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releases/certified-20.7
master
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${ noResults }
19699 Commits (cef5e9d4340f94de8801aaecfb004a080df8a68f)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
094efca065 |
Option L() is milliseconds, not seconds.
> Change the verbose output of option L() to say milliseconds and not seconds > as the value is in milliseconds. > > (closes issue #18264) > Reported by: jacco > Patches: > app_dial_patch.txt uploaded by lmadsen (license 10) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302917 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
39a167a12a |
Support greetingsfolder as documented in voicemail.conf.sample.
(closes issue #17870) Reported by: edhorton Patches: __20100816-app_voicemail-greetingsfolder-support.txt uploaded by lmadsen (license 10) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302833 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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7289113d93 |
Turn a noisy verbose message into a debug message.
This can drown your console if you're using the AMI over HTTP. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302788 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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1ce0e49b02 |
Merged revisions 302671 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r302671 | rmudgett | 2011-01-19 15:21:56 -0600 (Wed, 19 Jan 2011) | 15 lines DTMF transfer plays the wrong sounds for wrong number or other call failure. * Set the default for features.conf.sample xferfailsound option to "beeperr" as documented instead of "pbx-invalid" and corrected the use of it in DTMF blind transfer (#1). * Improved DTMF blind transfer handling of wrong numbers. Most of the concerns in this issue were taken care of by the patch for issue 17999: Issues with DTMF triggered attended transfers. (closes issue #18379) Reported by: gincantalupo Tested by: rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302693 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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5d7f358fc2 |
Merged revisions 302663 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r302663 | tilghman | 2011-01-19 15:20:28 -0600 (Wed, 19 Jan 2011) | 2 lines Add some API documentation ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302675 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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676b6cf22f |
Kill zombies.
When we ast_safe_fork() with a non-zero argument, we're expected to reap our own zombies. On a zero argument, however, the zombies are only reaped when there aren't any non-zero forked children alive. At other times, we accumulate zombies. This code is forward ported from res_agi in 1.4, so that forked children are always reaped, thus preventing an accumulation of zombie processes. (closes issue #18515) Reported by: ernied Patches: 20101221__issue18515.diff.txt uploaded by tilghman (license 14) Tested by: ernied git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302599 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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7d8cdc8c05 |
Don't call strlen() when we only need to look at the next character or two.
(closes issue #18042) Reported by: wdoekes Patches: astsvn-inefficient-ast-uri-decode.patch uploaded by wdoekes (license 717) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302554 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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faaebdfa2b |
Remove an extraneous \r\n at the end of a parking manager events.
(closes issue #18363) Reported by: clegall_proformatique Patches: asterisk_1.8_295998_parking_manager_events_format.patch uploaded by clegall proformatique (license 1139) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302551 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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aa527d8ab2 |
Properly handle partial reads from fgets() when handling AGIs.
When fgets() failed with EAGAIN, we were continually decrementing the available space left in our buffer, resulting in botched command handling. (closes issue #16032) Reported by: notahat Patches: agi_buffer_patch2.diff uploaded by fnordian (license 110) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302548 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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24fbdc445b |
Make sure that h_length is set when we short-circuit out of ast_gethostbyname.
(closes issue #16135) Reported by: thedavidfactor Patches: utils.patch uploaded by thedavidfactor (license 903) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302504 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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c9c8b434f0 |
Handle 'Resource temporarily unavailable' error more gracefully.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302461 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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ed7b735466 |
Remove references to priorityjumping from the sample extensions.conf.
Priority jumping was removed from pbx_config in r68970. (closes issue #18622) Reported by: kshumard Patches: extensions.conf.sample.patch uploaded by kshumard (license 92) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302416 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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28cbf8801c |
Merged revisions 302311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r302311 | mnicholson | 2011-01-18 15:35:03 -0600 (Tue, 18 Jan 2011) | 4 lines URI encode the user part of the contact header. ABE-2705 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302313 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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7ca6a0e2bd |
Convert device state callbacks to ao2 objects to fix a deadlock in chan_sip.
Lock scenario presented here: Thread 1 holds ast_rdlock_contexts &conlock holds handle_statechange hints holds handle_statechange hint waiting for cb_extensionstate Locked Here: chan_sip.c line 7428 (find_call) Thread 2 holds handle_request_do &netlock holds find_call sip_pvt_ptr waiting for ast_rdlock_contexts &conlock Locked Here: pbx.c line 9911 (ast_rdlock_contexts) Chan_sip has an established locking order of locking the sip_pvt and then getting the context lock. So the as stated by the summary, the operations in thread 2 have been modified to no longer require the context lock. (closes issue #18310) Reported by: one47 Patches: statecbs_ao2.mk2.patch uploaded by one47 (license 23), modified by me Review: https://reviewboard.asterisk.org/r/1072/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302265 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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f78288c30b |
Merged revisions 302172 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r302172 | rmudgett | 2011-01-18 12:04:36 -0600 (Tue, 18 Jan 2011) | 88 lines Issues with DTMF triggered attended transfers. Issue #17999 1) A calls B. B answers. 2) B using DTMF dial *2 (code in features.conf for attended transfer). 3) A hears MOH. B dial number C 4) C ringing. A hears MOH. 5) B hangup. A still hears MOH. C ringing. 6) A hangup. C still ringing until "atxfernoanswertimeout" expires. For v1.4 C will ring forever until C answers the dead line. (Issue #17096) Problem: When A and B hangup, C is still ringing. Issue #18395 SIP call limit of B is 1 1. A call B, B answered 2. B *2(atxfer) call C 3. B hangup, C ringing 4. Timeout waiting for C to answer 5. Recall to B fails because B has reached its call limit. Because B reached its call limit, it cannot do anything until the transfer it started completes. Issue #17273 Same scenario as issue 18395 but party B is an FXS port. Party B cannot do anything until the transfer it started completes. If B goes back off hook before C answers, B hears ringback instead of the expected dialtone. ********** Note for the issue #17273 and #18395 fix: DTMF attended transfer works within the channel bridge. Unfortunately, when either party A or B in the channel bridge hangs up, that channel is not completely hung up until the transfer completes. This is a real problem depending upon the channel technology involved. For chan_dahdi, the channel is crippled until the hangup is complete. Either the channel is not useable (analog) or the protocol disconnect messages are held up (PRI/BRI/SS7) and the media is not released. For chan_sip, a call limit of one is going to block that endpoint from any further calls until the hangup is complete. For party A this is a minor problem. The party A channel will only be in this condition while party B is dialing and when party B and C are conferring. The conversation between party B and C is expected to be a short one. Party B is either asking a question of party C or announcing party A. Also party A does not have much incentive to hangup at this point. For party B this can be a major problem during a blonde transfer. (A blonde transfer is our term for an attended transfer that is converted into a blind transfer. :)) Party B could be the operator. When party B hangs up, he assumes that he is out of the original call entirely. The party B channel will be in this condition while party C is ringing, while attempting to recall party B, and while waiting between call attempts. WARNING: The ATXFER_NULL_TECH conditional is a hack to fix the problem. It will replace the party B channel technology with a NULL channel driver to complete hanging up the party B channel technology. The consequences of this code is that the 'h' extension will not be able to access any channel technology specific information like SIP statistics for the call. ATXFER_NULL_TECH is not defined by default. ********** (closes issue #17999) Reported by: iskatel Tested by: rmudgett JIRA SWP-2246 (closes issue #17096) Reported by: gelo Tested by: rmudgett JIRA SWP-1192 (closes issue #18395) Reported by: shihchuan Tested by: rmudgett (closes issue #17273) Reported by: grecco Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1047/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302173 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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a16089f243 |
Merged revisions 293493 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 [^] ........ r293493 | twilson | 2010-11-01 09:58:00 -0500 (Mon, 01 Nov 2010) | 14 lines Only offer codecs both sides support for directmedia When using directmedia, Asterisk needs to limit the codecs offered to just the ones that both sides recognize, otherwise they may end up sending audio that the other side doesn't understand. (closes issue 0017403) Reported by: one47 Patches: sip_codecs_simplified4 uploaded by one47 (license 23) Tested by: one47, falves11 Review: https://reviewboard.asterisk.org/r/967/ [^] ........ Backporting a bugfix that should have been included. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302049 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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7781c78f06 |
Blocked revisions 301869 via svnmerge
........ r301869 | lmadsen | 2011-01-14 14:21:00 -0600 (Fri, 14 Jan 2011) | 7 lines Fix issue with cross-compiling failing (closes issue #18301) Reported by: abelbeck Patches: asterisk-1.4-bugid18301.patch.txt uploaded by abelbeck (license 946) Tested by: abelbeck, russellb ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@301887 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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d4a9e2c20a |
Add relationships to function documentation.
Fix amatuer type mistake git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@301848 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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62c5b3d9d4 |
Add relationships to function documentation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@301842 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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1566c39525 |
Add static entry for split Polycom 332 firmware.
(closes issue #18607) Reported by: cjacobsen Patches: polycom_331.diff uploaded by cjacobsen (license 1029) Tested by: lathama git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@301730 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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b381052ca0 |
Don't reject all SUBSCRIBE auth requests
When merging another SUBSCRIBE fix from 1.4, some braces were put in the wrong place. This patch fixes that. (closes issue #18597) Reported by: thsgmbh git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@301682 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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6adc202b2c |
Removed a usleep(1) that shouldn't be necessary in session_do, and removed the
ms_t member from the mansession_session structure. Merged revisions 301591 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r301591 | mnicholson | 2011-01-12 12:39:03 -0600 (Wed, 12 Jan 2011) | 5 lines Don't store the thread id for the manager session in the structure we pass to the thread for the manager session. ABE-2543 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@301594 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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ac9c6a3692 |
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/branches/1.6.2@301503 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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44abf69388 |
Fix a logic issue when passing context ARG
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@301310 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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423fa1a875 |
Merged revisions 301305 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r301305 | mnicholson | 2011-01-11 12:34:40 -0600 (Tue, 11 Jan 2011) | 4 lines Prevent buffer overflows in ast_uri_encode() ABE-2705 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@301307 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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b198610c96 |
SOUND_CACHE_DIR now defaults to empty
Sounds files included in the Asterisk tarball were being ignored and re-downloaded. Users wanting to cache the files can still override the setting using the --with-sounds-cache option. (closes issue #18589) Reported by: pabelanger Patches: issue18589.patch uploaded by pabelanger (license 224) Tested by: pabelanger Review: https://reviewboard.asterisk.org/r/1074/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@301220 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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5e7a60d065 |
Indicate log level argument for Log() is not optional
(closes issue #18586) Reported by: kshumard Patches: app_verbose.c.patch uploaded by kshumard (license 92) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@301176 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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15bf424dcb |
Initialize useropts/adminopts in case there is no column in the realtime DB.
(closes issue #18182) Reported by: dimas Patches: v1-18182.patch uploaded by dimas (license 88) Tested by: dimas git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@301089 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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1a3a5b1694 |
Fix regression causing forwarding voicemails to not work with file storage.
I had actually already fixed this in 295200 in 1.4 and thought it wasn't missing in the other branches for some reason. (closes issue #18358) Reported by: cabal95 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@301046 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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5068f61abd |
Merged revisions 300918 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r300918 | jpeeler | 2011-01-07 11:13:21 -0600 (Fri, 07 Jan 2011) | 7 lines Ensure good bye prompt in voicemail is played at the correct time. Specifically in the case of timing out but not leaving voicemail nothing should be heard. And when leaving voicemail it should be heard. ABE-2647 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@300951 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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c8ff5a6c26 |
Merged revisions 300621 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r300621 | tilghman | 2011-01-05 12:47:46 -0600 (Wed, 05 Jan 2011) | 10 lines Use the sanity check in place of the disconnect/connect cycle. The disconnect/connect cycle has the potential to cause random crashes. (closes issue #18243) Reported by: ks3 Patches: res_odbc.patch uploaded by ks3 (license 1147) Tested by: ks3 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@300622 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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1f28c82647 |
Change deprecated message to LOG_WARNING
Also removed latter part of message Discussed on #asterisk-dev git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@300574 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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f97b1161de |
Fix backwards and broken XML documentation.
(closes issue #18547) Reported by: jcovert Patches: xmldoc.c.patch uploaded by jcovert (license 551) chan_iax2.c.doc.patch uploaded by jcovert (license 551) chan_sip.c.patch uploaded by jcovert (license 551) chan_agent.c.patch uploaded by jcovert (license 551) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@300520 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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ad21f7cc19 |
Add some documentation to users.conf.sample.
(closes issue #18531) Reported by: lathama Patches: users.conf.sample2.diff uploaded by lathama (license 1028) Tested by: lathama git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@300431 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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806231d3b0 |
Merged revisions 300428 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r300428 | russell | 2011-01-04 14:56:04 -0600 (Tue, 04 Jan 2011) | 4 lines Update the autosupport script from Digium support. (closes AST-395) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@300429 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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2722ad135e |
Merged revisions 300216 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r300216 | twilson | 2011-01-04 11:11:48 -0600 (Tue, 04 Jan 2011) | 15 lines Don't authenticate SUBSCRIBE re-transmissions This only skips authentication on retransmissions that are already authenticated. A similar method is already used for INVITES. This is the kind of thing we end up having to do when we don't have a transaction layer... (closes issue #18075) Reported by: mdu113 Patches: diff.txt uploaded by twilson (license 396) Tested by: twilson, mdu113 Review: https://reviewboard.asterisk.org/r/1005/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@300298 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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828635332e |
Use correct variable for atxfercallbackretries config option.
* Misc formatting changes. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@300165 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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1aa72d01e0 |
Documentation typo
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@299864 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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4c916b5474 |
Merged revisions 299624 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r299624 | tilghman | 2010-12-25 04:04:06 -0600 (Sat, 25 Dec 2010) | 5 lines Move check for extension existence below variable inheritance, due to the possible use of an eswitch. (closes issue #16228) Reported by: jlaguilar ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@299625 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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76abd62c40 |
do not use progress which is for PRI and SS7, add mfcr2_progress member
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@299533 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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4e1350ed62 |
Enqueue AST_CONTROL_PROGRESS after AST_CONTROL_RINGING when MFC-R2 calls are accepted
(closes issue #18438) Reported by: mariner7 Tested by: moy git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@299530 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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205317e8b6 |
Resolve warnings by disambiguating the "s" extension as used by chan_dahdi from the "s" extension as used by the AEL macros.
(closes issue #18480) Reported by: nivek Patches: 20101215__issue18480__2.diff.txt uploaded by tilghman (license 14) Tested by: nivek git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@299448 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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077b8b9956 |
Merged revisions 299194,299198,299220 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r299194 | mnicholson | 2010-12-20 14:45:38 -0600 (Mon, 20 Dec 2010) | 6 lines Respond as soon as possible with a 202 Accepted to refer requests. This change also plugs a few memory leaks that can occur when parking sip calls. ABE-2656 ........ r299198 | mnicholson | 2010-12-20 15:00:44 -0600 (Mon, 20 Dec 2010) | 2 lines Remove changes to via processing that were not supposed to go into the last commit. ........ r299220 | mnicholson | 2010-12-20 15:21:39 -0600 (Mon, 20 Dec 2010) | 4 lines Use ast_free() instead of free() ABE-2656 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@299242 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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8eecdf7eac |
Documentation fix
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@299136 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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0916eb438c |
If a call was not answered, then the billsec was calculated unusually large.
Also, due to a copy and paste error, a request for the answer field would have given the start value, instead. (closes issue #18460) Reported by: joscas Patches: 20101215__issue18460.diff.txt uploaded by tilghman (license 14) Tested by: joscas git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@299130 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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d2619c264a |
Note that Park() timeout is milliseconds.
(closes issue #15758) Reported by: mmurdock Tested by: mmurdock, seanbright git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@299087 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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c448bc9800 |
Typos: recieved => received
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@299003 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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9fb5ffe7b5 |
Remove backtrace used for testing merge process
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@298962 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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fb52725167 |
Merged revisions 298905 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r298905 | tilghman | 2010-12-17 15:40:56 -0600 (Fri, 17 Dec 2010) | 6 lines Let Asterisk find better backtrace information with libbfd. The menuselect option BETTER_BACKTRACES, if enabled, will use libbfd to search for better symbol information within both the Asterisk binary, as well as loaded modules, to assist when using inline backtraces to track down problems. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@298957 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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d246d0beac |
Also include PTHREAD_LIBS and PTHREAD_CFLAGS for SQLite 3, as it's needed on some platforms.
(closes issue #18493) Reported by: pprindeville Patches: asterisk-1.8-sqlite3.patch uploaded by pprindeville (license 347) Tested by: pprindeville git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@298817 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |