3216 (closes issue #17393) Reported by: siby Tested by: rmudgett JIRA SWP-2727 Review: https://reviewboard.asterisk.org/r/1165/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@313545 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	453275054e	Fix detection of OpenSSL 1.0 (closes issue #19093) Reported by: tzafrir Patches: detect_openssl_10.diff uploaded by tzafrir (license 46) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@313277 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	afbfce2971	Stuck channel using FEATD_MF if caller hangs up at the right time. The cause was actually a caller hanging up just at the end of the Feature Group D DTMF tones that setup the call. The reason for this is a "guard timer" that's implemented using ast_safe_sleep(100). If the caller happens to hang up AFTER the final tone of the DTMF string but BEFORE the end of that ast_safe_sleep(), then ast_safe_sleep() will return non-zero. This causes the code to bounce to the end of ss_thread(), but it does NOT tear down the call properly. This should be a rare occurrence because the caller has to hang up at EXACTLY the right time. Nonetheless, it was happening quite regularly on the reporter's system. It's not easily reproducible, unless you purposely increase the guard-time to 2000 or more. Once you do that, you can reproduce it every time by watching the DTMF debug and hanging up just as it ends. Simply add an ast_hangup() before goto quit. (closes issue #15671) Reported by: jcromes Patches: issue15671.patch uploaded by pabelanger (license 224) Tested by: jcromes git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@313188 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	059e1a2dec	Limit the number of unauthenticated manager sessions and also limit the time they have to authenticate. AST-2011-005 (closes issue #18996) Reported by: tzafrir Tested by: mnicholson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@312761 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	cd24c868cb	Issues with ISDN calls changing B channels during call negotiations. The handling of the PROCEEDING message was not using the correct call structure if the B channel was changed. (The same for PROGRESS.) The call was also not hungup if the new B channel is not provisioned or is busy. * Made all call connection messages (SETUP_ACKNOWLEDGE, PROCEEDING, PROGRESS, ALERTING, CONNECT, CONNECT_ACKNOWLEDGE) ensure that they are using the correct structure and B channel. If there is any problem with the operations then the call is now hungup with an appropriate cause code. * Made miscellaneous messages (INFORMATION, FACILITY, NOTIFY) find the correct structure by looking for the call and not using the channel ID. NOTIFY is an exception with versions of libpri before v1.4.11 because a call pointer is not available for Asterisk to use. * Made all hangup messages (DISCONNECT, RELEASE, RELEASE_COMPLETE) find the correct structure by looking for the call and not using the channel ID. (closes issue #18313) Reported by: destiny6628 Tested by: rmudgett JIRA SWP-2620 (closes issue #18231) Reported by: destiny6628 Tested by: rmudgett JIRA SWP-2924 (closes issue #18488) Reported by: jpokorny JIRA SWP-2929 JIRA AST-437 (The issues fixed here are most likely causing this JIRA issue.) JIRA DAHDI-406 JIRA LIBPRI-33 (Stuck resetting flag likely fixed) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@312573 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	e7597d55df	app_voicemail: leave_vociemail doesn't use last_message_index to store next message trivial change to bring inline with 1.6.2 1.8 and trunk. The symptom was if msg0000 was missing, and the last was msg0004, the next msgnum would be msg0000 when it should have been msg0005 (issue #18998) Reported by: tootai Patches: bug18998.diff2.txt uploaded by alecdavis (license 585) Tested by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@312290 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	f6cd305277	Found some leaking file descriptors while looking at ast_FD_SETSIZE dead code. (issue #18969) Reported by: oej Patches: 20110315__issue18969__14.diff.txt uploaded by tilghman (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@312285 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	4de07be466	voicemail: get real last_message_index and count_messages, ODBC resequence change last_message_index to read the max msgnum stored in the database change count_messages to actually count the number of messages. last_message_index change: This fixed overwriting of the last message if msgnum=0 was missing. Previously every incoming message would overwrite msgnum=1. count_messages change: allows us to detect when requencing is required in opneA_mailbox. resequence enabled for ODBC storage: Assists with fixing up corrupt databases with gaps, but only when a user actively opens there mailboxes. (closes issue #18692,#18582,#19032) Reported by: elguero Patches: based on odbc_resequence_mailbox2.1.diff uploaded by elguero (license 37) Tested by: elguero, nivek, alecdavis Review: https://reviewboard.asterisk.org/r/1153/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@312174 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	752c3ef591	app_voicemail:close_mailbox imap_storage doesn't use last_msg_index git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@312119 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	c875064714	app_voicemail: close_mailbox needs to respect additional messages while mailbox is open. close_mailbox leave gaps in message sequence if messages are deleted and new messages arrive during this time, this is because the shuffle down to slot 0, only shuffles the number of pre-existing messages when mailbox is opened, ignoring new arrivals. Fix: in close_mailbox re-evaluate number of messages before the shuffle, this then includes new arrivals. Happens on filebased or ODBC storage. (issues #19032,#18582,#18692,#18998) Reported by: alecdavis,tootai,afosorio Review: https://reviewboard.asterisk.org/r/1153/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@312070 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	c0e4bdd609	Don't hold the pvt lock while streaming a file. ABE-2756 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@311203 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	9dbd90f447	Don't dec the usecount of an eventqent then use it. ABE-2756 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@311201 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	5510659c1f	Remove the provisional keepalive scheduler entry's reference to the pvt when we remove the scheduler entry. ABE-2756 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@311199 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago

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