asterisk

Commit Graph

Author	SHA1	Message	Date
Russell Bryant	c28e2d380c	Merged revisions 317429 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317429 \| russell \| 2011-05-05 17:11:19 -0500 (Thu, 05 May 2011) \| 5 lines Only display inband DTMF warning if inband DTMF detection is enabled. (closes issue #18901) Reported by: irroot ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317430 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	19b45ad446	Merged revisions 317425 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317425 \| russell \| 2011-05-05 16:53:13 -0500 (Thu, 05 May 2011) \| 7 lines Add missing ActioID handling to Events action. (closes issue #18949) Reported by: edersohe Patches: 0018949.patch uploaded by edersohe (license 1228) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317426 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Sean Bright	d508a921bf	Add some new editline bindings by default, and allow for user specified configuration. I excluded the part of this patch that used the HOME environment variable since the built-in editline library goes to great lengths to disallow that. Instead only settings the EDITRC environment variable will use a user specified file. Also, the default environment variable use to determine the edit more is AST_EDITMODE instead of AST_EDITOR (although the latter is still supported). (closes issue #15929) Reported by: kkm Patches: astcli-editrc-v2.diff uploaded by kkm (license 888) 015929-astcli-editrc-trunk.240324.diff uploaded by kkm (license 888) Tested by: seanbright git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317395 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Stefan Schmidt	19eb6c7384	Adding the Move to Front Hash functionality Moving a found object to the front of its bucket to reduce the necessary traversal steps to find an object. This change improves the search time on large system with many data or in link lists. (closes issue #19233) Reported by: schmidts Review: https://reviewboard.asterisk.org/r/1201/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316962 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Sean Bright	fe5938c51e	Merged revisions 316917-316919 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316917 \| seanbright \| 2011-05-04 22:23:28 -0400 (Wed, 04 May 2011) \| 5 lines Make sure that tcptls_session is properly initialized. (issue #18598) Reported by: ksn ........ r316918 \| seanbright \| 2011-05-04 22:25:20 -0400 (Wed, 04 May 2011) \| 5 lines Look at the correct buffer for our digest info instead of an empty one. (issue #18598) Reported by: ksn ........ r316919 \| seanbright \| 2011-05-04 22:30:45 -0400 (Wed, 04 May 2011) \| 10 lines Use the correct HTTP method when generating our digest, otherwise we always fail. When calculating the 'A2' portion of our digest for verification, we need the HTTP method that is currently in use. Unfortunately our mapping function was incorrect, resulting in invalid hashes being generated and, in turn, failures in authentication. (closes issue #18598) Reported by: ksn ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316920 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Sean Bright	34734f727f	Merged revisions 316663 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316663 \| seanbright \| 2011-05-04 10:35:05 -0400 (Wed, 04 May 2011) \| 8 lines Only return a single error via AMI when requesting a forbidden action. (closes issue #19216) Reported by: oej Patches: issue19216-1.8-r316204.patch uploaded by seanbright (license 71) Tested by: seanbright ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316664 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	f4417923ce	Merged revisions 316334 via svnmerge from 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	15 years ago
Russell Bryant	37aa52fd78	Merged revisions 316265 via svnmerge from 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	15 years ago
Sean Bright	a52395aaee	Merged revisions 316206 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316206 \| seanbright \| 2011-05-03 14:17:36 -0400 (Tue, 03 May 2011) \| 8 lines If we aren't interested in events, don't generate the FullyBooted event on AMI login. (closes issue #19089) Reported by: bklang Patches: issue19089-1.8-r316204.patch uploaded by seanbright (license 71) Tested by: seanbright ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316213 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	237d47b010	Clears exception flag during ast_read when func_jitterbuffer is enabled git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@315855 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	98f94daf88	Merged revisions 315810 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r315810 \| russell \| 2011-04-27 10:55:48 -0500 (Wed, 27 Apr 2011) \| 2 lines Set the copyright year to 2011 in the startup message. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@315811 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	8d2a71877a	Merged revisions 315644 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r315644 \| twilson \| 2011-04-26 14:39:01 -0700 (Tue, 26 Apr 2011) \| 32 lines Merged revisions 315643 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r315643 \| twilson \| 2011-04-26 14:27:44 -0700 (Tue, 26 Apr 2011) \| 25 lines Merged revisions 315596 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r315596 \| twilson \| 2011-04-26 14:16:10 -0700 (Tue, 26 Apr 2011) \| 18 lines 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/trunk@315670 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	24b6939496	Merged revisions 315645 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r315645 \| rmudgett \| 2011-04-26 17:14:31 -0500 (Tue, 26 Apr 2011) \| 21 lines The 'e' special extension fails to trigger in at least two cases. The 'e' extension is a fall back for the 'i', 't', or 'T' extensions if any of them do not exist. Many of the places the 'e' extension was supposed to be invoked fail because the priority was set wrong. There were two places where the 'e' extension was not even checked for fall back. * Made invoke the 'e' extension similarly to the previous 'i', 't', or 'T' extension check and added the 'e' extension as a fall back to the two missing locations. * Prioritized and optimized some hangup tests associated with the 'e' extension. (closes issue #19136) Reported by: kshumard Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1196/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@315649 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	079e794b1c	Merged revisions 314628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r314628 \| mnicholson \| 2011-04-21 13:24:05 -0500 (Thu, 21 Apr 2011) \| 27 lines Merged revisions 314620 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r314620 \| mnicholson \| 2011-04-21 13:22:19 -0500 (Thu, 21 Apr 2011) \| 20 lines Merged revisions 314607 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r314607 \| mnicholson \| 2011-04-21 13:19:21 -0500 (Thu, 21 Apr 2011) \| 14 lines Added limits to the number of unauthenticated sessions TCP based protocols are allowed to have open simultaneously. Also added timeouts for unauthenticated sessions where it made sense to do so. Unrelated, the manager interface now properly checks if the user has the "system" privilege before executing shell commands via the Originate action. AST-2011-005 AST-2011-006 (closes issue #18787) Reported by: kobaz (related to issue #18996) Reported by: tzafrir ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314666 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	7f23115ad2	New HD ConfBridge conferencing application. 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	15 years ago
David Vossel	18d591cb48	Introduction of the JITTERBUFFER dialplan function. Review: https://reviewboard.asterisk.org/r/1157/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314509 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	632cd26411	Merged revisions 314358 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r314358 \| twilson \| 2011-04-19 22:25:15 -0700 (Tue, 19 Apr 2011) \| 4 lines Initialize track pointer ast_reentrancy_init checks to see if it is NULL before initializing with calloc ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314359 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	02821fc5b4	Merged revisions 314251 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r314251 \| lmadsen \| 2011-04-19 10:42:10 -0500 (Tue, 19 Apr 2011) \| 8 lines Use SSLv23_client_method instead of old SSLv2 only. (closes issue #19095) (closes issue #19138) Reported by: tzafrir Patches: no_ssl2.diff uploaded by tzafrir (license 46) Tested by: russell, chazzam ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314252 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	4b4549106b	Merged revisions 314017 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r314017 \| dvossel \| 2011-04-18 08:41:06 -0500 (Mon, 18 Apr 2011) \| 17 lines 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/trunk@314018 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	05ddffccc4	Merged revisions 313860 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r313860 \| jrose \| 2011-04-15 10:08:05 -0500 (Fri, 15 Apr 2011) \| 17 lines Merged revisions 313859 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r313859 \| jrose \| 2011-04-15 09:58:37 -0500 (Fri, 15 Apr 2011) \| 10 lines Fix a Tab Completion bug that occurs due to multiple matches on a substring. Makes word_match function in cli.c repeat a search for a command string until a proper match is found or the string is searched to the last point. (closes issue #17494) Reported by: ffossard Review: https://reviewboard.asterisk.org/r/1180/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313867 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	ae2926b5d0	Add Device State Information CCSS for Generic Devices. Add Asterisk Device State information and callbacks to the Call Completion Supplemental Services for generic agents. There are currently not many devices that have native support for CCSS. Even as the devices become available there may be other reasons why one may choose to not take advantage of the native abilities and stick with the generic implementation. The generic implementation is quite capable and could be greatly enhanced by adding device state capabilities. A phone could then subscribe to the device state with a BLF key in conjunction with Asterisk hints. The advantages of the device state information would allow a single button to: request CCSS, cancel a CCSS request, and display the current state of a CCSS request. For example, you may have a single button that when not lit, there is no active CCSS request. When you press that button, the dialplan can query the DEVICE_STATE() associated with that caller to determine whether they should be calling CallCompletionRequest() or CallCompletionCancel(). If there is currently a pending request, then the dialplan would cancel it. This also has the advantage of showing the true state of a request, which is an asynchronous call, even when CallCompletionRequest() thinks it was successful. The actual request could ultimately fail. Once lit, further feedback can be provided to the caller about the current state of their request since it will be updated by the CCSS State Machine as appropriate. The DEVICE_STATE mapping is configurable since the BLF being used on a given phone type may vary. The idea is to allow some level of customization as to the phone's behavior. As an example, you may want the BLF key to go solid once you have requested a callback. You may then want the LED to blink (typically ringing) when either the callback is in process, which is a visual indication that the incoming call is the desired callback. You may want it to blink when the callee is ready but you are busy, giving you a visual indication that the target is available as you may want to get off the line so that the callback can be successful. Device state information is sent back via the ast_devstate_prov_add() callback for any generic CCSS device as it traverses through the state machine. You simply provide a map between CC_STATE values and the corresponding AST_DEVICE state values. You could then generate hints against these states similar to what is possible today with Custom Devstates or MeetMe states. For example, you may have an extension 3000 that is currently associated with device SIP/3000. You could then create a feature code for that extension that may look something like: exten => 823000,hint,ccss:sip/3000 You would then subscribe a BLF button to 823000 which would point to the dialplan that handled CCSS requests/cancels using the available DEVICE_STATE() information about ccss:sip/3000 to make the decision about what to do. (closes issue #18788) Reported by: p_lindheimer Patches: ccss.trunk.18788.patch uploaded by p lindheimer (license 558) Modified with final reviewboard comments. Tested by: p_lindheimer, loloski Review: https://reviewboard.asterisk.org/r/1105/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313744 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	c16d39ea83	Merged revisions 313588 via svnmerge from 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	15 years ago
Richard Mudgett	530afe7d97	Merged revisions 313366 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r313366 \| rmudgett \| 2011-04-11 17:27:25 -0500 (Mon, 11 Apr 2011) \| 2 lines Added "Connected Line ID" and "Connected Line ID Name" to "core show channel" output. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313367 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	68dd87ef0d	Merged revisions 313048 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r313048 \| jrose \| 2011-04-07 08:35:33 -0500 (Thu, 07 Apr 2011) \| 16 lines Merged revisions 313047 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r313047 \| jrose \| 2011-04-07 08:23:01 -0500 (Thu, 07 Apr 2011) \| 9 lines Makes parking lots clear and rebuild properly when features reload is invoked from CLI Before, default parkinglot in context parkedcalls with ext 700 would always be present and when reload was invoked, the previous parkinglots would not be cleared. (closes issue #18801) Reported by: mickecarlsson Review: https://reviewboard.asterisk.org/r/1161/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313049 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	a77fd545ab	Merged revisions 312766 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r312766 \| mnicholson \| 2011-04-05 09:14:50 -0500 (Tue, 05 Apr 2011) \| 22 lines Merged revisions 312764 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r312764 \| mnicholson \| 2011-04-05 09:13:07 -0500 (Tue, 05 Apr 2011) \| 15 lines Merged revisions 312761 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r312761 \| mnicholson \| 2011-04-05 09:10:34 -0500 (Tue, 05 Apr 2011) \| 8 lines Limit the number of unauthenticated manager sessions and also limit the time they have to authenticate. AST-2011-005 (closes issue #18996) Reported by: tzafrir Tested by: mnicholson ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312767 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	75594e6e4a	Merged revisions 312461 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r312461 \| rmudgett \| 2011-04-01 16:31:39 -0500 (Fri, 01 Apr 2011) \| 25 lines CallCompletionRequest()/CallCompletionCancel() exit non-zero if fail. The CallCompletionRequest()/CallCompletionCancel() dialplan applications exit nonzero on normal failure conditions. The nonzero exit causes the dialplan to hangup immediately. The dialplan author has no opportunity to report success/failure to the user. * Made always return zero so the dialplan can continue. * Made set CC_REQUEST_RESULT/CC_REQUEST_REASON and CC_CANCEL_RESULT/CC_CANCEL_REASON channel variables respectively. Also documented the values set. * Reduced the warning about no core instance in CallCompletionCancel() to a debug message. It is a normal event and should not be output at the WARNING level. (closes issue #18763) Reported by: p_lindheimer Patches: ccss.patch uploaded by p lindheimer (license 558) Modified Tested by: p_lindheimer, rmudgett JIRA SWP-3042 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312462 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	846cfa0ef0	New Feature for chan_dahdi. 4 length pattern matching. In chan_dahdi.conf, the user can now use length 4 patterns in addition to the usual length 2 patterns. The s ntax remains the same and the method used to track the pattern history will only change when using the length 4 patterns. (closes issue SWP-3250) Code: jrose rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312384 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	3731fd9ccc	Merged revisions 312286,312288 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r312286 \| tilghman \| 2011-04-01 05:44:33 -0500 (Fri, 01 Apr 2011) \| 2 lines Reload must react correctly against a possibly changed table, so dropping the conditional reload flag. ................ r312288 \| tilghman \| 2011-04-01 05:58:45 -0500 (Fri, 01 Apr 2011) \| 21 lines 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/trunk@312289 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	581bfad2f3	Merged revisions 311141 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r311141 \| mnicholson \| 2011-03-17 10:00:33 -0500 (Thu, 17 Mar 2011) \| 11 lines Merged revisions 311140 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r311140 \| mnicholson \| 2011-03-17 09:58:52 -0500 (Thu, 17 Mar 2011) \| 4 lines Don't write items to the manager socket twice. AST-2011-003 (closes issue 0018987) Reported by: ks-steven ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311142 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	4ae1cb9456	Merged revisions 310999 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r310999 \| twilson \| 2011-03-16 14:47:59 -0500 (Wed, 16 Mar 2011) \| 18 lines Merged revisions 310998 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r310998 \| twilson \| 2011-03-16 14:46:36 -0500 (Wed, 16 Mar 2011) \| 11 lines Fix crash on fdopen failure See security advisory AST-2011-004 (closes issue #18845) Reported by: cmaj Patches: patch-main-tcptls-1.8.3-rc2-open-session-crash-take2.diff.txt uploaded by cmaj (license 830) patch-main-tcptls-1.8.3-rc2-open-session-crash-take3.diff.txt uploaded by cmaj (license 830) Tested by: cmaj, twilson ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311001 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	d0846ea207	Merged revisions 310993 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r310993 \| twilson \| 2011-03-16 14:26:57 -0500 (Wed, 16 Mar 2011) \| 11 lines Merged revisions 310992 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r310992 \| twilson \| 2011-03-16 14:23:03 -0500 (Wed, 16 Mar 2011) \| 4 lines Don't keep trying to write to a closed connection See security advisory AST-2011-003. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311000 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	d958ca6953	Merged revisions 310902 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r310902 \| twilson \| 2011-03-16 12:19:57 -0500 (Wed, 16 Mar 2011) \| 43 lines Merged revisions 310889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r310889 \| twilson \| 2011-03-16 12:03:27 -0500 (Wed, 16 Mar 2011) \| 36 lines Merged revisions 310888 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r310888 \| twilson \| 2011-03-16 11:58:42 -0500 (Wed, 16 Mar 2011) \| 29 lines Don't delay DTMF in core bridge while listening for DTMF features This patch is mostly the work of Olle Johansson. I did some cleanup and added the silence generating code if transmit_silence is set. When a channel listens for DTMF in the core bridge, the outbound DTMF is not sent until we have received DTMF_END. For a long DTMF, this is a disaster. We send 4 seconds of DTMF to Asterisk, which sends no audio for those 4 seconds. Some products see this delay and the time skew on RTP packets that results and start ignoring the audio that is sent afterward. With this change, the DTMF_BEGIN frame is inspected and checked. If it matches a feature code, we wait for DTMF_END and activate the feature as before. If transmit_silence=yes in asterisk.conf, silence is sent if we paritally match a multi-digit feature. If it doesn't match a feature, the frame is forwarded along with the DTMF_END without delay. By doing it this way, DTMF is not delayed. (closes issue #15642) Reported by: jasonshugart Patches: issue_15652_dtmf_ast-1.4.patch.txt uploaded by twilson (license 396) Tested by: globalnetinc, jde (closes issue #16625) Reported by: sharvanek Review: https://reviewboard.asterisk.org/r/1092/ Review: https://reviewboard.asterisk.org/r/1125/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310941 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alec L Davis	858c11f075	Merged revisions 310781 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r310781 \| alecdavis \| 2011-03-15 14:00:55 +1300 (Tue, 15 Mar 2011) \| 10 lines core show locks: display ThreadID in hexadecimal Allow easier cross referencing of thread ID's with GDB backtraces (closes issue #18968) Reported by: alecdavis Patches: bug18968.diff.txt uploaded by alecdavis (license 585) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310833 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	de7280fc7d	Merged revisions 310636 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r310636 \| rmudgett \| 2011-03-14 11:50:59 -0500 (Mon, 14 Mar 2011) \| 39 lines Merged revisions 310635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r310635 \| rmudgett \| 2011-03-14 11:47:54 -0500 (Mon, 14 Mar 2011) \| 32 lines Merged revisions 310633 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r310633 \| rmudgett \| 2011-03-14 11:38:24 -0500 (Mon, 14 Mar 2011) \| 25 lines "Caller*ID failed checksum" on Wildcard TDM2400P and TDM410 The last character in the caller id message is getting a framing error. The checksum is the last character in the message. A framing error in the checksum could be because: 1) The sender did not send a full stop bit. 2) The sender cut off the FSK carrier too soon. 3) The sender opted to send zero of the specified zero to 10 trailing mark bits and round-off errors in the code resulted in the code not being where it thought it was in the demodulated bit stream. Bit 8 of 'b' is set when parity error. Bit 9 of 'b' is set when framing error. Made ignore the framing and parity error bits if the errored character is the checksum. We can tolerate a framing/parity error there. The checksum character validates the message. (closes issue #18474) Reported by: nivek Patches: callerid.c.1.patch uploaded by nivek (license 636) (with modifications) Tested by: nivek ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310637 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	16c5dda8ab	Fixes null reference bug introduced by audio hook changes that affects various OS distributions. Thanks David. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310547 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	6e36042f64	Mix Monitor: Now with r and t options. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310373 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alec L Davis	c7c0664bc4	Merged revisions 310287 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r310287 \| alecdavis \| 2011-03-11 19:47:44 +1300 (Fri, 11 Mar 2011) \| 17 lines remote_bridge_loop: prevent segfault when after transfer of IAX2 of DAHDI call If the channel condition is one of the following after breaking out of the loop, don't try to update_peer (where x = 0/1) 1). ZOMBIE 2). cx->tech_pvt != pvtx 3). gluex != ast_rtp_instance_get_glue(cx->tech->type)) (closes issue #18781) Reported by: alecdavis Patches: bug18781.diff3.txt uploaded by alecdavis (license 585) Tested by: alecdavis, ZX81 Review: https://reviewboard.asterisk.org/r/1128/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310288 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	254092f8f6	Merged revisions 310240 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r310240 \| twilson \| 2011-03-10 10:05:45 -0600 (Thu, 10 Mar 2011) \| 13 lines Add \r\n to remaining http headers passed to ast_http_send r309204 changed the behavior of ast_http_send. It now requires headers to be passed with trailing \r\n. This change updates the remaining instances in the code that did not pass the \r\n. (closes issue #18186) Reported by: nivaldomjunior Patches: res_phoneprov.c.diff uploaded by lathama (license 1028) manager.diff.txt uploaded by twilson (license 396) Tested by: lathama ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310241 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	6de1332214	Merged revisions 309808 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309808 \| tilghman \| 2011-03-06 18:54:42 -0600 (Sun, 06 Mar 2011) \| 14 lines 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/trunk@309809 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	798212c828	Merged revisions 309678 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309678 \| tilghman \| 2011-03-05 04:29:30 -0600 (Sat, 05 Mar 2011) \| 14 lines Merged revisions 309677 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309677 \| tilghman \| 2011-03-05 04:28:24 -0600 (Sat, 05 Mar 2011) \| 7 lines Missed part of the conversion when we started passing ppid to astcanary. (closes issue #18850) Reported by: viraptor Patches: canary_ppid.patch uploaded by viraptor (license 543) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309679 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	a7bfa10472	Add HangupRequest manager event, to specify when/where a channel gets hung up. (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	16 years ago
Jason Parker	dc616cfe2c	Merged revisions 309204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309204 \| qwell \| 2011-03-01 16:25:44 -0600 (Tue, 01 Mar 2011) \| 7 lines Fix consistency of CRLFs on HTTP headers that get sent out. (closes issue #18186) Reported by: nivaldomjunior Patches: 18186-httpheadernewline.diff uploaded by qwell (license 4) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309209 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	e5dc4c2d8e	Merged revisions 309035 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309035 \| tilghman \| 2011-02-28 05:10:28 -0600 (Mon, 28 Feb 2011) \| 15 lines 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/trunk@309036 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	642d6c306c	Merged revisions 308903 via svnmerge from 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	16 years ago
Terry Wilson	f88ef55d74	Merged revisions 308815 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r308815 \| twilson \| 2011-02-24 11:57:18 -0600 (Thu, 24 Feb 2011) \| 26 lines Merged revisions 308814 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r308814 \| twilson \| 2011-02-24 11:54:49 -0600 (Thu, 24 Feb 2011) \| 19 lines Merged revisions 308813 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r308813 \| twilson \| 2011-02-24 11:42:16 -0600 (Thu, 24 Feb 2011) \| 12 lines Don't broadcast FullyBooted to every AMI connection The FullyBooted event should not be sent to every AMI connection every time someone connects via AMI. It should only be sent to the user who just connected. (closes issue #18168) Reported by: FeyFre Patches: bug0018168.patch uploaded by FeyFre (license 1142) Tested by: FeyFre, twilson ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308816 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	0658c5ec2e	Merged revisions 308723 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r308723 \| mnicholson \| 2011-02-24 09:06:14 -0600 (Thu, 24 Feb 2011) \| 16 lines Merged revisions 308722 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r308722 \| mnicholson \| 2011-02-24 08:59:41 -0600 (Thu, 24 Feb 2011) \| 9 lines Merged revisions 308721 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r308721 \| mnicholson \| 2011-02-24 08:54:56 -0600 (Thu, 24 Feb 2011) \| 2 lines silence gcc 4.2 compiler warning ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308724 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	79e041f856	Fix compiler warning. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308624 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	d760e81f37	Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd 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	16 years ago
Andrew Latham	736133f874	Use ast_debug for console logging Guessed the log levels based on info that level 3 is the soft roof. Can we create a page / document to define the levels? git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308527 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	dad4fecc03	Merged revisions 308416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r308416 \| mnicholson \| 2011-02-21 09:02:20 -0600 (Mon, 21 Feb 2011) \| 19 lines Merged revisions 308414 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r308414 \| mnicholson \| 2011-02-21 09:00:22 -0600 (Mon, 21 Feb 2011) \| 12 lines Merged revisions 308413 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r308413 \| mnicholson \| 2011-02-21 08:57:15 -0600 (Mon, 21 Feb 2011) \| 5 lines Properly check the bounds of arrays when decoding UDPTL packets. Also, remove broken support for receiving UDPTL packets larger than 16k. That shouldn't ever happen anyway. AST-2011-002 FAX-281 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308417 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago

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3128 Commits (6d04d190dcf19adaa63e2c514b28ef679a181b00)