2728 Commits (5cf26dcdacd3153eebad1d41e2cdba2ff7638fbb)

Author	SHA1	Message	Date
Tilghman Lesher	5cf26dcdac	Implicitly sending a progress signal breaks some applications. ... Call Progress() in your dialplan if you explicitly want progress to be sent. (Reverts change 216430, closes issue #15957) Reported by: Pavel Troller on the Asterisk-Dev mailing list http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@220288 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	f4837ecbef	When IMAP variables were changed during a reload, Voicemail did not use the new values. ... This change introduces a configuration version variable, which ensures that connections with the old values are not reused but are allowed to expire normally. (closes issue #15934) Reported by: viniciusfontes Patches: 20090922__issue15934.diff.txt uploaded by tilghman (license 14) Tested by: viniciusfontes git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@219816 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	cefd4b7b03	If the user enters the same password as before, don't signal an error when the change does nothing. ... (closes issue #15492) Reported by: cbbs70a Patches: 20090713__issue15492.diff.txt uploaded by tilghman (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@218730 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	3913dcbc40	Ensure FollowMe sets language in channels it creates. ... Also, not in the original bug report, but related fields are accountcode and musicclass, and the inheritance of datastores. (closes issue #15372) Reported by: Romik Patches: 20090828__issue15372.diff.txt uploaded by tilghman (license 14) Tested by: cervajs git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@218577 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	12f92cf9c6	Don't say "Please try again" if we don't give the user another chance to try again. ... (issue #15055, SWP-129) Reported by: jthurman git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@218331 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	903460b848	Ensure we don't pickup ourselves when doing pickup by exten. ... (closes issue #15100) Reported by: lmsteffan Patches: (modified) pickup.patch uploaded by lmsteffan (license 779) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@218223 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	269854a47d	Don't ring another channel, if there's not enough time for a queue member to answer. ... (Fixes AST-228) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@217989 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	b10343fd19	When MOH is playing on the channel, announcements sent through the conference are not heard. ... (closes issue #14588) Reported by: voipas Patches: 20090716__issue14588__2.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen, twisted, tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@217156 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	05899c19a1	Make apps send PROGRESS control frame for early media and fix too early media issue in SIP ... The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI links *before* any call progress. The SIP channel receives these frames and by default signals 183 Session progress and starts sending media. This will cause phones to play silence and ignore the later 180 ringing message. A bad user experience. The fix is twofold: - We discovered that asterisk apps that support early media ("noanswer") did not send any PROGRESS frame to indicate early media. Fixed. - We introduce a setting in chan_sip so that users can disable any relay of media frames before the outbound channel actually indicates any sort of call progress. In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions of Asterisk, this will be enabled. We don't assume that it will change your Asterisk phone experience - only for the better. We encourage third-party application developers to make sure that if they have applications that wants to send early media, add a PROGRESS control frame transmission to make sure that all channel drivers actually will start sending early media. This has not been the default in Asterisk previous to this patch, so if you got inspiration from our code, you need to update accordingly. Sorry for the trouble and thanks for your support. This code has been running for a few months in a large scale installation (over 250 servers with PRI and/or BRI links to old PBX systems). That's no proof that this is an excellent patch, but, well, it's tested :-) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@216430 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Dwayne M. Hubbard	fad010ebe7	Use strrchr() so SoftHangup will correctly truncate multi-hyphen channel names ... In general channel names are in the form Foo/Bar-Z, but the channel name could have multiple hyphens and look like Foo/B-a-r-Z. Use strrchr to truncate the channel name at the last hyphen. (closes issue #15810) Reported by: dhubbard Patches: dw-softhangup-1.4.patch uploaded by dhubbard (license 733) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@215270 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	d581b4216a	Make all the symbols for the C-client callbacks global ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@213283 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	941ed1b2fe	Fixes memory leak caused by incorrectly freeing mixmonitor ... (closes issue #15699) Reported by: edantie Patches: mixmonitor.patch uploaded by edantie (license 862) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@213103 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	b35312b15b	This patch adds additional checking when generating queue log TRANSFER events. ... The additional checks prevent generation of false TRANSFER events in certain situations. (closes issue #14536) Reported by: aragon Patches: queue-log-xfer-fix1.diff uploaded by mnicholson (license 96) Tested by: aragon, mnicholson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@211953 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	63cc189747	AST-2009-005 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@211528 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	a56006702e	Resolve a deadlock involving app_chanspy and masquerades. ... (ABE-1936) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@211112 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	17656694c3	QUEUE_MEMBER_LIST _really_ wants the interface name, not the membername. ... This is a partial revert of revision 82590, which was an attempted cleanup, but in reality, it broke QUEUE_MEMBER_LIST, which has always been intended as a method by which component interfaces could be queried from the queue. Membername isn't useful here, because that field cannot be used to obtain further information about the member. See the documentation on QUEUE_MEMBER_LIST, RemoveQueueMember, QUEUE_MEMBER_PENALTY, and the various AMI commands which take a member argument for further justification. (closes issue #15664) Reported by: rain Patches: app_queue-queue_member_list.diff uploaded by rain (license 327) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@211038 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	ca0f026f41	Reverting index() fix, applying a different methodology, based upon developer discussions. ... (related to issue #15639) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@210066 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	a687e8c53f	Modify how Playtones() is used in Milliwatt() to resolve gain issue. ... When Milliwatt() was changed internally to use Playtones() so that the proper tone was used, it introduced a drop in gain in the output signal. So, use the playtones API directly and specify a volume argument such that the output matches the gain of the original Milliwatt() code. (closes issue #15386) Reported by: rue_mohr Patches: issue_15386.rev2.diff uploaded by russell (license 2) Tested by: rue_mohr git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@209838 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	0660bfbe74	Don't impose an arbitrary limit on member lines in queues.conf ... I know what some of you are thinking: "UGH! Mark, why are you using ast_strdup and ast_free for the string when you can just use ast_strdupa and let the memory free itself?! Have the bats been chewing on your brain again?" Based on past experiences, I don't like using ast_strdupa inside a loop. It's a good way to potentially exhaust stack space. Also, since this only happens when reloading queues, I don't think that heap allocations and frees are going to be a huge problem. (closes issue #15559) Reported by: amorsen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@208622 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	55d9c2ecaf	Do not log an ERROR if autoservice_stop() returns -1. ... This does not indicate an error. A return of -1 just means that the channel has been hung up. (reported in #asterisk-dev) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@208592 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	16d3415cf3	Prevent phantom calls to queue members. ... If a caller were to hang up while a periodic announcement or position were being said, the return value for those functions would incorrectly indicate that the caller was still in the queue. With these changes, the problem does not occur. (closes issue #14631) Reported by: latinsud Patches: queue_announce_ghost_call2.diff uploaded by latinsud (license 745) (with small modification from me) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@205349 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	a3848ec74c	Place unlock of mutex in an else block so that it does not get unlocked twice. ... (closes issue #15400) Reported by: aragon git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@204012 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Brooks	64e75ecf80	Fixing voicemail's error in checking max silence vs min message length ... Max silence was represented in milliseconds, yet vmminsecs (minmessage) was represented as seconds. Also, the inequality was reversed. The warning, if triggered, was "Max silence should be less than minmessage or you may get empty messages", which should have been logged if max silence was greater than minmessage, but the check was for less than. Also, conforming if statement to coding guidelines. closes issue #15331) Reported by: markd Review: https://reviewboard.asterisk.org/r/293/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@203719 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	86c204f34c	StopMixMonitor race condition (not giving up file immediately) ... StopMixMonitor only indicates to the MixMonitor thread to stop writing to the file. It does not guarantee that the recording's file handle is available to the dialplan immediately after execution. This results in a race condition. To resolve this, the filestream pointer is placed in a datastore on the channel. When StopMixMonitor is called, the datastore is retrieved from the channel and the filestream is closed immediately before returning to the dialplan. Documentation indicating the use of StopMixMonitor to free files has been updated as well. (closes issue #15259) Reported by: travisghansen Tested by: dvossel Review: https://reviewboard.asterisk.org/r/283/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@201423 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	94fa4d11b5	Improve support for media paths that can generate multiple frames at once. ... There are various media paths in Asterisk (codec translators and UDPTL, primarily) that can generate more than one frame to be generated when the application calling them expects only a single frame. This patch addresses a number of those cases, at least the primary ones to solve the known problems. In addition it removes the broken TRACE_FRAMES support, fixes a number of bugs in various frame-related API functions, and cleans up various code paths affected by these changes. https://reviewboard.asterisk.org/r/175/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@200991 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	48253ef901	Treat an empty FORWARD_CONTEXT the same way we treat a missing one. ... (closes issue #15056) Reported by: p_lindheimer Patches: 05292009_bug15056.diff uploaded by seanbright (license 71) Tested by: p_lindheimer git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@198251 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Leif Madsen	ad5f20b94b	Update MixMonitor documentation. ... Updated the MixMonitor documentation for the 'b' option so that it is more obvious that you must not optimize awat the Local channel when using this option. (issue #14829) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@197895 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	3268149a1f	Add flags to chanspy audiohook so that audio stays in sync. ... There are two flags being added to the chanspy audiohook here. One is the pre-existing AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set, we ensure that the read and write slinfactories on the audiohook do not skew beyond a certain tolerance. In addition, there is a new audiohook flag added here, AST_AUDIOHOOK_SMALL_QUEUE. With this flag set, we do not allow for a slinfactory to build up a substantial amount of audio before flushing it. For this particular issue, this means that the person spying on the call will hear the conversations in real time with very little delay in the audio. (closes issue #13745) Reported by: geoffs Patches: 13745.patch uploaded by mmichelson (license 60) Tested by: snblitz git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@197537 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	a2fd7f4e47	Fix handling of the 'state_interface' option of the 'queue add member' CLI ... command. This change relates to r184980, which was a backport of the state interface changes to app_queue from trunk. trunk and all of the 1.6.x branches are not affected. 'queue add member' allows for specifying an interface to use for device state when adding a queue member via CLI, but the validation code was not properly updated to reflect this optional argument. (closes issue #15198) Reported by: loloski Patches: 05272009_app_queue.diff uploaded by seanbright (license 71) Tested by: loloski git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@197024 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	65494bfdf7	Fix a bug where the MeetMe option 'D' did not actually prompt for the pin. ... (closes issue #15050) Reported by: pmhaddad git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@195635 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	6de96b9120	Ensure thread keys are initialized before attempting to access them. ... (closes issue #14889) Reported by: jaroth Patches: app_voicemail.c.patch uploaded by msirota (license 758) Tested by: msirota, BlargMaN git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@195520 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	efb22ba096	Add a similar dependency on SMDI for voicemail as already exists for ADSI. ... (closes issue #14846) Reported by: pj Patches: 20090413__bug14846__1.4.diff.txt uploaded by tilghman (license 14) 20090507__issue14846__1.6.0.diff.txt uploaded by tilghman (license 14) 20090507__issue14846__1.6.1.diff.txt uploaded by tilghman (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@195366 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	bec8573c37	This change modifies app_queue to properly generate CDR records in failure ... situations. This involves setting a proper cdr disposition coresponding to the given failure condition and ensuring the proper information is stored in the cdr record. (closes issue #13691) Reported by: dferrer Tested by: mnicholson (closes issue #13637) Reported by: atis Tested by: atis git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@194028 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	f6ba2472bd	Avoid initializing routines if the authentication fails. Fixes a crash (RR) issue. ... (closes issue #14508) Reported by: tiziano Patches: 20090221_2_wrongmailbox.diff.txt uploaded by tiziano (license 377) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@193955 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	8425d87bdf	Move 300 bytes around on the stack, to make more room for an extension buffer. ... This allows more concurrent extensions to be copied for a single voicemail, without creating a possibility of upsetting existing users, where a dialplan could run out of stack space where it had run fine before. Alternatively, we could have allocated off the heap, but that is a larger change and would have increased the chance for instability introduced by this change. This is really solved starting in 1.6.0.11, as the use of an ast_str buffer allows an unlimited number of extensions (up to available memory). We additionally create a new warning message when the buffer length is exceeded, permitting administrators to see an issue after the fact, whereas previously the list was silently truncated. (closes issue #14739) Reported by: p_lindheimer Patches: 20090417__bug14739.diff.txt uploaded by tilghman (license 14) Tested by: p_lindheimer git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@193755 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	6b15b32783	Fix a bug where the followme application would continue trying numbers after the caller hung up. ... (closes issue #13624) Reported by: sgenyuk git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@192429 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	b67282e2fd	Fix a bug which resulted from the Hebrew voicemail commit. ... This fixes a case where a certain message could get played twice. (closes issue #13155) Reported by: greenfieldtech Patches: app_voicemail.c.multi-lang-patch uploaded by greenfieldtech (license 369) Tested by: greenfieldtech git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@191778 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	972d9bf53c	Kevin has informed me that thi sort of thing is not necessary. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@191629 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	85a8916552	Move static buffers to outside for loops in app_chanspy. ... Similar to seanbright's commit 191422, this moves some static buffers to be defined outside of for loops since it is undefined if memory will be re-used or if the stack will grow with each iteration of the loop. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@191628 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	b5ec450104	Move the defintion of the a couple arrays out of loops. ... According to Kevin, it is unspecified as to whether a variable defined inside a block is allocated once by the compiler or for each pass through the block (loops being the only interesting case), so just define these before we get into our loop to be sure. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@191422 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	603a56aa69	Fix a crash in app_queue with very long member lists. ... A user reported via #asterisk that with very long lists of members, a crash occurs in ast_strdupa, so just use a single buffer and ast_copy_string instead of stack allocating copys of each interface name. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@191041 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	59ee389e31	Update CDR appropriately when AST_CAUSE_NO_ANSWER is set ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@189465 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	ef9ef40c19	Don't treat a NOANSWER like a CHANUNAVAIL ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@189463 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	caef916825	Umask should not be exported into global namespace. ... (closes issue #14912) Reported by: jcapp git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@188773 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	1cb43cfa75	Permit zero-length text messages in SIP. ... (Related to an issue posted to the -users list, subject "AEL2, BASE64_DECODE and hexadecimal") git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@187362 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	8e31a331b4	Fix a crash due to too few arguments to RetryDial. ... (closes issue #14852) Reported by: junky Patches: retry_fix.diff uploaded by junky (license 177) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@187135 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	56e7ca00c1	Fix Macro documentation to match current (and intended) behavior. ... (See -dev mailing list) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@186775 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	e2c30564bf	Revert commit 186445 because it causes the build to fail when IMAP_STORAGE is used. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@186565 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	3d9585f4ba	Found a conflict in the last commit, due to multiple targets ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@186445 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	6d08bad538	Distinguish in a sent email between simple sends and forwards. ... (closes issue #11678) Reported by: jamessan Patches: 20090330__bug11678.diff.txt uploaded by tilghman (license 14) Tested by: tilghman, lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@186415 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago