2852 Commits (1.4)

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	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
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
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
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
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
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
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
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
Tilghman Lesher	eafdcbacd4	Initialize column size to 0 to deal with a potential UnixODBC bug on 64-bit systems. ... (closes issue #18295) Reported by: pruiz git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@310140 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	744b5c4bbb	Don't try to free statically allocated memory. ... Note: compiling after ./configure --enable-dev-mode will keep these kinds of mistakes from being committed. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@309947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	aaf50c4688	Bug fix for MixMonitor involving filenames with '.' not in the extension ... Closes issue #18391) Reported by: pabelanger Patches:       bugfix.patch uploaded by jrose (license 1225) Tested by: jrose git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@309856 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Ruggles	fd2d1c2139	fix small memory leak ... fix small memory leak caused by a string allocation that wasn't freed (closes issue #18907) Reported by: andy11 Patches: asterisk_trunk-app_externalivr-leak.patch uploaded by andy11 (license 1224) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@309355 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	788e66cf20	Fix regression that changed behavior of queues when ringing a queue member. ... This reverts r298596, which was to fix a highly bizarre and contrived issue with a queue member that called into his own queue being transferred back into his own queue. I couldn't reproduce that issue in any way. I think one of the other recent transfer fixes actually fixed this. (closes issue #18747) Reported by: vrban git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@308002 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	d6eabdc0dd	fix this line again ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@306965 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	fb475c8c7e	clean this up, sorry my brain is not really working ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@306964 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	8bd2bfb587	Backup file storing message duration is not used with IMAP_STORAGE, remove code. ... The message duration is stored in the body of the email when using IMAP_STORAGE, so nothing needs to happen with the backup file. (closes issue #18718) Reported by: kerframil git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@306960 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	e9dcd4a9e2	make this safer and fully correct, pointed out by Steve Davis ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@306864 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	9fe4bc60b5	Minor AST_FRAME_TEXT related issues. ... * Include the null terminator in the buffer length. When the frame is queued it is copied. If the null terminator is not part of the frame buffer length, the receiver could see garbage appended onto it. * Add channel lock protection with ast_sendtext(). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@305888 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	e03c6ed738	Prevent a crash when dialing a technology with no destination (ex: Dial(SIP/)) ... chan_iax2 and other channel drivers already had code to prevent this. The attempt that app_dial was making to prevent it was not correct, so I fixed that. (closes issue #18371) Reported by: gbour Patches: 18371.patch uploaded by gbour (license 1162) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@305252 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	3c5fd20e90	Fix compilation when ODBC_STORAGE is defined. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@304952 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Sean Bright	1fb86f03c8	Backport MeetMe related reference leaks fixes from 1.6.2/1.8/trunk. ... I had forgotten that MeetMe in 1.4 also used astobj2, so backport the fixes where appropriate. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@304820 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	b30fe5496f	Fix voicemail sequencing for file based storage. ... A previous change was made to account for when the number of voicemail messages exceeds the max limit to be handled properly, but it caused gaps in the messages to not be properly handled. This has now been resolved. In later non 1.4 branches, it appears that resequencing wasn't even occurring due from what appears and accidental code removal. (closes issue #18498) Reported by: JJCinAZ Patches: bug18498v2.patch uploaded by jpeeler (license 325) (closes issue #18486) Reported by: bluefox Patches: bug18486.patch uploaded by jpeeler (license 325) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@303676 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	58a666ac5a	Fix channel redirect out of MeetMe() and other issues with channel softhangup. ... Mantis issue #18585 reports that a channel redirect out of MeetMe() stopped working properly. This issue includes a patch that resolves the issue by removing a call to ast_check_hangup() from app_meetme.c. I left that in my patch, as it doesn't need to be there. However, the rest of the patch fixes this problem with or without the change to app_meetme. The key difference between what happens before and after this patch is the effect of the END_OF_Q control frame. After END_OF_Q is hit in ast_read(), ast_read() will return NULL. With the ast_check_hangup() removed, app_meetme sees this which causes it to exit as intended. Checking ast_check_hangup() caused app_meetme to exit earlier in the process, and the target of the redirect saw the condition where ast_read() returned NULL. Removing ast_check_hangup() works around the issue in app_meetme, but doesn't solve the issue if another application did the same thing. There are also other edge cases where if an application finishes at the same time that a redirect happens, the target of the redirect will think that the channel hung up. So, I made some changes in pbx.c to resolve it at a deeper level. There