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2011-11-14Don't read past end of input when calling write()Terry Wilson
int blah = 1; ... write(chan->alertpipe[1], &blah, new_frames * sizeof(blah)) != (new_frames * sizeof(blah))) is only valid when new_frames == 1. Otherwise we start reading into adjacent variables declared on the stack. The read end discards what is read, so the values don't matter but it's not a good idea to read past where we want even though new_frames is almost always 1 and should never be large. This patch is basically taken out of kpfleming's eventfd branch, as he mentioned that he remembered fixing it there when I talked to him about this issue. Review: https://reviewboard.asterisk.org/r/1583/ ........ Merged revisions 345163 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345164 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345165 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-14Avoid unnecessary WARNING messageTerry Wilson
Add AST_CONTROL_UPDATE_RTP_PEER frame to be ignored here to avoid displaying a WARNING message. (closes issue ASTERISK-18610) Patch by: Kristijan_Vrban ........ Merged revisions 340878 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340879 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340880 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-11Update SHA1 code to RFC 6234Tzafrir Cohen
RFC 6234 is an update to RFC 3174 from which the code was originally taken. It has a slightly better code, and a better phrased license (simple 3-clause BSD). * main/sha1.c is sha1.c from RFC 6234 with formatting changes only. * include/asterisk/sha1.h merges sha.h and sha-private.h from RFC 6234. * Removed unused include of asterisk/sha1.h from main/channels.c Review: https://reviewboard.asterisk.org/r/1503/ Merge-From: http://svn.asterisk.org/svn/asterisk/branches/1.8@340263 Merge-From: http://svn.asterisk.org/svn/asterisk/branches/10@340280 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340283 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-22Merged revisions 337431 via svnmerge from Gregory Nietsky
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337431 | irroot | 2011-09-22 08:29:09 +0200 (Thu, 22 Sep 2011) | 25 lines Merged revisions 337430 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337430 | irroot | 2011-09-22 08:18:33 +0200 (Thu, 22 Sep 2011) | 19 lines Its possible to loose audio on ast_write when the channel is not transcoded correctly. in the case of DAHDI the channel is hungup. This patch tries to "fix" the problem and make the channel compatiable and warn the user of this problem. Please note there is a underlying problem with codec negotion this does not fix the problem it does try to rectify it and prevent loss of service. Review: https://reviewboard.asterisk.org/r/1442/ (closes issue ASTERISK-17541) (closes issue ASTERISK-18063) (issue ASTERISK-14384) (issue ASTERISK-17502) (issue ASTERISK-18325) (issue ASTERISK-18422) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337432 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-16Merged revisions 336307 via svnmerge from Jonathan Rose
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336307 | jrose | 2011-09-16 16:09:20 -0500 (Fri, 16 Sep 2011) | 20 lines Merged revisions 336294 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336294 | jrose | 2011-09-16 14:53:40 -0500 (Fri, 16 Sep 2011) | 13 lines 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/trunk@336311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-12Merged revisions 335434 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335434 | mnicholson | 2011-09-12 10:55:48 -0500 (Mon, 12 Sep 2011) | 13 lines Merged revisions 335433 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335433 | mnicholson | 2011-09-12 10:54:41 -0500 (Mon, 12 Sep 2011) | 6 lines Properly set caller_warning and callee_warning before we try to use them. ASTERISK-18199 Patch by: elguero Testing by: rtang ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-09Merged revisions 335078 via svnmerge from Matthew Jordan
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335078 | mjordan | 2011-09-09 11:27:01 -0500 (Fri, 09 Sep 2011) | 29 lines Merged revisions 335064 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335064 | mjordan | 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011) | 23 lines 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/trunk@335079 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-08-31Merged revisions 334010 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r334010 | rmudgett | 2011-08-31 10:23:11 -0500 (Wed, 31 Aug 2011) | 50 lines Merged revisions 334009 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334009 | rmudgett | 2011-08-31 10:20:31 -0500 (Wed, 31 Aug 2011) | 43 lines Call pickup race leaves orphaned channels or crashes. Multiple users attempting to pickup a call that has been forked to multiple extensions either crashes or fails a masquerade with a "bad things may happen" message. This is the scenario that is causing all the grief: 1) Pickup target is selected 2) target is marked as being picked up in ast_do_pickup() 3) target is unlocked by ast_do_pickup() 4) app dial or queue gets a chance to hang up losing calls and calls ast_hangup() on target 5) SINCE A MASQUERADE HAS NOT BEEN SETUP YET BY ast_do_pickup() with ast_channel_masquerade(), ast_hangup() completes successfully and the channel is no longer in the channels container. 6) ast_do_pickup() then calls ast_channel_masquerade() to schedule the masquerade on the dead channel. 7) ast_do_pickup() then calls ast_do_masquerade() on the dead channel 8) bad things happen while doing the masquerade and in the process ast_do_masquerade() puts the dead channel back into the channels container 9) The "orphaned" channel is visible in the channels list if a crash does not happen. This patch does the following: * Made ast_hangup() set AST_FLAG_ZOMBIE on a successfully hung-up channel and not release the channel lock until that has happened. * Made __ast_channel_masquerade() not setup a masquerade if either channel has AST_FLAG_ZOMBIE set. * Fix chan_agent misuse of AST_FLAG_ZOMBIE since it would no longer work. (closes issue ASTERISK-18222) Reported by: Alec Davis Tested by: rmudgett, Alec Davis, irroot, Karsten Wemheuer (closes issue ASTERISK-18273) Reported by: Karsten Wemheuer Tested by: rmudgett, Alec Davis, irroot, Karsten Wemheuer Review: https://reviewboard.asterisk.org/r/1400/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334011 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-08-29Merged revisions 333681 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r333681 | twilson | 2011-08-29 12:28:59 -0500 (Mon, 29 Aug 2011) | 7 lines Use realtime text when it is negotiated This patch make use of wirte_text() realtime text instead of send_text() if T.140 is in native formats. ASTERISK-17937 Review: https://reviewboard.asterisk.org/r/1356/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333689 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-08-08Merged revisions 331041 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r331041 | twilson | 2011-08-08 16:12:51 -0500 (Mon, 08 Aug 2011) | 6 lines Replace AMI Unlink events with Bridge events A previous update converted some of the Link and Unlink events to Bridge events, but a couple of Unlink events were missed. This patch rectifies the situation. (closes issues ASTERISK-17455) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-31Merged revisions 330369 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r330369 | rmudgett | 2011-07-30 18:57:56 -0500 (Sat, 30 Jul 2011) | 11 lines Merged revisions 330368 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330368 | rmudgett | 2011-07-30 18:56:29 -0500 (Sat, 30 Jul 2011) | 4 lines Remove some redundant locking code in ast_do_masquerade(). Also updated some comments. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@330370 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-30Merged revisions 330312 via svnmerge from Gregory Nietsky
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r330312 | irroot | 2011-07-30 17:34:41 +0200 (Sat, 30 Jul 2011) | 15 lines Merged revisions 330311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330311 | irroot | 2011-07-30 17:25:16 +0200 (Sat, 30 Jul 2011) | 9 lines prevent double masqurading channels when one is been hung up and deadlock avoidance is used. There is a race condition in ast_do_masquerade / ast_hangup (at least) Reported by me signed off by schmidts with input from David Vossel Review: https://reviewboard.asterisk.org/r/1323/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@330313 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-07Adds pass-through support for codec CELT.David Vossel
This patch adds pass-through support for CELT. CELT formats are defined in codecs.conf and can be configured to any sample rate a CELT endpoint supports. This patch also addresses a crash in channel.c resulting from a frame list being freed incorrectly. This crash was discovered while testing a CELT translator which had to split encoded audio into multiple frames. The codec translator is not a part of this patch, but may be contributed in the future. Review: https://reviewboard.asterisk.org/r/1294/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326855 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-29Merged revisions 325545 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325545 | mnicholson | 2011-06-29 11:18:39 -0500 (Wed, 29 Jun 2011) | 2 lines make framehooks prevent native bridging (for real this time) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325547 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-16Merged revisions 324048 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324048 | twilson | 2011-06-16 17:35:41 -0500 (Thu, 16 Jun 2011) | 8 lines 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/trunk@324050 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-13Merged revisions 323213 via svnmerge from Leif Madsen
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323213 | lmadsen | 2011-06-13 15:51:52 -0400 (Mon, 13 Jun 2011) | 6 lines Avoid dividing by zero with L() option to Dial() Reported by: nicolasom Patches: issue-17995.patch - nicolasom (License #5994) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323214 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-01Support routing text messages outside of a call.Russell Bryant
Asterisk now has protocol independent support for processing text messages outside of a call. Messages are routed through the Asterisk dialplan. SIP MESSAGE and XMPP are currently supported. There are options in sip.conf and jabber.conf that enable these features. There is a new application, MessageSend(). There are two new functions, MESSAGE() and MESSAGE_DATA(). Documentation will be available on the project wiki, wiki.asterisk.org. Thanks to Terry Wilson for the assistance with development and to David Vossel for helping with some additional testing. Review: https://reviewboard.asterisk.org/r/1042/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321546 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-26Use va_copy for stringfieldsTerry Wilson
The ast_string_field_build_va functions were written to take to separate va_lists to work around FreeBSD 4 not having va_copy defined. In the end, we don't support anything using gcc < 3 anyway because we use va_copy all over the place anyway. This patch just simplifies things by removing the second va_list function arguments in favor of va_copy. Review: https://reviewboard.asterisk.org/r/1233/ --This line, and those below, will be ignored-- M include/asterisk/stringfields.h M main/utils.c M main/channel.c git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320946 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-25Merged revisions 320823 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320823 | rmudgett | 2011-05-25 12:06:38 -0500 (Wed, 25 May 2011) | 18 lines The AMI Newstate event contains different information between v1.4 and v1.8. The addition of connected line support in v1.8 changes the behavior of the channel caller ID somewhat. The channel caller ID value no longer time shares with the connected line ID on outgoing call legs. The timing of some AMI events/responses output the connected line ID as caller ID. These party ID's are now separate. * The ConnectedLineNum and ConnectedLineName headers were added to many AMI events/responses if the CallerIDNum/CallerIDName headers were also present. (closes issue #18252) Reported by: gje Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1227/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320825 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-25Merged revisions 320796 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320796 | rmudgett | 2011-05-25 11:23:11 -0500 (Wed, 25 May 2011) | 17 lines 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/trunk@320820 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-20Merged revisions 320057 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320057 | rmudgett | 2011-05-20 11:43:02 -0500 (Fri, 20 May 2011) | 19 lines Crash while transferring a call during DTMF feature timeout. When a call is being attended transferred during the time between AST_FRAME_DTMF_BEGIN and AST_FRAME_DTMF_END, the transferred channel becomes a zombie (so tech data is not available), making ast_dtmf_stream() segfault when it tries to send the DTMF digit (at least with SIP channels). Patch based on feature-end-zombie.patch uploaded by Irontec (license 1256) * Check for zombies when ast_channel_bridge() returns. * Guarantee that the fo parameter value is initialized in ast_channel_bridge() before any returns. (closes issue #19116) Reported by: Irontec Tested by: rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320058 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-13Merged revisions 318921 via svnmerge from Brett Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318921 | bbryant | 2011-05-13 14:09:34 -0400 (Fri, 13 May 2011) | 8 lines Fixes a segmentation fault in dynamic hints when a channel technology isn't loaded for a hint. (closes issue #18495) Reported by: bertrand Tested by: bertrand ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318922 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-09Merged revisions 318142 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318142 | mnicholson | 2011-05-09 09:09:38 -0500 (Mon, 09 May 2011) | 9 lines Make indicate/control frames WRITE events on framehooks. Also, if a framehook returns a non-control frame, don't forward it to the channel. (closes issue #19251) Reported by: irroot Patches: (modified) framehook_indicate.patch2 uploaded by irroot (license 52) Tested by: irroot ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318143 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-03Merged revisions 316334 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316334 | dvossel | 2011-05-03 17:05:59 -0500 (Tue, 03 May 2011) | 8 lines Fixes framehook segfault on indicate (closes issue #19215) Reported by: irroot Patches: framehook_indicate.patch uploaded by irroot (license 52) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316335 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-03Merged revisions 316265 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316265 | russell | 2011-05-03 14:55:49 -0500 (Tue, 03 May 2011) | 5 lines 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/trunk@316293 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-27Clears exception flag during ast_read when func_jitterbuffer is enabledDavid Vossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@315855 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-21New HD ConfBridge conferencing application.David Vossel
Includes a new highly optimized and customizable ConfBridge application capable of mixing audio at sample rates ranging from 8khz-192khz. Review: https://reviewboard.asterisk.org/r/1147/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314598 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-20Introduction of the JITTERBUFFER dialplan function.David Vossel
Review: https://reviewboard.asterisk.org/r/1157/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314509 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-13Merged revisions 313588 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r313588 | rmudgett | 2011-04-13 11:31:50 -0500 (Wed, 13 Apr 2011) | 55 lines Merged revisions 313579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r313579 | rmudgett | 2011-04-13 11:29:49 -0500 (Wed, 13 Apr 2011) | 48 lines Merged revisions 313545 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r313545 | rmudgett | 2011-04-13 11:21:24 -0500 (Wed, 13 Apr 2011) | 41 lines 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/trunk@313606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-02Add HangupRequest manager event, to specify when/where a channel gets hung up.Jason Parker
(closes issue #18226) Reported by: clegall_proformatique Patches: asterisk_1.8_293157_hanguprequests.svn.patch uploaded by clegall proformatique (license 1139) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309300 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-24Merged revisions 308903 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r308903 | rmudgett | 2011-02-24 15:38:41 -0600 (Thu, 24 Feb 2011) | 9 lines Invalid read in ast_channel_set_caller_event(). Valgrind reported that ast_channel_set_caller_event() was reading data from a freed buffer when using the pre_set structure. Rearange things to pre-calculate the name and number pointer before updating the caller party structure to see if the name or number was changed. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308904 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd ↵David Vossel
audio ConfBridge, and other stuff -Functional changes 1. Dynamic global format list build by codecs defined in codecs.conf 2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf 3. Negotiation of SILK attributes in chan_sip. 4. SPEEX 32khz with translation 5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation using codec_resample.c 6. Various changes to RTP code required to properly handle the dynamic format list and formats with attributes. 7. ConfBridge now dynamically jumps to the best possible sample rate. This allows for conferences to take advantage of HD audio (Which sounds awesome) 8. Audiohooks are no longer limited to 8khz audio, and most effects have been updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT. 9. codec_resample now uses its own code rather than depending on libresample. -Organizational changes Global format list is moved from frame.c to format.c Various format specific functions moved from frame.c to format.c Review: https://reviewboard.asterisk.org/r/1104/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-07Pass a MCID request to the bridged channel.Richard Mudgett
Pass a MCID request to the bridged channel so the bridged channel can send it to the network. The ability to send the MCID request on an ISDN span is enabled with the new chan_dahdi.conf mcid_send option. JIRA SWP-2845 JIRA ABE-2736 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306755 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-04Replace ast_log(LOG_DEBUG, ...) with ast_debug()Paul Belanger
(closes issue #18556) Reported by: kkm Review: https://reviewboard.asterisk.org/r/1071/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306258 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-03Asterisk media architecture conversion - no more format bitfieldsDavid Vossel
This patch is the foundation of an entire new way of looking at media in Asterisk. The code present in this patch is everything required to complete phase1 of my Media Architecture proposal. For more information about this project visit the link below. https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal The primary function of this patch is to convert all the usages of format bitfields in Asterisk to use the new format and format_cap APIs. Functionally no change in behavior should be present in this patch. Thanks to twilson and russell for all the time they spent reviewing these changes. Review: https://reviewboard.asterisk.org/r/1083/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-03Merged revisions 305923 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r305923 | rmudgett | 2011-02-02 18:24:40 -0600 (Wed, 02 Feb 2011) | 24 lines Merged revisions 305889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r305889 | rmudgett | 2011-02-02 18:15:07 -0600 (Wed, 02 Feb 2011) | 17 lines Merged revisions 305888 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305888 | rmudgett | 2011-02-02 18:02:43 -0600 (Wed, 02 Feb 2011) | 8 lines 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(). * Fixed AMI SendText action ast_sendtext() return value check. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-24Merged revisions 303549 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303549 | russell | 2011-01-24 14:51:37 -0600 (Mon, 24 Jan 2011) | 45 lines 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/trunk@303551 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-12Merged revisions 301504 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301504 | jpeeler | 2011-01-12 12:12:08 -0600 (Wed, 12 Jan 2011) | 26 lines Merged revisions 301503 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r301503 | jpeeler | 2011-01-12 12:11:49 -0600 (Wed, 12 Jan 2011) | 19 lines Merged revisions 301502 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r301502 | jpeeler | 2011-01-12 12:10:42 -0600 (Wed, 12 Jan 2011) | 12 lines Fix CPU spike when pressing DTMF after agent login. The problem here is that DTMF was being continuously deferred and requeued since ast_safe_sleep is called in a loop. There are serveral other places in the code that sleeps and then loops in a similar fashion. Because of this fact I opted to not defer DTMF any more, which will not affect the original fix: https://reviewboard.asterisk.org/r/674 (closes issue #18130) Reported by: rgj ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301505 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-20Some scheduler API cleanup and improvements.Russell Bryant
Previously, I had added the ast_sched_thread stuff that was a generic scheduler thread implementation. However, if you used it, it required using different functions for modifying scheduler contents. This patch reworks how this is done and just allows you to optionally start a thread on the original scheduler context structure that has always been there. This makes it trivial to switch to the generic scheduler thread implementation without having to touch any of the other code that adds or removes scheduler entries. In passing, I made some naming tweaks to add ast_ prefixes where they were not there before. Review: https://reviewboard.asterisk.org/r/1007/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299091 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-07Merged revisions 297825 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r297825 | jpeeler | 2010-12-07 16:59:30 -0600 (Tue, 07 Dec 2010) | 26 lines Merged revisions 297824 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297824 | jpeeler | 2010-12-07 16:58:54 -0600 (Tue, 07 Dec 2010) | 19 lines Merged revisions 297823 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297823 | jpeeler | 2010-12-07 16:57:48 -0600 (Tue, 07 Dec 2010) | 12 lines Revert code that changed SSRC for DTMF. Some previous behavior was attempted to be restored, but mistakingly I did not realize that the previous behavior was incorrect. This fixes DTMF not being detected since DTMF shouldn't cause the SSRC to change. (related to issue #17404) (closes issue #18189) (closes issue #18352) Reported by: marcbou Tested by: cmbaker82 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@297826 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-24Merged revisions 296230 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296230 | russell | 2010-11-24 17:29:44 -0600 (Wed, 24 Nov 2010) | 20 lines Merged revisions 296221 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296221 | russell | 2010-11-24 17:28:19 -0600 (Wed, 24 Nov 2010) | 13 lines Merged revisions 296213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296213 | russell | 2010-11-24 17:26:43 -0600 (Wed, 24 Nov 2010) | 6 lines Make Asterisk less crashy. Since we might not put a new translation path on the channel, go ahead and set it to NULL right after destroying the old one to ensure we don't try to free an invalid translation path later on. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296235 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-24Merged revisions 296084 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296084 | russell | 2010-11-24 14:23:46 -0600 (Wed, 24 Nov 2010) | 26 lines Merged revisions 296083 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296083 | russell | 2010-11-24 14:23:11 -0600 (Wed, 24 Nov 2010) | 19 lines Merged revisions 296082 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296082 | russell | 2010-11-24 14:22:32 -0600 (Wed, 24 Nov 2010) | 12 lines Fix false reporting of an error by set_format(). In the case that the native format was able to be changed to match the new requested format, the code proceeded to attempt to build a translation path, anyway. The result would be NULL, since no translation path is necessary and resulted in this function thinking an error has occurred. This case is now specifically caught and no attempt to build a translation path is attempted. Thanks to our automated tests and bamboo.asterisk.org for catching this problem and making a whole lot of noise when things started failing. :-) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296085 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-24Merged revisions 296002 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296002 | russell | 2010-11-24 11:13:08 -0600 (Wed, 24 Nov 2010) | 52 lines Merged revisions 296001 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296001 | russell | 2010-11-24 11:03:16 -0600 (Wed, 24 Nov 2010) | 45 lines Merged revisions 296000 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296000 | russell | 2010-11-24 10:48:39 -0600 (Wed, 24 Nov 2010) | 38 lines 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/trunk@296034 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-22Merged revisions 295866 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r295866 | rmudgett | 2010-11-22 13:36:10 -0600 (Mon, 22 Nov 2010) | 60 lines Merged revisions 295843 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r295843 | rmudgett | 2010-11-22 13:28:23 -0600 (Mon, 22 Nov 2010) | 53 lines Merged revisions 295790 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295790 | rmudgett | 2010-11-22 12:46:26 -0600 (Mon, 22 Nov 2010) | 46 lines 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/trunk@295867 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-16Merged revisions 295282 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r295282 | rmudgett | 2010-11-16 17:02:36 -0600 (Tue, 16 Nov 2010) | 16 lines Merged revisions 295281 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r295281 | rmudgett | 2010-11-16 16:57:07 -0600 (Tue, 16 Nov 2010) | 9 lines Merged revisions 295280 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295280 | rmudgett | 2010-11-16 16:52:06 -0600 (Tue, 16 Nov 2010) | 1 line Dead code elimination in channel.c:ast_channel_bridge() variable who. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@295283 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-09Merged revisions 294466 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294466 | rmudgett | 2010-11-09 16:46:45 -0600 (Tue, 09 Nov 2010) | 1 line Allow ast_do_masquerade() failure to be reported again. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294467 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-09Merged revisions 294349 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294349 | rmudgett | 2010-11-09 10:55:32 -0600 (Tue, 09 Nov 2010) | 17 lines Analog lines do not transfer CONNECTED LINE or execute the interception macros. Add connected line update for sig_analog transfers and simplify the corresponding sig_pri and chan_misdn transfer code. Note that if you create a three-way call in sig_analog before transferring the call, the distinction of the caller/callee interception macros make little sense. The interception macro writer needs to be prepared for either caller/callee macro to be executed. The current implementation swaps which caller/callee interception macro is executed after a three-way call is created. Review: https://reviewboard.asterisk.org/r/996/ JIRA ABE-2589 JIRA SWP-2372 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294351 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08Merged revisions 294278 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r294278 | jpeeler | 2010-11-08 15:59:45 -0600 (Mon, 08 Nov 2010) | 23 lines Merged revisions 294277 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294277 | jpeeler | 2010-11-08 15:58:13 -0600 (Mon, 08 Nov 2010) | 16 lines Fix playback failure when using IAX with the timerfd module. To fix this issue the alert pipe will now be used when the timerfd module is in use. There appeared to be a race that was not solved by adding locking in the timerfd module, but needed to be there anyway. The race was between the timer being put in non-continuous mode in ast_read on the channel thread and the IAX frame scheduler queuing a frame which would enable continuous mode before the non-continuous mode event was read. This race for now is simply avoided. (closes issue #18110) Reported by: tpanton Tested by: tpanton I put tested by tpanton because it was tested on his hardware. Thanks for the remote access to debug this issue! ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294279 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-10-22Merged revisions 292704 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292704 | rmudgett | 2010-10-22 10:47:08 -0500 (Fri, 22 Oct 2010) | 19 lines Connected line is not updated when chan_dahdi/sig_pri or chan_misdn transfers a call. When a call is transfered by ECT or implicitly by disconnect in sig_pri or implicitly by disconnect in chan_misdn, the connected line information is not exchanged. The connected line interception macros also need to be executed if defined. The CALLER interception macro is executed for the held call. The CALLEE interception macro is executed for the active/ringing call. JIRA ABE-2589 JIRA SWP-2296 Patches: abe_2589_c3bier.patch uploaded by rmudgett (license 664) abe_2589_v1.8_v2.patch uploaded by rmudgett (license 664) Review: https://reviewboard.asterisk.org/r/958/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@292705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-10-13Merged revisions 291581 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r291581 | twilson | 2010-10-13 16:01:56 -0700 (Wed, 13 Oct 2010) | 35 lines Merged revisions 291580 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r291580 | twilson | 2010-10-13 15:58:43 -0700 (Wed, 13 Oct 2010) | 28 lines Merged revisions 291577 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291577 | twilson | 2010-10-13 15:45:15 -0700 (Wed, 13 Oct 2010) | 21 lines Don't ignore frames that have been queued when softhangup'd When an outgoing call is answered and hung up by the far end *very* quickly, we may not read any frames and therefor end up with a call that displays the wrong disposition/DIALSTATUS. The reason is because ast_queue_hangup() immediately sets the _softhangup flag on the channel and then queues the HANGUP control frame, but __ast_read refuses to read any frames if ast_check_hangup() indicates that a hangup request has been made (which it will if _softhangup is set). So, we end up losing control frames. This change makes __ast_read continue to read frames even if a soft hangup has been requested. It queues a hangup frame to make sure that __ast_read() will still eventually return NULL. Much thanks to David Vossel for all of the reviews, discussion, and help! (closes issue #16946) Reported by: davidw Review: https://reviewboard.asterisk.org/r/740/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@291657 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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