2285 Commits (3b2f5e7d4c890cf46995f3c6dece02f7f461d2ab)

Author	SHA1	Message	Date
Matthew Jordan	f13c3b3fd2	Fix for incorrect voicemail duration in external notifications ... This patch fixes an issue where the voicemail duration was being reported with a duration significantly less than the actual sound file duration. Voicemails that contained mostly silence were reporting the duration of only the sound in the file, as opposed to the duration of the file with the silence. This patch fixes this by having two durations reported in the __ast_play_and_record family of functions - the sound_duration and the actual duration of the file. The sound_duration, which is optional, now reports the duration of the sound in the file, while the actual full duration of the file is reported in the duration parameter. This allows the voicemail applications to use the sound_duration for minimum duration checking, while reporting the full duration to external parties if the voicemail is kept. (issue ASTERISK-2234) (closes issue ASTERISK-16981) Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1443 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@337118 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	02795f190e	Various changes to allow 1.8 to compile on Mac OS X Lion (10.7) ... * Makefile workaround for 10.6 extended to work on 10.7 and later. * Now uses the 'weak' symbol for Lion systems, which no longer support 'weak_import' Closes ASTERISK-17612. Closes ASTERISK-18213. Tested by: tilghman, oej. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@336733 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	21714a05b6	Fix bad RTP media bridges in directmedia calls on peers separated by multiple Asterisk nodes. ... In a situation involving devices on separate Asterisk trunks, the remote RTP bridge would break when starting a call with directmedia. This patch queues a new type of control frame so that our RTP bridge loop can properly detect when these situations occur and check to see if peers need to be updated in order to send their media to the proper location. (Closes issue ASTERISK-18340) Reported by: Thomas Arimont (Closes issue ASTERISK-17725) Reported by: kwk Tested by: twilson, jrose git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@336294 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	9910558356	Remove unnecessary libpri dependency checks in the configure script. ... Using the --with-pri option with the configure script generated an error about not having PRI_L2_PERSISTENCE if you did not have the absolute latest libpri SVN checkout installed. The AST_EXT_LIB_SETUP_DEPENDENT macro in the configure.ac script seems to be for libraries that are dependent upon other libraries and not necessarily for optional/added features within a library. (closes issue ASTERISK-18535) Reported by: Michael Keuter git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335911 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	5f1882731a	Fix a crash in res_ais. ... This patch resolves a crash observed in a load testing environment that involved the use of the res_ais module. I observed some crashes where the event delivery callback would get called, but the length parameter incidcating how much data there was to read was 0. The code assumed (with good reason I would think) that if this callback got called, there was an event available to read. However, if the rare case that there's nothing there, catch it and return instead of blowing up. More specifically, the change always ensure that the size of the received event in the cluster is always big enough to be a real ast_event. Review: https://reviewboard.asterisk.org/r/1423/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335497 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Jordan	7dc49195d8	Updated SIP 484 handling; added Incomplete control frame ... When a SIP phone uses the dial application and receives a 484 Address Incomplete response, if overlapped dialing is enabled for SIP, then the 484 Address Incomplete is forwarded back to the SIP phone and the HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete application dialplan logic was automatically triggered; now, explicit dialplan usage of the application is required. Additionally, this patch adds a new AST_CONTOL_FRAME type called AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame, it is an indication that the dialplan expects more digits back from the device. If the device supports overlap dialing it should attempt to notify the device that the dialplan is waiting for more digits; otherwise, it can handle the frame in a manner appropriate to the channel driver. (closes issue ASTERISK-17288) Reported by: Mikael Carlsson Tested by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/1416/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335064 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	6d49117675	Fix potential memory allocation failure crashes in config.c. ... * Added required checks to the returned memory allocation pointers to prevent crashes. * Made ast_include_rename() create a replacement ast_variable list node if the new filename is longer than the available space. Fixes potential crash and memory leak. * Factored out ast_variable_move() from ast_variable_update() so ast_include_rename() can also use it when creating a replacement ast_variable list node. * Made the filename stuffed at the end of the struct a minimum allocated size in ast_variable_new() in case ast_include_rename() changes the stored filename. * Constify struct char pointers pointing to strings stuffed at the end of the struct for: ast_variable, cache_file_mtime, and ast_config_map. * Factored out cfmtime_new() to remove inlined code and allow some struct pointers to become const. * Removed the list lock from struct cache_file_mtime that was never used. * Added doxygen comments to several structure elements and better documented what strings are stuffed at the struct end char array. * Reworked ast_config_text_file_save() and set_fn() to handle allocation failure of the include file scratch pad object tracking blank lines. * Made ast_config_text_file_save() fn[] declared with PATH_MAX to ensure it is long enough for any filename with path. Also reduced the number of container fileset buckets from a rediculus 180,000 to 1023. JIRA AST-618 Review: https://reviewboard.asterisk.org/r/1378/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@334296 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	b806cbbccb	Add some do not hold locks notes to channel.h ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@333785 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	b1f11e0df4	Revert previous commit. Not ready yet. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332945 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	c0ce03d77f	Signed ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332943 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Jordan	56549c96ab	Review: https://reviewboard.asterisk.org/r/1364/ ... This update adds a new AMI event, TestEvent, which is enabled when the TEST_FRAMEWORK compiler flag is defined. It also adds initial usage of this event to app_voicemail. The TestEvent AMI event is used extensively by the voicemail tests in the Asterisk Test Suite. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332817 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	582aacb3d8	Re-add support for spaces in pathnames, including now spaces in DESTDIR. ... This was initially added to 1.8 prior to release, primarily to support the standard paths on Mac OS X, but was partially reverted recently in Subversion, due to the lack of support for spaces in DESTDIR. This commit restores support for the standard paths on Mac OS X, and also includes support for spaces in DESTDIR. (closes issue ASTERISK-18290) Reported by: pabelanger Review: https://reviewboard.asterisk.org/r/1326/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332355 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	9328590ddb	Outgoing BRI calls fail when using Asterisk 1.8 with HA8, HB8, and B410P cards. ... France Telecom brings layer 2 and layer 1 down on BRI lines when the line is idle. When layer 1 goes down Asterisk cannot make outgoing calls and the HA8 and HB8 cards also get IRQ misses. The inability to make outgoing calls is because the line is in red alarm and Asterisk will not make calls over a line it considers unavailable. The IRQ misses for the HA8 and HB8 card are because the hardware is switching clock sources from the line which just brought layer 1 down to internal timing. There is a DAHDI option for the B410P card to not tell Asterisk that layer 1 went down so Asterisk will allow outgoing calls: "modprobe wcb4xxp teignored=1". There is a similar DAHDI option for the HA8 and HB8 cards: "modprobe wctdm24xxp bri_teignored=1". Unfortunately that will not clear up the IRQ misses when the telco brings layer 1 down. * Add layer 2 persistence option to customize the layer 2 behavior on BRI PTMP lines. The new option has three settings: 1) Use libpri default layer 2 setting. 2) Keep layer 2 up. Bring layer 2 back up when the peer brings it down. 3) Leave layer 2 down when the peer brings it down. Layer 2 will be brought up as needed for outgoing calls. JIRA AST-598 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332264 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	31a1b94622	Fixes some voicemail forwarding behavior based around prepend mode. ... Formerly, prepend forwarding would have the user record a message with no useful prompt and an expectation for the user to push a button on the phone when finished recording. If a length of silence was detected instead, the recording would be canceled and the user would re-enter the voicemail forwarding menu. Subsequent time-outs in prepend recording would also bug out in the sense that they would write over the original message and get sent to the recipient regardless of whether they timed out or were accepted. This patch fixes this issue and adds a prompt which will be played after a timeout informing the user that they needed to press a button. Currently, the sound files that we have are somewhat inadquate for this, so after the call we simply have Allison say "Please try again. Then press pound." which actually relies on two separate sound files. Just one would be more appropriate. reporter: Vlad Povorozniuc Review: https://reviewboard.asterisk.org/r/1327/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@329527 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Kinsey Moore	5905269669	RTP bridge away with inband DTMF and feature detection ... When deciding whether Asterisk was allowed to bridge the call away from the core, chan_sip did not take into account the usage of features on dialed channels that require monitoring of DTMF on channels utilizing inband DTMF. This would cause Asterisk to allow the call to be locally or remotely bridged, preventing access to the data required to detect activations of such features. (closes 17237) Review: https://reviewboard.asterisk.org/r/1302/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@328823 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	d72bfe9db3	Make AST_LIST_REMOVE safer ... AST_LIST_REMOVE shouldn't modify the element passed in if it isn't found. This commit also adds linked list unit tests. Review: https://reviewboard.asterisk.org/r/1321/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@328716 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	4a19bd7e74	Update chan_gtalk to work with changed GMail-based calls ... The messages sent by the GMail client have changed, but include the old-style messages as well. This patch checks for this case and uses the old-style offer. (closes issue ASTERISK-18084) Review: https://reviewboard.asterisk.org/r/1312/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@327682 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	83fad31713	DTMF wasn't being logged on connected consoles when enabled in logger.conf ... Previously in order for DTMF to be logged in a connected console session, the user would have to do logger set channel DTMF on. This corrects that so that it is on by default. This issue was caused by an off by one error incurred by a logger level count of 6 in logger.h where it should have been 7. (closes issue: ASTERISK-17974) Reported by: Luke H git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@324768 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	e1adc7cefa	Merged revisions 324634 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r324634 | dvossel | 2011-06-23 13:18:46 -0500 (Thu, 23 Jun 2011) | 13 lines Merged revisions 324627 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r324627 | dvossel | 2011-06-23 13:16:52 -0500 (Thu, 23 Jun 2011) | 7 lines 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/1.8@324652 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	0ada0bfea3	Stop sending IPv6 link-local scope-ids in SIP messages ... The idea behind the patch listed below was used, but in a more targeted manner. There are now address stringification functions for addresses that are meant to be sent to a remote party. Link-local scope-ids only make sense on the machine from which they originate and so are stripped in the new functions. There is also a host sanitization function added to chan_sip which is used for when peer and dialog tohost fields or sip_registry hostnames are used to craft a SIP message. Also added are some basic unit tests for netsock2 address parsing. (closes issue ASTERISK-17711) Reported by: ch_djalel Patches: asterisk-1.8.3.2-ipv6_ll_scope.patch uploaded by ch_djalel (license 1251) Review: https://reviewboard.asterisk.org/r/1278/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@324484 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	87908a3f5d	Fixes locking inversion issue in ast_async_goto() ... During this function we can not hold the "chan" lock while doing the masquerade, the explicit goto on the tmp chan, or the channel alloc. Instead we need to get the channel lock, store off information about the channel that we need, and then let the channel lock go for the remainder of the function. Review: https://reviewboard.asterisk.org/r/1275/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@324364 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	c84e7b911e	Lock the channel before calling the setoption callback ... The channel needs to be locked before calling these callback functions. Also, sip_setoption needs to lock the pvt and a check p->rtp is non-null before using it. Review: https://reviewboard.asterisk.org/r/1220/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@324048 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	c71181a78f	Make ARRAY_LEN() return the same type on x86 and x86_64 systems ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@323863 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	ee2920afba	Add rtpkeepalives back to 1.8 ... The RTP-engine conversion left out support for handling rtpkeepalives. This patch adds them back. (closes issue ASTERISK-17304) Reported by: lmadsen Review: https://reviewboard.asterisk.org/r/1226/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@323370 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	b304546811	Correct ast_db_deltree documentation ... ast_db_deltree returns -1 on error, otherwise the number of deletions git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@322865 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	aec1979e7f	Remove potential deadlock in call pickup race. ... Deadlock is possible in ast_do_pickup() when holding the target channel lock and trying to get the chan channel lock. Also, holding the target lock when calling ast_channel_masquerade() is not a good idea because that routine does deadlock avoidance. * Removed the need to hold the target lock after marking the target with a datastore and getting the connected line data off of the target channel. * Moved can_pickup() to ast_can_pickup() in features.c. Now all the call pickup methods use the same basic call pickup availability check. Review: https://reviewboard.asterisk.org/r/1234/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@322749 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	5f46b994f4	Adds ast_escape_encoded utility to properly handle escaping of quoted field before uri. ... This commit backports a feature in trunk affecting initreqprep so that display name won't be encoded improperly. Also includes unit tests for the ast_escape_quoted function. This patch gives 1.8 a much improved outlook in countries which don't use standard ASCII characters. (closes issue ASTERISK-16949) Reported by: Örn Arnarson Review: https://reviewboard.asterisk.org/r/1235/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@322585 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	c728c8d56b	Fixes level toggling for logger set levels since it was reversed ... (closes issue ASTERISK-17850) Reported by: Luke H Tested by: jrose, Luke H Review: https://reviewboard.asterisk.org/r/1244/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@322069 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	a32c86fb71	Constify subscription description parameter string. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@321813 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	8668ec7ef9	Update some comments. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@321517 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	9718442188	Fix issue with playback of H.261 video. ... (closes issue #19379) Reported by: neutrino88 Patches: videoprompt.patch uploaded by neutrino88 (license 297) (changes by russell) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@321335 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	7257b78f53	Update ast_sockaddr comment with an important note. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@321044 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	aa64eb1077	Give zombies a safe channel driver to use. ... Recent crashes from zombie channels suggests that they need a safe home to goto. When a masquerade happens, the physical part of the zombie channel is hungup. The hangup normally sets the channel private pointer to NULL. If someone then blindly does a callback to the channel driver, a crash is likely because the private pointer is NULL. The masquerade now sets the channel technology of zombie channels to the kill channel driver. Related to the following issues: (issue #19116) (issue #19310) Review: https://reviewboard.asterisk.org/r/1224/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@320796 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	95bf6f2fc3	Revert part of a change to the bridging API code ... The capabilities used in the bridging API are very different than the ones used for formats. When the conversion was made expanding the bit width of codecs, the bridging code was accidentally accosted in ways that it didn't deserve. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@319920 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	e8935ca8e4	Support gmime-2.4 ... (closes issue #18863) Reported by: tzafrir Patches: gmime-2.4-18.diff uploaded by tzafrir (license 46) Tested by: tzafrir Review: https://reviewboard.asterisk.org/r/1213/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@319085 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alec L Davis	87d80af96c	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/1.8@318671 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	a82f1bb995	Fix a bunch of compiler warnings generated by gcc 4.6.0. ... Most of these are -Wunused-but-set-variable, but there were a few others mixed in here, as well. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@316265 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	016370cc5c	Merged revisions 315502 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r315502 | tilghman | 2011-04-26 14:22:52 -0500 (Tue, 26 Apr 2011) | 21 lines Merged revisions 315501 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r315501 | tilghman | 2011-04-26 14:18:46 -0500 (Tue, 26 Apr 2011) | 14 lines 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/1.8@315503 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	79e275d265	AST_CONTROL_XXX comment changes. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@314417 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	2998c62fc4	sip codec negotiation of dynamic rtp payloads error fix ... This patch fixes how chan_sip handles dynamic rtp payload types it does not understand. At the moment if a dynamic payload's mime type does not match one we understand, the payload does not get removed from our payload table. As a result of this, the payload is set to whatever dynamic codec we use internally for that payload number on outgoing INVITES. This is incorrect. This patch fixes this by properly checking the rtpmap set function's return code to make sure it was found. The function can return both -1 and -2 depending on the source of the mismatch. We were just checking -1 explicitly. Review: https://reviewboard.asterisk.org/r/1169/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@314017 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	55aa84a6e9	Merged revisions 313278 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r313278 | lmadsen | 2011-04-11 14:33:03 -0500 (Mon, 11 Apr 2011) | 14 lines Merged revisions 313277 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r313277 | lmadsen | 2011-04-11 14:30:20 -0500 (Mon, 11 Apr 2011) | 6 lines 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/1.8@313279 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	b17b0a7fa8	Merged revisions 312287 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r312287 | tilghman | 2011-04-01 05:51:24 -0500 (Fri, 01 Apr 2011) | 14 lines Merged revisions 312285 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r312285 | tilghman | 2011-04-01 05:36:42 -0500 (Fri, 01 Apr 2011) | 7 lines 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/1.8@312288 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	56cd7709a5	Merged revisions 309251 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309251 | tilghman | 2011-03-01 19:06:02 -0600 (Tue, 01 Mar 2011) | 7 lines Revert previous 2 commits, and instead conditionally redefine the same macro used in flex 2.5.35 that clashed with our workaround. Not surprisingly, the workaround was exactly the same code as was provided by the Flex maintainers, albeit in two different places, in different macros. This should fix the FreeBSD builds, which have an older version of Flex. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@309808 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	1b78442e0d	Merged revisions 309033-309034 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309033 | tilghman | 2011-02-28 04:43:12 -0600 (Mon, 28 Feb 2011) | 4 lines A later version of flex already includes the fwrite workaround code, which if used twice causes a compilation error. Detect whether Flex will compile without the workaround; if so, suppress our workaround code. ........ r309034 | tilghman | 2011-02-28 05:07:52 -0600 (Mon, 28 Feb 2011) | 2 lines Clarify meaning, removing double negative (stupid!) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@309035 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	a5f6367057	No response sent for SIP CC subscribe/resubscribe request. ... Asterisk does not send a response if we try to subscribe for call completion after we have received a 180 Ringing. You can only subscribe for call completion when the call has been cleared. When we receive the 180 Ringing, for this call, its call-completion state is 'CC_AVAILABLE'. If we then send a subscribe message to Asterisk, it trys to change the call-completion state to 'CC_CALLER_REQUESTED'. Because this is an invalid state change, it just ignores the message. The only state Asterisk will accept our subscribe message is in the 'CC_CALLER_OFFERED' state. Asterisk will go into the 'CC_CALLER_OFFERED' when the SIP client clears the call by sending a CANCEL. Asterisk should always send a response. Even if its a negative one. The fix is to allow for the CCSS core to notify a CC agent that a failure has occurred when CC is requested. The "ack" callback is replaced with a "respond" callback. The "respond" callback has a parameter indicating either a successful response or a specific type of failure that may need to be communicated to the requester. (closes issue #18336) Reported by: GeorgeKonopacki Tested by: mmichelson, rmudgett JIRA SWP-2633 (closes issue #18337) Reported by: GeorgeKonopacki Tested by: mmichelson JIRA SWP-2634 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@307879 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Andrew Latham	5f9a130198	Documentation Updates. ... More updates to the removed doc folder and start updates to the man page. (issue #16505) Reported by: tzafrir Tested by: lathama git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@306826 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	1440b054eb	Use the non-specific API aliases, to avoid a problem with building the utils directory. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@305040 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	fd11f34692	Change mutex tracking so that it only consumes memory in the core mutex object when it's actually being used. ... This reduces the overall size of a mutex which was 3016 bytes before this back down to 216 bytes (this is on 64-bit Linux with a glibc-implemented mutex). The exactness of the numbers here may vary slightly based upon how mutexes are implemented on a platform, but the long and short of it is that prior to this commit, chan_iax2 held down 98MB of memory on a 64-bit system for nothing more than a table of 32767 locks. After this commit, the same table occupies a mere 7MB of memory. (closes issue #18194) Reported by: job Patches: 20110124__issue18194.diff.txt uploaded by tilghman (license 14) Tested by: tilghman Review: https://reviewboard.asterisk.org/r/1066 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@304950 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	193a80b513	Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@303907 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	cfc893a5bc	Merged revisions 303548 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303548 | russell | 2011-01-24 14:49:53 -0600 (Mon, 24 Jan 2011) | 38 lines Merged revisions 303546 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303546 | russell | 2011-01-24 14:32:21 -0600 (Mon, 24 Jan 2011) | 31 lines 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/1.8@303549 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago