mirror of https://github.com/asterisk/asterisk
master
22
23
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${ noResults }
17828 Commits (01cf5498d1d8977f50fbd5626b8fdf02a33b17b0)
| Author | SHA1 | Message | Date |
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01cf5498d1 |
Merged revisions 250051 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r250051 | lmadsen | 2010-03-02 16:09:27 -0500 (Tue, 02 Mar 2010) | 8 lines Update IMAP documentation. Update the IMAP documentation to make it clear that storing voicemails in the same folder as a large number of emails could potentially cause significant slow downs when writing or retrieving voicemails. (issue #16704) Reported by: TimeHider Tested by: lmadsen, TimeHider ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@250054 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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b84dbb846e |
Merged revisions 250045 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r250045 | lmadsen | 2010-03-02 15:52:19 -0500 (Tue, 02 Mar 2010) | 15 lines Merged revisions 250043 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r250043 | lmadsen | 2010-03-02 15:51:35 -0500 (Tue, 02 Mar 2010) | 7 lines Update documentation to clarify purpose of unanswered option. (closes issue #16267) Reported by: elsto Patches: cdr.conf.sample.patch.txt uploaded by lmadsen (license 10) Tested by: davidw, elsto ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@250049 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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32de04aec5 |
Merged revisions 250037 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r250037 | lmadsen | 2010-03-02 15:36:10 -0500 (Tue, 02 Mar 2010) | 4 lines Update documentation to not imply we support overriding options. (closes issue #16855) Reported by: davidw ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@250040 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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18e0570c41 |
revert ability to exit echo app
caused a regression, as only supported VOICE, not VIDEO etc. (issue #16880) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@249948 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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03012c2325 |
Merged revisions 249893 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r249893 | dvossel | 2010-03-02 13:08:38 -0600 (Tue, 02 Mar 2010) | 11 lines fixes adaptive jitterbuffer configuration When configuring the adaptive jitterbuffer, the target_extra value not only could not be set from the configuration, but was not even being set to its proper default. This value is required in order for the adaptive jitterbuffer to work correctly. To resolve this a config option has been added to expose this value to the conf files, and a default value is provided when no config specific value is present. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@249907 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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0f9cce8464 |
fixes ability to exit echo app
when called from a ISDN channel, null frames prevent '#' exit. Now only echo back VOICE and DTMF frames (closes issue #16880) Reported by: alecdavis Patches: based on echo_exit_1-6-1.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@249846 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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28f650fc27 |
Merged revisions 249672 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r249672 | seanbright | 2010-03-01 14:36:30 -0500 (Mon, 01 Mar 2010) | 18 lines Merged revisions 249671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r249671 | seanbright | 2010-03-01 14:35:01 -0500 (Mon, 01 Mar 2010) | 11 lines Fix crash in app_voicemail related to message counting. We were passing a 'struct inprocess **' and treating it like a 'struct inprocess *' causing a segfault. (closes issue #16921) Reported by: whardier Patches: 20100301_issue16921.patch uploaded by seanbright (license 71) Tested by: whardier ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@249673 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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30725488fa |
Merged revisions 249538 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r249538 | jpeeler | 2010-03-01 11:11:31 -0600 (Mon, 01 Mar 2010) | 18 lines Merged revisions 249536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r249536 | jpeeler | 2010-03-01 11:02:03 -0600 (Mon, 01 Mar 2010) | 11 lines Modify queued frames from local channels to not set the other side to up In this case, attended transfers were broken due to ast_feature_request_and_dial detecting the channel being set to up before the answer frame could be read and therefore failing to mark the channel as ready. This fix is a regression fix for 244785, which should continue to work properly as well. (closes issue #16816) Reported by: jamhed Tested by: jamhed, corruptor ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@249539 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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75df7e3e5f |
overlap receiving: automatically send CALL PROCEEDING when dialplan starts