are already places that unset the SOFTHANGUP_ASYNCGOTO flag in an attempt to abort the hangup process. My patch extends this to remove the END_OF_Q frame from the channel's read queue, making the "abort hangup" more complete. This same technique was used in every place where a softhangup flag was cleared. (closes issue #18585) Reported by: oej Tested by: oej, wedhorn, russell Review: https://reviewboard.asterisk.org/r/1082/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@303546 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	2c54f15171	Add new queue strategy to preserve behavior for when queue members moved to ao2. ... Add queue strategy called "rrordered" to mimic old behavior from when queue members were stored in a linked list. ABE-2707 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@303007 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	9f2050cc3d	Option L() is milliseconds, not seconds. ... Change the verbose output of option L() to say milliseconds and not seconds as the value is in milliseconds. (closes issue #18264) Reported by: jacco Patches: app_dial_patch.txt uploaded by lmadsen (license 10) Tested by: lmadsen, jacco git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@302916 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	25181e5bdb	change variable name to the same used in later branches ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@300924 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	93864bd03d	Ensure good bye prompt in voicemail is played at the correct time. ... Specifically in the case of timing out but not leaving voicemail nothing should be heard. And when leaving voicemail it should be heard. ABE-2647 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@300918 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tzafrir Cohen	24118f5839	Typos: recieved => received ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@299002 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	ccfbb0a30a	After recording only silence for a voicemail prepending, restore backup files. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@298683 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	71115232da	Fix improper hangup when doing an attended transfer to queue. ... Had to indicate ringing in wait_for_answer so the attended transfer code would not try and hang up the local channel it created, which would kill the call. ABE-2624 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@298596 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	4369b3c4ae	Don't create a Local channel if the target extension does not exist. ... (closes issue #18126) Reported by: junky Patches: followme.diff uploaded by junky (license 177) (partially restructured by me to avoid a possible memory leak) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@297689 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	92ea2179e5	Add "DAHDI" to a couple of app_meetme error messages. ... This is in response to some questions on IRC. To the user, there was nothing that made it obvious that this error had anything to do with DAHDI not being loaded. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@297228 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	25c2ccaac3	Properly restore backup information file when hanging up during message prepending. ... ABE-2654 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@296868 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	72d3cd8b47	Handle failures building translation paths more effectively. ... The problem scenario occurred on a heavily loaded system that was using the codec_dahdi module and exceeded the hardware transcoding capacity. The failure mode at that point was not good. The report came in to us as an Asterisk lock-up. The "core show locks" shows a ton of threads locked up (but no obvious deadlock). Upon deeper investigation, when the system is in this state, the CPU was maxed out. The CPU was being consumed by the Asterisk logger spewing messages on every audio frame for calls set up after transcoder capacity was reached. The purpose of this patch is to make Asterisk handle failures to create a translation path in a more graceful manner. If we can't translate, then the call just needs to be dropped, as it's not going to work. These are the changes: 1) In set_format() of channel.c (which is called by set_read_format() and set_write_format()), it was ignoring if ast_translator_build_path() failed and returned NULL. It now pays attention to that case and returns a result reflecting failure. With this change in place, the bridging code will immediately detect a failure and end the bridge instead of proceeding to try to bridge frames that can't be translated and making channel drivers freak out by sending them frames in a format they weren't expecting. 2) In ast_indicate_data() of channel.c, failure of ast_playtones_start() was ignored. It is now reflected in the return value of the function. This didn't turn out to have any affect on the bug, but seemed like a good change to leave in. 3) In app_dial(), when only sending a call to a single endpoint, it will attempt to do some bridging of its own of early audio. It uses make_compatible() when it's going to do this. However, it ignored failure from make compatible. So, even with the fix from #1, if there was early audio going through app_dial, there would still be a period of invalid frames passing through. After detecting failure here, Dial() exits. ABE-2658 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@296000 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	d32582d42f	The channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call. ... To recreate the problem: 1) Party A calls Party B 2) Invoke CLI "channel redirect" command to redirect channel call leg associated with A. 3) All associated channels are hung up. Note that if the CLI command were done on the channel call leg associated with B it works. This regression was a result of the fix for issue #16946 (https://reviewboard.asterisk.org/r/740/). The regression affects all features that use an async goto to execute the dialplan because of an external event: Channel redirect, AMI redirect, SIP REFER, and FAX detection. The struct ast_channel._softhangup code is a mess. The variable is used for several purposes that do not necessarily result in the call being hung up. I have added doxygen comments to describe how the various _softhangup bits are used. I have corrected all the places where the variable was tested in a non-bit oriented manner. The primary fix is the new AST_CONTROL_END_OF_Q frame. It acts as a weak hangup request so the soft hangup requests that do not normally result in a hangup do not hangup. JIRA SWP-2470 JIRA SWP-2489 (closes issue #18171) Reported by: SantaFox (closes issue #18185) Reported by: kwemheuer (closes issue #18211) Reported by: zahir_koradia (closes issue #18230) Reported by: vmarrone (closes issue #18299) Reported by: mbrevda (closes issue #18322) Reported by: nerbos Review: https://reviewboard.asterisk.org/r/1013/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@295790 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	3fc98d5530	Ensure original message duration is preserved when prepending a message. ... It seems the fix to issue 17103 was a little overzealous and removed the code that backed up the textfile containing the original message duration. This code has now been restored. (related to issue #17103) ABE-2654 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@295200 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	fe1ca1a8da	Fix regression causing abort in voicemail after opening a mailbox with no mesgs. ... In order to be more safe, some error handling code was changed to respect more error conditions including the potential memory allocation failure for deleted and heard message tracking introduced in 293004. However, last_message_index returns -1 for zero messages (perhaps as expected) and was triggering the stricter error checking. Because last_message_index is only called directly in one place, just return 0 from open_mailbox (for file based storage) when no messages are detected unless a real error has occurred. (closes issue #18240) Reported by: leobrown Patches: bug18240.1-6-2.diff.txt uploaded by alecdavis (license 585) Tested by: pabelanger git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@294903 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	969aac85dd	Fix inprocess_container in voicemail to correctly restrict max messages. ... The comparison function logic was off, so the number of sessions for a given mailbox were not being incremented properly. This problem caused the maximum number of messages per folder to not be respected when simultaneously leaving multiple voicemails just below the threshold. These problems should be fixed by the above, but just in case: Fixed resequence_mailbox to rely on the actual number of detected number of files in a directory rather than just assuming only 10 messages more than the maximum had been left. Also if more messages than the maximum are deleted they are actually removed now. The second purpose of this commit should have been separated out probably, but is related to the above. Again, if the number of messages in a given voicemail folder exceeds the maximum set limit make sure to allocate enough space for the deleted and heard index tracking array. A few random fixes: There was a forgotten decrement of the inprocess count in imap_store_file. When using IMAP storage, do not look in the directory where file based storage messages may still reside and influence the message count. Ensure to use only the first format in sendmail. ABE-2516 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@293004 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	c560e9994b	Record priv-recordintro as sln, not gsm ... This removes the gsm->sln step when transcoding priv-recordintro. (closes issue #18176) Reported by: pabelanger Patches: chan_sip.diff uploaded by pabelanger (license 224) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@292411 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	c7fd3b73d3	Fix improper operator key acceptance and clean up temp recording files. ... This is a fix for when pressing the operator key after recording an unavailable, busy, name, or temporary message in mailbox options. The operator key should not be accepted here, but should be allowed during the message recording. If the operator key is pressed during ensure the file is saved or deleted as apporopriate. Also, ensure removal of temporary recorded files after an early hang up or when message acceptance confirmation times out. ABE-2518 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@292223 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	e93fc235b1	When forwarding a message, a prepend means that the filesystem will always have a better copy. ... (closes issue #17803) Reported by: dpetersen Patches: 20100923__issue17803.diff.txt uploaded by tilghman (license 14) Tested by: dpetersen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@289873 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	c4f3f68e43	Fix a crash in app_sms. ... Since the data being passed to the generator callback is on the stack of the SMS() application, we must ensure that the generator is stopped before the application exits. ABE-2587 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@289424 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brett Bryant	3702c0cc02	Fix misvalidation of meetme pins in conjunction with the 'a' MeetMe flag. ... When using the 'a' MeetMe flag and having a user and admin pin setup for your conference, using the user pin would gain you admin priviledges. Also, when no user pin was set, an admin pin was, the 'a' MeetMe flag wasn't used, and the user tried to enter a conference then they were still prompted for a pin and forced to hit #. (closes issue #17908) Reported by: kuj Patches: pins_2.patch uploaded by kuj (license 1111) Tested by: kuj Review: [full review board URL with trailing slash] git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@287758 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	41969111bd	Blank columns should get set on reload, not ignored. ... (closes issue #16893) Reported by: haakon Patches: 20100818__issue16893.diff.txt uploaded by tilghman (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@287386 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	a91c96ba5d	whitespace fix ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@286956 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	fa2b4c0196	Ensure mailbox is not filled to capacity before doing message forwarding. ... Specifically, before prompting to record a prepended message the capacity is checked first. If the mailbox is full the extension will be reprompted. ABE-2517 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@286941 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago