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${ noResults }
2741 Commits (c79e242adabb4fe2de1781fae9ad76741d8c2061)
Author | SHA1 | Message | Date |
---|---|---|---|
|
c79e242ada |
fixes solaris segfault on dial with verbosity >= 3
(closes issue #16193) Reported by: asgaroth Patches: bug_16193_1.4.21.2_vers.diff uploaded by snuffy (license 35) Tested by: asgaroth, snuffy git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@231235 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
182c57ce17 |
fixes MixMonitor thread not exiting when StopMixMonitor is used
(closes issue #16152) Reported by: AlexMS Patches: stopmixmonitor_1.4.diff uploaded by dvossel (license 671) Tested by: dvossel, AlexMS Review: https://reviewboard.asterisk.org/r/424/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@230508 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
2ef94c5196 |
Document a limitation in the AVAILSTATUS variable from ChanIsAvail and provide
a workaround for it that does not change existing behavior. (closes issue #14426) Reported by: macli git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@229965 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
3c256882d6 |
This patch modifies the Dial application to monitor the calling channel for hangups while playing back announcements.
(closes issue #16005) Reported by: falves11 Patches: dial-announce-hangup-fix1.diff uploaded by mnicholson (license 96) Tested by: mnicholson, falves11 Review: https://reviewboard.asterisk.org/r/407/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@227827 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
ed413ec76c |
Fix a bug where the recorded privacy introduction file would not get removed if the caller hung up
while the called party had not yet answered. This was fixed by introducing an argument to the 'n' option which, when enabled, removes the introduction file under all scenarios. This was done to preserve the behavior that has existed for quite some time. (closes issue #14674) Reported by: ulogic Patches: bug14674.patch uploaded by jpeeler (license 325) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@226889 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
6e8a455534 |
Fix documentation for ast_softhangup() and correct the misuse thereof.
(closes issue #16103) Reported by: majorbloodnok git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@225105 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
8699a5f158 |
Suffix is not needed for a match
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@225103 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
926a033bf9 |
Do not attempt early media bridging (ie: direct RTP setup) if options are enabled that should prevent it.
(closes issue #14763) Reported by: cupotka git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@224565 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
e3464ac40a |
Ensure ringing continues for branched calls after progress is received
While waiting for an answer, don't send progress for branched calls for which ringing was sent. (closes issue #15028) Reported by: fnordian git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@223804 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
6429db49ba |
Remove a duplicate ao2_iterator_destroy().
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@223550 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
a9317f6cbe |
Fix potential memory leak in app_dial.c
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@223213 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
2ad7cb7e87 |
Fix ao2_iterator API to hold references to containers being iterated.
See Mantis issue for details of what prompted this change. Additional notes: This patch changes the ao2_iterator API in two ways: F_AO2I_DONTLOCK has become an enum instead of a macro, with a name that fits our naming policy; also, it is now necessary to call ao2_iterator_destroy() on any iterator that has been created. Currently this only releases the reference to the container being iterated, but in the future this could also release other resources used by the iterator, if the iterator implementation changes to use additional resources. (closes issue #15987) Reported by: kpfleming Review: https://reviewboard.asterisk.org/r/383/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@222152 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
5d982dda69 |
Avoid a deadlock in chanspy, just in case the spyee is masqueraded and chanspy_ds_chan_fixup() is called with the channel locked.
(closes issue #15965) Reported by: atis Patches: chanspy-deadlock-fix1.diff uploaded by mnicholson (license 96) Tested by: atis git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@220907 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
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 |
16 years ago |
|
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 |
16 years ago |
|
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 |
16 years ago |
|
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 |
16 years ago |
|
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 |
16 years ago |
|
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 |
16 years ago |
|
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 |
16 years ago |
|
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 |
16 years ago |
|
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 |
16 years ago |
|
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 |
16 years ago |
|
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 |
16 years ago |
|
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 |
16 years ago |
|
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 |
16 years ago |
|
63cc189747 |
AST-2009-005
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@211528 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
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 |
16 years ago |
|
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 |
16 years ago |
|
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 |
16 years ago |
|
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 |
16 years ago |
|
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 |
16 years ago |
|
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 |
16 years ago |
|
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 |
16 years ago |
|
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 |
16 years ago |
|
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 |
16 years ago |
|
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 |
16 years ago |
|
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 |
16 years ago |
|
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 |
16 years ago |
|
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 |
16 years ago |
|
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 |
16 years ago |
|
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 |
16 years ago |
|
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 |
16 years ago |
|
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 |
16 years ago |
|
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 |
16 years ago |
|
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 |
16 years ago |
|
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 |
16 years ago |
|
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 |
16 years ago |
|
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 |
16 years ago |
|
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 |
16 years ago |