Following Q.931 5.2.4 When the user has determined that sufficient call information has been received the user shall stop T302 and send CALL PROCEEDING to the network. Previously timeouts were possible if the dialplan took a long time to issue any response back to the network. Verified that our local TELCO also does the same. (issue #16789) Reported by: alecdavis Patches: overlap_receiving_trunk.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis (closes issue #16789) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@249364 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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dfa2501fda |
Merged revisions 249235 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r249235 | kpfleming | 2010-02-27 09:08:35 -0500 (Sat, 27 Feb 2010) | 9 lines Merged revisions 249234 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r249234 | kpfleming | 2010-02-27 09:07:59 -0500 (Sat, 27 Feb 2010) | 1 line add a reference to the now-published IAX2 RFC ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@249236 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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429e38a950 |
Merged revisions 249101 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r249101 | mmichelson | 2010-02-26 11:04:58 -0600 (Fri, 26 Feb 2010) | 14 lines Merged revisions 249100 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r249100 | mmichelson | 2010-02-26 11:04:29 -0600 (Fri, 26 Feb 2010) | 8 lines For T.38 reINVITEs treat a 606 the same as a 488. (closes issue #16792) Reported by: vrban Patches: t38_606.patch uploaded by vrban (license 756) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@249102 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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cc046c68ef |
Merged revisions 248952 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r248952 | jpeeler | 2010-02-25 17:09:54 -0600 (Thu, 25 Feb 2010) | 24 lines Merged revisions 248860 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r248860 | jpeeler | 2010-02-25 15:22:06 -0600 (Thu, 25 Feb 2010) | 18 lines Ensure that monitor recordings are written to the correct location (again) This is an extension to 248757. As such the dialplan test has been extended: exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n, dial(sip/5001) exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n, dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b) exten => 5042, n, dial(sip/5001) exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test3,m) exten => 5043, n, changemonitor(monitor_test4) exten => 5043, n, dial(sip/5001) exten => 5044, 1, monitor(wav,monitor_test4,m) exten => 5044, n, changemonitor(tmp/jeff/monitor_test5) ; this looks to fail by design and emits a warning exten => 5044, n, dial(sip/5001) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@248953 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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c9c753b7e2 |
Merged revisions 248946 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r248946 | mmichelson | 2010-02-25 16:41:48 -0600 (Thu, 25 Feb 2010) | 5 lines Fix incorrect ACL behavior when CIDR notation of "/0" is used. AST-2010-003 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@248947 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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d51611efee |
Merged revisions 248861 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r248861 | tilghman | 2010-02-25 15:22:39 -0600 (Thu, 25 Feb 2010) | 22 lines Merged revisions 248859 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r248859 | tilghman | 2010-02-25 15:21:05 -0600 (Thu, 25 Feb 2010) | 15 lines Some platforms clear /var/run at boot, which makes connecting a remote console... difficult. Previously, we only created the default /var/run/asterisk directory at install time. While we could create it in the init script, that would not work for those who start asterisk manually from the command line. So the safest thing to do is to create it as part of the Asterisk boot process. This also changes the ownership of the directory, because the pid and ctl files are created after we setuid/setgid. (closes issue #16802) Reported by: Brian Patches: 20100224__issue16802.diff.txt uploaded by tilghman (license 14) Tested by: tzafrir ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@248862 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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a58e335190 |
Merged revisions 248793 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r248793 | jpeeler | 2010-02-25 12:37:56 -0600 (Thu, 25 Feb 2010) | 22 lines Merged revisions 248757 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r248757 | jpeeler | 2010-02-25 12:06:54 -0600 (Thu, 25 Feb 2010) | 15 lines Ensure that monitor recordings are written to the correct location. Recordings should be placed in the monitor directory when a non-absolute path is used. Exact dialplan used for testing: exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n, dial(sip/5001) exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n, dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b) exten => 5042, n, dial(sip/5001) ABE-2101 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@248795 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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568e0b13ed |
Merged revisions 248584 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r248584 | tilghman | 2010-02-24 15:17:26 -0600 (Wed, 24 Feb 2010) | 14 lines Merged revisions 248582 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r248582 | tilghman | 2010-02-24 15:02:18 -0600 (Wed, 24 Feb 2010) | 7 lines Remove color code sequences from verbose messages that go to logfiles. (closes issue #16786) Reported by: dodo Patches: logger2.patch uploaded by dodo (license 989) Tested by: tilghman ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@248613 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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81b2dbe043 |
Merged revisions 248397 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r248397 | dvossel | 2010-02-23 10:34:39 -0600 (Tue, 23 Feb 2010) | 15 lines Merged revisions 248396 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r248396 | dvossel | 2010-02-23 10:26:05 -0600 (Tue, 23 Feb 2010) | 9 lines fixes invite with replaces deadlock (closes issue #16862) Reported by: pwalker Patches: replaces_deadlock_1.4 uploaded by dvossel (license 671) Tested by: pwalker, dvossel ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@248400 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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6f455cd645 |
Merged revisions 228798 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk (closes issue #16470) Reported by: kjotte ........ r228798 | tilghman | 2009-11-09 01:37:52 -0600 (Mon, 09 Nov 2009) | 14 lines Fix various problems detected with Valgrind. * chan_console accessed pvts after deallocation. * The module loader did not check usecount on shutdown, which led to chan_iax2 reading a timer that was already unloaded. (closes issue #16062) Reported by: alexanderheinz Patches: 20091109__issue16062.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@248008 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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ac29e44b08 |
Restore generated file from flex source
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@247933 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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c37946f41e |
Merged revisions 247914 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r247914 | rmudgett | 2010-02-19 11:33:33 -0600 (Fri, 19 Feb 2010) | 62 lines Merged revisions 247910 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r247910 | rmudgett | 2010-02-19 11:18:49 -0600 (Fri, 19 Feb 2010) | 55 lines Merged revision 247904 from https://origsvn.digium.com/svn/asterisk/be/branches/C.2-... .......... r247904 | rmudgett | 2010-02-19 10:49:44 -0600 (Fri, 19 Feb 2010) | 49 lines Make chan_misdn DTMF processing consistent with other channel technologies. The processing of DTMF tones on the receiving side of an ISDN channel is inconsistent with the way it is handled in other channels, especially DAHDI analog. This causes DTMF tones sent from an ISDN phone to be doubled at the connected party. We are using the following 2 options of misdn.conf 1) astdtmf=yes 2) senddtmf=yes Option one is necessary because the asterisk DSP DTMF detection is better than mISDN's internal DSP. Not as many false positives. Option two is necessary to transmit DTMF tones end to end when mISDN channels are connected to SIP channels with out of band DTMF for example. The symptom is that DTMF tones sent by an ISDN phone are doubled on the way through asterisk when two mISDN channels are connected with a Local channel in between or if it is bridged to an analog channel. The doubling of DTMF tones is because DTMF is passed inband to asterisk by the mISDN channel and passed out of band once again after the release of the DTMF tone. Passing it inband is wrong. Neither an analog channel nor SIP channel passes DTMF inband if configured to inband DTMF. Analog and SIP channels filter out the DTMF tones because they use the voice frames returned by ast_dsp_process. But chan_misdn passes the unfiltered input voice frames instead. To overcome one aspect of the problem, the doubling of DTMF tones when two mISDN channels are directly bridged, someone made an 'optimization', where in that case the DTMF tone passed out-of-band to the peer channel is not translated to an inband tone at the transmit side. This optimization is bad because it does not work in general. For example, analog channels or mISDN channels when bridged through an intermediary local channel will generate DTMF tones from out-of-band information. Also, of course, it must not be done when there is no inband DTMF available. This patch fixes the issue. Now chan_misdn will filter the received inband DTMF signal the same as other channel types. Another change included: No need to build an extra translation path because ast_process_dsp does it if required. Patches: misdn-dtmf.patch JIRA ABE-2080 ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@247922 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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0ab6e1116d |
Restore fwrite() line so ast_expr2f.c can compile.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@247919 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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9f3c3f79ba |
Blocked revisions 247915 via svnmerge
........ r247915 | dvossel | 2010-02-19 11:40:26 -0600 (Fri, 19 Feb 2010) | 7 lines handle_request_invite revise comment, fix coding guideline issues I'm working with this code right now trying to analyze a deadlock. This change is just to clean up a few things before I make a more complex patch. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@247918 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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24ae33ad6a |
Merged revisions 247841 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r247841 | tilghman | 2010-02-18 17:13:46 -0600 (Thu, 18 Feb 2010) | 7 lines Revert an errant part of a previous cleanup, to fix a memory corruption issue. (closes issue #16368) Reported by: thirionjwf Patches: res_speech.c.patch uploaded by thirionjwf (license 955) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@247842 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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cb59fb218e |
fixes dialog ref count crash isolated to the 1.6.0 branch
(closes issue #16375) Reported by: kobaz (closes issue #16796) Reported by: kobaz git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@247839 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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b39f12c917 |
Merged revisions 247787 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r247787 | tilghman | 2010-02-18 15:42:53 -0600 (Thu, 18 Feb 2010) | 17 lines If the peer record is from realtime, it could be set to 0, due to MySQL not representing NULL well in integer columns. NULL means the value is not specified for the column, which normally means the driver uses whatever is the default value. However, on MySQL, placing a NULL in either a float or integer column results in a retrieval of the 0 value. Hence, users get an errant error on load. This patch suppresses that error and makes the value as if it was not there. Note that this cannot be done in the realtime driver, because the lack of difference between NULL and 0 can only be intepreted correctly by the driver itself. If we did it in the realtime driver, then it would be effectively impossible to set any realtime field to 0, because it would act as if the field were unspecified and possibly take on a different value. (closes issue #16683) Reported by: wdoekes ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@247789 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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68193fd1e7 |
Blocked revisions 247770 via svnmerge
........ r247770 | dvossel | 2010-02-18 15:23:48 -0600 (Thu, 18 Feb 2010) | 9 lines fixes confbridge crash when no timing module is loaded. (closes issue #16471) Reported by: kjotte Patches: M16471.diff uploaded by junky (license 177) Tested by: kjotte, junky ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@247783 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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ddca01e0c4 |
Blocked revisions 247736 via svnmerge
........ r247736 | dvossel | 2010-02-18 14:58:41 -0600 (Thu, 18 Feb 2010) | 7 lines fixes Queue with C option crash (closes issue #16475) Reported by: okrief Patches: queue_crash.diff uploaded by dvossel (license 671) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@247739 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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caca82ce83 |
Merged revisions 247652 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r247652 | mnicholson | 2010-02-18 13:39:37 -0600 (Thu, 18 Feb 2010) | 13 lines Merged revisions 247651 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r247651 | mnicholson | 2010-02-18 13:38:09 -0600 (Thu, 18 Feb 2010) | 6 lines Copy the calling party's account code to the called party if they don't already have one. (closes issue #16331) Reported by: bluefox Tested by: mnicholson ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@247655 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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6005c9af91 |
Merged revisions 247509 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r247509 | lmadsen | 2010-02-18 11:54:43 -0500 (Thu, 18 Feb 2010) | 9 lines Merged revisions 247508 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r247508 | lmadsen | 2010-02-18 11:53:44 -0500 (Thu, 18 Feb 2010) | 1 line Add additional link to best practices document per jsmith. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@247510 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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1abd6cc1cc |
Merged revisions 247503 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r247503 | lmadsen | 2010-02-18 11:41:04 -0500 (Thu, 18 Feb 2010) | 18 lines Merged revisions 247502 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r247502 | lmadsen | 2010-02-18 11:38:17 -0500 (Thu, 18 Feb 2010) | 10 lines Add best practices documentation. (issue #16808) Reported by: lmadsen (issue #16810) Reported by: Nick_Lewis Tested by: lmadsen Review: https://reviewboard.asterisk.org/r/507/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@247504 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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5d5a79dea3 |
Merged revisions 247423 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r247423 | russell | 2010-02-17 22:20:11 -0600 (Wed, 17 Feb 2010) | 17 lines Merged revisions 247422 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r247422 | russell | 2010-02-17 22:19:01 -0600 (Wed, 17 Feb 2010) | 10 lines Tweak argument handling for wget in the sounds Makefile. 1) Fix the check to see if we are using wget to not be full of fail. The configure script populates this variable with the absolute path to wget if it is found, so it didn't work. 2) Allow some extra arguments to be passed in for wget. This is just a simple change to allow our Bamboo build script to tell wget to be quiet and not fill up our logs with download status output. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@247424 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Merged revisions 247335 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r247335 | mmichelson | 2010-02-17 15:22:40 -0600 (Wed, 17 Feb 2010) | 20 lines Fix two problems in ast_str functions found while writing a unit test. 1. The documentation for ast_str_set and ast_str_append state that the max_len parameter may be -1 in order to limit the size of the ast_str to its current allocated size. The problem was that the max_len parameter in all cases was a size_t, which is unsigned. Thus a -1 was interpreted as UINT_MAX instead of -1. Changing the max_len parameter to be ssize_t fixed this issue. 2. Once issue 1 was fixed, there was an off-by-one error in the case where we attempted to write a string larger than the current allotted size to a string when -1 was passed as the max_len parameter. When trying to write more than the allotted size, the ast_str's __AST_STR_USED was set to 1 higher than it should have been. Thanks to Tilghman for quickly spotting the offending line of code. Oh, and the unit test that I referenced in the top line of this commit will be added to reviewboard shortly. Sit tight... ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@247338 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 247169 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r247169 | mmichelson | 2010-02-17 10:24:54 -0600 (Wed, 17 Feb 2010) | 9 lines Merged revisions 247168 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r247168 | mmichelson | 2010-02-17 10:24:17 -0600 (Wed, 17 Feb 2010) | 3 lines Make sure that when autofill is disabled that callers not in the front of the queue cannot place calls. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@247170 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 247076 via svnmerge
........ r247076 | mmichelson | 2010-02-16 17:44:33 -0600 (Tue, 16 Feb 2010) | 12 lines Add va_end calls to __ast_str_helper. According to the man page for stdarg(3), "Each invocation of va_copy() must be matched by a corresponding invocation of va_end() in the same function." There were several cases in __ast_str_helper where va_copy was not matched with a corresponding call to va_end. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@247077 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 246985 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r246985 | mmichelson | 2010-02-16 15:15:38 -0600 (Tue, 16 Feb 2010) | 3 lines Add some clarifying documentation to the ast_str_set and ast_str_append functions. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@246986 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 246980 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r246980 | dvossel | 2010-02-16 14:54:48 -0600 (Tue, 16 Feb 2010) | 8 lines warning message if openssl support is missing while attempting tls connection (closes issue #16673) Reported by: michaesc Patches: tls_error_msg.diff uploaded by dvossel (license 671) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@246984 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 246899 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r246899 | dvossel | 2010-02-16 11:07:41 -0600 (Tue, 16 Feb 2010) | 16 lines fixes sample rate conversion issue with Monitor application When using ast_seekstream with the read/write streams of a monitor, the number of samples we are seeking must be of the same rate as the stream or the jump calculation will be incorrect. This patch adds logic to correctly convert the number of samples to jump to the sample rate the read/write stream is using. For example, if the call is G722 (16khz) and the read/write stream is recording a 8khz wav, seeking 320 samples of 16khz audio is not the same as seeking 320 samples of 8khz audio when performing the ast_seekstream on the stream. ABE-2044 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@246903 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 246710 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r246710 | tilghman | 2010-02-15 17:43:28 -0600 (Mon, 15 Feb 2010) | 12 lines Merged revisions 246709 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r246709 | tilghman | 2010-02-15 17:42:33 -0600 (Mon, 15 Feb 2010) | 5 lines Make the menuselect instructions correct by allowing 'make menuselect' to actually solve dependency problems. (Previously, it would fail out again with the same message about running 'make menuselect', which was NOT at all helpful.) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@246711 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 246546 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r246546 | dvossel | 2010-02-12 17:32:33 -0600 (Fri, 12 Feb 2010) | 21 lines Merged revisions 246545 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r246545 | dvossel | 2010-02-12 17:30:17 -0600 (Fri, 12 Feb 2010) | 16 lines lock channel during datastore removal On channel destruction the channel's datastores are removed and destroyed. Since there are public API calls to find and remove datastores on a channel, a lock should be held whenever datastores are removed and destroyed. This resolves a crash caused by a race condition in app_chanspy.c. (closes issue #16678) Reported by: tim_ringenbach Patches: datastore_destroy_race.diff uploaded by tim ringenbach (license 540) Tested by: dvossel ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@246549 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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3d1a273447 |
Fix some silly formatting that made my head hurt.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@246462 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 246204 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r246204 | tilghman | 2010-02-10 15:24:10 -0600 (Wed, 10 Feb 2010) | 2 lines Fussy compiler on another machine... ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@246205 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 246200 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r246200 | tilghman | 2010-02-10 15:19:35 -0600 (Wed, 10 Feb 2010) | 2 lines Fix weird issue with unit tests on optimized build - turned out to be a signing issue. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@246201 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 246116 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r246116 | dvossel | 2010-02-10 11:49:34 -0600 (Wed, 10 Feb 2010) | 14 lines Merged revisions 246115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r246115 | dvossel | 2010-02-10 11:44:20 -0600 (Wed, 10 Feb 2010) | 8 lines fixes random deadlock in app_queue with use_weight during reload (closes issue #16677) Reported by: tim_ringenbach Patches: app_queue_use_weight_deadlock.diff uploaded by tim ringenbach (license 540) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@246122 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 246070 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r246070 | jpeeler | 2010-02-10 10:47:37 -0600 (Wed, 10 Feb 2010) | 22 lines Change channel state on local channels for busy,answer,ring. Previously local channels channel state never changed. This became problematic when the state of the other side of the local channel was lost, for example during a masquerade. Changing the state of the local channel allows for the scenario to be detected when the channel state is set to ringing, but the peer isn't ringing. The specific problem scenario is described in 164201. Although this was noted on one of the issues, here is the tested dialplan verified to work: exten => 9700,1,Dial(Local/*9700@default&Local/0009700@default) exten => *9700,1,Set(GLOBAL(TESTCHAN)=${CHANNEL:0:${MATH(${LEN(${CHANNEL})}-1):0:2}}1) exten => *9700,n,wait(3) ;3 works, 1 did not exten => *9700,n,Dial(SIP/5001) exten => 0009700,1,Wait(1) ;1 works, 3 did not exten => 0009700,n,ChannelRedirect(${TESTCHAN},parkedcalls,701,1) (closes issue #14992) Reported by: davidw ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@246071 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 246022 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r246022 | tilghman | 2010-02-10 09:36:57 -0600 (Wed, 10 Feb 2010) | 2 lines Enable warnings on atypical conditions for the FILTER function (suggested by mmichelson on the -dev list). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@246023 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 245945 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r245945 | tilghman | 2010-02-10 08:06:12 -0600 (Wed, 10 Feb 2010) | 9 lines Merged revisions 245944 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r245944 | tilghman | 2010-02-10 07:37:13 -0600 (Wed, 10 Feb 2010) | 2 lines Include examples of FILTER usage in extension patterns where a "." may be a risk. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@245946 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 245793 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r245793 | dvossel | 2010-02-09 17:07:17 -0600 (Tue, 09 Feb 2010) | 18 lines Merged revisions 245792 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r245792 | dvossel | 2010-02-09 16:55:38 -0600 (Tue, 09 Feb 2010) | 12 lines Fixes iaxs and iaxsl size off by one issue. 2^15 = 32768 which is the maximum allowed iax2 callnumber. Creating the iaxs and iaxsl array of size 32768 means the maximum callnumber is actually out of bounds. This causes a nasty crash. (closes issue #15997) Reported by: exarv Patches: iax_fix.diff uploaded by dvossel (license 671) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@245796 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 245729 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r245729 | tilghman | 2010-02-09 12:06:30 -0600 (Tue, 09 Feb 2010) | 8 lines Ensure frames are only freed once. (closes issue #16361) Reported by: vlad Patches: 20100208__issue16361.diff.txt uploaded by tilghman (license 14) Tested by: kenny, bloodoff, misaksen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@245730 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 245680 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r245680 | kpfleming | 2010-02-09 10:24:52 -0600 (Tue, 09 Feb 2010) | 8 lines Don't offer MMR or JBIG transcoding during T.38 negotiation. After further discussion with Steve Underwood, we should not (yet) be offering to receive MMR or JBIG transcoded streams from T.38 endpoints. A future spandsp release will support those features, and then they can be enabled during negotiation ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@245681 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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6fd11d7446 |
Stop playing the message number multiple times.
Also remove some accidentally duplicated code, which may have been causing a memleak. This was caused by a bad merge. (closes issue #16579) Reported by: kue Patches: 0016525.patch uploaded by hokie21 (license 987) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@245627 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |