2010-03-04 Leif Madsen * Asterisk 1.6.0.26-rc1 released 2010-03-03 21:27 +0000 [r250612] Leif Madsen * doc/tex/localchannel.tex: Merged revisions 250609 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r250609 | lmadsen | 2010-03-03 16:22:55 -0500 (Wed, 03 Mar 2010) | 11 lines Update existing Local channel documentation. A complete re-write of the Local channel documentation has been performed, with the existing information from localchannel.txt and localchannel.tex merged in. (closes issue #16637) Reported by: kobaz Patches: localchannel.tex uploaded by lmadsen (license 10) localchannel.txt uploaded by lmadsen (license 10) Tested by: lmadsen, jsmith, mmichelson ........ 2010-03-03 19:07 +0000 [r250482] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 250481 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r250481 | jpeeler | 2010-03-03 13:06:06 -0600 (Wed, 03 Mar 2010) | 22 lines Merged revisions 250480 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r250480 | jpeeler | 2010-03-03 13:04:11 -0600 (Wed, 03 Mar 2010) | 15 lines Make sure to clear red alarm after polarity reversal. From the issue: The automatic overnight line tests (or manual ones) used on UK (BT) lines causes a red alarm on a dahdi / TDM400P connected channel. This is because the line uses voltage tests (battery loss) and polarity reversal. The polarity reversal causes chan_dahdi to initiate v23 CallerID processing but during this the event DAHDI_EVENT_NOALARM is ignored so that the alarm is never cleared. (closes issue #14163) Reported by: jedi98 Patches: chan_dahdi-1.4-inalarm.diff uploaded by jedi98 (license 653) Tested by: mattbrown, Chainsaw, mikeeccleston ........ ................ 2010-03-03 18:06 +0000 [r250265-250398] David Vossel * /, channels/chan_iax2.c: Merged revisions 250395 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r250395 | dvossel | 2010-03-03 12:03:19 -0600 (Wed, 03 Mar 2010) | 22 lines Merged revisions 250394 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r250394 | dvossel | 2010-03-03 12:02:27 -0600 (Wed, 03 Mar 2010) | 16 lines fixes problem with duplicate TXREQ packets When Asterisk receives an IAX2 TXREQ packet, try_transfer() will call store_by_transfercallno() to link the chan_iax2_pvt struct into iax_transfercallno_pvts. If a duplicate TXREQ packet is received for the same call, the pvt struct will be linked into iax_transfercallno_pvts multiple times. This patch fixes this. Thanks rain for debugging this and providing a patch! (closes issue #16904) Reported by: rain Patches: iax2-double-txreq-fix.diff uploaded by rain (license 327) Tested by: rain, dvossel ........ ................ * /, channels/chan_sip.c: Merged revisions 250246 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r250246 | dvossel | 2010-03-02 18:18:28 -0600 (Tue, 02 Mar 2010) | 2 lines fixes signed to unsigned int comparision issue for FaxMaxDatagram value. ........ 2010-03-02 21:11 +0000 [r250040-250054] Leif Madsen * doc/tex/imapstorage.tex, /: 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 ........ * configs/cdr.conf.sample: 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 ........ ................ * doc/tex/configuration.tex: 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 ........ 2010-03-02 19:45 +0000 [r249948] Alec L Davis * apps/app_echo.c: revert ability to exit echo app caused a regression, as only supported VOICE, not VIDEO etc. (issue #16880) 2010-03-02 19:20 +0000 [r249907] David Vossel * channels/chan_oss.c, channels/misdn_config.c, include/asterisk/abstract_jb.h, configs/alsa.conf.sample, channels/chan_jingle.c, channels/chan_usbradio.c, channels/chan_dahdi.c, channels/chan_skinny.c, configs/mgcp.conf.sample, main/abstract_jb.c, channels/chan_h323.c, channels/chan_alsa.c, configs/sip.conf.sample, channels/chan_mgcp.c, channels/chan_unistim.c, configs/console.conf.sample, configs/chan_dahdi.conf.sample, channels/chan_local.c, configs/oss.conf.sample, channels/chan_sip.c, /, configs/usbradio.conf.sample, configs/misdn.conf.sample, channels/chan_gtalk.c, channels/chan_console.c: 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. ........ 2010-03-02 09:16 +0000 [r249846] Alec L Davis * apps/app_echo.c: 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 2010-03-01 19:38 +0000 [r249673] Sean Bright * apps/app_voicemail.c, /: 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 ........ ................ 2010-03-01 17:13 +0000 [r249539] Jeff Peeler * channels/chan_local.c, /: 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 ........ ................ 2010-02-27 23:38 +0000 [r249364] Alec L Davis * channels/chan_dahdi.c: 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) 2010-02-27 14:09 +0000 [r249236] Kevin P. Fleming * /, channels/chan_iax2.c: 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 ........ ................ 2010-02-26 17:05 +0000 [r249102] Mark Michelson * /, channels/chan_sip.c: 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) ........ ................ 2010-02-25 23:11 +0000 [r248953] Jeff Peeler * res/res_monitor.c, /: 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) ........ ................ 2010-02-25 22:42 +0000 [r248947] Mark Michelson * /, main/acl.c: 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 ........ 2010-02-25 21:24 +0000 [r248862] Tilghman Lesher * main/asterisk.c, /: 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 ........ ................ 2010-02-25 18:46 +0000 [r248795] Jeff Peeler * res/res_monitor.c, /: 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 ........ ................ 2010-02-24 21:23 +0000 [r248613] Tilghman Lesher * /, main/logger.c: 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 ........ ................ 2010-02-23 16:51 +0000 [r248400] David Vossel * /, channels/chan_sip.c: 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 ........ ................ 2010-02-19 19:04 +0000 [r247933-248008] Tilghman Lesher * main/loader.c, /, channels/chan_console.c: 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 ........ * main/ast_expr2f.c: Restore generated file from flex source 2010-02-19 18:13 +0000 [r247919-247922] Richard Mudgett * channels/chan_misdn.c, /: 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 ................ ................ * main/ast_expr2f.c: Restore fwrite() line so ast_expr2f.c can compile. 2010-02-18 23:15 +0000 [r247842] Tilghman Lesher * res/res_speech.c, /: 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) ........ 2010-02-18 22:45 +0000 [r247839] David Vossel * channels/chan_sip.c: fixes dialog ref count crash isolated to the 1.6.0 branch (closes issue #16375) Reported by: kobaz (closes issue #16796) Reported by: kobaz 2010-02-18 21:47 +0000 [r247789] Tilghman Lesher * /, channels/chan_sip.c: 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 ........ 2010-02-18 19:45 +0000 [r247655] Matthew Nicholson * /, main/features.c: 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 ........ ................ 2010-02-18 16:56 +0000 [r247504-247510] Leif Madsen * README-SERIOUSLY.bestpractices.txt: 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. ........ ................ * README-SERIOUSLY.bestpractices.txt (added): 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/ ........ ................ 2010-02-18 04:20 +0000 [r247424] Russell Bryant * Makefile, /, sounds/Makefile: 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. ........ ................ 2010-02-17 21:35 +0000 [r246986-247338] Mark Michelson * /, main/utils.c, include/asterisk/strings.h: 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... ........ * /, apps/app_queue.c: 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. ........ ................ * /, include/asterisk/strings.h: 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. ........ 2010-02-16 21:07 +0000 [r246903-246984] David Vossel * main/tcptls.c, /: 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) ........ * main/channel.c, /: 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 ........ 2010-02-15 23:44 +0000 [r246711] Tilghman Lesher * Makefile, /: 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.) ........ ................ 2010-02-12 23:35 +0000 [r246549] David Vossel * main/channel.c, /: 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 ........ ................ 2010-02-12 18:57 +0000 [r246462] Jason Parker * main/channel.c: Fix some silly formatting that made my head hurt. 2010-02-10 21:27 +0000 [r246201-246205] Tilghman Lesher * /, funcs/func_strings.c: 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... ........ * /, funcs/func_strings.c: 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. ........ 2010-02-10 17:56 +0000 [r246122] David Vossel * /, apps/app_queue.c: 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) ........ ................ 2010-02-10 16:53 +0000 [r246071] Jeff Peeler * channels/chan_local.c, /: 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 ........ 2010-02-10 15:38 +0000 [r245946-246023] Tilghman Lesher * /, funcs/func_strings.c: 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). ........ * configs/extensions.conf.sample, /, funcs/func_strings.c: 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. ........ ................ 2010-02-09 23:14 +0000 [r245796] David Vossel * /, channels/chan_iax2.c: 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) ........ ................ 2010-02-09 18:09 +0000 [r245730] Tilghman Lesher * /, apps/app_fax.c: 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 ........ 2010-02-09 16:25 +0000 [r245681] Kevin P. Fleming * /, apps/app_fax.c: 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 ........ 2010-02-08 23:51 +0000 [r245627] Jason Parker * apps/app_voicemail.c: 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) 2010-02-08 22:46 +0000 [r245579] Tilghman Lesher * /, main/Makefile, channels/Makefile: Merged revisions 245578 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r245578 | tilghman | 2010-02-08 16:31:40 -0600 (Mon, 08 Feb 2010) | 12 lines Actually use _ASTLDFLAGS in the main/ and channels/ Makefiles. They were previously passed correctly, but they simply weren't used. This caused issues with various platforms whose builds needed to pass special linker flags via the configure script. (closes issue #16596) Reported by: pprindeville Patches: asterisk-1.6-astldflags.patch uploaded by pprindeville (license 347) Tested by: tilghman ........ 2010-02-08 20:42 +0000 [r245498] Jason Parker * main/ast_expr2.fl, /, main/ast_expr2f.c: Merged revisions 245497 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r245497 | qwell | 2010-02-08 14:41:05 -0600 (Mon, 08 Feb 2010) | 11 lines Merged revisions 245496 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r245496 | qwell | 2010-02-08 14:39:50 -0600 (Mon, 08 Feb 2010) | 4 lines Remove reference of documentation in source directory. People don't always build Asterisk from source (distro packages, anybody?). ........ ................ 2010-02-05 19:26 +0000 [r245093] Jeff Peeler * contrib/firmware (removed), /, LICENSE: Merged revisions 245090 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r245090 | jpeeler | 2010-02-05 13:26:22 -0600 (Fri, 05 Feb 2010) | 11 lines Merged revisions 245044 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r245044 | kpfleming | 2010-02-05 12:32:29 -0600 (Fri, 05 Feb 2010) | 5 lines Remove contrib/firmware directory as it is empty Remove explicit license for IAXy firmware as it is no longer included in the tree ........ ................ 2010-02-05 17:10 +0000 [r244928] Sean Bright * main/asterisk.c, /: Merged revisions 244927 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r244927 | seanbright | 2010-02-05 12:05:32 -0500 (Fri, 05 Feb 2010) | 9 lines Merged revisions 244926 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r244926 | seanbright | 2010-02-05 12:03:35 -0500 (Fri, 05 Feb 2010) | 1 line Update main copyright date. ........ ................ 2010-02-03 18:40 +0000 [r244506] Tilghman Lesher * channels/chan_dahdi.c, /: Merged revisions 244505 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r244505 | tilghman | 2010-02-03 12:34:29 -0600 (Wed, 03 Feb 2010) | 8 lines The chanvar= setting should inherit the entire list of variables, not just the first one. (closes issue #16359) Reported by: raarts Patches: dahdi-setvars.diff uploaded by raarts (license 937) Tested by: raarts ........ 2010-02-02 22:32 +0000 [r244447] David Vossel * main/udptl.c, /, channels/chan_sip.c, include/asterisk/udptl.h: Merged revisions 244443 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r244443 | dvossel | 2010-02-02 16:27:23 -0600 (Tue, 02 Feb 2010) | 18 lines fixes crash during T.38 negotiation caused by invalid or missing FaxMaxDatagram field AST-2010-001 (closes issue #16634) Reported by: krn (closes issue #16724) Reported by: barthpbx (closes issue #16517) Reported by: bklang (closes issue #16485) Reported by: elsto ........ 2010-02-02 Leif Madsen * Release Asterisk 1.6.0.22 * AST-2010-001: An attacker attempting to negotiate T.38 over SIP can remotely crash Asterisk by modifying the FaxMaxDatagram field of the SDP to contain either a negative or exceptionally large value. The same crash occurs when the FaxMaxDatagram field is omitted from the SDP as well. 2010-01-14 Leif Madsen * Release Asterisk 1.6.0.21 2010-01-08 Leif Madsen * Release Asterisk 1.6.0.21-rc1 2010-01-07 21:17 +0000 [r238494] Tilghman Lesher * channels/chan_oss.c, main/poll.c, channels/chan_usbradio.c, include/asterisk/utils.h, /, channels/chan_sip.c, channels/chan_alsa.c, channels/chan_console.c: Merged revisions 209400 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209400 | kpfleming | 2009-07-28 08:49:46 -0500 (Tue, 28 Jul 2009) | 3 lines Define side-effect-safe MIN and MAX macros and remove duplicate definitions from various files. (closes issue #16251) Reported by: asgaroth ........ 2010-01-07 20:22 +0000 [r238364-238441] David Vossel * /, channels/chan_iax2.c: Merged revisions 238412 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r238412 | dvossel | 2010-01-07 14:15:27 -0600 (Thu, 07 Jan 2010) | 16 lines Merged revisions 238411 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r238411 | dvossel | 2010-01-07 14:14:25 -0600 (Thu, 07 Jan 2010) | 10 lines fixes crash in "scheduled_destroy" in chan_iax A signed short was used to represent a callnumber. This is makes it possible to attempt to access the iaxs array with a negative index. (closes issue #16565) Reported by: jensvb ........ ................ * /, channels/chan_sip.c: Merged revisions 238405 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r238405 | dvossel | 2010-01-07 14:00:31 -0600 (Thu, 07 Jan 2010) | 8 lines Change in sip show channels display format allowing more digits for CID (closes issue #16459) Reported by: Rzadzins Patches: chan_sip_longer_cid.patch uploaded by Rzadzins (license 953) ........ * /, apps/app_queue.c: Merged revisions 238361 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r238361 | dvossel | 2010-01-07 12:58:23 -0600 (Thu, 07 Jan 2010) | 8 lines cli 'queue show' formatting fix. queue name was truncated over 12 characters (closes issue #16078) Reported by: RoadKill Patches: quequename_limit.patch uploaded by ppyy (license 906) Tested by: dvossel ........ 2010-01-06 21:48 +0000 [r238232] Tilghman Lesher * /, funcs/func_cdr.c: Merged revisions 238231 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r238231 | tilghman | 2010-01-06 15:45:17 -0600 (Wed, 06 Jan 2010) | 11 lines Merged revisions 238230 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r238230 | tilghman | 2010-01-06 15:41:55 -0600 (Wed, 06 Jan 2010) | 4 lines Revise documentation on disposition values to the actual values used. (closes issue #16289) Reported by: wdoekes ........ ................ 2010-01-06 20:38 +0000 [r238135-238182] Jeff Peeler * /, apps/app_meetme.c: Merged revisions 238181 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r238181 | jpeeler | 2010-01-06 14:37:18 -0600 (Wed, 06 Jan 2010) | 8 lines Fix misreverting from 177158. (closes issue #15725) Reported by: shanermn Patches: v1-15725.patch uploaded by dimas (license 88) Tested by: shanermn ........ * /, main/features.c: Merged revisions 238134 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r238134 | jpeeler | 2010-01-06 13:05:06 -0600 (Wed, 06 Jan 2010) | 10 lines Fix channel name comparison for bridge application. The channel name comparison was not comparing the whole string and therefore if one channel name was a substring of the other, the bridge would fail. (closes issue #16528) Reported by: telecos82 Patches: res_features_r236843.diff uploaded by telecos82 (license 687) ........ 2010-01-06 15:20 +0000 [r238011] Russell Bryant * /, apps/app_mp3.c: Merged revisions 238010 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r238010 | russell | 2010-01-06 09:19:10 -0600 (Wed, 06 Jan 2010) | 14 lines Merged revisions 238009 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r238009 | russell | 2010-01-06 09:18:22 -0600 (Wed, 06 Jan 2010) | 7 lines Resolve a crash due to an ast_frame not being fully initialized. (closes issue #16531) Reported by: john8675309 (closes SWP-615) ........ ................ 2010-01-06 06:51 +0000 [r237966] Tilghman Lesher * channels/chan_sip.c: Something clearly went wrong with a merge somewhere, because these are all duplicates (and therefore dead code). 2010-01-05 23:10 +0000 [r237843-237923] David Vossel * /, apps/app_queue.c: Merged revisions 237920 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r237920 | dvossel | 2010-01-05 17:08:50 -0600 (Tue, 05 Jan 2010) | 16 lines fixes holdtime playback issue in app_queue When reporting hold time, the number of seconds should be mod 60. Otherwise audio playback could be something like "2 minutes 123 seconds" rather than "2 minutes 3 seconds". Also, the "minute" sound file is missing, so for the moment until that file can be created the "minutes" file is used instead. (closes issue #16168) Reported by: nickilo Patches: patch-unified-trunk-rev-222176 uploaded by nickilo (license ) Tested by: nickilo, wonderg ........ * main/pbx.c, /: Merged revisions 237839 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r237839 | dvossel | 2010-01-05 13:29:47 -0600 (Tue, 05 Jan 2010) | 19 lines fixes subscriptions being lost after 'module reload' During a module reload if multiple extension configs are present, such as both extensions.conf and extensions.ael, watchers for one config's hints will be lost during the merging of the other config. This happens because hint watchers are only preserved for the current config being merged. The old context list is destroyed after the merging takes place, meaning any watchers that were not perserved will be removed. Now all hints are preserved during merging regardless of what config file is being merged. These hints are only restored if they are present within the new context list. (closes issue #16093) Reported by: jlaroff ........ 2010-01-05 17:19 +0000 [r237712] Russell Bryant * /, main/utils.c: Merged revisions 237699 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r237699 | russell | 2010-01-05 11:16:01 -0600 (Tue, 05 Jan 2010) | 14 lines Merged revisions 237697 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r237697 | russell | 2010-01-05 11:13:28 -0600 (Tue, 05 Jan 2010) | 7 lines Change a NOTICE log message to DEBUG where it belongs. (closes issue #16479) Reported by: alexrecarey (closes SWP-577) ........ ................ 2010-01-04 21:51 +0000 [r237407-237575] Tilghman Lesher * /, main/say.c: Merged revisions 237574 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r237574 | tilghman | 2010-01-04 15:48:20 -0600 (Mon, 04 Jan 2010) | 13 lines Merged revisions 237573 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r237573 | tilghman | 2010-01-04 15:45:46 -0600 (Mon, 04 Jan 2010) | 6 lines Bounds checking for input string (closes issue #16407) Reported by: qwell Patches: 20100104__issue16407.diff.txt uploaded by tilghman (license 14) ........ ................ * main/pbx.c, /: Merged revisions 237494 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r237494 | tilghman | 2010-01-04 14:59:01 -0600 (Mon, 04 Jan 2010) | 15 lines Merged revisions 237493 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r237493 | tilghman | 2010-01-04 14:57:35 -0600 (Mon, 04 Jan 2010) | 8 lines Regression in issue #15421 - Pattern matching (closes issue #16482) Reported by: wdoekes Patches: astsvn-16482-betterfix.diff uploaded by wdoekes (license 717) 20091223__issue16482.diff.txt uploaded by tilghman (license 14) Tested by: wdoekes, tilghman ........ ................ * main/config.c, /: Merged revisions 237414 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r237414 | tilghman | 2010-01-04 13:03:20 -0600 (Mon, 04 Jan 2010) | 2 lines Oops, didn't compile (thanks, kpfleming) ........ * main/config.c, /: Merged revisions 237410 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r237410 | tilghman | 2010-01-04 12:42:10 -0600 (Mon, 04 Jan 2010) | 7 lines Further reduce the encoded blank values back to blank in the realtime API. (closes issue #16533) Reported by: sergee Patches: 200100104__issue16533.diff.txt uploaded by tilghman (license 14) Tested by: sergee ........ * main/pbx.c, /, res/res_agi.c, include/asterisk/channel.h: Merged revisions 237406 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r237406 | tilghman | 2010-01-04 12:28:28 -0600 (Mon, 04 Jan 2010) | 23 lines Merged revisions 237405 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r237405 | tilghman | 2010-01-04 12:19:00 -0600 (Mon, 04 Jan 2010) | 16 lines Add a flag to disable the Background behavior, for AGI users. This is in a section of code that relates to two other issues, namely issue #14011 and issue #14940), one of which was the behavior of Background when called with a context argument that matched the current context. This fix broke FreePBX, however, in a post-Dial situation. Needless to say, this is an extremely difficult collision of several different issues. While the use of an exception flag is ugly, fixing all of the issues linked is rather difficult (although if someone would like to propose a better solution, we're happy to entertain that suggestion). (closes issue #16434) Reported by: rickead2000 Patches: 20091217__issue16434.diff.txt uploaded by tilghman (license 14) 20091222__issue16434__1.6.1.diff.txt uploaded by tilghman (license 14) Tested by: rickead2000 ........ ................ 2010-01-04 16:26 +0000 [r237324] Jeff Peeler * /, res/res_agi.c: Merged revisions 237323 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r237323 | jpeeler | 2010-01-04 10:24:51 -0600 (Mon, 04 Jan 2010) | 5 lines Fix timeout for AGI command speech recognize. (closes issue #16297) Reported by: semond ........ 2010-01-04 16:21 +0000 [r237320] Tilghman Lesher * channels/chan_local.c, /: Merged revisions 237319 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r237319 | tilghman | 2010-01-04 10:20:03 -0600 (Mon, 04 Jan 2010) | 10 lines Merged revisions 237318 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r237318 | tilghman | 2010-01-04 10:18:59 -0600 (Mon, 04 Jan 2010) | 3 lines It's also possible for the Local channel to directly execute an Application. Reviewboard: https://reviewboard.asterisk.org/r/452/ ........ ................ 2010-01-02 09:56 +0000 [r237137] Olle Johansson * /, channels/chan_sip.c: Merged revisions 237136 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r237136 | oej | 2010-01-02 10:54:22 +0100 (Lör, 02 Jan 2010) | 10 lines Merged revisions 237135 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r237135 | oej | 2010-01-02 10:52:30 +0100 (Lör, 02 Jan 2010) | 2 lines Release memory of the contact acl before unloading module ........ ................ 2009-12-30 22:00 +0000 [r236983] Tilghman Lesher * channels/chan_local.c, /: Merged revisions 236982 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r236982 | tilghman | 2009-12-30 15:59:18 -0600 (Wed, 30 Dec 2009) | 16 lines Merged revisions 236981 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236981 | tilghman | 2009-12-30 15:57:10 -0600 (Wed, 30 Dec 2009) | 9 lines Don't queue frames to channels that have no means to process them. (closes issue #15609) Reported by: aragon Patches: 20091230__issue16521__1.4__chan_local_only.diff.txt uploaded by tilghman (license 14) Tested by: aragon Review: https://reviewboard.asterisk.org/r/452/ ........ ................ 2009-12-30 21:12 +0000 [r236903] Jeff Peeler * /, utils/ael_main.c: Merged revisions 236902 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r236902 | jpeeler | 2009-12-30 15:09:28 -0600 (Wed, 30 Dec 2009) | 2 lines One more LOW_MEMORY compile fix. ........ 2009-12-30 17:55 +0000 [r236805-236849] Tilghman Lesher * /, cdr/cdr_adaptive_odbc.c: Merged revisions 236847 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r236847 | tilghman | 2009-12-30 11:53:29 -0600 (Wed, 30 Dec 2009) | 4 lines When the field is blank, don't warn about the field being unable to be coerced, just skip the column. (closes http://lists.digium.com/pipermail/asterisk-dev/2009-December/041362.html) Reported by Nic Colledge on the -dev list, fixed by me. ........ * /, channels/chan_sip.c: Merged revisions 236802 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r236802 | tilghman | 2009-12-29 17:05:45 -0600 (Tue, 29 Dec 2009) | 7 lines Shut down the SIP session timers more gracefully, in order to prevent a possible crash. (closes issue #16452) Reported by: corruptor Patches: 20091221__issue16452.diff.txt uploaded by tilghman (license 14) Tested by: corruptor ........ 2009-12-28 22:10 +0000 [r236714] Jason Parker * main/ast_expr2.c, /, main/ast_expr2.y: Merged revisions 236713 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r236713 | qwell | 2009-12-28 16:09:40 -0600 (Mon, 28 Dec 2009) | 8 lines Allow "REMAINDER" to function properly in expressions. (closes issue #16427) Reported by: wdoekes Patches: ast16-reminder-remainder.patch uploaded by wdoekes (license 717) Tested by: wdoekes ........ 2009-12-28 17:39 +0000 [r236668] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 236667 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r236667 | tilghman | 2009-12-28 11:37:46 -0600 (Mon, 28 Dec 2009) | 4 lines Use recommended option, not deprecated option. (closes issue #16515) Reported by: ManChicken ........ 2009-12-28 15:31 +0000 [r236511-236633] Sean Bright * include/asterisk/threadstorage.h, /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 236613 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r236613 | seanbright | 2009-12-28 10:22:54 -0500 (Mon, 28 Dec 2009) | 14 lines Merged revisions 236585 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236585 | seanbright | 2009-12-28 10:12:08 -0500 (Mon, 28 Dec 2009) | 7 lines Try a test compile to see if PTHREAD_ONCE_INIT requires extra braces. There was conditional code (based on build platform) to optioinally wrap PTHREAD_ONCE_INIT in braces that was removed since it is fixed in newer versions of Solaris/OpenSolaris, but I am still running into it on Solaris 10 x86 so add a configure-time check for it. ........ ................ * /, apps/app_meetme.c: Merged revisions 236510 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r236510 | seanbright | 2009-12-28 07:44:58 -0500 (Mon, 28 Dec 2009) | 19 lines Merged revisions 236509 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236509 | seanbright | 2009-12-28 07:43:36 -0500 (Mon, 28 Dec 2009) | 12 lines Avoid a crash with large numbers of MeetMe conferences. Similar to changes made to Queue(), when we have large numbers of conferences in meetme.conf (1000s) and we use alloca()/strdupa(), we can blow out the stack and crash, so instead just use a single fixed buffer. (closes issue #16509) Reported by: Kashif Raza Patches: 20091223_16509.patch uploaded by seanbright (license 71) Tested by: seanbright ........ ................ 2009-12-27 18:22 +0000 [r236435] Tilghman Lesher * contrib/init.d/rc.debian.asterisk, /: Merged revisions 236434 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r236434 | tilghman | 2009-12-27 12:20:53 -0600 (Sun, 27 Dec 2009) | 9 lines Merged revisions 236433 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236433 | tilghman | 2009-12-27 12:19:38 -0600 (Sun, 27 Dec 2009) | 2 lines Turn on colors in the daemon, since there's many requests for it on Ubuntu. ........ ................ 2009-12-26 15:29 +0000 [r236359] Kevin P. Fleming * /, sounds/Makefile: Merged revisions 236358 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r236358 | kpfleming | 2009-12-26 09:27:44 -0600 (Sat, 26 Dec 2009) | 9 lines Merged revisions 236357 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236357 | kpfleming | 2009-12-26 09:26:17 -0600 (Sat, 26 Dec 2009) | 1 line update to latest releases with zero uid/gid ........ ................ 2009-12-23 18:26 +0000 [r236187-236301] Tilghman Lesher * apps/app_stack.c, /: Merged revisions 236300 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r236300 | tilghman | 2009-12-23 12:25:27 -0600 (Wed, 23 Dec 2009) | 7 lines AGI may be invoked from outside the dialplan (closes issue #16510) Reported by: atis Patches: 20091223__issue16510.diff.txt uploaded by tilghman (license 14) Tested by: atis ........ * /, res/res_agi.c: Merged revisions 236186 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r236186 | tilghman | 2009-12-22 21:07:48 -0600 (Tue, 22 Dec 2009) | 11 lines Merged revisions 236184 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236184 | tilghman | 2009-12-22 20:55:24 -0600 (Tue, 22 Dec 2009) | 4 lines If EXEC only gets a single argument, don't crash when the second is used. (closes issue #16504) Reported by: bklang ........ ................ 2009-12-22 17:10 +0000 [r236066] David Vossel * /, channels/chan_sip.c: Merged revisions 236063 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r236063 | dvossel | 2009-12-22 11:00:08 -0600 (Tue, 22 Dec 2009) | 18 lines Merged revisions 236062 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236062 | dvossel | 2009-12-22 10:58:19 -0600 (Tue, 22 Dec 2009) | 11 lines fixes issue with p->method incorrectly set to ACK It is possible for a second ACK to come in for a retransmitted message. If an ack does not match an unacked message in our queue, restore the previous p->method as this ACK is completely ignored. (closes issue #16295) Reported by: omolenkamp Patches: issue16295_v2.diff uploaded by dvossel (license 671) ........ ................ 2009-12-21 19:54 +0000 [r235942] Jeff Peeler * res/res_monitor.c, /: Merged revisions 235941 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r235941 | jpeeler | 2009-12-21 13:54:20 -0600 (Mon, 21 Dec 2009) | 20 lines Merged revisions 235940 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r235940 | jpeeler | 2009-12-21 13:43:41 -0600 (Mon, 21 Dec 2009) | 13 lines Change Monitor to not assume file to write to does not contain pathing. 227944 changed the fname_base argument to always append the configured monitor path. This change was necessary to properly compare files for uniqueness. If a full path is given though, nothing needs to be appended and that is handled correctly now. (closes issue #16377) (closes issue #16376) Reported by: bcnit Patches: res_monitor.c-issue16376-1.patch uploaded by dant (license 670) ........ ................ 2009-12-21 17:11 +0000 [r235824] Tilghman Lesher * /, main/features.c: Merged revisions 235822 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r235822 | tilghman | 2009-12-21 11:00:46 -0600 (Mon, 21 Dec 2009) | 15 lines Merged revisions 235821 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r235821 | tilghman | 2009-12-21 10:45:03 -0600 (Mon, 21 Dec 2009) | 8 lines Send parking lot announcement to the channel which parked the call, not the park-ee. (closes issue #16234) Reported by: yeshuawatso Patches: 20091210__issue16234.diff.txt uploaded by tilghman (license 14) 20091221__issue16234__1.4.diff.txt uploaded by tilghman (license 14) Tested by: yeshuawatso ........ ................ 2009-12-18 22:58 +0000 [r235662] Jeff Peeler * main/channel.c, /, include/asterisk/cdr.h: Merged revisions 235660 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r235660 | jpeeler | 2009-12-18 16:51:37 -0600 (Fri, 18 Dec 2009) | 55 lines Merged revisions 235635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r235635 | jpeeler | 2009-12-18 16:29:51 -0600 (Fri, 18 Dec 2009) | 48 lines Correct CDR dispositions for BUSY/FAILED This patch is simple in that it reorders the disposition defines so that the fix for issue 12946 works properly (the default CDR disposition was changed to AST_CDR_NOANSWER). Also, the AST_CDR_FLAG_ORIGINATED flag was set in ast_call to ensure all CDR records are written. The side effects of CDR changes are scary, so I'm documenting the test cases performed to attempt to catch any regressions. The following tests were all performed using 1.4 rev 195881 vs head (235571) + patch: A calls B C calls B (busy) Hangup C Hangup A (Both SIP and features) A calls B A blind transfers to C Hangup C (Both SIP and features) A calls B A attended transfers to C Hangup C A calls B A attended transfers to C (SIP) C blind transfers to A (features) Hangup A All of the test scenario CDRs matched. The following tests were performed just with the patch to ensure proper operation (with unanswered=yes): exten =>s,1,Answer exten =>s,n,ResetCDR(w) exten =>s,n,ResetCDR(w) exten =>s,1,ResetCDR(w) exten =>s,n,ResetCDR(w) (closes issue #16180) Reported by: aatef Patches: bug16180.patch uploaded by jpeeler (license 325) ........ ................ 2009-12-18 22:42 +0000 [r235574-235657] Tilghman Lesher * /, configure, configure.ac: Merged revisions 235656 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r235656 | tilghman | 2009-12-18 16:40:46 -0600 (Fri, 18 Dec 2009) | 9 lines Merged revisions 235652 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r235652 | tilghman | 2009-12-18 16:39:30 -0600 (Fri, 18 Dec 2009) | 2 lines Revise verbiage, per #asterisk-dev discussion ........ ................ * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 235573 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r235573 | tilghman | 2009-12-18 15:19:43 -0600 (Fri, 18 Dec 2009) | 9 lines Merged revisions 235572 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r235572 | tilghman | 2009-12-18 15:18:16 -0600 (Fri, 18 Dec 2009) | 2 lines Point to the typical missing package, not the cryptic "termcap support". ........ ................ 2009-12-17 Leif Madsen * Release Asterisk 1.6.0.20 2009-12-09 Leif Madsen * Release Asterisk 1.6.0.20-rc1 2009-12-09 20:00 +0000 [r233841-233881] Russell Bryant * main/loader.c, /: Fix breakage of the "module load " CLI command. * main/loader.c, formats/format_ilbc.c, formats/format_vox.c, include/asterisk/module.h, formats/format_pcm.c, formats/format_h263.c, formats/format_g723.c, formats/format_h264.c, formats/format_jpeg.c, formats/format_g726.c, formats/format_gsm.c, formats/format_g729.c, main/editline/makelist.in, formats/format_sln.c, formats/format_wav.c, formats/format_ogg_vorbis.c, UPGRADE.txt, UPGRADE-1.6.txt, formats/format_wav_gsm.c, formats/format_sln16.c: Set a module load priority for format modules. A recent change to app_voicemail made it such that the module now assumes that all format modules are available while processing voicemail configuration. However, when autoloading modules, it was possible that app_voicemail was loaded before the format modules. Since format modules don't depend on anything, set a module load priority on them to ensure that they get loaded first when autoloading. This version of the patch is specific to Asterisk 1.4 and 1.6.0. These versions did not already support module load priority in the module API. This adds a trivial version of this which is just a module flag to include it in a pass before loading "everything". Thanks to mmichelson for the review! (closes issue #16412) Reported by: jiddings Tested by: russell Review: https://reviewboard.asterisk.org/r/445/ 2009-12-08 18:28 +0000 [r233729] Tilghman Lesher * /, res/res_musiconhold.c: Merged revisions 233718 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r233718 | tilghman | 2009-12-08 12:22:44 -0600 (Tue, 08 Dec 2009) | 8 lines Find another ref leak and change how we manage module references. (closes issue #16388) Reported by: parisioa Patches: 20091208__issue16388.diff.txt uploaded by tilghman (license 14) Tested by: parisioa, tilghman Review: https://reviewboard.asterisk.org/r/442/ ........ 2009-12-07 23:57 +0000 [r233617] Atis Lezdins * contrib/valgrind.supp, /: Merged revisions 233577 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r233577 | atis | 2009-12-08 01:10:13 +0200 (Tue, 08 Dec 2009) | 8 lines Fix compatibility with valgrind 3.3 and older. (noticed in issue #16388) Reported by: parisioa Patches: valgrind.supp uloaded by atis (license 242) Tested by: atis, parisioa ........ 2009-12-07 23:30 +0000 [r233475-233614] David Vossel * /, main/utils.c: Merged revisions 233611 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r233611 | dvossel | 2009-12-07 17:28:51 -0600 (Mon, 07 Dec 2009) | 4 lines fixes incorrect logic in ast_uri_encode issue #16299 ........ * /, channels/chan_sip.c: Merged revisions 233472 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r233472 | dvossel | 2009-12-07 12:08:46 -0600 (Mon, 07 Dec 2009) | 15 lines Merged revisions 233471 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r233471 | dvossel | 2009-12-07 12:07:38 -0600 (Mon, 07 Dec 2009) | 9 lines fixes missing Contact header angle brackets (closes issue #16298) Reported by: mgernoth Patches: reg_parse_issue_1.4.diff uploaded by dvossel (license 671) Tested by: dvossel ........ ................ 2009-12-07 16:16 +0000 [r233397] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 233394 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r233394 | mnicholson | 2009-12-07 10:14:42 -0600 (Mon, 07 Dec 2009) | 8 lines Do not reject SDP packets describing only non audio streams. (closes issue #16387) Reported by: zalex1953 Patches: media-level-c-fix1.diff uploaded by mnicholson (license 96) Tested by: mnicholson, zalex1953 ........ 2009-12-04 21:56 +0000 [r233284] David Vossel * configs/iax.conf.sample, /: Merged revisions 233280 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r233280 | dvossel | 2009-12-04 15:54:44 -0600 (Fri, 04 Dec 2009) | 14 lines Merged revisions 233279 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r233279 | dvossel | 2009-12-04 15:54:01 -0600 (Fri, 04 Dec 2009) | 7 lines clarify requirecalltoken option in iax.sample.conf (closes issue #16223) Reported by: bklang Patches: clarify-iax-requirecalltoken.patch uploaded by bklang (license 919) ........ ................ 2009-12-04 20:29 +0000 [r233236] Matthias Nick * /, pbx/pbx_config.c: Merged revisions 233093 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r233093 | mnick | 2009-12-04 11:15:47 -0600 (Fri, 04 Dec 2009) | 8 lines Parse global variables or expressions in hint extensions Parse global variables or expressions in hint extensions. Like: exten => 400,hint,DAHDI/i2/${GLOBAL(var)} (closes issue #16166) Reported by: rmudgett Tested by: mnick, rmudgett ........ 2009-12-04 20:11 +0000 [r233230] Russell Bryant * /: unblock a rev. 2009-12-04 17:39 +0000 [r233167] David Vossel * apps/app_voicemail.c, /: Merged revisions 233121 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r233121 | dvossel | 2009-12-04 11:22:31 -0600 (Fri, 04 Dec 2009) | 12 lines Merged revisions 233116 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r233116 | dvossel | 2009-12-04 11:21:34 -0600 (Fri, 04 Dec 2009) | 6 lines document and rename strip_control() in app_voicemail (closes issue #16291) Reported by: wdoekes ........ ................ 2009-12-04 17:20 +0000 [r233112] Russell Bryant * main/channel.c, /: Merged revisions 233100 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r233100 | russell | 2009-12-04 11:18:22 -0600 (Fri, 04 Dec 2009) | 14 lines Merged revisions 233092 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r233092 | russell | 2009-12-04 11:12:47 -0600 (Fri, 04 Dec 2009) | 7 lines Only do frame payload check for HOLD frames. This code was added for helping to debug the source of invalid HOLD frames. However, a side effect of this is that it will incorrectly report errors for frames that have an integer payload. Make the check for this block specific to the HOLD frame case. ........ ................ 2009-12-04 15:46 +0000 [r233047] Matthias Nick * main/dsp.c, /: Merged revisions 233046 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r233046 | mnick | 2009-12-04 09:38:33 -0600 (Fri, 04 Dec 2009) | 17 lines Merged revisions 233014 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r233014 | mnick | 2009-12-04 09:17:03 -0600 (Fri, 04 Dec 2009) | 11 lines Warning message gets displayed only once Added additional field 'int display_inband_dtmf_warning', which when set to '1' displays the warning ('Inband DTMF is not supported on codec %s. Use RFC2833'), and when set to '0' doesn't display the warning. Otherwise you would get hundreds of warnings every second. (closes issue #15769) Reported by: falves11 Patches: patch_15769_14.txt uploaded by mnick (license 874) Tested by: mnick, falves11 ........ ................ 2009-12-03 21:03 +0000 [r232864] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 232854 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r232854 | tilghman | 2009-12-03 14:47:07 -0600 (Thu, 03 Dec 2009) | 15 lines Merged revisions 232820 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r232820 | tilghman | 2009-12-03 14:10:19 -0600 (Thu, 03 Dec 2009) | 8 lines Deprecate "cz" in favor of "cs". Also, change the use of language codes so that language registers as a prefix, rather than an exact match. (closes issue #16272) Reported by: patrol-cz Patches: 20091203__issue16272.diff.txt uploaded by tilghman (license 14) ........ ................ 2009-12-03 14:47 +0000 [r232811] David Ruggles * apps/app_externalivr.c: Merged revisions 232587 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r232587 | diruggles | 2009-12-02 17:17:22 -0500 (Wed, 02 Dec 2009) | 12 lines Prevent double closing of FDs by EIVR This caused a problem when asterisk was under heavy load and running both AGI and EIVR applications. EIVR would close an FD at which point it would be considered freed and be used by a new AGI instance the second close would then close the FD now in use by AGI. (closes issue #16305) Reported by: diLLec Tested by: thedavidfactor, diLLec Review: https://reviewboard.asterisk.org/r/436/ ........ 2009-12-03 00:32 +0000 [r232699] Tilghman Lesher * /, res/res_musiconhold.c: Merged revisions 232660-232661 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r232660 | tilghman | 2009-12-02 18:08:55 -0600 (Wed, 02 Dec 2009) | 19 lines Fix multiple issues with musiconhold, which led to classes not getting destroyed properly. * Classes are now tracked past removal from the core container, and module removal is actively prevented until all references are freed. * A hanging reference stored in the channel has been removed. This could have caused a mismatch and the music state not properly cleared, if two or more reloads occurred between MOH being stopped and MOH being restarted. * In certain circumstances, duplicate classes were possible. * A race existed at reload time between a process being killed and the thread responsible for reading from the related pipe respawning that process. * Several reference counts have also been corrected. At least one could have caused deleted classes to stick around forever, consuming resources. This originally manifested as MOH external processes that were not killed at reload time. (closes issue #16279, closes issue #16207) Reported by: parisioa, dcabot Patches: 20091202__issue16279__2.diff.txt uploaded by tilghman (license 14) Tested by: parisioa, tilghman ........ r232661 | tilghman | 2009-12-02 18:09:36 -0600 (Wed, 02 Dec 2009) | 2 lines Remove debugging line ........ 2009-12-02 22:03 +0000 [r232577-232583] Jeff Peeler * main/manager.c, /: Merged revisions 232582 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r232582 | jpeeler | 2009-12-02 16:02:43 -0600 (Wed, 02 Dec 2009) | 14 lines Merged revisions 232581 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r232581 | jpeeler | 2009-12-02 15:57:42 -0600 (Wed, 02 Dec 2009) | 7 lines Send ack (response/message) after receiving manager action userevent (closes issue #16264) Reported by: dimas Patches: event-ack.patch uploaded by dimas (license 88) ........ ................ * main/manager.c, /: Merged revisions 232576 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r232576 | jpeeler | 2009-12-02 15:32:50 -0600 (Wed, 02 Dec 2009) | 8 lines Make manager response to "Action: events" finish with empty line (closes issue #16275) Reported by: vnovy Patches: manager.c.diff uploaded by vnovy (license 922) ........ 2009-12-02 17:08 +0000 [r232357] Joshua Colp * /, apps/app_amd.c: Merged revisions 232356 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r232356 | file | 2009-12-02 13:06:54 -0400 (Wed, 02 Dec 2009) | 12 lines Merged revisions 232355 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r232355 | file | 2009-12-02 13:04:52 -0400 (Wed, 02 Dec 2009) | 5 lines Fix a bug where if you hung up very quickly after calling AMD it would overwrite the AMDSTATUS of HANGUP with TOOLONG. (closes issue #16239) Reported by: CGMChris ........ ................ 2009-12-02 17:03 +0000 [r232354] David Vossel * /, main/acl.c: Merged revisions 232351 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r232351 | dvossel | 2009-12-02 11:00:15 -0600 (Wed, 02 Dec 2009) | 12 lines Merged revisions 232350 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r232350 | dvossel | 2009-12-02 10:59:18 -0600 (Wed, 02 Dec 2009) | 6 lines ast_outaddrfor doesn't do htons() on port, looks odd in strace. (closes issue #16290) Reported by: wdoekes ........ ................ 2009-12-02 16:41 +0000 [r232346] Joshua Colp * /, channels/chan_sip.c: Merged revisions 232345 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r232345 | file | 2009-12-02 12:40:14 -0400 (Wed, 02 Dec 2009) | 7 lines Add support for handling the 415 Unsupported media type response like we do for a 488 Not acceptable here response. (closes issue #16186) Reported by: atis Patches: sip_t38_response_415.patch uploaded by atis (license 242) ........ 2009-12-02 15:45 +0000 [r232272] David Vossel * funcs/func_groupcount.c, /: Merged revisions 232269 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r232269 | dvossel | 2009-12-02 09:42:54 -0600 (Wed, 02 Dec 2009) | 15 lines Merged revisions 232268 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r232268 | dvossel | 2009-12-02 09:41:36 -0600 (Wed, 02 Dec 2009) | 9 lines fixes segfault in func_groupcount closes issue #16337) Reported by: Parantido Patches: issue_16337.diff uploaded by dvossel (license 671) Tested by: Parantido, dvossel ........ ................ 2009-12-02 00:49 +0000 [r232092] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 232091 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r232091 | jpeeler | 2009-12-01 18:45:18 -0600 (Tue, 01 Dec 2009) | 17 lines Merged revisions 232090 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r232090 | jpeeler | 2009-12-01 18:42:58 -0600 (Tue, 01 Dec 2009) | 10 lines Do not modify the gain settings on data calls. (The digital flag actually represents a data call.) (closes issue #15972) Reported by: udosw Patches: transcap_digital_fix.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis ........ ................ 2009-12-01 23:39 +0000 [r232009-232013] Russell Bryant * /, funcs/func_lock.c: Merged revisions 232012 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r232012 | russell | 2009-12-01 17:38:34 -0600 (Tue, 01 Dec 2009) | 2 lines Fix a build error on FreeBSD. ........ * /, main/file.c: Merged revisions 232008 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r232008 | russell | 2009-12-01 17:27:53 -0600 (Tue, 01 Dec 2009) | 9 lines Merged revisions 232007 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r232007 | russell | 2009-12-01 17:25:36 -0600 (Tue, 01 Dec 2009) | 2 lines Fix a warning pointed out by buildbot. ........ ................ 2009-12-01 21:57 +0000 [r231928] Jeff Peeler * main/channel.c, /: Merged revisions 231927 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r231927 | jpeeler | 2009-12-01 15:54:21 -0600 (Tue, 01 Dec 2009) | 19 lines Merged revisions 231911 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231911 | jpeeler | 2009-12-01 15:29:31 -0600 (Tue, 01 Dec 2009) | 12 lines Fix crash with invalid frame data The crash was happening as a result of a frame containing an invalid data pointer, but was set with data length of zero. The few times the issue was reproduced it _seemed_ that the frame was queued properly, that is the data pointer was set to NULL. I never could reproduce the crash so as a last resort the crash has been fixed, but a check in __ast_read has been added to give as much information about the source of problematic frames in the future. (closes issue #16058) Reported by: atis ........ ................ 2009-12-01 21:22 +0000 [r231879] David Vossel * main/pbx.c, /: Merged revisions 231867 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r231867 | dvossel | 2009-12-01 15:20:19 -0600 (Tue, 01 Dec 2009) | 9 lines Merged revisions 231853 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231853 | dvossel | 2009-12-01 15:14:31 -0600 (Tue, 01 Dec 2009) | 3 lines WaitExten m option with no parameters generates frame with zero datalen but non-null data ptr ........ ................ 2009-12-01 15:51 +0000 [r231744] Matthew Nicholson * /, main/file.c: Merged revisions 231741 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r231741 | mnicholson | 2009-12-01 09:47:36 -0600 (Tue, 01 Dec 2009) | 9 lines Merged revisions 231740 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231740 | mnicholson | 2009-12-01 09:34:57 -0600 (Tue, 01 Dec 2009) | 2 lines Ignore unknown formats in ast_format_str_reduce() and return an error if no know formats are found. ........ ................ 2009-11-30 21:52 +0000 [r231693] Kevin P. Fleming * main/udptl.c, /, channels/chan_sip.c, include/asterisk/udptl.h: Merged revisions 231692 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r231692 | kpfleming | 2009-11-30 15:47:42 -0600 (Mon, 30 Nov 2009) | 22 lines Another round of UDPTL stack fixes/improvements: 1) Allow users of UDPTL stack to associate a character-string tag with a UDPTL session, so that log/error/debug messages generated by the UDPTL stack can be 'connected' to the endpoint that caused them to be generated. 2) Improve comments (and process) of calculating the far end's maximum IFP size when redundancy mode is in use for error correction. 3) When an IFP larger than the calculated 'far max IFP' size is presented for writing, truncate it rather than putting in the buffer and allowing the buffer to overflow; this will cause the ends to retrain to a lower bit rate that produces IFPs of an appropriate size if possible, and if not possible, the FAX transfer will fail completely. In these cases, it is due to the one endpoint supplying a T38FaxMaxDatagram value that is improperly calculated and is too low to be of use; we have configuration options available to override this behavior. 4) Eliminate use of T38FaxMaxDatagram value in udptl.conf; it is no longer needed. ........ 2009-11-30 21:37 +0000 [r231691] Matthew Nicholson * apps/app_voicemail.c, include/asterisk/file.h, /, main/file.c, main/app.c: Merged revisions 231688 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r231688 | mnicholson | 2009-11-30 15:31:55 -0600 (Mon, 30 Nov 2009) | 15 lines Merged revisions 231614 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231614 | mnicholson | 2009-11-30 15:11:44 -0600 (Mon, 30 Nov 2009) | 8 lines Remove duplicate entries from voicemail format lists. This prevents app_voicemail from entering an infinite loop when the same format is specified twice in the format list. (closes issue #15625) Reported by: Shagg63 Tested by: mnicholson Review: https://reviewboard.asterisk.org/r/429/ ........ ................ 2009-11-30 20:45 +0000 [r231603] Joshua Colp * /, channels/chan_sip.c: Merged revisions 231602 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r231602 | file | 2009-11-30 16:44:30 -0400 (Mon, 30 Nov 2009) | 5 lines When receiving SDP that matches the version of the last one do not treat it as a fatal error. (closes issue #16238) Reported by: seandarcy ........ 2009-11-30 18:58 +0000 [r231517-231560] David Vossel * /, apps/app_queue.c: Merged revisions 231556 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r231556 | dvossel | 2009-11-30 12:55:07 -0600 (Mon, 30 Nov 2009) | 11 lines app_queue crashes randomly, often during call-transfers This patch adds a ref to the queue_ent object's parent call_queue in queue_exec() so the call_queue won't be destroyed while the the queue_ent still holds a pointer to it. (closes issue 0015686) Tested by: dvossel, aragon ........ * main/rtp.c, /: Merged revisions 231491 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r231491 | dvossel | 2009-11-30 11:28:28 -0600 (Mon, 30 Nov 2009) | 17 lines Merged revisions 231441 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231441 | dvossel | 2009-11-30 11:14:08 -0600 (Mon, 30 Nov 2009) | 11 lines fixes crash caused by RTP comfort noise payload greater than 24 bytes AST-2009-010 (closes issue #16242) Reported by: amorsen Patches: issue16242.diff uploaded by oej (license 306) Tested by: amorsen, oej, dvossel ........ ................ 2009-11-25 22:34 +0000 [r231300] Tilghman Lesher * main/channel.c, /: Merged revisions 231299 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r231299 | tilghman | 2009-11-25 16:33:02 -0600 (Wed, 25 Nov 2009) | 9 lines Merged revisions 231298 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231298 | tilghman | 2009-11-25 16:31:57 -0600 (Wed, 25 Nov 2009) | 2 lines After a frame duplication failure, unlock the channel before returning. ........ ................ 2009-11-25 15:48 +0000 [r231192] Matthew Nicholson * /, pbx/pbx_lua.c: Merged revisions 231189 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r231189 | mnicholson | 2009-11-25 09:42:48 -0600 (Wed, 25 Nov 2009) | 4 lines Load pbx_lua with global symbols to allow linking with other lua libraries. Found by Maxim Litnitskiy. ........ 2009-11-24 18:53 +0000 [r231096] Jeff Peeler * /, main/features.c: Merged revisions 231095 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r231095 | jpeeler | 2009-11-24 12:50:36 -0600 (Tue, 24 Nov 2009) | 11 lines Fix erroneous hangup extension execution ast_spawn_extension behaves differently from 1.4 in that hangups and extensions that do not exist do not return an error, whereas in 1.6 it does. This is now taken into account so that the AST_FLAG_BRIDGE_HANGUP_RUN flag gets set properly. (closes issue #16106) Reported by: ajohnson Tested by: ajohnson ........ 2009-11-23 15:46 +0000 [r230882] Joshua Colp * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 230881 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r230881 | file | 2009-11-23 09:45:45 -0600 (Mon, 23 Nov 2009) | 7 lines Change fax detection in chan_sip so it behaves as one would expect. Internally the way T.38 is negotiated has changed and the option no longer reflects a behavior that is valid. It will now look for a CNG tone on received calls and if present send the call to the 'fax' extension. It is then up to the application or channel to request the switch over to T.38. ........ 2009-11-23 15:35 +0000 [r230782-230878] Kevin P. Fleming * /, channels/chan_sip.c: Merged revisions 230877 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r230877 | kpfleming | 2009-11-23 09:34:16 -0600 (Mon, 23 Nov 2009) | 9 lines Merged revisions 230839 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r230839 | kpfleming | 2009-11-23 09:09:24 -0600 (Mon, 23 Nov 2009) | 1 line Correct fix for issue #16268... the reporter's original patch was very close to correct. ........ ................ * /, channels/chan_sip.c: Merged revisions 230773 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r230773 | kpfleming | 2009-11-23 08:15:48 -0600 (Mon, 23 Nov 2009) | 12 lines Merged revisions 230772 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r230772 | kpfleming | 2009-11-23 08:13:56 -0600 (Mon, 23 Nov 2009) | 5 lines Ensure that SDP parsing does not ignore the last line of the SDP. (closes issue #16268) Reported by: sgimeno ........ ................ 2009-11-20 22:37 +0000 [r230729] David Vossel * /, channels/chan_iax2.c: Merged revisions 230726 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r230726 | dvossel | 2009-11-20 16:35:54 -0600 (Fri, 20 Nov 2009) | 7 lines fixes iax2 show cache locking error, thanks alecdavis! (closes issue #16094) Reported by: alecdavis Patches: bug16094.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis, dvossel ........ 2009-11-20 21:09 +0000 [r230631] Matthew Nicholson * /, main/features.c: Merged revisions 230628 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r230628 | mnicholson | 2009-11-20 15:01:10 -0600 (Fri, 20 Nov 2009) | 15 lines Merged revisions 230627 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r230627 | mnicholson | 2009-11-20 14:53:06 -0600 (Fri, 20 Nov 2009) | 8 lines Copy the peer CDR's userfield to the bridge CDR if it exists. This is necessary for the recordagentcalls option in chan_agent to store the recorded file name in the bridge CDR. (closes issue #14590) Reported by: msetim Patches: queue_agent_userfield.patch uploaded by Laureano (license 265) Tested by: Laureano, mnicholson ........ ................ 2009-11-20 17:37 +0000 [r230512-230587] David Vossel * /, include/asterisk/audiohook.h, main/audiohook.c: Merged revisions 230583 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r230583 | dvossel | 2009-11-20 11:26:20 -0600 (Fri, 20 Nov 2009) | 6 lines audiohook signal trigger on every status change (issue #14618) Review: https://reviewboard.asterisk.org/r/434/ ........ * apps/app_mixmonitor.c, /: Merged revisions 230509 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r230509 | dvossel | 2009-11-19 15:26:21 -0600 (Thu, 19 Nov 2009) | 17 lines Merged revisions 230508 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r230508 | dvossel | 2009-11-19 15:22:46 -0600 (Thu, 19 Nov 2009) | 10 lines 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/ ........ ................ 2009-11-30 Leif Madsen * Release Asterisk 1.6.0.19 * AST-2009-010 * SDP parser regression fix (issue #16268, issue #16238) 2009-11-18 Leif Madsen * Release Asterisk 1.6.0.18 2009-11-13 Leif Madsen * Release Asterisk 1.6.0.18-rc3 2009-11-13 15:56 +0000 [r229913] Joshua Colp * /, channels/chan_sip.c: Merged revisions 229912 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r229912 | file | 2009-11-13 09:56:16 -0600 (Fri, 13 Nov 2009) | 2 lines Fix T.38 negotiation regression introduced with the SDP parser changes. ........ 2009-11-12 16:49 +0000 [r229673] David Vossel * funcs/func_audiohookinherit.c, /: Merged revisions 229670 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r229670 | dvossel | 2009-11-12 10:44:39 -0600 (Thu, 12 Nov 2009) | 12 lines Merged revisions 229669 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r229669 | dvossel | 2009-11-12 10:41:49 -0600 (Thu, 12 Nov 2009) | 6 lines fixes merging error, datastore was being freed in the wrong function. (closes issue #16219) Reported by: aragon ........ ................ 2009-11-11 19:51 +0000 [r229475-229500] David Brooks * main/pbx.c: Merged revisions 229499 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r229499 | dbrooks | 2009-11-11 13:48:18 -0600 (Wed, 11 Nov 2009) | 15 lines Merged revisions 229498 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r229498 | dbrooks | 2009-11-11 13:46:19 -0600 (Wed, 11 Nov 2009) | 8 lines Solaris doesn't like NULL going to ast_log Solaris will crash if NULL is passed to ast_log. This simple patch simply uses S_OR to get around this. (closes issue #15392) Reported by: yrashk ........ ................ * /, apps/app_softhangup.c: Merged revisions 229460 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r229460 | dbrooks | 2009-11-11 12:13:56 -0600 (Wed, 11 Nov 2009) | 7 lines Flags not initialized in app_softhangup.c, causing undefined behavior Trivial patch [kobaz] to initialize an ast_flags = {0} (closes issue #16129) Reported by: kobaz ........ 2009-11-10 22:17 +0000 [r229363] Tilghman Lesher * main/pbx.c, /: Merged revisions 229361 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r229361 | tilghman | 2009-11-10 16:14:22 -0600 (Tue, 10 Nov 2009) | 19 lines Merged revisions 229360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r229360 | tilghman | 2009-11-10 16:09:16 -0600 (Tue, 10 Nov 2009) | 12 lines If two pattern classes start with the same digit and have the same number of characters, they will compare equal. The example given in the issue report is that of [234] and [246], which have these characteristics, yet they are clearly not equivalent. The code still uses these two characteristics, yet when the two scores compare equal, an additional check will be done to compare all characters within the class to verify equality. (closes issue #15421) Reported by: jsmith Patches: 20091109__issue15421__2.diff.txt uploaded by tilghman (license 14) Tested by: jsmith, thedavidfactor ........ ................ 2009-11-10 22:03 +0000 [r229357] David Ruggles * doc/externalivr.txt: Merged revisions 229356 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r229356 | diruggles | 2009-11-10 17:01:50 -0500 (Tue, 10 Nov 2009) | 16 lines Merged revisions 229355 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r229355 | diruggles | 2009-11-10 16:45:15 -0500 (Tue, 10 Nov 2009) | 9 lines Fix ExternalIVR Documentation Remove documentation for event that doesn't function (closes issue #16220) Reported by: thedavidfactor Patches: externalivr.txt.20091110.1622.patch uploaded by thedavidfactor (license 903) ........ ................ 2009-11-10 21:31 +0000 [r229352] Tilghman Lesher * apps/app_stack.c, /: Merged revisions 229351 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r229351 | tilghman | 2009-11-10 15:22:50 -0600 (Tue, 10 Nov 2009) | 7 lines When GOSUB is invoked within an AGI, it may not exit correctly. (closes issue #16216) Reported by: atis Patches: 20091110__atis_work.diff.txt uploaded by tilghman (license 14) Tested by: atis ........ 2009-11-10 20:07 +0000 [r229283] Joshua Colp * /, codecs/codec_g726.c: Merged revisions 229282 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r229282 | file | 2009-11-10 16:06:13 -0400 (Tue, 10 Nov 2009) | 15 lines Merged revisions 229281 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r229281 | file | 2009-11-10 16:03:14 -0400 (Tue, 10 Nov 2009) | 8 lines Remove broken support for direct transcoding between G.726 RFC3551 and G.726 AAL2. On some systems the translation core would actually consider g726aal2 -> g726 -> signed linear to be a quicker path then g726aal2 -> signed linear which exposed this problem. (closes issue #15504) Reported by: globalnetinc ........ ................ 2009-11-10 17:54 +0000 [r229234] David Vossel * /, channels/chan_iax2.c: Merged revisions 229168 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r229168 | dvossel | 2009-11-10 11:16:49 -0600 (Tue, 10 Nov 2009) | 15 lines Merged revisions 229167 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r229167 | dvossel | 2009-11-10 11:15:57 -0600 (Tue, 10 Nov 2009) | 9 lines don't crash on log message in solaris AST-2009-006 (closes issue #16206) Reported by: bklang Tested by: bklang ........ ................ 2009-11-10 17:37 +0000 [r229229] David Ruggles * doc/externalivr.txt: Merged revisions 229228 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r229228 | diruggles | 2009-11-10 12:33:47 -0500 (Tue, 10 Nov 2009) | 18 lines Merged revisions 229191 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r229191 | diruggles | 2009-11-10 12:23:59 -0500 (Tue, 10 Nov 2009) | 11 lines Document ExternalIVR event tag collision ExternalIVR uses the D tag for two different event types. This documents that behavior and how to differentiate between the two cases. Also includes a minor spelling fix and clarification (closes issue #16211) Reported by: thedavidfactor Patches: externalivr.txt.20091109.1507.patch uploaded by thedavidfactor (license 903) ........ ................ 2009-11-10 15:41 +0000 [r229100] Matthew Nicholson * channels/chan_sip.c: Reverted revision 202006. (closes issue #16175) Reported by: paul-tg 2009-11-10 11:19 +0000 [r229057] Gavin Henry * contrib/scripts/asterisk.ldap-schema, /: Merged revisions 229050 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r229050 | ghenry | 2009-11-10 11:16:10 +0000 (Tue, 10 Nov 2009) | 20 lines Schema file additions * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses to allow standalone dialplan, account and mailbox entries (STRUCTURAL) * Added new Fields: - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir, - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap, - AstAccountVideoSupport, AstAccountIgnoreSDPVersion * Removed redundant IPaddr (there's already IPAddress) - Gives more configuration Flags for SIP-Users available (tested) - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses without extensibleObject (which really should be the last resort); gives also additional possibilities for LDAP-filter (closes issue #15874) Reported by: Medozas Patches: asterisk.ldap-schema.patch uploaded by Medozas (license 41) Tested by: Medozas, suretec ........ 2009-11-09 Leif Madsen * Release Asterisk 1.6.0.18-rc2 2009-11-09 15:39 +0000 [r228898] Leif Madsen * main/channel.c: Merged revisions 228897 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228897 | lmadsen | 2009-11-09 09:38:38 -0600 (Mon, 09 Nov 2009) | 14 lines Merged revisions 228896 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228896 | lmadsen | 2009-11-09 09:37:43 -0600 (Mon, 09 Nov 2009) | 6 lines Update WARNING message. Update a WARNING message to give a suggested fix when encountered. (closes issue #16198) Reported by: atis Tested by: atis ........ ................ 2009-11-09 14:58 +0000 [r228861] Matthew Nicholson * /, include/asterisk/lock.h: Merged revisions 228858 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228858 | mnicholson | 2009-11-09 08:37:07 -0600 (Mon, 09 Nov 2009) | 15 lines Merged revisions 228827 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228827 | mnicholson | 2009-11-09 08:16:03 -0600 (Mon, 09 Nov 2009) | 8 lines Perform limited bounds checking when destroying ast_mutex_t structures to make sure we don't try to use negative indices. (closes issue #15588) Reported by: zerohalo Patches: 20090820__issue15588.diff.txt uploaded by tilghman (license 14) Tested by: zerohalo ........ ................ 2009-11-06 22:38 +0000 [r228696] David Vossel * main/channel.c, /: Merged revisions 228693 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228693 | dvossel | 2009-11-06 16:35:44 -0600 (Fri, 06 Nov 2009) | 16 lines Merged revisions 228692 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228692 | dvossel | 2009-11-06 16:33:27 -0600 (Fri, 06 Nov 2009) | 9 lines fixes audiohook write crash occuring in chan_spy whisper mode. After writing to the audiohook list in ast_write(), frames were being freed incorrectly. Under certain conditions this resulted in a double free crash. (closes issue #16133) Reported by: wetwired ........ ................ 2009-11-06 20:42 +0000 [r228651] Matthew Nicholson * funcs/func_base64.c, /, main/utils.c: Merged revisions 228620 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228620 | mnicholson | 2009-11-06 13:47:11 -0600 (Fri, 06 Nov 2009) | 15 lines Merged revisions 228378 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228378 | mnicholson | 2009-11-06 10:26:59 -0600 (Fri, 06 Nov 2009) | 8 lines Properly handle '=' while decoding base64 messages and null terminate strings returned from BASE64_DECODE. (closes issue #15271) Reported by: chappell Patches: base64_fix.patch uploaded by chappell (license 8) Tested by: kobaz ........ ................ 2009-11-06 18:40 +0000 [r228549] Joshua Colp * /, channels/chan_sip.c: Merged revisions 228548 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228548 | file | 2009-11-06 14:37:59 -0400 (Fri, 06 Nov 2009) | 11 lines Merged revisions 228547 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228547 | file | 2009-11-06 14:32:58 -0400 (Fri, 06 Nov 2009) | 4 lines Don't overwrite caller ID name on a trunk with the configured fullname when using users.conf (issue ABE-1989) ........ ................ 2009-11-06 Leif Madsen * Release Asterisk 1.6.0.18-rc1 2009-11-06 17:53 +0000 [r228479-228500] Joshua Colp * /, doc/tex/localchannel.tex: Merged revisions 228499 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r228499 | file | 2009-11-06 13:52:00 -0400 (Fri, 06 Nov 2009) | 2 lines Fix the localchannel.tex file. ........ * channels/chan_sip.c: Fix a logic flaw I introduced when I was testing stuff out. 2009-11-06 17:10 +0000 [r228423] David Vossel * /, codecs/codec_ilbc.c: Merged revisions 228420 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228420 | dvossel | 2009-11-06 11:09:01 -0600 (Fri, 06 Nov 2009) | 19 lines Merged revisions 228418 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228418 | dvossel | 2009-11-06 11:07:13 -0600 (Fri, 06 Nov 2009) | 13 lines fixes segfault in iLBC For reasons not yet known, it appears possible for an ast_frame to have a datalen greater than zero while the actual data is NULL during Packet Loss Concealment. Most codecs don't support PLC so this doesn't affect them. This patch catches the malformed frame and prevents the crash from occuring. Additional efforts to determine why it is possible for a frame to look like this are still being investigated. (issue #16979) ........ ................ 2009-11-06 16:56 +0000 [r228411-228415] Joshua Colp * channels/chan_sip.c: Fix a crash caused by freeing a dialog directly instead of using dialog_unref. (closes issue #16097) Reported by: steinwej Patches: no_RTP.diff uploaded by steinwej (license 841) * /, main/abstract_jb.c: Merged revisions 228410 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228410 | file | 2009-11-06 12:42:23 -0400 (Fri, 06 Nov 2009) | 14 lines Merged revisions 228409 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228409 | file | 2009-11-06 12:41:20 -0400 (Fri, 06 Nov 2009) | 7 lines Fix a bug caused by a partially invalid frame (from the jitterbuffer) passing through the Asterisk core. (closes issue #15560) Reported by: jvandal (closes issue #15709) Reported by: covici ........ ................ 2009-11-06 15:44 +0000 [r228271-228342] David Vossel * /, main/astfd.c: Merged revisions 228339 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228339 | dvossel | 2009-11-06 09:42:46 -0600 (Fri, 06 Nov 2009) | 12 lines Merged revisions 228338 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228338 | dvossel | 2009-11-06 09:41:41 -0600 (Fri, 06 Nov 2009) | 5 lines fixes crash in astfd.c (closes issue #15981) Reported by: slavon ........ ................ * funcs/func_audiohookinherit.c, /: Merged revisions 228268 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r228268 | dvossel | 2009-11-06 09:04:24 -0600 (Fri, 06 Nov 2009) | 9 lines fixes memory leak in func_audiohookinherit.c (closes issue #15394) Reported by: boroda Patches: bug15394_memoryleak_diff2.txt uploaded by dbrooks (license 790) Tested by: dbrooks, boroda ........ 2009-11-05 21:24 +0000 [r228190] Jeff Peeler * apps/app_chanspy.c, /: Merged revisions 228189 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r228189 | jpeeler | 2009-11-05 15:23:06 -0600 (Thu, 05 Nov 2009) | 11 lines Fix the fix for chanspy option o In 224178, I assumed the uploaded patch was correct as it had received positive feedback. The flags were being checked in the incorrect location. Upon testing the fix this time it was also found that the flags from the dialplan weren't being copied to the chanspy_translation_helper. (closes issue #16167) Reported by: marhbere ........ 2009-11-05 19:39 +0000 [r228146] David Brooks * channels/chan_misdn.c, /: Merged revisions 228145 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228145 | dbrooks | 2009-11-05 13:34:50 -0600 (Thu, 05 Nov 2009) | 16 lines Merged revisions 228078 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228078 | dbrooks | 2009-11-05 12:59:41 -0600 (Thu, 05 Nov 2009) | 9 lines chan_misdn Asterisk 1.4.27-rc2 crash Crash related to chan_misdn connection. Patch submitted by gknispel_proformatique, tested by francesco_r. "I have many crash since i have upgraded to Asterisk 1.4.27-rc2. Attached a full bt." This patch zeros out an ast_frame. (closes issue #16041) Reported by: francesco_r ........ ................ 2009-11-05 19:17 +0000 [r228081] Jason Parker * channels/chan_vpb.cc, /: Merged revisions 228080 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228080 | qwell | 2009-11-05 13:16:29 -0600 (Thu, 05 Nov 2009) | 15 lines Merged revisions 228079 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228079 | qwell | 2009-11-05 13:14:25 -0600 (Thu, 05 Nov 2009) | 8 lines Fix crash on VPB exception when no hardware is present. (closes issue #14970) Reported by: tzafrir Patches: vpb_exception.diff uploaded by tzafrir (license 46) Tested by: markwaters ........ ................ 2009-11-04 23:52 +0000 [r227946] Jeff Peeler * res/res_monitor.c, /: Merged revisions 227945 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r227945 | jpeeler | 2009-11-04 17:50:59 -0600 (Wed, 04 Nov 2009) | 21 lines Merged revisions 227944 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227944 | jpeeler | 2009-11-04 17:47:08 -0600 (Wed, 04 Nov 2009) | 14 lines Fix incorrect filename comparsion after monitor file change The logic to detect if a requested file is indeed a different file from the current file was incorrect. The main issue being confusion of the use of filename_base which was previously set without pathing information and then compared to another full path. Robust file comparison logic has been added to properly check if two files are the same even if symlinks are used. (closes issue #15313) Reported by: caspy Patches: 20091103__issue15313__1.4.diff.txt uploaded by jpeeler (license 325) but mostly tilghman's work ........ ................ 2009-11-04 21:15 +0000 [r227763-227833] Matthew Nicholson * apps/app_dial.c, /: Merged revisions 227829 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r227829 | mnicholson | 2009-11-04 15:03:33 -0600 (Wed, 04 Nov 2009) | 17 lines Merged revisions 227827 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227827 | mnicholson | 2009-11-04 14:52:27 -0600 (Wed, 04 Nov 2009) | 10 lines 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/ ........ ................ * channels/chan_sip.c: Modify the SDP parsing code to parse session and media level items separately. With the new code, media level proprieties should no longer be confused with session level proprieties. This change also reorganizes some of the SDP parsing code which should make it easier to manage in the future. (closes issue #14994) Reported by: frawd 2009-11-04 19:27 +0000 [r227717-227743] Joshua Colp * /, static-http/prototype.js: Merged revisions 227739 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r227739 | file | 2009-11-04 15:26:19 -0400 (Wed, 04 Nov 2009) | 12 lines Merged revisions 227735 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227735 | file | 2009-11-04 15:25:37 -0400 (Wed, 04 Nov 2009) | 5 lines Fix a security issue where it may be possible for someone to execute a cross-site AJAX request exploit. (AST-2009-009) ........ ................ * /, channels/chan_sip.c: Merged revisions 227712 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r227712 | file | 2009-11-04 15:20:46 -0400 (Wed, 04 Nov 2009) | 12 lines Merged revisions 227700 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227700 | file | 2009-11-04 15:17:39 -0400 (Wed, 04 Nov 2009) | 5 lines Fix a security issue where sending a REGISTER with a differing username in the From URI and Authorization header would reveal whether it was valid or not. (AST-2009-008) ........ ................ 2009-11-03 20:00 +0000 [r227373] Jason Parker * Makefile, /, main/Makefile: Merged revisions 227372 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r227372 | qwell | 2009-11-03 13:59:46 -0600 (Tue, 03 Nov 2009) | 9 lines Fix some build issues on Solaris. (closes issue #14517) (SWP-109) Reported by: asgaroth Patches: bug_14517.diff uploaded by snuffy (license 35) Tested by: asgaroth, snuffy, dougm, qwell ........ 2009-11-03 19:49 +0000 [r227362-227369] Leif Madsen * apps/app_controlplayback.c, /: Merged revisions 227368 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r227368 | lmadsen | 2009-11-03 13:48:53 -0600 (Tue, 03 Nov 2009) | 8 lines Change warning message to debug message. app_controlplayback outputs a warning, when in fact it is normal. (closes issue #16071) Reported by: atis Patches: controlplayback_warning.patch uploaded by atis (license 242) ........ * configs/extensions.conf.sample, /: Merged revisions 227361 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r227361 | lmadsen | 2009-11-03 13:25:18 -0600 (Tue, 03 Nov 2009) | 11 lines Additional fixes to the extensions.conf.sample file. Update the extensions.conf.sample [stdexten] context so that we use the variable instead of requiring it to be passed explicitly. Also updated uses of the [stdexten] context throughout. (closes issue #15858) Reported by: pprindeville Patches: stdexten-context-update.txt uploaded by lmadsen (license 10) Tested by: pprindeville ........ 2009-11-03 18:05 +0000 [r227278] Richard Mudgett * channels/chan_dahdi.c: Merged revisions 227275 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227275 | rmudgett | 2009-11-03 11:55:47 -0600 (Tue, 03 Nov 2009) | 4 lines Make sure the outgoing flag is cleared if a new channel fails to get created for outgoing calls. This is the relevant portion of asterisk/trunk -r226648 ........ 2009-11-03 15:37 +0000 [r227168] Joshua Colp * /, channels/chan_sip.c: Merged revisions 227167 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r227167 | file | 2009-11-03 11:37:08 -0400 (Tue, 03 Nov 2009) | 12 lines Merged revisions 227166 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227166 | file | 2009-11-03 11:36:16 -0400 (Tue, 03 Nov 2009) | 5 lines Fix a bug where an RPID header could be generated with a blank username in the URI. (closes issue #15909) Reported by: kobaz ........ ................ 2009-11-03 15:24 +0000 [r227163] Leif Madsen * configs/extensions.conf.sample, /: Merged revisions 227162 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r227162 | lmadsen | 2009-11-03 09:19:47 -0600 (Tue, 03 Nov 2009) | 7 lines Update extensions.conf.sample file to fix incorrect extensions. (closes issue #15857) Reported by: pprindeville Patches: stdexten.patch#2 uploaded by pprindeville (license 347) Tested by: pprindeville ........ 2009-11-03 11:21 +0000 [r227102] Olle Johansson * /, channels/chan_sip.c: Merged revisions 227091 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r227091 | oej | 2009-11-03 12:11:15 +0100 (Tis, 03 Nov 2009) | 15 lines Merged revisions 227088 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227088 | oej | 2009-11-03 11:29:59 +0100 (Tis, 03 Nov 2009) | 7 lines Use proper response code when violating Contact ACL's. https://reviewboard.asterisk.org/r/415/ Thanks kpfleming for a quick review. (EDVX-003) ........ ................ 2009-11-02 21:05 +0000 [r226975-226976] David Brooks * channels/chan_sip.c: SIP channel name uniqueness SIP channel names were supposed to be unique by way of a name suffix derived from the pointer to the channel's private data. Uniqueness was preserved on 32-bit systems, but not on 64-bit systems. This patch, as suggested by kpfleming, replaces this suffix with a simple incremented unsigned int. (closes issue #15152) Reported by: palbrecht Review: https://reviewboard.asterisk.org/r/420/ * /: SIP channel name uniqueness SIP channel names were supposed to be unique by way of a name suffix derived from the pointer to the channel's private data. Uniqueness was preserved on 32-bit systems, but not on 64-bit systems. This patch, as suggested by kpfleming, replaces this suffix with a simple incremented unsigned int. (closes issue #15152) Reported by: palbrecht Review: https://reviewboard.asterisk.org/r/420/ 2009-11-02 18:09 +0000 [r226891] Joshua Colp * apps/app_dial.c, /: Merged revisions 226890 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r226890 | file | 2009-11-02 14:08:54 -0400 (Mon, 02 Nov 2009) | 18 lines Merged revisions 226889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226889 | file | 2009-11-02 14:08:11 -0400 (Mon, 02 Nov 2009) | 11 lines 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) ........ ................ 2009-11-02 17:16 +0000 [r226813] Tilghman Lesher * /, contrib/init.d/rc.redhat.asterisk: Merged revisions 226812 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r226812 | tilghman | 2009-11-02 11:15:31 -0600 (Mon, 02 Nov 2009) | 15 lines Merged revisions 226811 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226811 | tilghman | 2009-11-02 11:14:20 -0600 (Mon, 02 Nov 2009) | 8 lines Don't allow two separate instances of safe_asterisk when restarting from the init script. (closes issue #14562) Reported by: davidw Patches: Initially 20091022__issue14562.diff.txt uploaded by tilghman (license 14) Modified to 20091030__Issue14562_diff.txt uploaded by davidw (license 780) Tested by: davidw ........ ................ 2009-10-29 18:14 +0000 [r226533] Joshua Colp * channels/chan_local.c, /, doc/tex/localchannel.tex: Merged revisions 226532 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r226532 | file | 2009-10-29 15:13:42 -0300 (Thu, 29 Oct 2009) | 13 lines Merged revisions 226531 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226531 | file | 2009-10-29 15:11:26 -0300 (Thu, 29 Oct 2009) | 6 lines Add an option to enabling passing music on hold start and stop requests through instead of acting on them in chan_local. (closes issue #14709) Reported by: dimas ........ ................ 2009-10-28 20:17 +0000 [r226381-226387] Leif Madsen * configs/sip.conf.sample: Merged revisions 226384 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r226384 | lmadsen | 2009-10-28 15:11:07 -0500 (Wed, 28 Oct 2009) | 17 lines Merged revisions 226382 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226382 | lmadsen | 2009-10-28 15:06:13 -0500 (Wed, 28 Oct 2009) | 9 lines Update documentation in sip.conf.sample. Update the documentation in sip.conf.sample in order to make it more clear that directmedia/canreinvite do not cause Asterisk to ignore reINVITEs. It is only used to stop Asterisk from generating a reINVITE, but does not stop it from accepting them if necessary. (closes issue #15644) Reported by: lmadsen ........ ................ * /, doc/tex/channelvariables.tex: Merged revisions 226378 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r226378 | lmadsen | 2009-10-28 14:50:00 -0500 (Wed, 28 Oct 2009) | 15 lines Merged revisions 226377 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226377 | lmadsen | 2009-10-28 14:48:29 -0500 (Wed, 28 Oct 2009) | 7 lines Update CALLINGSUBADDR channel variable documentation. (closes issue #15734) Reported by: alecdavis Patches: channelvariables.tex.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis ........ ................ 2009-10-28 18:05 +0000 [r226167-226306] Tilghman Lesher * /, include/asterisk/linkedlists.h: Merged revisions 226305 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r226305 | tilghman | 2009-10-28 13:04:05 -0500 (Wed, 28 Oct 2009) | 9 lines Merged revisions 226304 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226304 | tilghman | 2009-10-28 13:02:25 -0500 (Wed, 28 Oct 2009) | 2 lines Fix documentation (pointed out by TheDavidFactor on #-dev) ........ ................ * main/manager.c, /: Merged revisions 226159 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r226159 | tilghman | 2009-10-27 15:22:07 -0500 (Tue, 27 Oct 2009) | 14 lines Merged revisions 226138 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226138 | tilghman | 2009-10-27 15:16:49 -0500 (Tue, 27 Oct 2009) | 7 lines Manager output is not always NULL-terminated, so force a NULL at the end of the filestream. (closes issue #15495) Reported by: pdf Patches: 20090916__issue15495.diff.txt uploaded by tilghman (license 14) Tested by: pdf ........ ................ 2009-10-26 23:13 +0000 [r226019] Tzafrir Cohen * /, configure, configure.ac: detect ARM Linux EABI OSARCH as linux-gnu instead of linux-gnueabi * Set OSARCH to linux-gnu even if host_os is linux-gnueabi * When checking if we are Linux, check OSARCH rather than host_os The newer ARM ABI ("EABI") shows the OS name 'linux-gnueabi' rather than 'linux-gnu' . This patch sets OSARCH to be 'linux-gnu' even in such a case. OSARCH is tested for the value of 'linux-gnu' in one or two places in the tree. This patch also fixes the check libcap to check for $OSARCH rather than $host_os . See also: http://wiki.debian.org/ArmEabiPort Merged revisions 225957 via svnmerge from http://svn.digium.com/svn/asterisk/branches/1.4 Merged revisions 226018 via svnmerge from http://svn.digium.com/svn/asterisk/trunk 2009-10-26 15:46 +0000 [r225869] Kevin P. Fleming * apps/app_fax.c: Backport audio handling loop fixes from trunk version of app_fax. This backport resolves some issues handling audio frames during FAX processing, and ensures that the FAX application doesn't accidentally get notified of a T.38 switchover at the end of a successful FAX. (issue #16127) 2009-10-23 14:05 +0000 [r225583] Kevin P. Fleming * Makefile, /: Merged revisions 225582 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r225582 | kpfleming | 2009-10-23 09:02:42 -0500 (Fri, 23 Oct 2009) | 17 lines Merged revisions 225581 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225581 | kpfleming | 2009-10-23 09:00:01 -0500 (Fri, 23 Oct 2009) | 10 lines Don't force menuselect.makeopts to be rebuilt on every build. For some reason the menuselect.makeopts file was listed as PHONY in the Makefile, resulting in 'make' needing to rebuild it for every build. This then resulted in the embedded module rules being rebuilt on every build, which can be slow and is unnecessary. This patch fixes the problem by properly allowing 'make' to know when the menuselect.makeopts file needs to be rebuilt (defining the proper dependencies). ........ ................ 2009-10-22 21:53 +0000 [r225486] Leif Madsen * doc/valgrind.txt, contrib/valgrind.supp (added): Merged revisions 225485 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r225485 | lmadsen | 2009-10-22 16:52:30 -0500 (Thu, 22 Oct 2009) | 19 lines Merged revisions 225484 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225484 | lmadsen | 2009-10-22 16:51:52 -0500 (Thu, 22 Oct 2009) | 11 lines Clean valgrind output by suppressing false errors. Update valgrind.txt documentation and add valgrind.supp file in order to allow those who are creating valgrind output to have less false errors in the logfile. (closes issue #16007) Reported by: atis Patches: valgrind.txt.diff uploaded by atis (license 242) asterisk2.supp uploaded by atis (license 242) Tested by: atis, amorsen ........ ................ 2009-10-22 17:13 +0000 [r225361] Tilghman Lesher * main/pbx.c, /, apps/app_meetme.c, include/asterisk/channel.h: Merged revisions 225360 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r225360 | tilghman | 2009-10-22 12:11:23 -0500 (Thu, 22 Oct 2009) | 11 lines Merged revisions 225105 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225105 | tilghman | 2009-10-21 11:02:12 -0500 (Wed, 21 Oct 2009) | 4 lines Fix documentation for ast_softhangup() and correct the misuse thereof. (closes issue #16103) Reported by: majorbloodnok ........ ................ 2009-10-21 22:10 +0000 [r225310-225311] David Vossel * /, channels/chan_iax2.c: Merged revisions 225307 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r225307 | dvossel | 2009-10-21 16:58:46 -0500 (Wed, 21 Oct 2009) | 20 lines Merged revisions 225243 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225243 | dvossel | 2009-10-21 15:58:08 -0500 (Wed, 21 Oct 2009) | 13 lines IAX2: VNAK loop caused by signaling frames with no destination call number It is possible for the PBX thread to queue up signaling frames before a destination call number is received. This can result in signaling frames being sent out with no destination call number. Since recent versions of Asterisk require accurate destination callnumbers for all Full Frames, this can cause a VNAK loop to occur. To resolve this no signaling frames are sent until a destination callnumber is received, and destination call numbers are now only required for iax_pvt matching when the frame is an ACK. Review: https://reviewboard.asterisk.org/r/413/ ........ ................ * configs/iax.conf.sample, /, channels/chan_sip.c, configs/sip.conf.sample, channels/chan_iax2.c: Merged revisions 225033 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r225033 | dvossel | 2009-10-21 09:39:10 -0500 (Wed, 21 Oct 2009) | 27 lines Merged revisions 225032 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225032 | dvossel | 2009-10-21 09:37:04 -0500 (Wed, 21 Oct 2009) | 20 lines IAX/SIP shrinkcallerid option The shrinking of caller id removes '(', ' ', ')', non-trailing '.', and '-' from the string. This means values such as 555.5555 and test-test result in 555555 and testtest. There are instances, such as Skype integration, where a specific value is passed via caller id that must be preserved unmodified. This patch makes the shrinking of caller id optional in chan_sip and chan_iax in order to support such cases. By default this option is on to preserve previous expected behavior. (closes issue #15940) Reported by: dimas Patches: v2-15940.patch uploaded by dimas (license 88) 15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671) Tested by: dvossel Review: https://reviewboard.asterisk.org/r/408/ ........ ................ 2009-10-21 03:15 +0000 [r224933] Russell Bryant * include/asterisk/translate.h, main/dsp.c, main/frame.c, /, main/translate.c, include/asterisk/dsp.h, codecs/codec_dahdi.c, include/asterisk/frame.h: Merged revisions 224932 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r224932 | russell | 2009-10-20 22:09:04 -0500 (Tue, 20 Oct 2009) | 12 lines Merged revisions 224931 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224931 | russell | 2009-10-20 21:59:54 -0500 (Tue, 20 Oct 2009) | 5 lines Isolate frames returned from a DSP instance or codec translator. The reasoning for these changes are the same as what I wrote in the commit message for rev 222878. ........ ................ 2009-10-20 22:10 +0000 [r224857] Tilghman Lesher * /, main/audiohook.c: Merged revisions 224856 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r224856 | tilghman | 2009-10-20 17:09:07 -0500 (Tue, 20 Oct 2009) | 12 lines Merged revisions 224855 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224855 | tilghman | 2009-10-20 17:07:11 -0500 (Tue, 20 Oct 2009) | 5 lines Pay attention to the return value of the manipulate function. While this looks like an optimization, it prevents a crash from occurring when used with certain audiohook callbacks (diagnosed with SVN trunk, backported to 1.4 to keep the source consistent across versions). ........ ................ 2009-10-20 17:48 +0000 [r224775] Joshua Colp * /, main/features.c: Merged revisions 224774 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r224774 | file | 2009-10-20 14:47:34 -0300 (Tue, 20 Oct 2009) | 12 lines Merged revisions 224773 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224773 | file | 2009-10-20 14:46:37 -0300 (Tue, 20 Oct 2009) | 5 lines Add support for relaying early media in the features attended transfer option. (closes issue #14828) Reported by: licedey ........ ................ 2009-10-19 23:50 +0000 [r224672] Kevin P. Fleming * main/rtp.c, /: Merged revisions 224671 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r224671 | kpfleming | 2009-10-19 18:47:39 -0500 (Mon, 19 Oct 2009) | 14 lines Merged revisions 224670 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224670 | kpfleming | 2009-10-19 18:44:07 -0500 (Mon, 19 Oct 2009) | 7 lines Correct timestamp calculations when RTP sample rates over 8kHz are used. While testing some endpoints that support 16kHz and 32kHz sample rates, some log messages were generated due to calc_rxstamp() computing timestamps in a way that produced odd results, so this patch sanitizes the result of the computations. ........ ................ 2009-10-19 19:50 +0000 [r224568] Joshua Colp * apps/app_dial.c, /: Merged revisions 224567 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r224567 | file | 2009-10-19 16:49:09 -0300 (Mon, 19 Oct 2009) | 12 lines Merged revisions 224565 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224565 | file | 2009-10-19 16:47:50 -0300 (Mon, 19 Oct 2009) | 5 lines Do not attempt early media bridging (ie: direct RTP setup) if options are enabled that should prevent it. (closes issue #14763) Reported by: cupotka ........ ................ 2009-10-19 00:12 +0000 [r224449] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 224448 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r224448 | tilghman | 2009-10-18 19:05:56 -0500 (Sun, 18 Oct 2009) | 3 lines Allow ODBC storage to be queried with multiple mailboxes. This corrects an issue reported on the -users list. ........ 2009-10-17 02:03 +0000 [r224332-224337] Jeff Peeler * channels/chan_dahdi.c: fix typo, sorry * channels/chan_dahdi.c, /: Merged revisions 224331 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r224331 | jpeeler | 2009-10-16 20:36:08 -0500 (Fri, 16 Oct 2009) | 20 lines Merged revisions 224330 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224330 | jpeeler | 2009-10-16 20:32:47 -0500 (Fri, 16 Oct 2009) | 13 lines Fix stale caller id data from being reported in AMI NewChannel event The problem here is that chan_dahdi is designed in such a way to set certain values in the dahdi_pvt only once. One of those such values is the configured caller id data in chan_dahdi.conf. For PRI, the configured caller id data could be overwritten during a call. Instead of saving the data and restoring, it was decided that for all non-analog channels it was simply best to not set the configured caller id in the first place and also clear it at the end of the call. (closes issue #15883) Reported by: jsmith ........ ................ 2009-10-16 20:48 +0000 [r224262] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 224261 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r224261 | rmudgett | 2009-10-16 15:40:57 -0500 (Fri, 16 Oct 2009) | 25 lines Merged revisions 224260 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224260 | rmudgett | 2009-10-16 15:25:23 -0500 (Fri, 16 Oct 2009) | 18 lines Never released PRI channels when using Busy() or Congestion() dialplan apps. When the Busy() or Congestion() application is used towards ISDN (an ISDN progress is sent), the responding ISDN Disconnect or Release may contain the ISDN cause user busy or one of the congestion causes. In chan_dahdi.c these causes will only set the needbusy or needcongestion flags and not activate the softhangup procedure. Unfortunately only the latter can interrupt the endless wait loop of Busy()/Congestion(). Result: PRI channels staying in state busy for the rest of asterisk life or until the other end times out and forces the call to clear. (in issue 0014292) Reported by: tomaso Patches: disc_rel_userbusy.patch uploaded by tomaso (license 564) (This patch is unrelated to the issue.) ........ ................ 2009-10-15 15:57 +0000 [r224179] Jeff Peeler * apps/app_chanspy.c, /: Merged revisions 224178 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r224178 | jpeeler | 2009-10-15 10:57:14 -0500 (Thu, 15 Oct 2009) | 11 lines Readd removed ability to allow listening to one side of the call in app_chanspy (Option o) (closes issue #15675) Reported by: john8675309 Patches: issue15675patchtrunk.txt uploaded by dbrooks (license 790) Tested by: jgutierrez on users list: http://lists.digium.com/pipermail/asterisk-users/2009-October/239155.html ........ 2009-10-12 23:50 +0000 [r223833] Jeff Peeler * apps/app_dial.c, /: Merged revisions 223832 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r223832 | jpeeler | 2009-10-12 18:48:09 -0500 (Mon, 12 Oct 2009) | 15 lines Merged revisions 223804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223804 | jpeeler | 2009-10-12 18:12:50 -0500 (Mon, 12 Oct 2009) | 8 lines 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 ........ ................ 2009-10-12 21:07 +0000 [r223759] David Vossel * configs/iax.conf.sample, /: Merged revisions 223756 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r223756 | dvossel | 2009-10-12 15:58:27 -0500 (Mon, 12 Oct 2009) | 5 lines Clarifies trunkmaxsize, trunkfreq, and trunkmtu iax2 options SWP-151 ........ 2009-10-12 14:28 +0000 [r223653] Kevin P. Fleming * /, channels/chan_sip.c, apps/app_fax.c: Merged revisions 223652 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r223652 | kpfleming | 2009-10-12 09:25:29 -0500 (Mon, 12 Oct 2009) | 13 lines Remove automatic switching from T.38 to voice mode in chan_sip. chan_sip has some code to automatically switch from T.38 mode to voice mode when a voice frame is written to the channel while it is in T.38 mode; this was intended to handle the situation when a FAX transmission has ended and the channel is not yet hung up, but is causing problems at the beginning of FAX sessions as well when there are still voice frames 'in flight' at the time the T.38 negotiation completes. This patch removes the automatic switchover, and changes app_fax to explicitly switch off T.38 mode when the FAX transmission process ends. (closes issue #16025) Reported by: jamicque ........ 2009-10-11 17:27 +0000 [r223488] Russell Bryant * main/autoservice.c, /: Merged revisions 223487 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r223487 | russell | 2009-10-11 12:25:42 -0500 (Sun, 11 Oct 2009) | 17 lines Merged revisions 223485-223486 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223485 | russell | 2009-10-11 12:22:52 -0500 (Sun, 11 Oct 2009) | 6 lines Don't use data outside of its scope. The purpose of this code was to have a hangup frame put on the list of deferred frames. However, the code that read the hangup frame was outside of the scope of where the hangup frame was declared. ........ r223486 | russell | 2009-10-11 12:25:06 -0500 (Sun, 11 Oct 2009) | 2 lines Remove some unnecessary code. ........ ................ 2009-10-09 23:08 +0000 [r223404] Jeff Peeler * channels/chan_dahdi.c, channels/chan_h323.c: Fix interpretation of PRIREDIRECTIONREASON set by chan_sip. This commit is the simplest way to solve a problem that has already been solved in trunk with the "COLP/CONP and Redirecting party information into Asterisk" commit. In trunk the redirection reason is translated into a generic redirect reason. I would have had to do the same fix except chan_sip never reads PRIREDIRECTREASON. So both chan_dahdi and chan_h323 have been modified to interpret the one different redirect reason of "no-answer" properly and set the ISDN reason code 2 of "no reply". (closes issue #15033) Reported by: steinwej 2009-10-09 20:59 +0000 [r223331] Kevin P. Fleming * /, apps/app_fax.c: Merged revisions 223330 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r223330 | kpfleming | 2009-10-09 15:58:44 -0500 (Fri, 09 Oct 2009) | 10 lines Initiate T.38 switchover when acting as called party, regardless of FAX direction. SendFAX() and ReceiveFAX() can be given options to indicate whether they should act as the calling or called party; this mode should be used to decide whether to initiate a switchover to T.38, not the direction that the FAX transfer will take place. (closes issue #16039) Reported by: jamicque ........ 2009-10-09 18:36 +0000 [r223276] Matthew Nicholson * main/channel.c, /: Merged revisions 223273 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r223273 | mnicholson | 2009-10-09 13:34:08 -0500 (Fri, 09 Oct 2009) | 14 lines Merged revisions 223225 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223225 | mnicholson | 2009-10-09 13:20:11 -0500 (Fri, 09 Oct 2009) | 8 lines Signal timeouts by returning AST_CONTROL_RINGING when originating calls. (closes issue #15104) Reported by: nblasgen Patches: manager-timeout1.diff uploaded by mnicholson (license 96) Tested by: nblasgen, mnicholson ........ ................ 2009-10-09 18:20 +0000 [r223226] Mark Michelson * apps/app_dial.c, /: Merged revisions 223215 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r223215 | mmichelson | 2009-10-09 13:17:34 -0500 (Fri, 09 Oct 2009) | 9 lines Recorded merge of revisions 223213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223213 | mmichelson | 2009-10-09 13:17:12 -0500 (Fri, 09 Oct 2009) | 3 lines Fix potential memory leak in app_dial.c ........ ................ 2009-10-09 17:57 +0000 [r223210] David Vossel * /, channels/chan_sip.c: Merged revisions 223206 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r223206 | dvossel | 2009-10-09 12:53:37 -0500 (Fri, 09 Oct 2009) | 16 lines Merged revisions 223205 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223205 | dvossel | 2009-10-09 12:52:35 -0500 (Fri, 09 Oct 2009) | 10 lines fixes sip registration using authuser in user.conf (closes issue #14954) Reported by: tornblad Tested by: mmichelson, tornblad, dvossel ........ ................ 2009-10-09 17:27 +0000 [r223172] Matthew Nicholson * cdr/cdr_sqlite3_custom.c, /: Merged revisions 223136 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r223136 | mnicholson | 2009-10-09 12:14:38 -0500 (Fri, 09 Oct 2009) | 8 lines Don't close the sqlite database when reloading. Only close the database when unloading. (closes issue #15953) Reported by: frawd Patches: sqlite3_rev220097.diff uploaded by frawd (license 610) Tested by: frawd ........ 2009-10-09 17:11 +0000 [r223091-223135] David Vossel * /, channels/chan_sip.c: Merged revisions 223132 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r223132 | dvossel | 2009-10-09 11:54:02 -0500 (Fri, 09 Oct 2009) | 9 lines 'auth=' did not parse md5 secret correctly (closes issue #15949) Reported by: ebroad Patches: authparsefix.patch uploaded by ebroad (license 878) 15949_trunk.diff uploaded by dvossel (license 671) Tested by: ebroad ........ * /, channels/chan_sip.c: Merged revisions 223088 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r223088 | dvossel | 2009-10-09 10:49:30 -0500 (Fri, 09 Oct 2009) | 14 lines p->peerauth is always empty in transmit_register() When using callbackextension or specifing the peer name in a registration string, the peer's specific auth settings set by the "auth=" strings within the peer definition are not used by the registration. Thanks to ebroad for reporting the issue and providing the patch. (closes issue #15955) Reported by: ebroad Patches: regauthfix.patch uploaded by ebroad (license 878) ........ 2009-10-08 19:54 +0000 [r222881] Russell Bryant * include/asterisk/file.h, main/frame.c, /, main/file.c, include/asterisk/frame.h: Merged revisions 222880 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r222880 | russell | 2009-10-08 14:52:03 -0500 (Thu, 08 Oct 2009) | 51 lines Merged revisions 222878 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222878 | russell | 2009-10-08 14:45:47 -0500 (Thu, 08 Oct 2009) | 44 lines Make filestream frame handling safer by isolating frames before returning them. This patch is related to a number of issues on the bug tracker that show crashes related to freeing frames that came from a filestream. A number of fixes have been made over time while trying to figure out these problems, but there re still people seeing the crash. (Note that some of these bug reports include information about other problems. I am specifically addressing the filestream frame crash here.) I'm still not clear on what the exact problem is. However, what is _very_ clear is that we have seen quite a few problems over time related to unexpected behavior when we try to use embedded frames as an optimization. In some cases, this optimization doesn't really provide much due to improvements made in other areas. In this case, the patch modifies filestream handling such that the embedded frame will not be returned. ast_frisolate() is used to ensure that we end up with a completely mallocd frame. In reality, though, we will not actually have to malloc every time. For filestreams, the frame will almost always be allocated and freed in the same thread. That means that the thread local frame cache will be used. So, going this route doesn't hurt. With this patch in place, some people have reported success in not seeing the crash anymore. (SWP-150) (AST-208) (ABE-1834) (issue #15609) Reported by: aragon Patches: filestream_frisolate-1.4.diff2.txt uploaded by russell (license 2) Tested by: aragon, russell (closes issue #15817) Reported by: zerohalo Tested by: zerohalo (closes issue #15845) Reported by: marhbere Review: https://reviewboard.asterisk.org/r/386/ ........ ................ 2009-10-08 19:42 +0000 [r222876] David Vossel * main/netsock.c, /, include/asterisk/netsock.h: Merged revisions 222873 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r222873 | dvossel | 2009-10-08 14:35:30 -0500 (Thu, 08 Oct 2009) | 6 lines fixes an ast_netsock_list memory leak. ABE-1998 Review: https://reviewboard.asterisk.org/r/395/ ........ 2009-10-08 16:47 +0000 [r222693-222800] Richard Mudgett * channels/misdn_config.c, /: Merged revisions 222799 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r222799 | rmudgett | 2009-10-08 11:44:33 -0500 (Thu, 08 Oct 2009) | 19 lines Merged revisions 222797 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222797 | rmudgett | 2009-10-08 11:33:06 -0500 (Thu, 08 Oct 2009) | 12 lines Fix memory leak if chan_misdn config parameter is repeated. Memory leak when the same config option is set more than once in an misdn.conf section. Why must this be considered? Templates! Defining a template with default port options and later adding to or overriding some of them. Patches: memleak-misdn.patch JIRA ABE-1998 ........ ................ * channels/chan_misdn.c, /: Merged revisions 222692 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r222692 | rmudgett | 2009-10-07 16:56:36 -0500 (Wed, 07 Oct 2009) | 21 lines Merged revisions 222691 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222691 | rmudgett | 2009-10-07 16:51:24 -0500 (Wed, 07 Oct 2009) | 14 lines chan_misdn.c:process_ast_dsp() memory leak misdn.conf: astdtmf must be set to "yes". With "no", buffer loss does not occur. The translated frame "f2" when passing through ast_dsp_process() is not freed whenever it is not used further in process_ast_dsp(). Then in the end it is never ever freed. Patches: translate.patch JIRA ABE-1993 ........ ................ 2009-10-07 18:35 +0000 [r222605] Sean Bright * main/pbx.c: Properly initialize ast_devstate_aggregate so we don't crash sporadically. This looks like it was just missed during a merge. (closes issue #15841) Reported by: amorsen Patches: ast_devstate_aggregate_init-in-ast_extension_state2.patch uploaded by amorsen (license 676) Tested by: amorsen (closes issue #15852) Reported by: amorsen Tested by: amorsen, farisraouf 2009-10-07 17:47 +0000 [r222546] David Vossel * /, channels/chan_sip.c: Merged revisions 222543 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r222543 | dvossel | 2009-10-07 12:44:52 -0500 (Wed, 07 Oct 2009) | 14 lines Merged revisions 222542 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222542 | dvossel | 2009-10-07 12:41:21 -0500 (Wed, 07 Oct 2009) | 8 lines crash on transfer handle_invite_replaces() attempts to uplock a pvt's owner channel without first verifing that it exists. (issue #16027) ........ ................ 2009-10-07 17:32 +0000 [r222541] Tilghman Lesher * res/ael/pval.c: Small typo (thanks, jpeeler) 2009-10-06 23:57 +0000 [r222302-222464] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 222463 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r222463 | jpeeler | 2009-10-06 18:56:01 -0500 (Tue, 06 Oct 2009) | 14 lines Merged revisions 222462 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222462 | jpeeler | 2009-10-06 18:51:19 -0500 (Tue, 06 Oct 2009) | 8 lines Add missing unlock(s) in dahdi_read (two cases in trunk) (closes issue #15683) Reported by: alecdavis ........ ................ * channels/chan_dahdi.c: Fix potential crash when entire span request is received. The variable index used in this scenario for accessing the dahdi_pvts was wrong and was most likely copied from the several other places it is used correctly. (closes issue #15998) Reported by: tsearle Patches: dahdi_reset_crash.patch uploaded by tsearle (license 373) Modified: branches/1.4/channels/chan_dahdi.c * channels/chan_dahdi.c, /: Merged revisions 222351 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r222351 | jpeeler | 2009-10-06 15:35:19 -0500 (Tue, 06 Oct 2009) | 9 lines Fix 222298 (crash during destruction of second channel when variable set with setvar). I mistakenly reasoned that setvar would be used on all channels. Since it can be set per channel, give each dahdi channel a copy of the variable. (related to #15899) ........ * channels/chan_dahdi.c, /: Merged revisions 222298 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r222298 | jpeeler | 2009-10-06 14:24:59 -0500 (Tue, 06 Oct 2009) | 9 lines Fix crash during destruction of second channel when variable set with setvar. The setvar line in chan_dahdi.conf is shared among all the channels, so make sure to only free the resources only when the last channel is destroyed. (closes issue #15899) Reported by: tzafrir ........ 2009-10-06 19:19 +0000 [r222279] Tilghman Lesher * res/ael/pval.c, /: Recorded merge of revisions 222273 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r222273 | tilghman | 2009-10-06 14:17:11 -0500 (Tue, 06 Oct 2009) | 5 lines When we call a gosub routine, the variables should be scoped to avoid contaminating the caller. This affected the ~~EXTEN~~ hack, where a subroutine might have changed the value before it was used in the caller. Patch by myself, tested by ebroad on #asterisk ........ 2009-11-04 Leif Madsen * Release Asterisk 1.6.0.17 * AST-2009-008 and AST-2009-009 2009-10-06 Leif Madsen * Release Asterisk 1.6.0.16-rc2 2009-10-06 01:33 +0000 [r222111-222185] Kevin P. Fleming * main/astobj2.c, /, funcs/func_dialgroup.c, include/asterisk/astobj2.h, res/res_phoneprov.c, channels/chan_console.c, res/res_musiconhold.c, apps/app_queue.c, channels/chan_iax2.c: Merged revisions 222176 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r222176 | kpfleming | 2009-10-05 20:24:24 -0500 (Mon, 05 Oct 2009) | 27 lines Recorded merge of revisions 222152 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05 Oct 2009) | 20 lines 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/ ........ ................ * main/udptl.c, /, channels/chan_sip.c, configs/udptl.conf.sample, UPGRADE.txt, configs/sip.conf.sample: Recorded merge of revisions 222110 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r222110 | kpfleming | 2009-10-05 14:45:00 -0500 (Mon, 05 Oct 2009) | 25 lines Allow non-compliant T.38 endpoints to be supportable via configuration option. Many T.38 endpoints incorrectly send the maximum IFP frame size they can accept as the T38FaxMaxDatagram value in their SDP, when in fact this value is supposed to be the maximum UDPTL payload size (datagram size) they can accept. If the value they supply is small enough (a commonly supplied value is '72'), T.38 UDPTL transmissions will likely fail completely because the UDPTL packets will not have enough room for a primary IFP frame and the redundancy used for error correction. If this occurs, the Asterisk UDPTL stack will emit log messages warning that data loss may occur, and that the value may need to be overridden. This patch extends the 't38pt_udptl' configuration option in sip.conf to allow the administrator to override the value supplied by the remote endpoint and supply a value that allows T.38 FAX transmissions to be successful with that endpoint. In addition, in any SIP call where the override takes effect, a debug message will be printed to that effect. This patch also removes the T38FaxMaxDatagram configuration option from udptl.conf.sample, since it has not actually had any effect for a number of releases. In addition, this patch cleans up the T.38 documentation in sip.conf.sample (which incorrectly documented that T.38 support was passthrough only). (issue #15586) Reported by: globalnetinc ........ 2009-10-02 17:37 +0000 [r222038] David Vossel * /, channels/chan_iax2.c: Merged revisions 222030 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r222030 | dvossel | 2009-10-02 12:34:07 -0500 (Fri, 02 Oct 2009) | 9 lines Merged revisions 222026 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222026 | dvossel | 2009-10-02 12:32:13 -0500 (Fri, 02 Oct 2009) | 3 lines Removes unnecessary unlock, clarifies a memcpy. ........ ................ 2009-10-02 17:00 +0000 [r221972] Tilghman Lesher * main/astobj2.c, /: Merged revisions 221971 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r221971 | tilghman | 2009-10-02 11:59:57 -0500 (Fri, 02 Oct 2009) | 9 lines Merged revisions 221970 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221970 | tilghman | 2009-10-02 11:58:03 -0500 (Fri, 02 Oct 2009) | 2 lines Ensure the result of the hash function is positive. Negative array offsets suck. ........ ................ 2009-10-02 13:04 +0000 [r221963] Sean Bright * funcs/func_strings.c: Revert XML docs that ended up in the 1.6.0 and 1.6.1 branches during a merge. 2009-10-02 03:05 +0000 [r221921] Tilghman Lesher * /, main/logger.c: Merged revisions 221920 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r221920 | tilghman | 2009-10-01 22:04:34 -0500 (Thu, 01 Oct 2009) | 4 lines Initialize a variable that we check immediately upon startup. (closes issue #15973) Reported by: atis ........ 2009-10-02 01:20 +0000 [r221853] Richard Mudgett * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h, /: Merged revisions 221844 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r221844 | rmudgett | 2009-10-01 20:09:31 -0500 (Thu, 01 Oct 2009) | 33 lines Merged revisions 221769 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221769 | rmudgett | 2009-10-01 18:18:28 -0500 (Thu, 01 Oct 2009) | 26 lines Occasionally losing use of B channels in chan_misdn. I have not been able to reproduce the problem of losing channels. However, I have seen in the code a reentrancy problem that might give these symptoms. The reentrancy patch does several things: 1) Guards B channel and B channel structure allocation. 2) Makes the B channel structure find routines more precise in locating records. 3) Never leave a B channel allocated if we received cause 44. The last item may cause temporary outgoing call problems, but they should clear when the line becomes idle. (closes issue #15490) Reported by: slutec18 Patches: issue15490_channel_alloc_reentrancy.patch uploaded by rmudgett (license 664) Tested by: rmudgett, slutec18 (closes issue #15458) Reported by: FabienToune Patches: issue15458_channel_alloc_reentrancy.patch uploaded by rmudgett (license 664) Tested by: FabienToune, rmudgett, slutec18 ........ ................ 2009-10-02 00:03 +0000 [r221778] Tilghman Lesher * main/asterisk.c, main/rtp.c, /, main/say.c: Merged revisions 221777 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r221777 | tilghman | 2009-10-01 18:59:15 -0500 (Thu, 01 Oct 2009) | 9 lines Merged revisions 221776 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221776 | tilghman | 2009-10-01 18:53:12 -0500 (Thu, 01 Oct 2009) | 2 lines Fix a bunch of off-by-one errors ........ ................ 2009-10-01 21:04 +0000 [r221745] David Vossel * /, channels/chan_sip.c: Merged revisions 221697 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r221697 | dvossel | 2009-10-01 14:33:33 -0500 (Thu, 01 Oct 2009) | 9 lines outbound tls connections were not defaulting to port 5061 (closes issue #15854) Reported by: dvossel Patches: sip_port_config_trunk.diff uploaded by dvossel (license 671) Tested by: dvossel ........ 2009-10-01 20:34 +0000 [r221742] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 221705 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r221705 | tilghman | 2009-10-01 15:09:46 -0500 (Thu, 01 Oct 2009) | 2 lines Revision 220906 (a merge from 1.4) was not merged correctly, causing a problem with non-dynamic peers. ........ 2009-10-01 20:19 +0000 [r221712] David Vossel * channels/chan_sip.c: Fixes issue with non dynamic hosts not being set for peers 2009-10-01 17:09 +0000 [r221662] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 221554,221589 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r221554 | oej | 2009-10-01 02:00:04 -0500 (Thu, 01 Oct 2009) | 3 lines Simplify code for porturi, use TRUE/FALSE constructs when it's just TRUE or FALSE. ................ r221589 | mnicholson | 2009-10-01 10:26:20 -0500 (Thu, 01 Oct 2009) | 9 lines Merged revisions 221588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221588 | mnicholson | 2009-10-01 10:24:00 -0500 (Thu, 01 Oct 2009) | 2 lines Use unsigned ints for portinuri flags. ........ ................ 2009-10-01 16:18 +0000 [r221598] Kevin P. Fleming * main/udptl.c, /, configs/udptl.conf.sample, UPGRADE.txt: Merged revisions 221592 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r221592 | kpfleming | 2009-10-01 11:16:09 -0500 (Thu, 01 Oct 2009) | 12 lines Remove ability to control T.38 FAX error correction from udptl.conf. chan_sip has had the ability to control T.38 FAX error correction mode on a per-peer (or global) basis for a couple of releases now, which is where it should have been all along. This patch removes the ability to configure it in udptl.conf, but issues a warning if the user tries to do, telling them to look at sip.conf.sample for how to configure it now. For any SIP peers that are T.38 enabled in sip.conf, there is already a default for FEC error correction even if the user does not specify any mode, so this change will not turn off error correction by default, it will have the same default value that has been in the udptl.conf sample file. ........ 2009-09-30 23:08 +0000 [r221486] Matthew Nicholson * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 221432 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r221432 | mnicholson | 2009-09-30 15:40:20 -0500 (Wed, 30 Sep 2009) | 17 lines Merged revisions 221360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221360 | mnicholson | 2009-09-30 14:36:06 -0500 (Wed, 30 Sep 2009) | 10 lines Fix SRV lookup and Request-URI generation in chan_sip. This patch adds a new field "portinuri" to the sip dialog struct and the sip peer struct. That field is used during RURI generation to determine if the port should be included in the RURI. It is also used in some places to determine if an SRV lookup should occur. (closes issue #14418) Reported by: klaus3000 Tested by: klaus3000, mnicholson Review: https://reviewboard.asterisk.org/r/369/ ........ ................ 2009-09-30 20:02 +0000 [r221369] Matthias Nick * configs/cdr_custom.conf.sample, /, funcs/func_strings.c: Merged revisions 221368 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r221368 | mnick | 2009-09-30 14:42:36 -0500 (Wed, 30 Sep 2009) | 23 lines Merged revisions 221153,221157,221303 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221153 | mnick | 2009-09-30 10:37:39 -0500 (Wed, 30 Sep 2009) | 2 lines check bounds - prevents for buffer overflow ........ r221157 | mnick | 2009-09-30 10:41:46 -0500 (Wed, 30 Sep 2009) | 8 lines added a new dialplan function 'CSV_QUOTE' and changed the cdr_custom.sample.conf (closes issue #15471) Reported by: dkerr Patches: csv_quote_14.txt uploaded by mnick (license ) Tested by: mnick ........ r221303 | mnick | 2009-09-30 14:02:00 -0500 (Wed, 30 Sep 2009) | 2 lines changed the prototype definition of csv_quote ........ ................ 2009-09-30 18:50 +0000 [r221301] Terry Wilson * main/rtp.c, /, channels/chan_sip.c, include/asterisk/rtp.h, configs/sip.conf.sample: Merged revisions 221266 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r221266 | twilson | 2009-09-30 12:52:30 -0500 (Wed, 30 Sep 2009) | 32 lines Merged revisions 221086 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009) | 25 lines Change the SSRC by default when our media stream changes Be default, change SSRC when doing an audio stream changes Asterisk doesn't honor marker bit when reinvited to already-bridged RTP streams,resulting in far-end stack discarding packets with "old" timestamps that areactually part of a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is a reinvite, unless the 'constantssrc' is set to true in sip.conf. The original issue reported to Digium support detailed the following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based Application Server Call comes in fromITSP, Asterisk dials the app server which sends a re-invite back toAsterisk--not to negotiate to send media directly to the ITSP, but to indicatethat it's changing the stream it's sending to Asterisk. The app servergenerates a new SSRC, sequence numbers, timestamps, and sets the marker bit on the new stream. Asterisk passes through the teimstamp of the new stream, butdoes not reset the SSRC, sequence numbers, or set the marker bit. When the timestamp on the new stream is older than the timestamp on the originalstream, the ITSP (which doesn't know there has been any change) discards the newframes because it thinks they are too old. This patch addresses this by changing the SSRC on a stream update unless constantssrc=true is set in sip.conf. Review: https://reviewboard.asterisk.org/r/374/ ........ ................ 2009-09-30 16:57 +0000 [r221202] Tilghman Lesher * main/channel.c, /: Merged revisions 221201 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r221201 | tilghman | 2009-09-30 11:56:42 -0500 (Wed, 30 Sep 2009) | 14 lines Merged revisions 221200 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221200 | tilghman | 2009-09-30 11:55:21 -0500 (Wed, 30 Sep 2009) | 7 lines Avoid a potential NULL dereference. (closes issue #15865) Reported by: kobaz Patches: 20090915__issue15865.diff.txt uploaded by tilghman (license 14) Tested by: kobaz ........ ................ 2009-09-30 14:52 +0000 [r221087] Sean Bright * apps/app_voicemail.c, /: Merged revisions 221085 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r221085 | seanbright | 2009-09-30 10:47:58 -0400 (Wed, 30 Sep 2009) | 9 lines Clarify documentation for VoiceMailMain()'s a() option. We require box numbers, not names as the documentation implies. (issue #14740) Reported by: pj Patches: __20090729-app_voicemail-documentation.patch uploaded by lmadsen (license 10) Tested by: seanbright, lmadsen ........ 2009-09-30 04:34 +0000 [r220976-221045] Tilghman Lesher * /, funcs/func_lock.c: Recorded merge of revisions 221044 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r221044 | tilghman | 2009-09-29 23:32:36 -0500 (Tue, 29 Sep 2009) | 8 lines Allow locks to be inherited through a masquerade without causing starvation. (closes issue #14859) Reported by: atis Patches: 20090821__issue14859.diff.txt uploaded by tilghman (license 14) 20090925__issue14859__1.6.1.diff.txt uploaded by tilghman (license 14) Tested by: atis, tilghman ........ * /, channels/chan_sip.c: Merged revisions 220906 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r220906 | tilghman | 2009-09-29 14:57:37 -0500 (Tue, 29 Sep 2009) | 16 lines Merged revisions 220873 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r220873 | tilghman | 2009-09-29 12:59:26 -0500 (Tue, 29 Sep 2009) | 9 lines Reduce CPU usage related to building a peer merely for devicestates. This fixes a 100% CPU problem in the SIP driver, found by profiling the driver while the problem was occurring. (closes issue #14309) Reported by: pkempgen Patches: 20090924__issue14309.diff.txt uploaded by tilghman (license 14) Tested by: pkempgen, vrban ........ ................ 2009-09-29 20:25 +0000 [r220940] Matthew Nicholson * apps/app_chanspy.c: 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 2009-09-29 17:04 +0000 [r220834] Jeff Peeler * apps/app_voicemail.c, /: Merged revisions 220833 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r220833 | jpeeler | 2009-09-29 11:58:29 -0500 (Tue, 29 Sep 2009) | 12 lines Make deletion of temporary greetings work properly with IMAP_STORAGE When imapgreetings was set to yes, the message was being deleted but wasn't actually being expunged. When imapgreetings was set to no, the file based message was not being deleted at all. All good now! (closes issue #14949) Reported by: noahisaac Patches: vm_tempgreeting_removal.patch uploaded by noahisaac (license 748), modified by me ........ 2009-09-28 19:13 +0000 [r220723] Sean Bright * /, Makefile.rules: Merged revisions 220721 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r220721 | seanbright | 2009-09-28 15:11:20 -0400 (Mon, 28 Sep 2009) | 10 lines Merged revisions 220717 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r220717 | seanbright | 2009-09-28 15:09:25 -0400 (Mon, 28 Sep 2009) | 3 lines When selecting DONT_OPTIMIZE in menuselect, explicitly pass -O0 to the compiler so we override any default optimization levels for a particular install. ........ ................ 2009-09-26 15:11 +0000 [r220587] Tilghman Lesher * /, include/asterisk/aes.h: Merged revisions 220586 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r220586 | tilghman | 2009-09-26 10:10:28 -0500 (Sat, 26 Sep 2009) | 2 lines Allow AES to compile, when OpenSSL is not present. ........ 2009-09-24 20:42 +0000 [r220372] David Vossel * main/tcptls.c, /: Merged revisions 220365 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r220365 | dvossel | 2009-09-24 15:37:20 -0500 (Thu, 24 Sep 2009) | 8 lines fixes tcptls_session memory leak caused by ref count error (closes issue #15939) Reported by: dvossel Review: https://reviewboard.asterisk.org/r/375/ ........ 2009-09-24 19:42 +0000 [r220290] Tilghman Lesher * apps/app_playback.c, main/pbx.c, /, apps/app_disa.c: Merged revisions 220289 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r220289 | tilghman | 2009-09-24 14:41:02 -0500 (Thu, 24 Sep 2009) | 13 lines Merged revisions 220288 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r220288 | tilghman | 2009-09-24 14:39:41 -0500 (Thu, 24 Sep 2009) | 6 lines 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 ........ ................ 2009-09-24 18:22 +0000 [r220101-220219] Sean Bright * Makefile, /: Merged revisions 220217 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r220217 | seanbright | 2009-09-24 14:19:41 -0400 (Thu, 24 Sep 2009) | 9 lines Merged revisions 220213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r220213 | seanbright | 2009-09-24 14:18:18 -0400 (Thu, 24 Sep 2009) | 1 line Resolve parallel build warnings. Reported by Klaus Darilion on the asterisk-dev mailing list. ........ ................ * Makefile, build_tools/mkpkgconfig, /: Merged revisions 220100 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r220100 | seanbright | 2009-09-24 10:44:08 -0400 (Thu, 24 Sep 2009) | 9 lines Merged revisions 220099 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r220099 | seanbright | 2009-09-24 10:41:57 -0400 (Thu, 24 Sep 2009) | 2 lines Remove the remaining bashisms in the Makefile/mkpkgconfig ........ ................ 2009-09-24 08:37 +0000 [r220029] Michiel van Baak * build_tools/mkpkgconfig, /: Merged revisions 220028 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r220028 | mvanbaak | 2009-09-24 10:36:18 +0200 (Thu, 24 Sep 2009) | 14 lines Merged revisions 220027 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r220027 | mvanbaak | 2009-09-24 10:33:50 +0200 (Thu, 24 Sep 2009) | 7 lines mkpkgconfig does not need bash so make it use /bin/sh This fixes building on all systems that don't have bash at /bin/bash Reported by _ys on #asterisk-dev Tested by _ys on #asterisk-dev ........ ................ 2009-09-22 21:47 +0000 [r219819] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 219818 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219818 | tilghman | 2009-09-22 16:43:22 -0500 (Tue, 22 Sep 2009) | 17 lines Merged revisions 219816 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219816 | tilghman | 2009-09-22 16:37:03 -0500 (Tue, 22 Sep 2009) | 10 lines 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 ........ ................ 2009-09-21 17:03 +0000 [r219724] David Vossel * /, channels/chan_iax2.c: Merged revisions 219721 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219721 | dvossel | 2009-09-21 11:59:05 -0500 (Mon, 21 Sep 2009) | 9 lines Merged revisions 219720 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219720 | dvossel | 2009-09-21 11:55:53 -0500 (Mon, 21 Sep 2009) | 3 lines Reverting merge 219520. This change was not necessary. ........ ................ 2009-09-20 18:20 +0000 [r219663] Tilghman Lesher * /, main/file.c: Merged revisions 219654 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219654 | tilghman | 2009-09-20 12:55:49 -0500 (Sun, 20 Sep 2009) | 15 lines Merged revisions 219653 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219653 | tilghman | 2009-09-20 12:52:05 -0500 (Sun, 20 Sep 2009) | 8 lines Really stop the stream, when ast_closestream() is called. (closes issue #15129) Reported by: bmh Patches: 20090918__issue15129.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/372/ ........ ................ 2009-09-19 03:06 +0000 [r219588] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 219587 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219587 | russell | 2009-09-18 21:59:52 -0500 (Fri, 18 Sep 2009) | 13 lines Merged revisions 219586 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219586 | russell | 2009-09-18 21:51:13 -0500 (Fri, 18 Sep 2009) | 6 lines Make sure the iax_pvt exists before dereferencing it. This fixes the latest crash posted on issue 15609. (issue #15609) ........ ................ 2009-09-18 23:23 +0000 [r219454-219523] David Vossel * /, channels/chan_iax2.c: Merged revisions 219520 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219520 | dvossel | 2009-09-18 18:20:58 -0500 (Fri, 18 Sep 2009) | 15 lines Merged revisions 219519 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219519 | dvossel | 2009-09-18 18:19:50 -0500 (Fri, 18 Sep 2009) | 9 lines iax2 frame double free The iax frame's retrans sched id was written over right before iax2_frame_free was called. In iax2_frame_free that retrans id is used to delete the sched item. By writing over the retrans field before the sched item could be deleted, it was possible for a retransmit to occur on a freed frame. ........ ................ * /, channels/chan_sip.c: Merged revisions 219451 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219451 | dvossel | 2009-09-18 11:20:41 -0500 (Fri, 18 Sep 2009) | 20 lines Merged revisions 219450 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219450 | dvossel | 2009-09-18 11:19:15 -0500 (Fri, 18 Sep 2009) | 14 lines via-header branches not updated correctly on INVITE INVITE requests must always contain a new unique branch id. When a new branch id is created for an INVITE, the dialog's invite_branch variable must be updated so CANCEL requests use the correct branch id. (closes issue #15262) Reported by: maniax Patches: asterisk-1.6.1.0-sip-branch.patch uploaded by tweety (license 608) invite_new_branch_trunk.diff uploaded by dvossel (license 671) Tested by: maniax, dvossel ........ ................ 2009-09-18 13:57 +0000 [r219413] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 219412 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r219412 | tilghman | 2009-09-18 08:54:51 -0500 (Fri, 18 Sep 2009) | 6 lines Missing value setting line for maxsecs/maxmessage (closes issue #15696) Reported by: fhackenberger Patches: maxsecs.patch uploaded by fhackenberger (license 592) ........ 2009-09-17 22:36 +0000 [r219365] Joshua Colp * /, channels/chan_sip.c: Merged revisions 219324 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219324 | mmichelson | 2009-09-17 17:22:01 -0500 (Thu, 17 Sep 2009) | 12 lines Merged revisions 219320 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219320 | mmichelson | 2009-09-17 17:20:50 -0500 (Thu, 17 Sep 2009) | 6 lines Send a 100 Trying response when we detect a spiral. This was problematic during spiral tests at SIPit... along with some other things as well. ........ ................ 2009-09-17 22:01 +0000 [r219305] David Vossel * /, channels/chan_sip.c: Merged revisions 219304 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219304 | dvossel | 2009-09-17 16:59:21 -0500 (Thu, 17 Sep 2009) | 27 lines Merged revisions 219303 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219303 | dvossel | 2009-09-17 16:29:37 -0500 (Thu, 17 Sep 2009) | 21 lines INVITE w/Replaces deadlock fix This patch cleans up the locking logic in chan_sip.c's handle_invite_replaces() function as well as making use of ast_do_masquerade() rather than forcing the masquerade on an ast_read(). The code had several redundant unlocks that would result in 'freed more times than we've locked!' errors. I cleaned these up as well as moving all the unlock logic to the end of the function. This patch should also resolve the issue people were having with the replacecall channel never being unlocked with one legged calls. (closes issue #15151) Reported by: irroot Patches: invite_w_replaces_1.4.diff uploaded by dvossel (license 671) Tested by: irroot, dvossel Review: https://reviewboard.asterisk.org/r/371/ ........ ................ 2009-09-17 19:58 +0000 [r219265] Joshua Colp * /, channels/chan_sip.c: Merged revisions 219264 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r219264 | file | 2009-09-17 14:57:39 -0500 (Thu, 17 Sep 2009) | 2 lines Ensure no spaces exist before "refresher=" when doing the comparison. ........ 2009-09-17 Leif Madsen * Released Asterisk 1.6.0.16-rc1 2009-09-17 15:44 +0000 [r219198] Matthew Nicholson * main/channel.c, /, include/asterisk/cdr.h, include/asterisk/channel.h: Merged revisions 219139 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219139 | mnicholson | 2009-09-17 10:18:01 -0500 (Thu, 17 Sep 2009) | 17 lines Merged revisions 219136 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219136 | mnicholson | 2009-09-17 09:58:39 -0500 (Thu, 17 Sep 2009) | 10 lines Prevent a potential race condition and crash when hanging up a channel by removing the channel from the channel list before begining channel tear down. This fix may potentially cause problems with CDR backends that access the channel a CDR is associated with via the channel list. This fix makes the channel unavabile at the time when the CDR backend is invoked. This has been documented in include/asterisk/cdr.h. (closes issue #15316) Reported by: vmarrone Tested by: mnicholson Review: https://reviewboard.asterisk.org/r/362/ ........ ................ 2009-09-16 23:52 +0000 [r219064] Tilghman Lesher * main/config.c, configs/extensions.conf.sample, /: Merged revisions 219061 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219061 | tilghman | 2009-09-16 18:42:12 -0500 (Wed, 16 Sep 2009) | 15 lines Merged revisions 219023 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219023 | tilghman | 2009-09-16 18:21:53 -0500 (Wed, 16 Sep 2009) | 8 lines Properly deal with quotes in the arguments of '#exec' includes. (closes issue #15583) Reported by: pkempgen Patches: 20090726__issue15583.diff.txt uploaded by tilghman (license 14) 20090726__issue15583-1.4-4.diff.txt uploaded by pkempgen (license 169) Tested by: pkempgen ........ ................ 2009-09-16 19:26 +0000 [r218935] Mark Michelson * /, channels/chan_sip.c: Merged revisions 218933 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r218933 | mmichelson | 2009-09-16 14:25:36 -0500 (Wed, 16 Sep 2009) | 12 lines Reverse order of args to fread. This way, we don't always write a null byte into byte 1 of the buffer (closes issue #15905) Reported by: ebroad Patches: freadfix.patch uploaded by ebroad (license 878) Tested by: ebroad ........ 2009-09-16 18:44 +0000 [r218931] Joshua Colp * /, channels/chan_sip.c: Merged revisions 218918 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r218918 | file | 2009-09-16 13:31:47 -0500 (Wed, 16 Sep 2009) | 5 lines On TCP and TLS connections do not attempt to stop retransmission of the packet internally. This was preventing responses from being properly processed because the packet was not being found causing handle_response to return prematurely. ........ 2009-09-16 18:11 +0000 [r218869] David Brooks * main/pbx.c, /: Merged revisions 218868 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r218868 | dbrooks | 2009-09-16 13:06:42 -0500 (Wed, 16 Sep 2009) | 20 lines Merged revisions 218867 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218867 | dbrooks | 2009-09-16 13:00:45 -0500 (Wed, 16 Sep 2009) | 13 lines Fixes CID pattern matching behavior to mirror that of extension pattern matching. Pattern matching for extensions uses a type of scoring system, giving values for specificity to each character in the pattern. Unfortunately, this is done character by character, in order. This does lead to some less specific patterns being first in line for matching, but it will usually get the job done. This patch merely brings CID matching to the same level as extension matching. This patch does not attempt to tackle the problem shared by extension matching. (closes issue #14708) Reported by: klaus3000 ........ ................ 2009-09-16 13:36 +0000 [r218800] Russell Bryant * contrib/firmware/iax/iaxy.bin (removed), /, UPGRADE.txt: Merged revisions 218799 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r218799 | russell | 2009-09-16 08:34:41 -0500 (Wed, 16 Sep 2009) | 16 lines Merged revisions 218798 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218798 | russell | 2009-09-16 08:33:43 -0500 (Wed, 16 Sep 2009) | 9 lines Remove the IAXy firmware from Asterisk. The firmware can now be found on downloads.digium.com, where the rest of our binary downloads live. This was the last part of our Asterisk tarballs that was considered non-free by Debian. :-) (closes issue #15838) Reported by: paravoid ........ ................ 2009-09-15 22:39 +0000 [r218732] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 218731 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r218731 | tilghman | 2009-09-15 17:33:10 -0500 (Tue, 15 Sep 2009) | 13 lines Merged revisions 218730 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218730 | tilghman | 2009-09-15 17:27:41 -0500 (Tue, 15 Sep 2009) | 6 lines 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) ........ ................ 2009-09-15 19:31 +0000 [r218690] David Vossel * /, channels/chan_sip.c: Merged revisions 218687 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r218687 | dvossel | 2009-09-15 14:22:37 -0500 (Tue, 15 Sep 2009) | 2 lines upward bound checking for port string to int conversion ........ 2009-09-15 16:21 +0000 [r218601] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 218586 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r218586 | mnicholson | 2009-09-15 11:15:02 -0500 (Tue, 15 Sep 2009) | 15 lines Merged revisions 218578 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218578 | mnicholson | 2009-09-15 11:03:54 -0500 (Tue, 15 Sep 2009) | 8 lines Send request contact header field with response to registrer queries instead of the address of record. (closes issue #14438) Reported by: ravindrad Patches: regquerypatch uploaded by ravindrad (license 684) Tested by: ravindrad ........ ................ 2009-09-15 16:05 +0000 [r218580] Tilghman Lesher * /, apps/app_followme.c: Merged revisions 218579 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r218579 | tilghman | 2009-09-15 11:04:41 -0500 (Tue, 15 Sep 2009) | 16 lines Merged revisions 218577 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218577 | tilghman | 2009-09-15 11:01:17 -0500 (Tue, 15 Sep 2009) | 9 lines 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 ........ ................ 2009-09-15 15:42 +0000 [r218505-218573] Mark Michelson * /, channels/chan_sip.c: Merged revisions 218566 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r218566 | mmichelson | 2009-09-15 10:40:14 -0500 (Tue, 15 Sep 2009) | 4 lines Use a better method of ensuring null-termination of the buffer while reading the SDP when using TCP. ........ * /, channels/chan_sip.c: Merged revisions 218499,218504 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r218499 | mmichelson | 2009-09-15 09:59:50 -0500 (Tue, 15 Sep 2009) | 3 lines Fix off-by-one error when reading SDP sent over TCP. ........ r218504 | mmichelson | 2009-09-15 10:05:53 -0500 (Tue, 15 Sep 2009) | 3 lines Ensure that SDP read from TCP socket is null-terminated. ........ 2009-09-15 15:03 +0000 [r218501] Kevin P. Fleming * /, sounds/Makefile: Merged revisions 218500 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r218500 | kpfleming | 2009-09-15 11:02:21 -0400 (Tue, 15 Sep 2009) | 9 lines Merged revisions 218497 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218497 | kpfleming | 2009-09-15 10:55:58 -0400 (Tue, 15 Sep 2009) | 1 line Use proper hostname for downloading sound files. ........ ................ 2009-09-14 22:49 +0000 [r218431] Jeff Peeler * channels/chan_dahdi.c: Merged revisions 218430 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r218430 | jpeeler | 2009-09-14 17:38:25 -0500 (Mon, 14 Sep 2009) | 18 lines Merged revisions 218401 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218401 | jpeeler | 2009-09-14 16:47:11 -0500 (Mon, 14 Sep 2009) | 11 lines Fix handling of DAHDI_EVENT_REMOVED event to prevent crash in do_monitor. After talking to rmudgett about some of his recent iflist locking changes, it was determined that the only place that would destroy a channel without being explicitly to do so was in handle_init_event. The loop to walk the interface list has been modified to wait to destroy the channel until the dahdi_pvt of the channel to be destroyed is no longer needed. (closes issue #15378) Reported by: samy ........ ................ 2009-09-14 19:49 +0000 [r218362] Tilghman Lesher * apps/app_voicemail.c, /, configs/voicemail.conf.sample, sounds/Makefile: Merged revisions 218361 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r218361 | tilghman | 2009-09-14 14:29:48 -0500 (Mon, 14 Sep 2009) | 11 lines Recorded merge of revisions 218331 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218331 | tilghman | 2009-09-14 14:16:35 -0500 (Mon, 14 Sep 2009) | 4 lines 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 ........ ................ 2009-09-14 15:22 +0000 [r218244] Matthew Nicholson * /, apps/app_directed_pickup.c: Merged revisions 218224 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r218224 | mnicholson | 2009-09-14 09:57:23 -0500 (Mon, 14 Sep 2009) | 14 lines Merged revisions 218223 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218223 | mnicholson | 2009-09-14 09:53:57 -0500 (Mon, 14 Sep 2009) | 8 lines 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) ........ ................ 2009-09-13 19:10 +0000 [r218216] Tzafrir Cohen * channels/chan_phone.c, /: gcc 4.4: Remove a nop memset size 0 that annoys gcc This memset doesn't write beyond the end of the buffer. (tmpbuf has size of 4). Merged revisions 218184 via svnmerge from http://svn.digium.com/svn/asterisk/trunk 2009-09-12 13:10 +0000 [r218108] Michiel van Baak * main/rtp.c: Use the ip for the new 'rtp set debug ip '. Since 1.6.X still has the deprecated 'rtp debug ip ' this patch is different from the fix that went into trunk (closes issue #15711) Reported by: davidw Patches: 2009082800-rtpdebug.diff.txt uploaded by mvanbaak (license 7) Tested by: davidw 2009-09-11 05:58 +0000 [r217920-218051] Tilghman Lesher * /, apps/app_queue.c: Merged revisions 217990 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r217990 | tilghman | 2009-09-10 18:54:51 -0500 (Thu, 10 Sep 2009) | 10 lines Merged revisions 217989 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r217989 | tilghman | 2009-09-10 18:52:22 -0500 (Thu, 10 Sep 2009) | 3 lines Don't ring another channel, if there's not enough time for a queue member to answer. (Fixes AST-228) ........ ................ * contrib/scripts/iax-friends.sql, /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions 217916 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r217916 | tilghman | 2009-09-10 18:12:16 -0500 (Thu, 10 Sep 2009) | 2 lines Make calltoken support work with realtime users and peers. ........ 2009-09-10 22:31 +0000 [r217858-217913] David Vossel * channels/chan_sip.c: sip peer matching by address only with TCP/TLS This patch removes the contact header matching logic and adds logic to match all tcp/tls connections by ip only. Thanks to oej for finding the issue and suggesting solutions. Review: https://reviewboard.asterisk.org/r/355/ * /, channels/chan_iax2.c: Merged revisions 217807 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r217807 | dvossel | 2009-09-10 16:07:47 -0500 (Thu, 10 Sep 2009) | 28 lines Merged revisions 217806 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r217806 | dvossel | 2009-09-10 16:06:07 -0500 (Thu, 10 Sep 2009) | 22 lines IAX2 encryption regression The IAX2 Call Token security patch inadvertently broke the use of encryption due to the reorganization of code in the socket_process() function. When encryption is used, an incoming full frame must first be decrypted before the information elements can be parsed. The security release mistakenly moved IE parsing before decryption in order to process the new Call Token IE. To resolve this, decryption of full frames is once again done before looking into the frame. This involves searching for an existing callno, checking the pvt to see if encryption is turned on, and decrypting the packet before the internal fields of the full frame are accessed. (closes issue #15834) Reported by: karesmakro Patches: iax2_encryption_fix_1.4.diff uploaded by dvossel (license 671) Tested by: dvossel, karesmakro Review: https://reviewboard.asterisk.org/r/355/ ........ ................ 2009-09-10 19:53 +0000 [r217736] mnick : * /, res/res_musiconhold.c: Merged revisions 217730 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r217730 | mnick | 2009-09-10 14:39:41 -0500 (Thu, 10 Sep 2009) | 17 lines Sets the correct musicclass after an announcement (closes issue #15279) Reported by: mbeckwell Patches: patch.txt uploaded by mnick (license ) Tested by: mnick (closes issue #15832) Reported by: mbeckwell Patches: patch.txt uploaded by mnick (license 874) Tested by: mnick ........ 2009-09-10 12:16 +0000 [r217596] Olle Johansson * /, channels/chan_sip.c: Merged revisions 217593 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r217593 | oej | 2009-09-10 14:06:55 +0200 (Tor, 10 Sep 2009) | 8 lines Include ActionID in all events that are responsed to AMI Action SIPShowRegistry (closes issue #15868) Reported by: nic_bellamy Patches: manager_SIPshowregistry_actionid.patch uploaded by nic bellamy (license 299) ........ 2009-09-09 20:15 +0000 [r217484] Tzafrir Cohen * /, res/res_phoneprov.c: gcc 4.4 fix: union instead of cast gcc 4.4 has more strict rules for aliasing. It doesn't like a struct sockaddr_in pointer pointing to a struct sockaddr. So we make it a union. Merged revisions 217445 via svnmerge from http://svn.digium.com/svn/asterisk/trunk 2009-09-09 11:33 +0000 [r217405] Olle Johansson * /, channels/chan_sip.c: Merged revisions 217368 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r217368 | oej | 2009-09-09 12:39:43 +0200 (Ons, 09 Sep 2009) | 2 lines Not having any TLS session to write to is a serious XMIT_ERROR. ........ 2009-09-08 21:45 +0000 [r217281] Kevin P. Fleming * configure: Commit regenerated configure script that I missed earlier. 2009-09-08 20:31 +0000 [r217209] Tilghman Lesher * /, apps/app_meetme.c: Merged revisions 217199 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r217199 | tilghman | 2009-09-08 15:28:41 -0500 (Tue, 08 Sep 2009) | 14 lines Merged revisions 217156 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r217156 | tilghman | 2009-09-08 15:01:45 -0500 (Tue, 08 Sep 2009) | 7 lines 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 ........ ................ 2009-09-08 16:38 +0000 [r217075] Kevin P. Fleming * /, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 217074 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r217074 | kpfleming | 2009-09-08 11:37:28 -0500 (Tue, 08 Sep 2009) | 9 lines Ensure that the default autoconf CFLAGS are not used. A recent change to the configure script that allows the user to specify CFLAGS and/or LDFLAGS to the script had the unfortunate side effect of letting autoconf's default CFLAGS (-g -O2) feed in to the rest of the build system, thereby overriding the DONT_OPTIMIZE setting in menuselect. That problem is now corrected. ........ 2009-09-08 15:35 +0000 [r217034] Tilghman Lesher * /, res/res_limit.c: Merged revisions 217033 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r217033 | tilghman | 2009-09-08 10:30:18 -0500 (Tue, 08 Sep 2009) | 4 lines Remove what appears to be an unnecessary define. (closes issue #15851) Reported by: tzafrir ........ 2009-09-08 14:28 +0000 [r216996] David Vossel * /, channels/chan_sip.c: Merged revisions 216993 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216993 | dvossel | 2009-09-08 09:26:30 -0500 (Tue, 08 Sep 2009) | 14 lines caller id number empty parse_uri was not being given the correct scheme's, as a result, uri parsing did not parse the username correctly. One of the side effects of this is an empty caller id. (closes issue #15839) Reported by: ebroad Patches: blank_cidv2.patch uploaded by ebroad (license 878) parse_uri_fix.diff uploaded by dvossel (license 671) Tested by: ebroad, dvossel ........ 2009-09-07 16:38 +0000 [r216645-216843] Olle Johansson * /, channels/chan_sip.c: Merged revisions 216842 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216842 | oej | 2009-09-07 18:35:12 +0200 (MÃ¥n, 07 Sep 2009) | 2 lines Make sure we reset global_exclude_static at channel reload ........ * /, channels/chan_sip.c: Merged revisions 216695 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216695 | oej | 2009-09-07 15:06:19 +0200 (MÃ¥n, 07 Sep 2009) | 8 lines If there is no session timer in the INVITE, set it to default value (not unset minimum = -1) Patch by oej closes issue #15621 Reported by: fnordian Tested by: atis ........ * configs/sip.conf.sample: fix documentation so it agrees with code * channels/chan_sip.c, CHANGES: Add doc and turn off premature media filter by default * apps/app_playback.c, main/pbx.c, /, channels/chan_sip.c, apps/app_disa.c, configs/sip.conf.sample: Merged revisions 216438 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r216438 | oej | 2009-09-04 16:02:34 +0200 (Fre, 04 Sep 2009) | 35 lines Merged revisions 216430 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27 lines 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 :-) ........ ................ 2009-09-04 19:32 +0000 [r216595] Sean Bright * apps/app_voicemail.c, /: Merged revisions 216593 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216593 | seanbright | 2009-09-04 15:29:02 -0400 (Fri, 04 Sep 2009) | 1 line Use ast_free() instead of free(). ........ 2009-09-04 17:32 +0000 [r216548] Tilghman Lesher * main/pbx.c, /, UPGRADE-1.6.txt: Merged revisions 216547 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216547 | tilghman | 2009-09-04 12:31:44 -0500 (Fri, 04 Sep 2009) | 3 lines Enable turning off the application delimiter warning with the 'dontwarn' option. Suggested on the -dev list, and implemented in an alternate way by me. ........ 2009-09-04 15:07 +0000 [r216507] Michiel van Baak * /, main/utils.c: Merged revisions 216506 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r216506 | mvanbaak | 2009-09-04 17:05:05 +0200 (Fri, 04 Sep 2009) | 9 lines Merged revisions 216435 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r216435 | mvanbaak | 2009-09-04 15:56:10 +0200 (Fri, 04 Sep 2009) | 2 lines make asterisk compile under devmode with DEBUG_THREADS enabled on OpenBSD ........ ................ 2009-09-04 10:49 +0000 [r216265] Russell Bryant * doc/IAX2-security.txt (added), /: Merged revisions 216264 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r216264 | russell | 2009-09-04 05:48:44 -0500 (Fri, 04 Sep 2009) | 16 lines Merged revisions 216263 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r216263 | russell | 2009-09-04 05:48:00 -0500 (Fri, 04 Sep 2009) | 9 lines Merged revisions 216262 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r216262 | russell | 2009-09-04 05:47:37 -0500 (Fri, 04 Sep 2009) | 2 lines Add a plain text version of the IAX2 security document. ........ ................ ................ 2009-09-04 06:11 +0000 [r216223] Michiel van Baak * main/astobj2.c, /: Merged revisions 216222 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216222 | mvanbaak | 2009-09-04 08:08:33 +0200 (Fri, 04 Sep 2009) | 3 lines make sure 'start' is always initialized. Makes asterisk compile with --enable-dev-mode ........ 2009-09-03 19:42 +0000 [r216011-216097] Russell Bryant * UPGRADE.txt: tweak * /, UPGRADE.txt: Merged revisions 216092 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r216092 | russell | 2009-09-03 14:38:35 -0500 (Thu, 03 Sep 2009) | 16 lines Merged revisions 216085 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r216085 | russell | 2009-09-03 14:36:46 -0500 (Thu, 03 Sep 2009) | 9 lines Merged revisions 216080 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r216080 | russell | 2009-09-03 14:35:23 -0500 (Thu, 03 Sep 2009) | 2 lines Add a note about IAX2 to UPGRADE.txt. ........ ................ ................ * /, doc/IAX2-security.pdf (added): Merged revisions 216009 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r216009 | russell | 2009-09-03 13:45:54 -0500 (Thu, 03 Sep 2009) | 16 lines Merged revisions 216008 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r216008 | russell | 2009-09-03 13:44:58 -0500 (Thu, 03 Sep 2009) | 9 lines Merged revisions 216005 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r216005 | russell | 2009-09-03 13:42:24 -0500 (Thu, 03 Sep 2009) | 2 lines Add IAX2 security document related to AST-2009-006. ........ ................ ................ 2009-09-03 18:40 +0000 [r216003] David Vossel * channels/iax2-parser.c, main/astobj2.c, configs/iax.conf.sample, include/asterisk/acl.h, channels/iax2-parser.h, /, include/asterisk/astobj2.h, channels/iax2.h, main/acl.c, channels/chan_iax2.c: Merged revisions 215955 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r215955 | dvossel | 2009-09-03 11:31:54 -0500 (Thu, 03 Sep 2009) | 6 lines Merge code associated with AST-2009-006 (closes issue #12912) Reported by: rathaus Tested by: tilghman, russell, dvossel, dbrooks ........ 2009-09-03 Leif Madsen * Asterisk 1.6.0.15 released * AST-2009-006 2009-08-28 Leif Madsen * Asterisk 1.6.0.14 released 2009-08-11 Tilghman Lesher * Released 1.6.0.14-rc1 2009-08-10 19:51 +0000 [r211551-211587] Tilghman Lesher * doc/CODING-GUIDELINES, /: Merged revisions 211584 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r211584 | tilghman | 2009-08-10 14:49:41 -0500 (Mon, 10 Aug 2009) | 9 lines Merged revisions 211583 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r211583 | tilghman | 2009-08-10 14:48:48 -0500 (Mon, 10 Aug 2009) | 1 line Conversion specifiers, not format specifiers ........ ................ * res/res_config_curl.c, apps/app_waitforring.c, channels/chan_misdn.c, funcs/func_channel.c, apps/app_macro.c, pbx/pbx_config.c, apps/app_chanspy.c, apps/app_mixmonitor.c, res/res_odbc.c, main/asterisk.c, apps/app_voicemail.c, doc/CODING-GUIDELINES, utils/muted.c, apps/app_meetme.c, main/utils.c, cdr/cdr_pgsql.c, res/res_musiconhold.c, apps/app_followme.c, channels/misdn_config.c, utils/frame.c, main/channel.c, main/cdr.c, res/ael/pval.c, funcs/func_enum.c, channels/chan_phone.c, apps/app_osplookup.c, apps/app_setcallerid.c, main/manager.c, funcs/func_odbc.c, apps/app_minivm.c, res/res_agi.c, res/res_config_ldap.c, apps/app_rpt.c, channels/chan_mgcp.c, apps/app_adsiprog.c, funcs/func_dialplan.c, main/dnsmgr.c, channels/chan_sip.c, res/res_limit.c, apps/app_waitforsilence.c, agi/eagi-test.c, apps/app_waituntil.c, main/acl.c, apps/app_queue.c, channels/chan_oss.c, agi/eagi-sphinx-test.c, channels/chan_usbradio.c, res/snmp/agent.c, pbx/pbx_dundi.c, apps/app_sms.c, utils/extconf.c, apps/app_verbose.c, apps/app_stack.c, apps/app_dahdibarge.c, funcs/func_rand.c, apps/app_readfile.c, main/frame.c, /, apps/app_record.c, funcs/func_strings.c, cdr/cdr_adaptive_odbc.c, apps/app_alarmreceiver.c, channels/chan_iax2.c, main/indications.c, main/config.c, main/cli.c, pbx/pbx_loopback.c, channels/chan_dahdi.c, pbx/pbx_spool.c, res/res_smdi.c, channels/chan_skinny.c, main/features.c, main/http.c, main/pbx.c, apps/app_privacy.c, codecs/codec_speex.c, funcs/func_math.c, channels/chan_agent.c, apps/app_morsecode.c, apps/app_disa.c, funcs/func_cut.c, channels/iax2-provision.c, pbx/dundi-parser.c, apps/app_talkdetect.c: AST-2009-005 2009-08-10 14:10 +0000 [r211348] Joshua Colp * /, channels/chan_sip.c: Merged revisions 211347 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r211347 | file | 2009-08-10 11:07:44 -0300 (Mon, 10 Aug 2009) | 5 lines Fix retrieval of the port used for the video stream when adding SDP to a SIP message. (closes issue #15121) Reported by: jsmith ........ 2009-08-09 15:43 +0000 [r211233-211276] Tilghman Lesher * /, main/astfd.c: Merged revisions 211275 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r211275 | tilghman | 2009-08-09 10:42:02 -0500 (Sun, 09 Aug 2009) | 9 lines Merged revisions 211274 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r211274 | tilghman | 2009-08-09 10:41:01 -0500 (Sun, 09 Aug 2009) | 2 lines Small oops. Clear the flags which have been checked. ........ ................ * apps/app_stack.c, /: Merged revisions 211232 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r211232 | tilghman | 2009-08-09 02:11:22 -0500 (Sun, 09 Aug 2009) | 4 lines Check for NULL frame, before dereferencing pointer. (closes issue #15617) Reported by: rain ........ 2009-08-07 20:14 +0000 [r211114] Russell Bryant * apps/app_chanspy.c, /: Merged revisions 211113 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r211113 | russell | 2009-08-07 15:12:21 -0500 (Fri, 07 Aug 2009) | 11 lines Recorded merge of revisions 211112 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r211112 | russell | 2009-08-07 15:11:31 -0500 (Fri, 07 Aug 2009) | 4 lines Resolve a deadlock involving app_chanspy and masquerades. (ABE-1936) ........ ................ 2009-08-07 18:18 +0000 [r211044] Tilghman Lesher * /, apps/app_queue.c: Merged revisions 211040 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r211040 | tilghman | 2009-08-07 13:17:41 -0500 (Fri, 07 Aug 2009) | 21 lines Merged revisions 211038 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r211038 | tilghman | 2009-08-07 13:16:28 -0500 (Fri, 07 Aug 2009) | 14 lines 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) ........ ................ 2009-08-07 13:08 +0000 [r210993] Kevin P. Fleming * main/udptl.c, /: Merged revisions 210992 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r210992 | kpfleming | 2009-08-07 08:08:00 -0500 (Fri, 07 Aug 2009) | 13 lines Workaround broken T.38 endpoints that offer tiny MaxDatagram sizes. Some T.38 endpoints treat T38FaxMaxDatagram as the maximum IFP size that should be sent to them, rather than the maximum packet payload size. If such an endpoint also requests UDPRedundancy as the error correction mode, we'll end up calculating a tiny maximum IFP size, so small as to be unusable. This patch sets a lower bound on what we'll consider the remote's maximum IFP size to be, assuming that endpoints that do this really can accept larger packets than they've offered to accept. (closes issue #15649) Reported by: dazza76 ........ 2009-08-06 21:46 +0000 [r210909-210915] Tilghman Lesher * main/channel.c, /: Merged revisions 210914 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r210914 | tilghman | 2009-08-06 16:46:01 -0500 (Thu, 06 Aug 2009) | 14 lines Merged revisions 210913 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r210913 | tilghman | 2009-08-06 16:45:01 -0500 (Thu, 06 Aug 2009) | 7 lines Because channel information can be accessed outside of the channel thread, we must lock the channel prior to modifying it. (closes issue #15397) Reported by: caspy Patches: 20090714__issue15397.diff.txt uploaded by tilghman (license 14) Tested by: caspy ........ ................ * apps/app_stack.c, include/asterisk/app.h, /, main/app.c: Merged revisions 210908 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r210908 | tilghman | 2009-08-06 16:29:26 -0500 (Thu, 06 Aug 2009) | 9 lines Allow Gosub to recognize quote delimiters without consuming them. (closes issue #15557) Reported by: rain Patches: 20090723__issue15557.diff.txt uploaded by tilghman (license 14) Tested by: rain Review: https://reviewboard.asterisk.org/r/316/ ........ 2009-08-06 17:47 +0000 [r210818] Joshua Colp * /, channels/chan_sip.c: Merged revisions 210817 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r210817 | file | 2009-08-06 14:47:04 -0300 (Thu, 06 Aug 2009) | 11 lines Accept additional T.38 reinvites after an initial one has been handled. Discussion of this subject has yielded that it is not actually acceptable to change T.38 parameters after the initial reinvite but declining is harsh and can cause the fax to fail when it may be possible to allow it to continue. This patch changes things so that additional T.38 reinvites are accepted but parameter changes ignored. This gives the fax a fighting chance. (closes issue #15610) Reported by: huangtx2009 ........ 2009-08-05 20:07 +0000 [r210647] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 210640 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r210640 | rmudgett | 2009-08-05 14:40:03 -0500 (Wed, 05 Aug 2009) | 21 lines Merged revisions 210575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r210575 | rmudgett | 2009-08-05 14:18:56 -0500 (Wed, 05 Aug 2009) | 14 lines Dialplan starts execution before the channel setup is complete. * Issue 15655: For the case where dialing is complete for an incoming call, dahdi_new() was asked to start the PBX and then the code set more channel variables. If the dialplan hungup before these channel variables got set, asterisk would likely crash. * Fixed potential for overlap incoming call to erroneously set channel variables as global dialplan variables if the ast_channel structure failed to get allocated. * Added missing set of CALLINGSUBADDR in the dialing is complete case. (closes issue #15655) Reported by: alecdavis ........ ................ 2009-08-05 18:57 +0000 [r210568] Leif Madsen * doc/tex/imapstorage.tex, /: Merged revisions 210564 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r210564 | lmadsen | 2009-08-05 13:49:58 -0500 (Wed, 05 Aug 2009) | 19 lines Merged revisions 210563 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r210563 | lmadsen | 2009-08-05 13:46:21 -0500 (Wed, 05 Aug 2009) | 11 lines Update imapstorage.txt documentation. Updated the imapstorage.txt documentation to reflect that issues with c-client versions older than 2007 seem to cause crashing issues that are not seen with more recent versions. Documentation has been updated to reflect this. (closes issue #14496) Reported by: vbcrlfuser Patches: __20090727-imap-documentation-patch.txt uploaded by lmadsen (license 10) Tested by: lmadsen, mmichelson, dbrooks ........ ................ 2009-08-04 14:54 +0000 [r210239] Kevin P. Fleming * Makefile, /: Merged revisions 210238 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r210238 | kpfleming | 2009-08-04 09:53:00 -0500 (Tue, 04 Aug 2009) | 16 lines Merged revisions 210237 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r210237 | kpfleming | 2009-08-04 09:51:39 -0500 (Tue, 04 Aug 2009) | 10 lines Eliminate spurious compiler warnings from system headers on *BSD platforms. Ensure that system headers located in /usr/local/include are actually treated as system headers by the compiler, and not as local headers which are subject to warnings from the -Wundef compiler option and others. (closes issue #15606) Reported by: mvanbaak ........ ................ 2009-08-01 11:31 +0000 [r209840-209896] Russell Bryant * main/db1-ast/mpool/mpool.c, /: Merged revisions 209887 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r209887 | russell | 2009-08-01 06:29:25 -0500 (Sat, 01 Aug 2009) | 12 lines Merged revisions 209879 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r209879 | russell | 2009-08-01 06:27:25 -0500 (Sat, 01 Aug 2009) | 5 lines Resolve a valgrind warning about a read from uninitialized memory. (issue #15396) Reported by: aragon ........ ................ * apps/app_milliwatt.c, /: Merged revisions 209839 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r209839 | russell | 2009-08-01 06:02:07 -0500 (Sat, 01 Aug 2009) | 20 lines Merged revisions 209838 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r209838 | russell | 2009-08-01 05:59:05 -0500 (Sat, 01 Aug 2009) | 13 lines 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 ........ ................ 2009-08-01 01:13 +0000 [r209762] Kevin P. Fleming * channels/misdn/isdn_lib.c, utils/frame.c, /, main/Makefile, channels/misdn/ie.c, main/event.c: Merged revisions 209760-209761 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r209760 | kpfleming | 2009-07-31 20:03:07 -0500 (Fri, 31 Jul 2009) | 13 lines Merged revisions 209759 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r209759 | kpfleming | 2009-07-31 19:52:00 -0500 (Fri, 31 Jul 2009) | 7 lines Minor changes inspired by testing with latest GCC. The latest GCC (what will become 4.5.x) has a few new warnings, that in these cases found some either downright buggy code, or at least seriously poorly designed code that could be improved. ........ ................ r209761 | kpfleming | 2009-07-31 20:04:06 -0500 (Fri, 31 Jul 2009) | 1 line Revert accidental Makefile change. ................ 2009-07-31 21:56 +0000 [r209712] Russell Bryant * /, main/event.c: Merged revisions 209711 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209711 | russell | 2009-07-31 16:53:31 -0500 (Fri, 31 Jul 2009) | 2 lines Fix some places where ast_event_type was used instead of ast_event_ie_type. ........ 2009-07-30 16:37 +0000 [r209555-209587] David Brooks * channels/chan_dahdi.c: Merged revisions 209554 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209554 | dbrooks | 2009-07-30 11:07:05 -0500 (Thu, 30 Jul 2009) | 6 lines Fixes numerous spelling errors. Patch submitted by alecdavis. (closes issue #15595) Reported by: alecdavis ........ * include/asterisk/abstract_jb.h, contrib/init.d/rc.debian.asterisk, /, apps/app_sms.c, codecs/lpc10/pitsyn.c, channels/chan_console.c: Merged revisions 209554 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209554 | dbrooks | 2009-07-30 11:07:05 -0500 (Thu, 30 Jul 2009) | 6 lines Fixes numerous spelling errors. Patch submitted by alecdavis. (closes issue #15595) Reported by: alecdavis ........ 2009-07-28 12:01 +0000 [r209394] Kevin P. Fleming * apps/app_fax.c: Correct error in backport of latest app_fax fixes. 2009-07-28 00:19 +0000 [r209325] Tilghman Lesher * /, sounds/sounds.xml: Merged revisions 209317 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r209317 | tilghman | 2009-07-27 19:14:12 -0500 (Mon, 27 Jul 2009) | 9 lines Merged revisions 209315 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r209315 | tilghman | 2009-07-27 19:12:03 -0500 (Mon, 27 Jul 2009) | 2 lines Publish French extra sounds ........ ................ 2009-07-27 21:44 +0000 [r209259-209280] Kevin P. Fleming * /, apps/app_fax.c: Merged revisions 209279 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209279 | kpfleming | 2009-07-27 16:43:36 -0500 (Mon, 27 Jul 2009) | 7 lines Cleanup T.38 negotiation changes. Convert LOG_NOTICE messages about T.38 negotiation in debug level 1 messages, clean up some looping logic, and correct an improper use of ast_free() for freeing an ast_frame. ........ * /, apps/app_fax.c: Merged revisions 209256 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209256 | kpfleming | 2009-07-27 16:21:43 -0500 (Mon, 27 Jul 2009) | 10 lines Make T.38 switchover in ReceiveFAX synchronous. In receive mode, if the channel that ReceiveFAX is running on supports T.38, we should *always* attempt to switch T.38, rather than listening for an incoming CNG tone and only triggering on that. The channel may be using a low-bitrate codec that distorts the CNG tone, the sending FAX endpoint may not send CNG at all, or there could be a variety of other reasons that we don't detect it, but in all those cases if T.38 is available we certainly want to use it. ........ 2009-07-27 20:23 +0000 [r209221] David Brooks * channels/chan_dahdi.c, channels/chan_vpb.cc, res/res_smdi.c, /, include/asterisk/module.h, main/features.c, res/res_agi.c, res/res_jabber.c: Merged revisions 209098 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209098 | dbrooks | 2009-07-27 11:33:50 -0500 (Mon, 27 Jul 2009) | 6 lines Fixing typos. Replaces "recieved" with "received" and "initilize" with "initialize" (closes issue #15571) Reported by: alecdavis ........ 2009-07-27 20:16 +0000 [r209133-209198] Mark Michelson * /, res/res_musiconhold.c: Merged revisions 209197 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209197 | mmichelson | 2009-07-27 15:11:42 -0500 (Mon, 27 Jul 2009) | 9 lines Honor channel's music class when using realtime music on hold. (closes issue #15051) Reported by: alexh Patches: 15051.patch uploaded by mmichelson (license 60) Tested by: alexh ........ * main/udptl.c, /, configs/udptl.conf.sample: Merged revisions 209132 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r209132 | mmichelson | 2009-07-27 12:50:04 -0500 (Mon, 27 Jul 2009) | 24 lines Merged revisions 209131 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r209131 | mmichelson | 2009-07-27 12:44:06 -0500 (Mon, 27 Jul 2009) | 18 lines Allow for UDPTL to use only even-numbered ports if desired. There are some VoIP providers out there that will not accept SDP offers with odd numbered UDPTL ports. While it is my personal opinion that these VoIP providers are misinterpreting RFC 2327, it really is not a big deal to play along with their silly little games. Of course, since restricting UDPTL ports to only even numbers reduces the range of available ports by half, so the option to use only even port numbers is off by default. A user can enable the behavior by setting use_even_ports=yes in udptl.conf. (closes issue #15182) Reported by: CGMChris Patches: 15182.patch uploaded by mmichelson (license 60) Tested by: CGMChris ........ ................ 2009-07-27 16:06 +0000 [r209061] David Brooks * res/res_smdi.c, pbx/pbx_dundi.c: Just replacing typos "recieved" with "received". From issue #15360, forgot to apply to trunk and other branches. 2009-07-27 15:39 +0000 [r209057] Kevin P. Fleming * Makefile, /: Merged revisions 209056 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209056 | kpfleming | 2009-07-27 10:38:59 -0500 (Mon, 27 Jul 2009) | 10 lines Restore explicit export of ASTCFLAGS/ASTLDFLAGS and underscore-variants to sub-makes. During the recent Makefile improvements I made, it seemed the 'make' was automatically carrying down the ASTCFLAGS/ASTLDFLAGS settings to sub-makes, so I removed the explict export of them. However, there are some circumstances where make does this, and some where it does not, so I've brought them back to ensure they are always exported. I also removed an extraneous double setting of _ASTLDFLAGS on *BSD platforms. ........ 2009-07-27 01:21 +0000 [r208925] Jeff Peeler * /, main/translate.c, channels/chan_iax2.c: Merged revisions 208924 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208924 | jpeeler | 2009-07-26 20:20:37 -0500 (Sun, 26 Jul 2009) | 9 lines Merged revisions 208923 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208923 | jpeeler | 2009-07-26 20:18:31 -0500 (Sun, 26 Jul 2009) | 2 lines Fix logic errors from 208746 ........ ................ 2009-07-25 06:24 +0000 [r208752] Jeff Peeler * /, channels/chan_skinny.c, main/translate.c, channels/chan_iax2.c: Merged revisions 208749 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208749 | jpeeler | 2009-07-25 01:23:18 -0500 (Sat, 25 Jul 2009) | 13 lines Merged revisions 208746 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208746 | jpeeler | 2009-07-25 01:19:50 -0500 (Sat, 25 Jul 2009) | 7 lines Fix compiling under dev-mode with gcc 4.4.0. Mostly trivial changes, but I did not know of any other way to fix the "dereferencing type-punned pointer will break strict-aliasing rules" error without creating a tmp variable in chan_skinny. ........ ................ 2009-07-24 18:49 +0000 [r208594] Russell Bryant * apps/app_dial.c, /: Merged revisions 208593 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208593 | russell | 2009-07-24 13:42:32 -0500 (Fri, 24 Jul 2009) | 14 lines Merged revisions 208592 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208592 | russell | 2009-07-24 13:38:24 -0500 (Fri, 24 Jul 2009) | 7 lines 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) ........ ................ 2009-07-24 18:31 +0000 [r208589] Mark Michelson * /, channels/chan_sip.c: Merged revisions 208588 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208588 | mmichelson | 2009-07-24 13:31:04 -0500 (Fri, 24 Jul 2009) | 16 lines Merged revisions 208587 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208587 | mmichelson | 2009-07-24 13:26:50 -0500 (Fri, 24 Jul 2009) | 10 lines Only send a BYE when hanging up a channel that is up. For cases where Asterisk sends an INVITE and receives a non 2XX final response, Asterisk would follow the INVITE transaction by immediately sending a BYE, which was unnecessary. (closes issue #14575) Reported by: chris-mac ........ ................ 2009-07-24 15:04 +0000 [r208468-208549] Kevin P. Fleming * main/udptl.c, /, channels/chan_sip.c, include/asterisk/udptl.h: Merged revisions 208548 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r208548 | kpfleming | 2009-07-24 10:02:53 -0500 (Fri, 24 Jul 2009) | 8 lines Resolve a T.38 negotiation issue left over from the udptl-updates merge. The udptl-updates branch that was merged yesterday failed to properly send back T.38 SDP responses with the correct error correction mode, if the incoming SDP from the other end caused us to change error correction modes. This patch corrects that situation. ........ * UPGRADE.txt: Use correct formatting for T.38 change note in UPGRADE.txt * main/rtp.c, main/channel.c, main/udptl.c, main/frame.c, /, channels/chan_sip.c, apps/app_fax.c, UPGRADE.txt, include/asterisk/udptl.h, include/asterisk/frame.h: Merged revisions 208464 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r208464 | kpfleming | 2009-07-23 16:57:24 -0500 (Thu, 23 Jul 2009) | 46 lines Rework of T.38 negotiation and UDPTL API to address interoperability problems Over the past couple of months, a number of issues with Asterisk negotiating (and successfully completing) T.38 sessions with various endpoints have been found. This patch attempts to address many of them, primarily focused around ensuring that the endpoints' MaxDatagram size is honored, and in addition by ensuring that T.38 session parameter negotiation is performed correctly according to the ITU T.38 Recommendation. The major changes here are: 1) T.38 applications in Asterisk (app_fax) only generate/receive IFP packets, they do not ever work with UDPTL packets. As a result of this, they cannot be allowed to generate packets that would overflow the other endpoints' MaxDatagram size after the UDPTL stack adds any error correction information. With this patch, the application is told the maximum *IFP* size it can generate, based on a calculation using the far end MaxDatagram size and the active error correction mode on the T.38 session. The same is true for sending *our* MaxDatagram size to the remote endpoint; it is computed from the value that the application says it can accept (for a single IFP packet) combined with the active error correction mode. 2) All treatment of T.38 session parameters as 'capabilities' in chan_sip has been removed; these parameters are not at all like audio/video stream capabilities. There are strict rules to follow for computing an answer to a T.38 offer, and chan_sip now follows those rules, using the desired parameters from the application (or channel) that wants to accept the T.38 negotiation. 3) chan_sip now stores and forwards ast_control_t38_parameters structures for tracking 'our' and 'their' T.38 session parameters; this greatly simplifies negotiation, especially for pass-through calls. 4) Since T.38 negotiation without specifying parameters or receiving the final negotiated parameters is not very worthwhile, the AST_CONTROL_T38 control frame has been removed. A note has been added to UPGRADE.txt about this removal, since any out-of-tree applications that use it will no longer function properly until they are upgraded to use AST_CONTROL_T38_PARAMETERS. Review: https://reviewboard.asterisk.org/r/310/ ........ 2009-07-23 19:35 +0000 [r208389] Mark Michelson * /, channels/chan_sip.c: Merged revisions 208388 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208388 | mmichelson | 2009-07-23 14:34:49 -0500 (Thu, 23 Jul 2009) | 24 lines Merged revisions 208386 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208386 | mmichelson | 2009-07-23 14:24:21 -0500 (Thu, 23 Jul 2009) | 17 lines Fix a problem where a 491 response could be sent out of dialog. This generalizes the fix for issue 13849. The initial fix corrected the problem that Asterisk would reply with a 491 if a reinvite were received from an endpoint and we had not yet received an ACK from that endpoint for the initial INVITE it had sent us. This expansion also allows Asterisk to appropriately handle an INVITE with authorization credentials if Asterisk had not received an ACK from the previous transaction in which Asterisk had responded to an unauthorized INVITE with a 407. (closes issue #14239) Reported by: klaus3000 Patches: 14239.patch uploaded by mmichelson (license 60) Tested by: klaus3000 ........ ................ 2009-07-23 19:23 +0000 [r208384] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 208383 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208383 | jpeeler | 2009-07-23 14:21:50 -0500 (Thu, 23 Jul 2009) | 12 lines Merged revisions 208380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208380 | jpeeler | 2009-07-23 14:19:53 -0500 (Thu, 23 Jul 2009) | 6 lines Only set the priindication setting when not performing a reload (closes issue #14696) Reported by: fdecher ........ ................ 2009-07-23 16:30 +0000 [r208264-208316] Mark Michelson * /, channels/chan_sip.c: Merged revisions 208314 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208314 | mmichelson | 2009-07-23 11:29:37 -0500 (Thu, 23 Jul 2009) | 9 lines Merged revisions 208312 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208312 | mmichelson | 2009-07-23 11:29:18 -0500 (Thu, 23 Jul 2009) | 3 lines Remove inaccurate XXX comment. ........ ................ * /, channels/chan_sip.c: Merged revisions 208263 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208263 | mmichelson | 2009-07-23 10:46:34 -0500 (Thu, 23 Jul 2009) | 15 lines Merged revisions 208262 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208262 | mmichelson | 2009-07-23 10:43:07 -0500 (Thu, 23 Jul 2009) | 8 lines Properly handle 183 responses which do not contain an SDP. (closes issue #15442) Reported by: ffloimair Patches: 15442.patch uploaded by mmichelson (license 60) Tested by: tkarl, ffloimair ........ ................ 2009-07-21 22:47 +0000 [r207947] Tilghman Lesher * /, funcs/func_strings.c: Merged revisions 207946 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207946 | tilghman | 2009-07-21 17:45:32 -0500 (Tue, 21 Jul 2009) | 15 lines Merged revisions 207945 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207945 | tilghman | 2009-07-21 17:38:54 -0500 (Tue, 21 Jul 2009) | 8 lines Force an error if a blank is passed to QUOTE (because the documentation states the argument is not optional). This change makes URIENCODE and QUOTE behave similarly, since the documentation states that the argument is not optional, for both. (closes issue #15439) Reported by: pkempgen Patches: 20090706__issue15439.diff.txt uploaded by tilghman (license 14) ........ ................ 2009-07-21 20:27 +0000 [r207783-207860] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 207854 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207854 | jpeeler | 2009-07-21 15:26:02 -0500 (Tue, 21 Jul 2009) | 16 lines Merged revisions 207827 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207827 | jpeeler | 2009-07-21 15:16:55 -0500 (Tue, 21 Jul 2009) | 9 lines Wait for wink before dialing when using E&M wink signaling There was already code for other signaling types in dahdi_handle_event to handle dialing if a dial operation dial string was present. Simply add SIG_EMWINK to the list. (closes issue #14434) Reported by: araasch ........ ................ * channels/chan_dahdi.c: Revert r207636, this approach could potentially block for an unacceptable amount of time. 2009-07-21 14:30 +0000 [r207725] Mark Michelson * main/manager.c, /: Merged revisions 207723 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207723 | mmichelson | 2009-07-21 09:29:40 -0500 (Tue, 21 Jul 2009) | 11 lines Merged revisions 207714 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207714 | mmichelson | 2009-07-21 09:26:00 -0500 (Tue, 21 Jul 2009) | 5 lines Document default timeout for AMI originations. AST-224 ........ ................ 2009-07-21 13:39 +0000 [r207683] Kevin P. Fleming * funcs/Makefile, codecs/lpc10/Makefile, main/db1-ast/Makefile, Makefile, agi/Makefile, codecs/Makefile, utils/Makefile, /, main/Makefile, codecs/gsm/Makefile, Makefile.moddir_rules, Makefile.rules, pbx/Makefile, res/Makefile, channels/Makefile: Merged revisions 207680 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207680 | kpfleming | 2009-07-21 08:28:04 -0500 (Tue, 21 Jul 2009) | 18 lines Merged revisions 207647 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207647 | kpfleming | 2009-07-21 08:04:44 -0500 (Tue, 21 Jul 2009) | 12 lines Ensure that user-provided CFLAGS and LDFLAGS are honored. This commit changes the build system so that user-provided flags (in ASTCFLAGS and ASTLDFLAGS) are supplied to the compiler/linker *after* all flags provided by the build system itself, so that the user can effectively override the build system's flags if desired. In addition, ASTCFLAGS and ASTLDFLAGS can now be provided *either* in the environment before running 'make', or as variable assignments on the 'make' command line. As a result, the use of COPTS and LDOPTS is no longer necessary, so they are no longer documented, but are still supported so as not to break existing build systems that supply them when building Asterisk. ........ ................ 2009-07-21 04:38 +0000 [r207636] Jeff Peeler * channels/chan_dahdi.c: Wait for wink before dialing when using E&M wink signaling This patch adds a new dahdi_wait function to specifically wait for the wink event. If the wink is not eventually received the channel is hung up. (closes issue #14434) Reported by: araasch Patches: emwinkmod uploaded by araasch (license 693) 2009-07-20 19:55 +0000 [r207425] Mark Michelson * /, channels/chan_sip.c: Merged revisions 207424 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207424 | mmichelson | 2009-07-20 14:48:12 -0500 (Mon, 20 Jul 2009) | 39 lines Merged revisions 207423 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207423 | mmichelson | 2009-07-20 14:39:59 -0500 (Mon, 20 Jul 2009) | 33 lines Answer video SDP offers properly when videosupport is not enabled. Copied from Review board: In issue 12434, the reporter describes a situation in which audio and video is offered on the call, but because videosupport is disabled in sip.conf, Asterisk gives no response at all to the video offer. According to RFC 3264, all media offers should have a corresponding answer. For offers we do not intend to actually reply to with meaningful values, we should still reply with the port for the media stream set to 0. In this patch, we take note of what types of media have been offered and save the information on the sip_pvt. The SDP in the response will take into account whether media was offered. If we are not otherwise going to answer a media offer, we will insert an appropriate m= line with the port set to 0. It is important to note that this patch is pretty much a bandage being applied to a broken bone. The patch *only* helps for situations where video is offered but videosupport is disabled and when udptl_pt is disabled but T.38 is offered. Asterisk is not guaranteed to respond to every media offer. Notable cases are when multiple streams of the same type are offered. The 2 media stream limit is still present with this patch, too. In trunk and the 1.6.X branches, things will be a bit different since Asterisk also supports text in SDPs as well. (closes issue #12434) Reported by: mnnojd Review: https://reviewboard.asterisk.org/r/311 Review: https://reviewboard.asterisk.org/r/313 ........ ................ 2009-07-20 16:37 +0000 [r207362] Russell Bryant * main/channel.c, /: Merged revisions 207361 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207361 | russell | 2009-07-20 11:36:15 -0500 (Mon, 20 Jul 2009) | 16 lines Merged revisions 207360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207360 | russell | 2009-07-20 11:26:24 -0500 (Mon, 20 Jul 2009) | 9 lines Only do the chan->fdno check in ast_read() in a developer build. I changed this check to only happen in a dev-mode build. I also added a comment explaining what is going on. I also made it so that detection of this situation does not affect ast_read() operation. (closes issue #14723) Reported by: seadweller ........ ................ 2009-07-18 01:35 +0000 [r207286] Richard Mudgett * channels/misdn/isdn_lib.c, channels/misdn_config.c, channels/misdn/isdn_lib_intern.h, channels/misdn/isdn_lib.h, doc/tex/misdn.tex, channels/chan_misdn.c, main/callerid.c, configs/misdn.conf.sample: Merged revisions 145293,158010 from https://origsvn.digium.com/svn/asterisk/branches/1.4 to make merging easier. These changes are already on trunk. ................ r145293 | rmudgett | 2008-09-30 18:55:24 -0500 (Tue, 30 Sep 2008) | 54 lines channels/chan_misdn.c channels/misdn/isdn_lib.c * Miscellaneous other fixes from trunk to make merging easier later. ........ r145200 | rmudgett | 2008-09-30 16:00:54 -0500 (Tue, 30 Sep 2008) | 7 lines * Miscellaneous formatting changes to make v1.4 and trunk more merge compatible in the mISDN area. channels/chan_misdn.c * Eliminated redundant code in cb_events() EVENT_SETUP ........ r144257 | crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008) | 9 lines improved helptext of misdn_set_opt. ........ r142181 | rmudgett | 2008-09-09 12:30:52 -0500 (Tue, 09 Sep 2008) | 1 line Cleaned up comment ........ r138738 | rmudgett | 2008-08-18 16:07:28 -0500 (Mon, 18 Aug 2008) | 30 lines channels/chan_misdn.c * Made bearer2str() use allowed_bearers_array[] * Made use the causes.h defines instead of hardcoded numbers. * Made use Asterisk presentation indicator values if either of the mISDN presentation or screen options are negative. * Updated the misdn_set_opt application option descriptions. * Renamed the awkward Caller ID presentation misdn_set_opt application option value not_screened to restricted. Deprecated the not_screened option value. channels/misdn/isdn_lib.c * Made use the causes.h defines instead of hardcoded numbers. * Fixed some spelling errors and typos. * Added all defined facility code strings to fac2str(). channels/misdn/isdn_lib.h * Added doxygen comments to struct misdn_bchannel. channels/misdn/isdn_lib_intern.h * Added doxygen comments to struct misdn_stack. channels/misdn_config.c configs/misdn.conf.sample * Updated the mISDN presentation and screen parameter descriptions. doc/misdn.txt (doc/tex/misdn.tex) * Updated the misdn_set_opt application option descriptions. * Fixed some spelling errors and typos. ................ r158010 | rmudgett | 2008-11-19 19:46:09 -0600 (Wed, 19 Nov 2008) | 9 lines Merged revision 157977 from https://origsvn.digium.com/svn/asterisk/team/group/issue8824 ........ Fixes JIRA ABE-1726 The dial extension could be empty if you are using MISDN_KEYPAD to control ISDN provider features. ................ 2009-07-17 19:38 +0000 [r207097-207157] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 207156 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207156 | jpeeler | 2009-07-17 14:37:38 -0500 (Fri, 17 Jul 2009) | 14 lines Merged revisions 207155 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207155 | jpeeler | 2009-07-17 14:36:19 -0500 (Fri, 17 Jul 2009) | 7 lines Fix format specifier to print out an unsigned long long. Yep, it's even ifdefed out code. But it made it to the RR list... (closes issue #14726) Reported by: lmadsen ........ ................ * configs/chan_dahdi.conf.sample, /: Merged revisions 207095 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r207095 | jpeeler | 2009-07-17 14:16:35 -0500 (Fri, 17 Jul 2009) | 2 lines Update some missing allowed options for overlapdial ........ 2009-07-17 17:53 +0000 [r206871-207032] David Vossel * /, channels/chan_sip.c: Merged revisions 207029 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r207029 | dvossel | 2009-07-17 12:51:44 -0500 (Fri, 17 Jul 2009) | 6 lines sip option flags handled incorrectly (closes issue #15376) Reported by: Takehiko Ooshima Tested by: dvossel, Takehiko_Ooshima ........ * /, channels/chan_sip.c: Merged revisions 206939 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206939 | dvossel | 2009-07-17 11:13:22 -0500 (Fri, 17 Jul 2009) | 20 lines Merged revisions 206938 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206938 | dvossel | 2009-07-17 11:05:06 -0500 (Fri, 17 Jul 2009) | 14 lines SIP incorrect From: header information when callpres is prohib Some ITSP make use of the "Anonymous" display name to detect a requirement to withhold caller id across the PSTN. This does not work if the display name is "Unknown". (closes issue #14465) Reported by: Nick_Lewis Patches: chan_sip.c-callerpres.patch uploaded by Nick (license 657) chan_sip.c-callerpres_trunk.patch uploaded by dvossel (license 671) Tested by: Nick_Lewis, dvossel ........ ................ * configs/iax.conf.sample, /: Merged revisions 206873 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206873 | dvossel | 2009-07-16 16:33:51 -0500 (Thu, 16 Jul 2009) | 12 lines Merged revisions 206872 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206872 | dvossel | 2009-07-16 16:33:19 -0500 (Thu, 16 Jul 2009) | 6 lines error in iax.conf related IP-based access control (closes issue #15518) Reported by: pkempgen ........ ................ * /, main/callerid.c: Merged revisions 206868 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206868 | dvossel | 2009-07-16 16:25:22 -0500 (Thu, 16 Jul 2009) | 14 lines Merged revisions 206867 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206867 | dvossel | 2009-07-16 16:24:16 -0500 (Thu, 16 Jul 2009) | 8 lines avoid segfault caused by user error If the CALLERPRES() dialplan function is set to nothing, a segfault occurs. This is user error to begin with, but I'd rather see a cli warning message than have Asterisk crash on me. ........ ................ 2009-07-16 16:52 +0000 [r206809] Tilghman Lesher * funcs/func_realtime.c, /: Merged revisions 206808 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206808 | tilghman | 2009-07-16 11:51:05 -0500 (Thu, 16 Jul 2009) | 13 lines Merged revisions 206807 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206807 | tilghman | 2009-07-16 11:27:35 -0500 (Thu, 16 Jul 2009) | 6 lines Fix a memory leak. (closes issue #15517) Reported by: adomjan Patches: func_realtime.c-ast_variable_destroy.diff uploaded by adomjan (license 487) ........ ................ 2009-07-15 22:06 +0000 [r206775] David Vossel * /, channels/chan_sip.c: Merged revisions 206768 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r206768 | dvossel | 2009-07-15 17:04:13 -0500 (Wed, 15 Jul 2009) | 8 lines Session timer were not activated if Supported header field in INVITE had both "timer" and other options. (closes issue #15403) Reported by: makoto Patches: sip-session-timer.patch uploaded by makoto (license ........ 2009-07-15 21:34 +0000 [r206762] Richard Mudgett * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h, /: Merged revisions 206707 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206707 | rmudgett | 2009-07-15 16:14:41 -0500 (Wed, 15 Jul 2009) | 33 lines Merged revisions 206706 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r206706 | rmudgett | 2009-07-15 15:44:55 -0500 (Wed, 15 Jul 2009) | 26 lines Merged revision 206700 from https://origsvn.digium.com/svn/asterisk/be/branches/C.2-... .......... Fixed chan_misdn crash because mISDNuser library is not thread safe. With Asterisk the mISDNuser library is driven by two threads concurrently: 1. channels/misdn/isdn_lib.c::manager_event_handler() 2. channels/misdn/isdn_lib.c::misdn_lib_isdn_event_catcher() Calls into the library are done concurrently and recursively from isdn_lib.c. Both threads can fiddle with the master/child layer3_proc_t lists. One thread may traverse the list when the other interrupts it and then removes the list element which the first thread was currently handling. This is exactly what caused the crash. About 60 calls were needed to a Gigaset CX475 before it occurred once. This patch adds locking when calling into the mISDNuser library. This also fixes some cb_log calls with wrong port parameter. JIRA ABE-1913 Patches: misdn-locking.patch (Modified with mostly cosmetic changes) .......... ................ ................ 2009-07-15 20:21 +0000 [r206705] David Vossel * /, channels/chan_sip.c: Merged revisions 206702 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r206702 | dvossel | 2009-07-15 15:20:01 -0500 (Wed, 15 Jul 2009) | 10 lines callerid(num) is wrong when username is missing A domain only sip uri would return 123.123.123.123 as callid num. Now, if the username is missing from a uri, the callerid num field is left empty. (closes issue #15476) Reported by: viraptor ........ 2009-07-15 16:02 +0000 [r206637] Sean Bright * /, codecs/codec_dahdi.c: Merged revisions 206636 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206636 | seanbright | 2009-07-15 12:00:24 -0400 (Wed, 15 Jul 2009) | 9 lines Merged revisions 206635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206635 | seanbright | 2009-07-15 11:57:51 -0400 (Wed, 15 Jul 2009) | 1 line Only print debug info in codec_dahdi if we are asking for it. ........ ................ 2009-07-14 20:22 +0000 [r206585] Tilghman Lesher * /, contrib/scripts/meetme.sql: Recorded merge of revisions 206567 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r206567 | tilghman | 2009-07-14 15:14:45 -0500 (Tue, 14 Jul 2009) | 6 lines Document all meetme realtime fields, and in the process, make some field lengths more consistent. (closes issue #15493) Reported by: lasko Patches: meetme.diff uploaded by lasko (license 833) ........ 2009-07-14 18:17 +0000 [r206555] Richard Mudgett * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, channels/chan_misdn.c, /: Merged revisions 206489 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206489 | rmudgett | 2009-07-14 12:01:48 -0500 (Tue, 14 Jul 2009) | 35 lines Merged revisions 206487 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206487 | rmudgett | 2009-07-14 11:44:47 -0500 (Tue, 14 Jul 2009) | 28 lines Fixes several call transfer issues with chan_misdn. * issue #14355 - Crash if attempt to transfer a call to an application. Masquerade the other pair of the four asterisk channels involved in the two calls. The held call already must be a bridged call (not an applicaton) or it would have been rejected. * issue #14692 - Held calls are not automatically cleared after transfer. Allow the core to initate disconnect of held calls to the ISDN port. This also fixes a similar case where the party on hold hangs up before being transferred or taken off hold. * JIRA ABE-1903 - Orphaned held calls left in music-on-hold. Do not simply block passing the hangup event on held calls to asterisk core. * Fixed to allow held calls to be transferred to ringing calls. Previously, held calls could only be transferred to connected calls. * Eliminated unused call states to simplify hangup code. * Eliminated most uses of "holded" because it is not a word. (closes issue #14355) (closes issue #14692) Reported by: sodom Patches: misdn_xfer_v14_r205839.patch uploaded by rmudgett (license 664) Tested by: rmudgett ........ ................ 2009-07-14 14:54 +0000 [r206387] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 206386 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206386 | russell | 2009-07-14 09:51:44 -0500 (Tue, 14 Jul 2009) | 20 lines Merged revisions 206385 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r206385 | russell | 2009-07-14 09:48:00 -0500 (Tue, 14 Jul 2009) | 13 lines Merged revisions 206384 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r206384 | russell | 2009-07-14 09:45:47 -0500 (Tue, 14 Jul 2009) | 6 lines Ensure apathetic replies are sent out on the proper socket. chan_iax2 supports multiple address bindings. The send_apathetic_reply() function did not attempt to send its response on the same socket that the incoming message came in on. ........ ................ ................ 2009-07-14 01:25 +0000 [r206369] Richard Mudgett * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged revisions 206341 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206341 | rmudgett | 2009-07-13 19:48:59 -0500 (Mon, 13 Jul 2009) | 11 lines Merged revisions 206284 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206284 | rmudgett | 2009-07-13 19:17:28 -0500 (Mon, 13 Jul 2009) | 4 lines Fix some memory leaks in chan_misdn. JIRA ABE-1911 ........ ................ 2009-07-10 22:50 +0000 [r206017] David Vossel * /, channels/chan_sip.c: Merged revisions 205985 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r205985 | dvossel | 2009-07-10 16:42:10 -0500 (Fri, 10 Jul 2009) | 16 lines SIP register not using peer's outbound proxy If callbackextension is defined for a peer it successfully causes a registration to occur, but the registration ignores the outboundproxy settings for the peer. This patch allows the peer to be passed to obproxy_get() in transmit_register(). (closes issue #14344) Reported by: Nick_Lewis Patches: callbackextension_peer_trunk.diff uploaded by dvossel (license 671) Tested by: dvossel Review: https://reviewboard.asterisk.org/r/294/ ........ 2009-07-10 18:44 +0000 [r205940] Kevin P. Fleming * main/udptl.c, /: Merged revisions 205939 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r205939 | kpfleming | 2009-07-10 13:44:09 -0500 (Fri, 10 Jul 2009) | 1 line Update comments about the level of T.38 support in Asterisk. ........ 2009-07-10 17:44 +0000 [r205879-205880] Mark Michelson * channels/chan_sip.c: Fix build. * /, channels/chan_sip.c: Merged revisions 205878 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205878 | mmichelson | 2009-07-10 12:39:57 -0500 (Fri, 10 Jul 2009) | 30 lines Merged revisions 205877 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r205877 | mmichelson | 2009-07-10 12:39:13 -0500 (Fri, 10 Jul 2009) | 23 lines Merged revisions 205776 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205776 | mmichelson | 2009-07-10 10:56:45 -0500 (Fri, 10 Jul 2009) | 16 lines Merged revisions 205775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul 2009) | 10 lines Ensure that outbound NOTIFY requests are properly routed through stateful proxies. With this change, we make note of Record-Route headers present in any SUBSCRIBE request that we receive so that our outbound NOTIFY requests will have the proper Route headers in them. (closes issue #14725) Reported by: ibc ........ ................ ................ ................ 2009-07-10 16:48 +0000 [r205843] David Vossel * /, channels/chan_sip.c: Merged revisions 205840 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205840 | dvossel | 2009-07-10 11:42:04 -0500 (Fri, 10 Jul 2009) | 37 lines Merged revisions 205804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205804 | dvossel | 2009-07-10 11:23:59 -0500 (Fri, 10 Jul 2009) | 31 lines SIP registration auth loop caused by stale nonce If an endpoint sends two registration requests in a very short period of time with the same nonce, both receive 401 responses from Asterisk, each with a different nonce (the second 401 containing the current nonce and the first one being stale). If the endpoint responds to the first 401, it does not match the current nonce so Asterisk sends a third 401 with a newly generated nonce (which updates the current nonce)... Now if the endpoint responds to the second 401, it does not match the current nonce either and Asterisk sends a fourth 401 with a newly generated nonce... This loop goes on and on. There appears to be a simple fix for this. If the nonce from the request does not match our nonce, but is a good response to a previous nonce, instead of sending a 401 with a newly generated nonce, use the current one instead. This breaks the loop as the nonce is not updated until a response is received. Additional logic has been added to make sure no nonce can be responded to twice though. (closes issue #15102) Reported by: Jamuel Patches: patch-bug_0015102 uploaded by Jamuel (license 809) nonce_sip.diff uploaded by dvossel (license 671) Tested by: Jamuel Review: https://reviewboard.asterisk.org/r/289/ ........ ................ 2009-07-10 15:57 +0000 [r205777] Mark Michelson * /, channels/chan_sip.c: Merged revisions 205776 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205776 | mmichelson | 2009-07-10 10:56:45 -0500 (Fri, 10 Jul 2009) | 16 lines Merged revisions 205775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul 2009) | 10 lines Ensure that outbound NOTIFY requests are properly routed through stateful proxies. With this change, we make note of Record-Route headers present in any SUBSCRIBE request that we receive so that our outbound NOTIFY requests will have the proper Route headers in them. (closes issue #14725) Reported by: ibc ........ ................ 2009-07-10 15:35 +0000 [r205771] Kevin P. Fleming * /, apps/app_fax.c: Merged revisions 205770 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r205770 | kpfleming | 2009-07-10 10:28:11 -0500 (Fri, 10 Jul 2009) | 12 lines Fix some remaining T.38 negotiation problems in app_fax. Revision 205696 did not quite fix all the issues with the T.38 negotiation changes and app_fax; this patch corrects them, along with a couple of other minor issues. (closes issue #15480) Reported by: dimas Patches: test2-15480.patch uploaded by dimas (license 88) ........ 2009-07-09 23:46 +0000 [r205729] Richard Mudgett * channels/chan_dahdi.c: Merged revisions 205728 via svn merge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205728 | rmudgett | 2009-07-09 18:37:53 -0500 (Thu, 09 Jul 2009) | 21 lines No audio on calls from Asterisk to various ISDN devices until DTMF sent by caller. Add missing clearing of the dialing flag when the ISDN call is CONNECTED. (i.e. When libpri generates the event PRI_EVENT_ANSWER.) (closes issue #15420) Reported by: scottbmilne Patches: bug15420-1.4.25.1-diff2.txt uploaded by alecdavis (license 585) Tested by: scottbmilne, alecdavis (closes issue #15416) Reported by: avinoash (closes issue #15389) Reported by: alecdavis This patch should also fix the following issue: (issue #15205) Reported by: vinsik ........ 2009-07-09 21:26 +0000 [r205697] Kevin P. Fleming * /, channels/chan_sip.c, apps/app_fax.c, include/asterisk/frame.h: Merged revisions 205696 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r205696 | kpfleming | 2009-07-09 16:20:23 -0500 (Thu, 09 Jul 2009) | 16 lines Repair ability of SendFAX/ReceiveFAX to respond to T.38 switchover. Recent changes in T.38 negotiation in Asterisk caused these applications to not respond when the other endpoint initiated a switchover to T.38; this resulted in the T.38 switchover failing, and the FAX attempt to be made using an audio connection, instead of T.38 (which would usually cause the FAX to fail completely). This patch corrects this problem, and the applications will now correctly respond to the T.38 switchover request. In addition, the response will include the appopriate T.38 session parameters based on what the other end offered and what our end is capable of. (closes issue #14849) Reported by: afosorio ........ 2009-07-09 16:21 +0000 [r205597-205608] David Vossel * include/asterisk/time.h, /: Merged revisions 205600 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205600 | dvossel | 2009-07-09 11:19:09 -0500 (Thu, 09 Jul 2009) | 9 lines Merged revisions 205599 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205599 | dvossel | 2009-07-09 11:18:09 -0500 (Thu, 09 Jul 2009) | 2 lines Changing ast_samp2tv to not use floating point. ........ ................ * main/rtp.c, /, channels/chan_iax2.c, include/asterisk/frame.h: Merged revisions 205479 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205479 | dvossel | 2009-07-08 18:19:09 -0500 (Wed, 08 Jul 2009) | 16 lines Merged revisions 205471 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205471 | dvossel | 2009-07-08 18:15:54 -0500 (Wed, 08 Jul 2009) | 10 lines Fixes 8khz assumptions Many calculations assume 8khz is the codec rate. This is not always the case. This patch only addresses chan_iax.c and res_rtp_asterisk.c, but I am sure there are other areas that make this assumption as well. Review: https://reviewboard.asterisk.org/r/306/ ........ ................ 2009-07-09 08:32 +0000 [r205533] Michiel van Baak * /, main/ssl.c: Merged revisions 205532 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r205532 | mvanbaak | 2009-07-09 10:31:24 +0200 (Thu, 09 Jul 2009) | 5 lines pthread_self returns a pthread_t which is not an unsigned int on all pthread implementations. Casting it to an unsigned int fixes compiler warnings. Tested on OpenBSD and Linux both 32 and 64 bit ........ 2009-07-08 22:17 +0000 [r205415] David Vossel * include/asterisk/devicestate.h, main/pbx.c, /, main/devicestate.c, include/asterisk/pbx.h: Merged revisions 205412 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205412 | dvossel | 2009-07-08 17:15:06 -0500 (Wed, 08 Jul 2009) | 12 lines Merged revisions 205409 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205409 | dvossel | 2009-07-08 16:35:12 -0500 (Wed, 08 Jul 2009) | 6 lines moving ast_devstate_to_extenstate to pbx.c from devicestate.c ast_devstate_to_extenstate belongs in pbx.c. This change fixes a compile time error with chan_vpb as well. ........ ................ 2009-07-08 19:27 +0000 [r205351] Mark Michelson * /, apps/app_queue.c: Merged revisions 205350 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205350 | mmichelson | 2009-07-08 14:26:55 -0500 (Wed, 08 Jul 2009) | 20 lines Merged revisions 205349 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205349 | mmichelson | 2009-07-08 14:26:13 -0500 (Wed, 08 Jul 2009) | 14 lines 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) ........ ................ 2009-07-08 18:20 +0000 [r205296] Jason Parker * config.guess, config.sub, /: Merged revisions 205291 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205291 | qwell | 2009-07-08 13:19:46 -0500 (Wed, 08 Jul 2009) | 9 lines Merged revisions 205288 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205288 | qwell | 2009-07-08 13:19:03 -0500 (Wed, 08 Jul 2009) | 1 line Update config.guess and config.sub from the savannah.gnu.org git repo. ........ ................ 2009-07-08 17:01 +0000 [r205224] Tilghman Lesher * main/say.c: oops, fixing build 2009-07-08 16:56 +0000 [r205220] David Vossel * include/asterisk/time.h, /: Merged revisions 205216 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205216 | dvossel | 2009-07-08 11:54:24 -0500 (Wed, 08 Jul 2009) | 17 lines Merged revisions 205215 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205215 | dvossel | 2009-07-08 11:53:40 -0500 (Wed, 08 Jul 2009) | 10 lines ast_samp2tv needs floating point for 16khz audio In ast_samp2tv(), (1000000 / _rate) = 62.5 when _rate is 16000. The .5 is currently stripped off because we don't calculate using floating points. This causes madness with 16khz audio. (issue ABE-1899) Review: https://reviewboard.asterisk.org/r/305/ ........ ................ 2009-07-08 16:28 +0000 [r205200] Tilghman Lesher * /, main/say.c: Merged revisions 205196 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205196 | tilghman | 2009-07-08 11:27:50 -0500 (Wed, 08 Jul 2009) | 9 lines Merged revisions 205188 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205188 | tilghman | 2009-07-08 11:26:15 -0500 (Wed, 08 Jul 2009) | 2 lines Add redirection warnings for the invalid language codes previously removed. ........ ................ 2009-07-08 15:56 +0000 [r205139-205152] Russell Bryant * /, main/ssl.c: Merged revisions 205151 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r205151 | russell | 2009-07-08 10:56:28 -0500 (Wed, 08 Jul 2009) | 2 lines Use tabs instead of spaces for indentation. ........ * main/asterisk.c, /, main/Makefile, res/res_crypto.c, main/ssl.c (added), include/asterisk/_private.h, res/res_jabber.c: Merged revisions 205120 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r205120 | russell | 2009-07-08 10:17:19 -0500 (Wed, 08 Jul 2009) | 16 lines Move OpenSSL initialization to a single place, make library usage thread-safe. While doing some reading about OpenSSL, I noticed a couple of things that needed to be improved with our usage of OpenSSL. 1) We had initialization of the library done in multiple modules. This has now been moved to a core function that gets executed during Asterisk startup. We already link OpenSSL into the core for TCP/TLS functionality, so this was the most logical place to do it. 2) OpenSSL is not thread-safe by default. However, making it thread safe is very easy. We just have to provide a couple of callbacks. One callback returns a thread ID. The other handles locking. For more information, start with the "Is OpenSSL thread-safe?" question on the FAQ page of openssl.org. ........ 2009-07-08 14:35 +0000 [r205117] David Vossel * channels/chan_sip.c, include/asterisk/sched.h: SIP Dialog ref counting This patch adds reference counting for sip dialogs into 1.6.0. When proc_session_timer() is called from the scheduler thread it has no guarantee the session timer's dialog won't be freed from underneath it. Now the session timer holds a reference to the dialog, preventing it from being destroyed during the middle of proc_session_timer(). (closes issue #13623) Reported by: Nik Soggia Review: https://reviewboard.asterisk.org/r/302/ 2009-07-06 15:17 +0000 [r204980] Tilghman Lesher * main/say.c: Restore Hungarian (mistakenly removed during merge) 2009-07-06 13:39 +0000 [r204949] Kevin P. Fleming * main/channel.c, /: Merged revisions 204948 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r204948 | kpfleming | 2009-07-06 08:38:29 -0500 (Mon, 06 Jul 2009) | 7 lines Improve handling of AST_CONTROL_T38 and AST_CONTROL_T38_PARAMETERS for non-T.38-capable channels. This change allows applications that request T.38 negotiation on a channel that does not support it to get the proper indication that it is not supported, rather than thinking that negotiation was started when it was not. ........ 2009-07-02 22:03 +0000 [r204836] Richard Mudgett * channels/chan_misdn.c, /: Merged revisions 204835 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r204835 | rmudgett | 2009-07-02 17:01:28 -0500 (Thu, 02 Jul 2009) | 17 lines Merged revisions 204834 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204834 | rmudgett | 2009-07-02 16:59:43 -0500 (Thu, 02 Jul 2009) | 10 lines Removed confusing warning message "Got Busy in Connected State" If an incoming mISDN call is answered with the Answer application and a subsequent Dial gets a busy endpoint then it is valid for that already connected channel to get the busy indication. Asterisk will play the busy tones until the dialplan plays something else or hangs up the call. (closes issue #11974) Reported by: fvdb ........ ................ 2009-07-02 18:07 +0000 [r204652-204754] David Vossel * include/asterisk/devicestate.h, main/pbx.c, /, main/devicestate.c: Merged revisions 204710 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r204710 | dvossel | 2009-07-02 11:03:44 -0500 (Thu, 02 Jul 2009) | 21 lines Merged revisions 204681 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204681 | dvossel | 2009-07-02 10:05:57 -0500 (Thu, 02 Jul 2009) | 14 lines Improved mapping of extension states from combined device states. This fixes a few issues with incorrect extension states and adds a cli command, core show device2extenstate, to display all possible state mappings. (closes issue #15413) Reported by: legart Patches: exten_helper.diff uploaded by dvossel (license 671) Tested by: dvossel, legart, amilcar Review: https://reviewboard.asterisk.org/r/301/ ........ ................ * channels/chan_sip.c: removes fake dialog_unref and dialog_ref function calls. dialog_unref() and dialog_ref() in 1.6.0 where only place holders for reference counting once it was implemented. The functions did nothing but return the pointer on ref and NULL on unref. These calls have been removed to make way for a patch that actually does dialog ref counting. 2009-06-30 21:21 +0000 [r204581] Tilghman Lesher * /, main/say.c, UPGRADE.txt: Merged revisions 204563 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r204563 | tilghman | 2009-06-30 15:41:04 -0500 (Tue, 30 Jun 2009) | 13 lines Merged revisions 204556 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204556 | tilghman | 2009-06-30 15:23:51 -0500 (Tue, 30 Jun 2009) | 6 lines More incorrect language codes, plus ensuring that regionalizations use the specified language, and not English for grammar. (closes issue #15022) Reported by: greenfieldtech Patches: 20090519__issue15022.diff.txt uploaded by tilghman (license 14) ........ ................ 2009-06-30 18:50 +0000 [r204476] Jason Parker * /, main/say.c: Merged revisions 204475 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r204475 | qwell | 2009-06-30 13:48:35 -0500 (Tue, 30 Jun 2009) | 9 lines Merged revisions 204474 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204474 | qwell | 2009-06-30 13:47:06 -0500 (Tue, 30 Jun 2009) | 1 line Fix ast_say_counted_noun to correctly handle Polish. Fix a comment typo in passing. ........ ................ 2009-06-30 18:44 +0000 [r204471] Tilghman Lesher * apps/app_voicemail.c, /, main/say.c, UPGRADE.txt: Recorded merge of revisions 204470 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r204470 | tilghman | 2009-06-30 13:36:24 -0500 (Tue, 30 Jun 2009) | 18 lines Recorded merge of revisions 204469 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204469 | tilghman | 2009-06-30 13:23:35 -0500 (Tue, 30 Jun 2009) | 11 lines "tw" is the language specification for Twi (from Ghana) not Taiwanese. (closes issue #15346) Reported by: volivier Patches: 20090617__issue15346__1.4.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__trunk.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__1.6.0.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__1.6.1.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__1.6.2.diff.txt uploaded by tilghman (license 14) Tested by: volivier ........ ................ 2009-06-29 22:52 +0000 [r204248-204302] Mark Michelson * /, channels/chan_sip.c: Merged revisions 204301 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r204301 | mmichelson | 2009-06-29 17:50:35 -0500 (Mon, 29 Jun 2009) | 15 lines Merged revisions 204300 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204300 | mmichelson | 2009-06-29 17:45:34 -0500 (Mon, 29 Jun 2009) | 9 lines Add error message so that it is clear why a SIP peer was not processed when a DNS lookup fails on a host or outboundproxy. (closes issue #13432) Reported by: p_lindheimer Patches: outboundproxy.patch uploaded by p (license 558) ........ ................ * /, channels/chan_sip.c: Merged revisions 204247 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r204247 | mmichelson | 2009-06-29 16:48:54 -0500 (Mon, 29 Jun 2009) | 32 lines Merged revisions 204243,204246 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204243 | mmichelson | 2009-06-29 16:23:43 -0500 (Mon, 29 Jun 2009) | 22 lines Fix a problem where chan_sip would ignore "old" but valid responses. chan_sip has had a problem for quite a long time that would manifest when Asterisk would send multiple SIP responses on the same dialog before receiving a response. The problem occurred because chan_sip only kept track of the highest outgoing sequence number used on the dialog. If Asterisk sent two requests out, and a response arrived for the first request sent, then Asterisk would ignore the response. The result was that Asterisk would continue retransmitting the requests and ignoring the responses until the maximum number of retransmissions had been reached. The fix here is to rearrange the code a bit so that instead of simply comparing the sequence number of the response to our latest outgoing sequence number, we walk our list of outstanding packets and determine if there is a match. If there is, we continue. If not, then we ignore the response. In doing this, I found a few completely useless variables that I have now removed. (closes issue #11231) Reported by: flefoll Review: https://reviewboard.asterisk.org/r/298 ........ r204246 | mmichelson | 2009-06-29 16:37:05 -0500 (Mon, 29 Jun 2009) | 3 lines Fix build oops. ........ ................ 2009-06-27 01:14 +0000 [r203910] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 203909 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r203909 | rmudgett | 2009-06-26 20:07:52 -0500 (Fri, 26 Jun 2009) | 23 lines Merged revisions 203908 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203908 | rmudgett | 2009-06-26 19:55:12 -0500 (Fri, 26 Jun 2009) | 16 lines The ISDN CPE side should not exclusively pick B channels normally. Before this patch, Asterisk unconditionally picked B channels exclusively on the CPE side and normally allowed alternative B channels on the network side. Now Asterisk does the opposite. Reasons for the CPE side to normally not pick B channels exclusively: * For CPE point-to-multipoint mode (i.e. phone side), the CPE side does not have enough information to exclusively pick B channels. (There may be other devices on the line.) * Q.931 gives preference to the network side picking B channels. * Some telcos require the CPE side to not pick B channels exclusively. (closes issue #14383) Reported by: mbrancaleoni ........ ................ 2009-06-26 22:12 +0000 [r203855] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 203853 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r203853 | jpeeler | 2009-06-26 17:11:31 -0500 (Fri, 26 Jun 2009) | 12 lines Merged revisions 203848 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203848 | jpeeler | 2009-06-26 17:09:19 -0500 (Fri, 26 Jun 2009) | 5 lines Make sure to recreate the dahdi pseudo channel after dahdi restart (closes issue #14477) Reported by: timking ........ ................ 2009-06-26 21:25 +0000 [r203780-203818] Russell Bryant * /, main/file.c: Merged revisions 203802 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r203802 | russell | 2009-06-26 16:21:48 -0500 (Fri, 26 Jun 2009) | 22 lines Merged revisions 203785 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203785 | russell | 2009-06-26 16:16:39 -0500 (Fri, 26 Jun 2009) | 15 lines Don't fast forward past the end of a message. This is nice change for users of the voicemail application. If someone gets a little carried away with fast forwarding through a message, they can easily get to the end and accidentally exit the voicemail application by hitting the fast forward key during the following prompt. This adds some safety by not allowing a fast forward past the end of a message. (closes issue #14554) Reported by: lacoursj Patches: 21761.patch uploaded by lacoursj (license 707) Tested by: lacoursj ........ ................ * /, channels/chan_sip.c: Merged revisions 203779 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203779 | russell | 2009-06-26 15:45:00 -0500 (Fri, 26 Jun 2009) | 5 lines Ensure the TCP read buffer is fully initialized before handling each packet. (closes issue #14452) Reported by: umberto71 ........ 2009-06-26 20:16 +0000 [r203722] David Brooks * apps/app_voicemail.c, /: Merged revisions 203721 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203721 | dbrooks | 2009-06-26 15:13:51 -0500 (Fri, 26 Jun 2009) | 16 lines 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/ ........ 2009-06-26 19:54 +0000 [r203717] Jeff Peeler * channels/chan_dahdi.c: reverse whitespace change 203711 that was based on looking at sig_analog (which has about a 1000 line indentation change that is not worth doing here) 2009-06-26 19:49 +0000 [r203716] David Vossel * /, channels/chan_iax2.c: Merged revisions 203710 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203710 | dvossel | 2009-06-26 14:47:11 -0500 (Fri, 26 Jun 2009) | 7 lines moving debug message from level 0 to 1. (closes issue #15404) Reported by: leobrown Patches: iax_codec_debug.patch uploaded by leobrown (license 541) ........ 2009-06-26 19:47 +0000 [r203711] Jeff Peeler * channels/chan_dahdi.c: whitespace fix 2009-06-26 19:29 +0000 [r203701] Joshua Colp * main/rtp.c, main/channel.c, main/frame.c, /, channels/chan_sip.c, apps/app_fax.c, configs/sip.conf.sample, include/asterisk/frame.h: Merged revisions 203699 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203699 | file | 2009-06-26 16:27:24 -0300 (Fri, 26 Jun 2009) | 2 lines Improve T.38 negotiation by exchanging session parameters between application and channel. ........ 2009-06-26 19:25 +0000 [r203698] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 203672 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203672 | jpeeler | 2009-06-26 14:03:25 -0500 (Fri, 26 Jun 2009) | 16 lines Check if polarityonanswerdelay has elapsed before setting a channel as answered after a polarity reversal. Previously on a polarity switch event chan_dahdi would set the channel immediately as answered. This would cause problems if a polarity reversal occurred when the line was picked up as the dial would not have yet occurred. Now if the polarity reversal occurs before delay has elapsed after coming off hook or an answer, it is ignored. Also, some refactoring was done in _handle_event. (closes issue #13917) Reported by: alecdavis Patches: chan_dahdi.bug13917.feb09.diff2.txt uploaded by alecdavis (license 585) Tested by: alecdavis ........ 2009-06-25 21:47 +0000 [r203447] David Vossel * main/ast_expr2.fl, main/ast_expr2.c, /: Merged revisions 203444 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203444 | dvossel | 2009-06-25 16:45:32 -0500 (Thu, 25 Jun 2009) | 4 lines fixes a few redundant conditions (issue #15269) ........ 2009-06-25 21:18 +0000 [r203387] Terry Wilson * main/cli.c, /: Merged revisions 203381 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r203381 | twilson | 2009-06-25 16:15:11 -0500 (Thu, 25 Jun 2009) | 11 lines Merged revisions 203380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203380 | twilson | 2009-06-25 16:13:10 -0500 (Thu, 25 Jun 2009) | 4 lines I didn't see that Mark already fixed the underlying issue! Yay for removing useless code. ........ ................ 2009-06-25 21:06 +0000 [r203117-203377] Russell Bryant * /, main/features.c: Merged revisions 203376 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r203376 | russell | 2009-06-25 16:04:55 -0500 (Thu, 25 Jun 2009) | 16 lines Merged revisions 203375 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203375 | russell | 2009-06-25 16:02:18 -0500 (Thu, 25 Jun 2009) | 9 lines Fix a case where CDR answer time could be before the start time involving parking. (closes issue #13794) Reported by: davidw Patches: 13794.patch uploaded by murf (license 17) 13794.patch.160 uploaded by murf (license 17) Tested by: murf, dbrooks ........ ................ * /, channels/chan_sip.c: Merged revisions 203116 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r203116 | russell | 2009-06-25 11:04:10 -0500 (Thu, 25 Jun 2009) | 18 lines Merged revisions 203115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203115 | russell | 2009-06-25 11:02:16 -0500 (Thu, 25 Jun 2009) | 11 lines Resolve a crash related to a T.38 reinvite race condition. This change resolves a crash observed locally during some T.38 testing. A call was set up using a call file, and when the T.38 reinvite came in, the channel state was still AST_STATE_DOWN. The reason is explained by a comment in the code that previously lived in the handling of AST_STATE_RINGING. This change modifies the logic to handle the same race condition for any channel state that is not UP. (closes ABE-1895) ........ ................ 2009-06-24 21:18 +0000 [r203044] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 203037 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r203037 | rmudgett | 2009-06-24 16:08:55 -0500 (Wed, 24 Jun 2009) | 15 lines Merged revisions 203036 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203036 | rmudgett | 2009-06-24 16:01:43 -0500 (Wed, 24 Jun 2009) | 8 lines Improved chan_dahdi.conf pritimer error checking. Valid format is: pritimer=timer_name,timer_value * Fixed segfault if the ',' is missing. * Completely check the range returned by pri_timer2idx() to prevent possible access outside array bounds. ........ ................ 2009-06-24 18:29 +0000 [r202968] Mark Michelson * /, channels/chan_sip.c: Merged revisions 202967 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r202967 | mmichelson | 2009-06-24 13:29:10 -0500 (Wed, 24 Jun 2009) | 9 lines Merged revisions 202966 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202966 | mmichelson | 2009-06-24 13:28:47 -0500 (Wed, 24 Jun 2009) | 3 lines Use the handy UNLINK macro instead of hand-coding the same thing in-line. ........ ................ 2009-06-24 18:09 +0000 [r202926] Joshua Colp * /, channels/chan_sip.c: Merged revisions 202925 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r202925 | file | 2009-06-24 15:08:17 -0300 (Wed, 24 Jun 2009) | 2 lines Ensure the default settings are applied for T.38 when we set it up for a peer. ........ 2009-06-23 22:09 +0000 [r202763] Matthew Fredrickson * channels/chan_dahdi.c, /: Merged revisions 202761 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r202761 | mattf | 2009-06-23 17:08:43 -0500 (Tue, 23 Jun 2009) | 1 line I could have sworn I committed this patch ages ago, but... bug fix with setting NAI properly on linksets in certain situations. ........ 2009-06-23 21:26 +0000 [r202754] Ryan Brindley * main/config.c, /: Merged revisions 202753 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r202753 | rbrindley | 2009-06-23 16:25:17 -0500 (Tue, 23 Jun 2009) | 9 lines If we delete the info, lets also delete the lines (closes issue #14509) Reported by: timeshell Patches: 20090504__bug14509.diff.txt uploaded by tilghman (license 14) Tested by: awk, timeshell ........ 2009-06-23 16:40 +0000 [r202675] David Vossel * /, channels/chan_sip.c: Merged revisions 202672 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r202672 | dvossel | 2009-06-23 11:31:30 -0500 (Tue, 23 Jun 2009) | 18 lines Merged revisions 202671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202671 | dvossel | 2009-06-23 11:28:46 -0500 (Tue, 23 Jun 2009) | 12 lines MWI NOTIFY contains a wrong URI if Asterisk listens to non-standard port and transport (closes issue #14659) Reported by: klaus3000 Patches: patch_chan_sip_fixMWIuri_1.4.txt uploaded by klaus3000 (license 65) mwi_port-transport_trunk.diff uploaded by dvossel (license 671) Tested by: dvossel, klaus3000 Review: https://reviewboard.asterisk.org/r/288/ ........ ................ 2009-06-22 20:12 +0000 [r202498] Russell Bryant * main/channel.c, /: Merged revisions 202497 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r202497 | russell | 2009-06-22 15:11:04 -0500 (Mon, 22 Jun 2009) | 11 lines Merged revisions 202496 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202496 | russell | 2009-06-22 15:08:53 -0500 (Mon, 22 Jun 2009) | 4 lines Report CallerID change during a masquerade. Reported by: markster ........ ................ 2009-06-22 16:30 +0000 [r202471] Sean Bright * cdr/cdr_sqlite3_custom.c, /: Merged revisions 202417 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r202417 | seanbright | 2009-06-22 12:09:50 -0400 (Mon, 22 Jun 2009) | 4 lines Fix lock usage in cdr_sqlite3_custom to avoid potential crashes during reload. Pointed out by Russell while working on the CEL branch. ........ 2009-06-22 16:06 +0000 [r202416] Russell Bryant * /, channels/chan_sip.c: Merged revisions 202415 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r202415 | russell | 2009-06-22 11:05:08 -0500 (Mon, 22 Jun 2009) | 9 lines Merged revisions 202414 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202414 | russell | 2009-06-22 11:00:00 -0500 (Mon, 22 Jun 2009) | 2 lines Make Polycom subscription type override check more explicit. ........ ................ 2009-06-22 15:05 +0000 [r202338-202344] Mark Michelson * /, channels/chan_sip.c: Merged revisions 202343 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r202343 | mmichelson | 2009-06-22 09:58:24 -0500 (Mon, 22 Jun 2009) | 36 lines Merged revisions 202341-202342 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202341 | mmichelson | 2009-06-22 09:42:55 -0500 (Mon, 22 Jun 2009) | 26 lines Fix a situation in which Asterisk would not stop retransmitting 487s. If a CANCEL were received by Asterisk, we would send a 487 in response to the original INVITE and a 200 OK for the CANCEL. If there were a network hiccup which caused the 200 OK and the 487 to be lost, then the UA communicating with Asterisk may try to retransmit its CANCEL. Asterisk's response to this used to be to try sending another 487 to the canceled INVITE and another 200 OK to the CANCEL. The problem here is that the originally-sent 487 was sent "reliably" meaning that it will be retransmitted until it is received properly. So when we receive the second CANCEL it is likely that the first batch of 487s we sent is still going strong and reaches the UA. The result was that the second set of 487s would be retransmitted constantly until the maximum number of retries had been reached. The fix for this is that if we receive a second CANCEL for an INVITE, then we cancel the retransmission of the first set of 487s and start a second set. This causes the dialog to be terminated reasonably. (closes issue #14584) Reported by: klaus3000 Patches: 14584_v2.patch uploaded by mmichelson (license 60) Tested by: klaus3000 ........ r202342 | mmichelson | 2009-06-22 09:44:58 -0500 (Mon, 22 Jun 2009) | 3 lines Remove an extra debug line left from previous commit. ........ ................ * /, channels/chan_sip.c: Merged revisions 202337 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r202337 | mmichelson | 2009-06-22 09:35:09 -0500 (Mon, 22 Jun 2009) | 31 lines Merged revisions 202336 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202336 | mmichelson | 2009-06-22 09:34:05 -0500 (Mon, 22 Jun 2009) | 25 lines Fix a possible infinite loop in SDP parsing during glare situation. There was a while loop in get_ip_and_port_from_sdp which was controlled by a call to get_sdp_iterate. The loop would exit either if what we were searching for was found or if the return was NULL. The problem is that get_sdp_iterate never returns NULL. This means that if what we were searching for was not present, the loop would run infinitely. This modification of the loop fixes the problem. (closes issue #15213) Reported by: schmidts (closes issue #15349) Reported by: samy (closes issue #14464) Reported by: pj (closes issue #15345) Reported by: aragon Patches: sip_inf_loop.patch uploaded by mmichelson (license 60) Tested by: aragon ........ ................ 2009-06-21 16:14 +0000 [r202259-202263] Russell Bryant * cdr/cdr_manager.c, /: Merged revisions 202262 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r202262 | russell | 2009-06-21 11:11:48 -0500 (Sun, 21 Jun 2009) | 2 lines Fix possibility of crashiness during reload in custom fields handling. ........ * cdr/cdr_manager.c, /: Merged revisions 202258 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r202258 | russell | 2009-06-21 11:00:23 -0500 (Sun, 21 Jun 2009) | 2 lines Standardize return values of load_config() so reload() doesn't report an error on success. ........ 2009-06-20 19:14 +0000 [r202184] Sean Bright * /, apps/app_fax.c: Merged revisions 202183 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r202183 | seanbright | 2009-06-20 15:09:47 -0400 (Sat, 20 Jun 2009) | 5 lines Fix version detection for API changes in spandsp. (closes issue #15355) Reported by: deuffy ........ 2009-06-19 21:07 +0000 [r202006] Matthew Nicholson * channels/chan_sip.c: Added deadlock protection to try_suggested_sip_codec in chan_sip.c. Review: https://reviewboard.asterisk.org/r/287/ 2009-06-19 20:27 +0000 [r201997] David Vossel * /, channels/chan_iax2.c: Merged revisions 201994 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r201994 | dvossel | 2009-06-19 15:24:37 -0500 (Fri, 19 Jun 2009) | 14 lines Merged revisions 201993 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201993 | dvossel | 2009-06-19 15:22:02 -0500 (Fri, 19 Jun 2009) | 8 lines timestamp was being converted to host order as a short rather than a long (closes issue #15361) Reported by: ffloimair Patches: ts_issue.diff uploaded by dvossel (license 671) ........ ................ 2009-06-19 00:44 +0000 [r201786-201830] Tilghman Lesher * /, main/features.c: Merged revisions 201829 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r201829 | tilghman | 2009-06-18 19:43:41 -0500 (Thu, 18 Jun 2009) | 13 lines Merged revisions 201828 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201828 | tilghman | 2009-06-18 19:40:41 -0500 (Thu, 18 Jun 2009) | 6 lines If the "h" extension fails, give it another chance in main/pbx.c. If the "h" extension fails, give it another chance in main/pbx.c, when it returns from the bridge code. Fixes an issue where the "h" extension may occasionally not fire, when a Dial is executed from a Macro. Debugged in #asterisk with user tompaw. ........ ................ * /, apps/Makefile: Merged revisions 201783 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201783 | tilghman | 2009-06-18 15:52:36 -0500 (Thu, 18 Jun 2009) | 6 lines One of the changes in 1.6.1 was to allow app_directory to use functionality within app_voicemail for directory functions. It is therefore no longer necessary for app_directory to be linked against the ODBC libraries (and it never was necessary for app_directory to be linked against IMAP, though it was). ........ 2009-06-18 16:58 +0000 [r201682] David Vossel * channels/misdn/isdn_lib.c, main/asterisk.c, utils/conf2ael.c, main/ast_expr2.c, utils/stereorize.c, codecs/gsm/src/gsm_destroy.c, /, channels/h323/ast_h323.cxx, main/ast_expr2f.c, res/ael/ael_lex.c, utils/ael_main.c, utils/extconf.c, pbx/pbx_config.c, res/res_config_ldap.c: Merged revisions 201678 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201678 | dvossel | 2009-06-18 11:37:42 -0500 (Thu, 18 Jun 2009) | 11 lines fixes some memory leaks and redundant conditions (closes issue #15269) Reported by: contactmayankjain Patches: patch.txt uploaded by contactmayankjain (license 740) memory_leak_stuff.trunk.diff uploaded by dvossel (license 671) Tested by: contactmayankjain, dvossel ........ 2009-06-18 15:32 +0000 [r201612] Russell Bryant * /, res/res_musiconhold.c: Merged revisions 201610 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r201610 | russell | 2009-06-18 10:27:10 -0500 (Thu, 18 Jun 2009) | 36 lines Merged revisions 201600 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201600 | russell | 2009-06-18 10:24:31 -0500 (Thu, 18 Jun 2009) | 29 lines Fix memory corruption and leakage related reloads of non files mode MoH classes. For Music on Hold classes that are not files mode, meaning that we are executing an application that will feed us audio data, we use a thread to monitor the external application and read audio from it. This thread also makes use of the MoH class object. In the MoH class destructor, we used pthread_cancel() to ask the thread to exit. Unfortunately, the code did not wait to ensure that the thread actually went away. What needed to be done is a pthread_join() to ensure that the thread fully cleans up before we proceed. By adding this one line, we resolve two significant problems: 1) Since the thread was never joined, it never fully goes away. So, on every reload of non-files mode MoH, an unused thread was sticking around. 2) There was a race condition here where the application monitoring thread could still try to access the MoH class, even though the thread executing the MoH reload has already destroyed it. (issue #15109) Reported by: jvandal (issue #15123) Reported by: axisinternet (issue #15195) Reported by: amorsen (issue AST-208) ........ ................ 2009-06-17 20:10 +0000 [r201459-201463] Mark Michelson * /, channels/chan_sip.c: Merged revisions 201462 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201462 | mmichelson | 2009-06-17 15:10:01 -0500 (Wed, 17 Jun 2009) | 12 lines Fix problem with no audio due to ignoring the SDP. A recent change to our SDP version comparison made audio not function on some calls. This was because of a test wherein we were trying to see if an unsigned value was less than 0. This is a dumb comparison and arguably the compiler should have warned about it. Alas, though, it slipped past. Now it's fixed by changing the variable to be a signed type. Found by several developers. Tested by mnicholson and dbrooks. ........ * main/channel.c, /: Merged revisions 201458 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r201458 | mmichelson | 2009-06-17 15:04:12 -0500 (Wed, 17 Jun 2009) | 15 lines Merged revisions 201450 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201450 | mmichelson | 2009-06-17 14:59:31 -0500 (Wed, 17 Jun 2009) | 9 lines Change the datastore traversal in ast_do_masquerade to use a safe list traversal. It is possible for datastore fixup functions to remove the datastore from the list and free it. In particular, the queue_transfer_fixup in app_queue does this. While I don't yet know of this causing any crashes, it certainly could. Found while discussing a separate issue with Brian Degenhardt. ........ ................ 2009-06-17 19:55 +0000 [r201449] David Vossel * apps/app_mixmonitor.c, /: Merged revisions 201445 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r201445 | dvossel | 2009-06-17 14:45:35 -0500 (Wed, 17 Jun 2009) | 25 lines Merged revisions 201423 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201423 | dvossel | 2009-06-17 14:28:12 -0500 (Wed, 17 Jun 2009) | 19 lines 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/ ........ ................ 2009-06-17 19:35 +0000 [r201443] David Brooks * /, channels/chan_sip.c: Merged revisions 201381 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r201381 | dbrooks | 2009-06-17 14:15:07 -0500 (Wed, 17 Jun 2009) | 16 lines Merged revisions 201380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201380 | dbrooks | 2009-06-17 13:45:50 -0500 (Wed, 17 Jun 2009) | 9 lines Checks for NULL sip_pvt pointer in chan_sip.c->acf_channel_read() Zombie channels could be passed, and chan_sip.c wasn't checking for it. Could crash Asterisk. Now checking for NULL pointer. (closes issue #15330) Reported by: okrief Tested by: dbrooks ........ ................ 2009-06-17 12:05 +0000 [r201263] Kevin P. Fleming * /, include/asterisk/linkedlists.h: Merged revisions 201262 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r201262 | kpfleming | 2009-06-17 07:04:17 -0500 (Wed, 17 Jun 2009) | 15 lines Merged revisions 201261 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201261 | kpfleming | 2009-06-17 07:03:25 -0500 (Wed, 17 Jun 2009) | 9 lines Correct AST_LIST_APPEND_LIST behavior when list to be appended is empty. When the list to be appended is empty, and the list to be appended to is *not*, AST_LIST_APPEND_LIST would actually cause the target list to become broken, and no longer have a pointer to its last entry. This patch fixes the problem. (reported by Stanislaw Pitucha on the asterisk-dev mailing list) ........ ................ 2009-06-16 22:31 +0000 [r201226] David Vossel * /, channels/chan_sip.c: Merged revisions 201223 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201223 | dvossel | 2009-06-16 17:29:30 -0500 (Tue, 16 Jun 2009) | 2 lines fix issue with build_contact introduced by the "SIP trasnport type issues" commit ........ 2009-06-16 19:34 +0000 [r201093] Kevin P. Fleming * apps/app_chanspy.c, apps/app_mixmonitor.c, main/channel.c, main/autoservice.c, main/frame.c, /, apps/app_meetme.c, configure, main/slinfactory.c, autoconf/ast_gcc_attribute.m4, configure.ac, include/asterisk/linkedlists.h, main/file.c, include/asterisk/channel.h, include/asterisk/frame.h: Merged revisions 201056,201090 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r201056 | kpfleming | 2009-06-16 13:54:30 -0500 (Tue, 16 Jun 2009) | 18 lines Merged revisions 200991 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun 2009) | 11 lines 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/ ........ ................ r201090 | kpfleming | 2009-06-16 14:27:12 -0500 (Tue, 16 Jun 2009) | 5 lines Another minor fix to compiler attribute checking. Defaulting to 'static' for the function scope was bad... so remove it. ................ 2009-06-16 17:11 +0000 [r200992] David Vossel * /, channels/chan_sip.c: Merged revisions 200946 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r200946 | dvossel | 2009-06-16 11:03:30 -0500 (Tue, 16 Jun 2009) | 32 lines SIP transport type issues What this patch addresses: 1. ast_sip_ouraddrfor() by default binds to the UDP address/port reguardless if the sip->pvt is of type UDP or not. Now when no remapping is required, ast_sip_ouraddrfor() checks the sip_pvt's transport type, attempting to set the address and port to the correct TCP/TLS bindings if necessary. 2. It is not necessary to send the port number in the Contact header unless the port is non-standard for the transport type. This patch fixes this and removes the todo note. 3. In sip_alloc(), the default dialog built always uses transport type UDP. Now sip_alloc() looks at the sip_request (if present) and determines what transport type to use by default. 4. When changing the transport type of a sip_socket, the file descriptor must be set to -1 and in some cases the tcptls_session's ref count must be decremented and set to NULL. I've encountered several issues associated with this process and have created a function, set_socket_transport(), to handle the setting of the socket type. (closes issue #13865) Reported by: st Patches: dont_add_port_if_tls.patch uploaded by Kristijan (license 753) 13865.patch uploaded by mmichelson (license 60) tls_port_v5.patch uploaded by vrban (license 756) transport_issues.diff uploaded by dvossel (license 671) Tested by: mmichelson, Kristijan, vrban, jmacz, dvossel Review: https://reviewboard.asterisk.org/r/278/ ........ 2009-06-16 16:34 +0000 [r200986] Kevin P. Fleming * /, configure, autoconf/ast_gcc_attribute.m4, configure.ac: Merged revisions 200985 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r200985 | kpfleming | 2009-06-16 11:32:36 -0500 (Tue, 16 Jun 2009) | 7 lines Fix problems with new compiler attribute checking in configure script. The last changes to ast_gcc_attribute.m4 caused some problems checking for various attributes, because the scope of the symbol the attribute is applied to can be important; this patch allows the scope to be specified for the check. ........ 2009-06-16 16:02 +0000 [r200945] Michiel van Baak * apps/app_voicemail.c, /: Merged revisions 200943 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r200943 | mvanbaak | 2009-06-16 17:51:36 +0200 (Tue, 16 Jun 2009) | 9 lines add FILE_STORAGE to Voicemail Build Options Voicemail can only use one storage module at the moment. Because it's unclear that selecting one of the storage modules in menuselect will disable filesystem storage we now have a FILE_STORAGE option that conflicts with the other modules. (closes issue #15333) ........ 2009-06-16 01:33 +0000 [r200724-200767] Kevin P. Fleming * /, configure, include/asterisk/autoconfig.h.in, autoconf/ast_gcc_attribute.m4: Merged revisions 200764 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r200764 | kpfleming | 2009-06-15 20:28:08 -0500 (Mon, 15 Jun 2009) | 11 lines Ensure that configure-script testing for compiler attributes actually works. The configure script tests for compiler attributes didn't actually enable enough warnings or provide a proper test harness to determine whether the compiler supports the attribute in question or not; this caused gcc 4.1 to report that it supports 'weakref', but it doesn't actually support it in the way that is needed for our optional API mechanism. The new configure script test will properly distinguish between full support and partial support for this attribute, among others. ........ * /, CHANGES: Merged revisions 200726 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r200726 | kpfleming | 2009-06-15 20:03:22 -0500 (Mon, 15 Jun 2009) | 6 lines Document the new automatic 'ignoresdpversion' behavior. Asterisk will now automatically ignore incorrect incoming SDP version numbers when necessary to complete a T.38 re-INVITE operation. ........ * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 165180,200689 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r165180 | mnicholson | 2008-12-17 12:49:12 -0600 (Wed, 17 Dec 2008) | 14 lines This patch adds a new 'ignoresdpversion' option to sip.conf. When this is enabled (either globally or for a specific peer), chan_sip will treat any SDP data it receives as new data and update the media stream accordingly. By default, Asterisk will only modify the media stream if the SDP session version received is different from the current SDP session version. This option is required to interoperate with devices that have non-standard SDP session version implementations (observed by toc on the bug tracker with Microsoft OCS which always uses 0 as the session version). http://reviewboard.digium.com/r/94/ (closes issue #13958) Reported by: toc Tested by: toc ........ r200689 | kpfleming | 2009-06-15 15:42:38 -0500 (Mon, 15 Jun 2009) | 12 lines Accept T.38 re-INVITE responses with invalid SDP versions. This commit changes the 'incoming SDP version' check logic a bit more; when 'ignoresdpversion' is *not* set for a peer, if we initiate a re-INVITE to switch to T.38, we'll always accept the peer's SDP response, even if they don't properly increment the SDP version number as they should. If this situation occurs, a warning message will be generated suggesting that the peer's configuration be changed to include the 'ignoresdpversion' configuration option (although ideally they'd fix their SIP implementation to be RFC compliant). AST-221 ........ 2009-06-15 15:22 +0000 [r200515] Mark Michelson * /, channels/chan_sip.c: Merged revisions 200514 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r200514 | mmichelson | 2009-06-15 10:22:11 -0500 (Mon, 15 Jun 2009) | 11 lines Merged revisions 200513 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200513 | mmichelson | 2009-06-15 10:21:46 -0500 (Mon, 15 Jun 2009) | 5 lines Add INFO to our allowed methods so that endpoints know they may send it to us. AST-223 ........ ................ 2009-06-12 19:08 +0000 [r200362] Mark Michelson * main/channel.c, /: Merged revisions 200361 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r200361 | mmichelson | 2009-06-12 14:07:51 -0500 (Fri, 12 Jun 2009) | 16 lines Merged revisions 200360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200360 | mmichelson | 2009-06-12 14:06:41 -0500 (Fri, 12 Jun 2009) | 10 lines Suppress a warning message and give a better return code when generating inband ringing after a call is answered. (closes issue #15158) Reported by: madkins Patches: 15158.patch uploaded by mmichelson (license 60) Tested by: madkins ........ ................ 2009-06-11 22:42 +0000 [r200228] Sean Bright * Makefile, /: Merged revisions 199781 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r199781 | seanbright | 2009-06-09 14:08:53 -0400 (Tue, 09 Jun 2009) | 2 lines Fix all of the parallel build warnings issued when running make -j#. ........ 2009-06-11 21:18 +0000 [r200149] Mark Michelson * /, channels/chan_sip.c: Merged revisions 200146 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r200146 | mmichelson | 2009-06-11 16:17:14 -0500 (Thu, 11 Jun 2009) | 5 lines Fix a crash due to a potentially NULL p->options. Thanks to mnicholson for pointing it out. ........ 2009-06-11 12:16 +0000 [r200040] Leif Madsen * /, build_tools/make_version_c, build_tools/make_version_h: Merged revisions 200039 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r200039 | lmadsen | 2009-06-11 08:15:09 -0400 (Thu, 11 Jun 2009) | 8 lines Fix path for .flavor and .version (issue #14737) Reported by: davidw Patches: flavor.patch uploaded by davidw (license 780) Tested by: davidw ........ 2009-06-10 20:29 +0000 [r199994] David Brooks * main/pbx.c, /: Fixes the argument order in definition of new_find_extension(). In the definition of new_find_extension(), the arguments 'callerid' and 'label' were swapped. The prototype declaration and all calls to the function are ordered 'callerid' then 'label', but the function itself was ordered 'label' then 'callerid'. (closes issue #15303) Reported by: JimDickenson 2009-06-10 20:20 +0000 [r199975] Mark Michelson * channels/chan_sip.c: The 1.6.0 branch was missing all invite_branch logic. It has now been added. 2009-06-10 16:13 +0000 [r199858] Sean Bright * include/asterisk/utils.h, /: Merged revisions 199857 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r199857 | seanbright | 2009-06-10 12:10:23 -0400 (Wed, 10 Jun 2009) | 9 lines Merged revisions 199856 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199856 | seanbright | 2009-06-10 12:08:35 -0400 (Wed, 10 Jun 2009) | 2 lines __WORDSIZE is not available on all platforms, so use sizeof(void *) instead. ........ ................ 2009-06-09 20:54 +0000 [r199821] David Vossel * /, channels/chan_sip.c: Merged revisions 199818 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r199818 | dvossel | 2009-06-09 15:47:57 -0500 (Tue, 09 Jun 2009) | 11 lines CLI NOTIFY sending wrong transport type. SIP's cli NOTIFY command only used UDP rather than copying the transport type from the peer. (closes issue #15283) Reported by: jthurman Patches: sip-notify-tcp-svn199728.patch uploaded by jthurman (license 614) Tested by: jthurman, dvossel ........ 2009-06-08 19:39 +0000 [r199632] Sean Bright * include/asterisk/utils.h, /: Merged revisions 199630 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r199630 | seanbright | 2009-06-08 15:33:09 -0400 (Mon, 08 Jun 2009) | 32 lines Merged revisions 199626,199628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199626 | seanbright | 2009-06-08 15:24:32 -0400 (Mon, 08 Jun 2009) | 21 lines Increase the size of our thread stack on 64 bit processors. We were setting the stack size for each thread to 240KB regardless of architecture, which meant that in some scenarios we actually had less available stack space on 64 bit processors (pointers use 8 bytes instead of 4). So now we calculate the stack size we reserve based on the platform's __WORDSIZE, which gives us: 32 bit -> 240KB 64 bit -> 496KB 128 bit -> 1008KB (that's right, we're ready for 128 bit processors) Patch typed by me but written by several members of #asterisk-dev, including Kevin, Tilghman, and Qwell. (closes issue #14932) Reported by: jpiszcz Patches: 06052009_issue14932.patch uploaded by seanbright (license 71) Tested by: seanbright ........ r199628 | seanbright | 2009-06-08 15:28:33 -0400 (Mon, 08 Jun 2009) | 2 lines Fix a typo in the stack size calculation just introduced. ........ ................ 2009-06-05 21:37 +0000 [r199301] David Vossel * main/pbx.c, /: Merged revisions 199298 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r199298 | dvossel | 2009-06-05 16:21:22 -0500 (Fri, 05 Jun 2009) | 21 lines Merged revisions 199297 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199297 | dvossel | 2009-06-05 16:19:56 -0500 (Fri, 05 Jun 2009) | 14 lines Fixes issue with hints giving unexpected results. Hints with two or more devices that include ONHOLD gave unexpected results. (closes issue #15057) Reported by: p_lindheimer Patches: onhold_trunk.diff uploaded by dvossel (license 671) pbx.c.1.4.patch uploaded by p (license 558) devicestate.c.trunk.patch uploaded by p (license 671) Tested by: p_lindheimer, dvossel Review: https://reviewboard.asterisk.org/r/254/ ........ ................ 2009-06-05 13:51 +0000 [r199228] Mark Michelson * channels/chan_dahdi.c, /: Merged revisions 199227 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r199227 | mmichelson | 2009-06-05 08:51:08 -0500 (Fri, 05 Jun 2009) | 14 lines Correct "dahdi show channels" output when specifying a group. Since a DAHDI channel may belong to multiple groups, we need to use a bitwise and instead of equivalence to determine whether to display the channel information. (closes issue #15248) Reported by: gentian Patches: 15248.patch uploaded by mmichelson (license 60) Tested by: gentian ........ 2009-06-04 19:16 +0000 [r199142] David Vossel * /, channels/chan_iax2.c: Merged revisions 199139 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r199139 | dvossel | 2009-06-04 14:10:16 -0500 (Thu, 04 Jun 2009) | 9 lines Merged revisions 199138 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199138 | dvossel | 2009-06-04 14:00:15 -0500 (Thu, 04 Jun 2009) | 3 lines Additional updates to AST-2009-001 ........ ................ 2009-08-11 Tilghman Lesher * Asterisk 1.6.0.13 released * channels/chan_sip.c: Bad merge from 1.6.0 branch 2009-08-10 Tilghman Lesher * Asterisk 1.6.0.12 released * AST-2009-005 2009-06-05 Leif Madsen * Asterisk 1.6.0.10 released 2009-06-04 David Vossel * channels/chan_iax2.c: Additional updates for AST-2009-001 2009-06-04 David Vossel * channels/chan_iax2.c: REGAUTH loop fix related to AST-2009-001 2009-04-06 Leif Madsen * Release Asterisk 1.6.0.9 2009-04-03 16:27 -0500 [r186517] Mark Michelson * Remove an invalid call to free memory. A bad merge from trunk to 1.6.0 meant freeing memory that should not be freed. In trunk, pkt->data is an ast_str, but in 1.6.0, it is allocated in the same chunk of memory as the sip_pkt. This only affects 1.6.0. (closes issue #14819) Reported by: cwolff09 2009-04-02 Leif Madsen * Release Asterisk 1.6.0.8 2009-04-02 Tilghman Lesher * Fix for security issue AST-2009-003 2009-03-30 Leif Madsen * Release Asterisk 1.6.0.7 2009-03-19 Leif Madsen * Release Asterisk 1.6.0.7-rc2 2009-03-19 15:40 +0000 [r183066-183109] Joshua Colp * /, channels/chan_sip.c: Merged revisions 183108 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r183108 | file | 2009-03-19 12:37:23 -0300 (Thu, 19 Mar 2009) | 11 lines Improve our triggering of a T38 switchover internally when triggered by a received reinvite. Previously we reached across the channel bridge to get the other party's SIP dialog structure in order to trigger an outgoing reinvite. This is extremely dangerous to do and only works if bridged to another SIP channel. This patch changes this to use the T38 control frame method of requesting a switchover. This change also causes the SIP channel driver to propogate back whether the switchover worked or not instead of blindly accepting the incoming T38 reinvite. Review: http://reviewboard.digium.com/r/200/ ........ * main/channel.c, /: Merged revisions 183057 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r183057 | file | 2009-03-18 19:22:56 -0300 (Wed, 18 Mar 2009) | 6 lines Fix an issue where a T38 control frame would get dropped. If two channels were bridged together using a generic bridge the T38 control frame would get passed up instead of being indicated on the other channel. ........ 2009-03-18 21:19 +0000 [r183029] Jeff Peeler * /, channels/h323/ast_h323.cxx: Merged revisions 183028 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r183028 | jpeeler | 2009-03-18 16:18:27 -0500 (Wed, 18 Mar 2009) | 4 lines Add some code removed by mistake from commit 182722 that works around a file descriptor leak in versions of PWLib prior to 1.12.0. ........ 2009-03-18 14:24 +0000 [r182945] Russell Bryant * main/poll.c, main/io.c, main/channel.c, channels/chan_skinny.c, configure, apps/app_mp3.c, res/res_agi.c, include/asterisk/poll-compat.h, channels/chan_alsa.c, main/asterisk.c, apps/app_nbscat.c, /, main/Makefile, include/asterisk/autoconfig.h.in, configure.ac, include/asterisk/io.h, main/utils.c, include/asterisk/channel.h: Merged revisions 182847 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r182847 | russell | 2009-03-17 21:28:55 -0500 (Tue, 17 Mar 2009) | 52 lines Merged revisions 182810 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines Fix cases where the internal poll() was not being used when it needed to be. We have seen a number of problems caused by poll() not working properly on Mac OSX. If you search around, you'll find a number of references to using select() instead of poll() to work around these issues. In Asterisk, we've had poll.c which implements poll() using select() internally. However, we were still getting reports of problems. vadim investigated a bit and realized that at least on his system, even though we were compiling in poll.o, the system poll() was still being used. So, the primary purpose of this patch is to ensure that we're using the internal poll() when we want it to be used. The changes are: 1) Remove logic for when internal poll should be used from the Makefile. Instead, put it in the configure script. The logic in the configure script is the same as it was in the Makefile. Ideally, we would have a functionality test for the problem, but that's not actually possible, since we would have to be able to run an application on the _target_ system to test poll() behavior. 2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll() throughout the source tree to ast_poll(). I feel that it is good practice to give the API call a new name when we are changing its behavior and not using the system version directly in all cases. So, normally, ast_poll() is just redefined to poll(). On systems where AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll(). 4) Change poll() in main/poll.c to be ast_internal_poll(). It's worth noting that any code that still uses poll() directly will work fine (if they worked fine before). So, for example, out of tree modules that are using poll() will not stop working or anything. However, for modules to work properly on Mac OSX, ast_poll() needs to be used. (closes issue #13404) Reported by: agalbraith Tested by: russell, vadim http://reviewboard.digium.com/r/198/ ........ ................ 2009-03-17 20:51 +0000 [r182723] Jeff Peeler * channels/h323/compat_h323.cxx, /, channels/h323/ast_h323.cxx, configure, autoconf/ast_check_openh323.m4, channels/h323/compat_h323.h, channels/chan_h323.c, channels/h323/ast_h323.h, channels/h323/chan_h323.h: Merged revisions 182722 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182722 | jpeeler | 2009-03-17 15:47:31 -0500 (Tue, 17 Mar 2009) | 15 lines Allow H.323 Plus library to be used in addition to the OpenH323 library Chan_h323 can now be compiled against both the previously supported versions of OpenH323 as well as the current H.323 Plus (version 1.20.2). The configure script has been modified to look in the default install location of h323 to hopefully help avoid using the environment variables OPENH323DIR and PWLIBDIR. Also, the CLI command "h323 show version" has been added which indicates which version of h323 is in use. (closes issue #11261) Reported by: vhatz Patches: asterisk-1.6.0.6-h323plus.patch uploaded by jthurman (license 614) ........ 2009-03-17 15:27 +0000 [r182569] Russell Bryant * main/channel.c, /: Merged revisions 182553 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182553 | russell | 2009-03-17 10:22:12 -0500 (Tue, 17 Mar 2009) | 5 lines Tweak the handling of the frame list inside of ast_answer(). This does not change any behavior, but moves the frames from the local frame list back to the channel read queue using an O(n) algorithm instead of O(n^2). ........ 2009-03-17 15:00 +0000 [r182526-182532] Kevin P. Fleming * main/channel.c, /: Merged revisions 182530 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182530 | kpfleming | 2009-03-17 09:59:33 -0500 (Tue, 17 Mar 2009) | 2 lines correct logic flaw in ast_answer() changes in r182525 ........ * main/channel.c, /, main/features.c, include/asterisk/channel.h: Merged revisions 182525 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182525 | kpfleming | 2009-03-17 09:38:11 -0500 (Tue, 17 Mar 2009) | 11 lines Improve behavior of ast_answer() to not lose incoming frames ast_answer(), when supplied a delay before returning to the caller, use ast_safe_sleep() to implement the delay. Unfortunately during this time any incoming frames are discarded, which is problematic for T.38 re-INVITES and other sorts of channel operations. When a delay is not passed to ast_answer(), it still delays for up to 500 milliseconds, waiting for media to arrive. Again, though, it discards any control frames, or non-voice media frames. This patch rectifies this situation, by storing all incoming frames during the delay period on a list, and then requeuing them onto the channel before returning to the caller. http://reviewboard.digium.com/r/196/ ........ 2009-03-17 05:53 +0000 [r182451] Tilghman Lesher * main/db.c, /: Merged revisions 182450 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r182450 | tilghman | 2009-03-17 00:51:54 -0500 (Tue, 17 Mar 2009) | 14 lines Merged revisions 182449 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182449 | tilghman | 2009-03-17 00:50:52 -0500 (Tue, 17 Mar 2009) | 7 lines Fix race in astdb The underlying db1 implementation does not fully isolate the pages retrieved from astdb, so the lock protecting accesses needs to be extended until the copy from the shared memory structure is done. (closes issue #14682) Reported by: makoto ........ ................ 2009-03-16 17:52 +0000 [r182283] David Vossel * /, channels/chan_iax2.c: Merged revisions 182282 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r182282 | dvossel | 2009-03-16 12:49:58 -0500 (Mon, 16 Mar 2009) | 13 lines Merged revisions 182281 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182281 | dvossel | 2009-03-16 12:47:42 -0500 (Mon, 16 Mar 2009) | 7 lines Randomize IAX2 encryption padding The 16-32 byte random padding at the beginning of an encrypted IAX2 frame turns out to not be all that random at all. This patch calls ast_random to fill the padding buffer with random data. The padding is randomized at the beginning of every encrypted call and for every encrypted retransmit frame. Review: http://reviewboard.digium.com/r/193/ ........ ................ 2009-03-16 17:36 +0000 [r182212-182279] Tilghman Lesher * /, funcs/func_env.c: Merged revisions 182278 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182278 | tilghman | 2009-03-16 12:33:38 -0500 (Mon, 16 Mar 2009) | 7 lines Fix an off-by-one error in the FILE() function, and extend FILE()'s length parameter to work like variable substitution. Previously, FILE() returned one less character than specified, due to the terminating NULL. Both the offset and length parameters now behave identically to the way variable substitution offsets and lengths also work. (closes issue #14670) Reported by: BMC ........ * channels/chan_local.c, /: Merged revisions 182211 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r182211 | tilghman | 2009-03-16 10:50:55 -0500 (Mon, 16 Mar 2009) | 14 lines Merged revisions 182208 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182208 | tilghman | 2009-03-16 10:39:15 -0500 (Mon, 16 Mar 2009) | 7 lines Fixup glare detection, to fix a memory leak of a local pvt structure. (closes issue #14656) Reported by: caspy Patches: 20090313__bug14656__2.diff.txt uploaded by tilghman (license 14) Tested by: caspy ........ ................ 2009-03-16 13:59 +0000 [r182172] Joshua Colp * main/channel.c, /: Merged revisions 182171 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182171 | file | 2009-03-16 10:58:24 -0300 (Mon, 16 Mar 2009) | 7 lines Fix a memory leak in the ast_answer / __ast_answer API call. For a channel that is not yet answered this API call will wait until a voice frame is received on the channel before returning. It does this by waiting for frames on the channel and reading them in. The frames read in were not freed when they should have been. ........ 2009-03-13 21:26 +0000 [r182064-182122] Mark Michelson * /, apps/app_queue.c: Merged revisions 182121 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182121 | mmichelson | 2009-03-13 16:26:20 -0500 (Fri, 13 Mar 2009) | 6 lines Change faulty comparison used when announcing average hold minutes and seconds (closes issue #14227) Reported by: caspy ........ * /, main/features.c: Merged revisions 182029 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r182029 | mmichelson | 2009-03-13 12:26:43 -0500 (Fri, 13 Mar 2009) | 41 lines Merged revisions 181990 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181990 | mmichelson | 2009-03-13 12:12:32 -0500 (Fri, 13 Mar 2009) | 35 lines Check the DYNAMIC_FEATURES of both the chan and peer when interpreting DTMF. Dynamic features defined in the applicationmap section of features.conf allow one to specify whether the caller, callee, or both have the ability to use the feature. The documentation in the features.conf.sample file could be interpreted to mean that one only needs to set the DYNAMIC_FEATURES channel variable on the calling channel in order to allow for the callee to be able to use the features which he should have permission to use. However, the DYNAMIC_FEATURES variable would only be read from the channel of the participant that pressed the DTMF sequence to activate the feature. The result of this was that the callee was unable to use dynamic features unless the dialplan writer had taken measures to be sure that the DYNAMIC_FEATURES variable was set on the callee's channel. This commit changes the behavior of ast_feature_interpret to concatenate the values of DYNAMIC_FEATURES from both parties involved in the bridge. The features themselves determine who has permission to use them, so there is no reason to believe that one side of the bridge could gain the ability to perform an action that they should not have the ability to perform. Kevin Fleming pointed out on the asterisk-users list that the typical way that this was worked around in the past was by setting _DYNAMIC_FEATURES on the calling channel so that the value would be inherited by the called channel. While this works, the documentation alone is not enough to figure out why this is necessary for the callee to be able to use dynamic features. In this particular case, changing the code to match the documentation is safe, easy, and will generally make things easier for people for future installations. This bug was originally reported on the asterisk-users list by David Ruggles. (closes issue #14657) Reported by: mmichelson Patches: 14657.patch uploaded by mmichelson (license 60) ........ ................ 2009-03-13 17:28 +0000 [r182036] Joshua Colp * /, channels/chan_sip.c: Merged revisions 182022 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182022 | file | 2009-03-13 14:25:09 -0300 (Fri, 13 Mar 2009) | 7 lines Fix an issue with requesting a T38 reinvite before the call is answered. The code responsible for sending the T38 reinvite did not check if an INVITE was already being handled. This caused things to get confused and the call to fail. The code now defers sending the T38 reinvite until the current INVITE is done being handled. (issue AST-191) ........ 2009-03-12 21:44 +0000 [r181770-181848] Mark Michelson * /, apps/app_queue.c: Merged revisions 181846 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r181846 | mmichelson | 2009-03-12 16:43:51 -0500 (Thu, 12 Mar 2009) | 3 lines Run the macro on the queue member's channel when he answers, not the caller's channel. ........ * /, channels/chan_sip.c: Merged revisions 181769 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181769 | mmichelson | 2009-03-12 13:30:58 -0500 (Thu, 12 Mar 2009) | 28 lines Merged revisions 181768 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181768 | mmichelson | 2009-03-12 13:29:48 -0500 (Thu, 12 Mar 2009) | 22 lines Properly send a 487 on an INVITE we have not responded to if we receive a BYE. If we receive an INVITE from an endpoint and then later receive a BYE from that same endpoint before we have sent a final response for the INVITE, then we need to respond to the INVITE with a 487. There was logic in the code prior to this commit which seemed to exist solely to handle this situation, but there was one condition in an if statement which was incorrect. The only way we would send a 487 was if the sip_pvt had no owner channel. This made no sense since we created the owner channel when we received the INVITE, meaning that the majority of the time we would never send the 487. The 487 being sent should not rely on whether we have created a channel. Its delivery should be dependent on the current state of the initial INVITE transaction. With this commit, that logic is now correctly in place. (closes issue #14149) Reported by: legranjl Patches: 14149.patch uploaded by mmichelson (license 60) Tested by: legranjl ........ ................ 2009-03-12 17:58 +0000 [r181732] Tilghman Lesher * /, configure, main/translate.c: Merged revisions 181731 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r181731 | tilghman | 2009-03-12 12:32:13 -0500 (Thu, 12 Mar 2009) | 9 lines Adjust translation table column widths based upon the translation times. Previously, only 5 columns were displayed, and if a translation time exceeded 99,999 useconds, it would be displayed as 0, instead of its actual time. (closes issue #14532) Reported by: pj Patches: 20090311__bug14532.diff.txt uploaded by tilghman (license 14) Tested by: pj ........ 2009-03-12 16:57 +0000 [r181613-181666] Joshua Colp * /, res/res_musiconhold.c: Merged revisions 181665 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181665 | file | 2009-03-12 13:56:58 -0300 (Thu, 12 Mar 2009) | 9 lines Merged revisions 181664 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181664 | file | 2009-03-12 13:56:20 -0300 (Thu, 12 Mar 2009) | 2 lines Fix incorrect usage of strncasecmp... I really meant to use strcasecmp. ........ ................ * /, res/res_musiconhold.c: Merged revisions 181661 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181661 | file | 2009-03-12 13:53:52 -0300 (Thu, 12 Mar 2009) | 19 lines Merged revisions 181659-181660 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181659 | file | 2009-03-12 13:50:37 -0300 (Thu, 12 Mar 2009) | 8 lines Fix another scenario where depending on configuration the stream would not get read. For custom commands we don't know whether the audio is coming from a stream or not so we are going to have to read the data despite no channels. (closes issue #14416) Reported by: caspy ........ r181660 | file | 2009-03-12 13:52:45 -0300 (Thu, 12 Mar 2009) | 2 lines Fix logic flaw in previous commit. ........ ................ * /, res/res_musiconhold.c: Merged revisions 181656 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181656 | file | 2009-03-12 13:32:20 -0300 (Thu, 12 Mar 2009) | 17 lines Merged revisions 181655 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181655 | file | 2009-03-12 13:29:19 -0300 (Thu, 12 Mar 2009) | 10 lines Fix issue with streaming MOH failing if nobody is listening. When a music class is setup to actually provide music on hold from a stream we need to constantly read audio from it since it will constantly be providing audio. This is now done despite there being no channels listening to it. (closes issue #14416) Reported by: caspy ........ ................ * apps/app_dial.c, /: Merged revisions 181612 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r181612 | file | 2009-03-12 10:24:12 -0300 (Thu, 12 Mar 2009) | 5 lines Fix crash when sleep and retries argument was not given to RetryDial application. (closes issue #14647) Reported by: sherpya ........ 2009-03-12 01:04 +0000 [r181543] Richard Mudgett * /, build_tools/make_version: Merged revisions 181542 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r181542 | rmudgett | 2009-03-11 20:00:29 -0500 (Wed, 11 Mar 2009) | 1 line Use the correct branch integrated property when generating the version string ........ 2009-03-11 23:19 +0000 [r181509] Michiel van Baak * /, configs/sip.conf.sample: Merged revisions 181499 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk Provide correct hint to debug SIP trouble in the default config (closes issue #14646) Reported by: strk 2009-03-11 22:22 +0000 [r181450] Jason Parker * /, configure, configure.ac: Merged revisions 181444 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181444 | qwell | 2009-03-11 17:20:13 -0500 (Wed, 11 Mar 2009) | 11 lines Merged revisions 181436 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181436 | qwell | 2009-03-11 17:18:42 -0500 (Wed, 11 Mar 2009) | 4 lines Allow prefix to set localstatedir (when used and different from the default). This is similar to the /etc change that was made for the non-FreeBSD case. ........ ................ 2009-03-11 22:15 +0000 [r181425-181429] Russell Bryant * main/channel.c, /: Merged revisions 181428 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r181428 | russell | 2009-03-11 17:14:55 -0500 (Wed, 11 Mar 2009) | 2 lines Make handling of the BRIDGEPVTCALLID variable thread-safe. ........ * main/channel.c, /: Merged revisions 181424 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181424 | russell | 2009-03-11 16:49:29 -0500 (Wed, 11 Mar 2009) | 17 lines Merged revisions 181423 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181423 | russell | 2009-03-11 16:42:58 -0500 (Wed, 11 Mar 2009) | 9 lines Make code that updates BRIDGEPEER variable thread-safe. It is not safe to read the name field of an ast_channel without the channel locked. This patch fixes some places in channel.c where this was being done, and lead to crashes related to masquerades. (closes issue #14623) Reported by: guillecabeza ........ ................ 2009-03-11 17:37 +0000 [r181372] David Vossel * channels/iax2-parser.h, /, channels/chan_iax2.c: Merged revisions 181371 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181371 | dvossel | 2009-03-11 12:34:57 -0500 (Wed, 11 Mar 2009) | 17 lines Merged revisions 181340 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181340 | dvossel | 2009-03-11 12:25:31 -0500 (Wed, 11 Mar 2009) | 11 lines encrypted IAX2 during packet loss causes decryption to fail on retransmitted frames If an iax channel is encrypted, and a retransmit frame is sent, that packet's iseqno is updated while it is encrypted. This causes the entire frame to be corrupted. When the corrupted frame is sent, the other side decrypts it and sends a VNAK back because the decrypted frame doesn't make any sense. When we get the VNAK, we look through the sent queue and send the same corrupted frame causing a loop. To fix this, encrypted frames requiring retransmission are decrypted, updated, then re-encrypted. Since key-rotation may change the key held by the pvt struct, the keys used for encryption/decryption are held within the iax_frame to guarantee they remain correct. (closes issue #14607) Reported by: stevenla Tested by: dvossel Review: http://reviewboard.digium.com/r/192/ ........ ................ 2009-03-11 17:28 +0000 [r181297-181352] Joshua Colp * /, channels/chan_sip.c: Merged revisions 181345 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181345 | file | 2009-03-11 14:26:40 -0300 (Wed, 11 Mar 2009) | 21 lines Merged revisions 181328 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181328 | file | 2009-03-11 14:22:52 -0300 (Wed, 11 Mar 2009) | 14 lines Fix issue where an attended transfer could not be completed under a rare scenario. When completing an attended transfer chan_sip does a check to make sure the extension in the URI portion of the Refer-To header is a local valid extension. We don't actually need to check this since we know for sure the other channel is already up and talking to the extension. Some devices do not put the extension in the Refer-To header either, which can cause the extension check to fail. We now no longer do this check if it is an attended transfer. (closes issue #14628) Reported by: sverre Patches: 14628.diff uploaded by file (license 11) ........ ................ * /, channels/chan_sip.c: Merged revisions 181296 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181296 | file | 2009-03-11 13:40:48 -0300 (Wed, 11 Mar 2009) | 16 lines Merged revisions 181295 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181295 | file | 2009-03-11 13:36:50 -0300 (Wed, 11 Mar 2009) | 9 lines Fix a problem with inband DTMF detection on outgoing SIP calls when dtmfmode=auto. When dtmfmode was set to auto the inband DTMF detector was not setup on outgoing SIP calls. This caused inband DTMF detection to fail. The inband DTMF detector is now setup for both dtmfmode inband and auto. (closes issue #13713) Reported by: makoto ........ ................ 2009-03-11 15:54 +0000 [r181137-181284] Jeff Peeler * channels/h323/ast_h323.cxx: add missing header file * utils/extconf.c: Fix merge oops from 181137 * utils/Makefile, include/asterisk/utils.h, include/asterisk/astmm.h, /, channels/chan_sip.c, channels/h323/ast_h323.cxx, utils/extconf.c: Merged revisions 181135 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r181135 | jpeeler | 2009-03-10 23:06:44 -0500 (Tue, 10 Mar 2009) | 20 lines Fix malloc debug macros to work properly with h323. The main problem here was that cstdlib was undefining free thereby causing the proper debug macros to not be used. ast_h323.cxx has been changed to call ast_free instead to avoid the issue. A few other issues were addressed: - There were a few instances of functions improperly passing ast_free instead of ast_free_ptr. - Some clean up was done to avoid the debug macros intentionally being redefined. (copied below from Kevin's commit, appreciate the help) - disable astmm.h from doing anything when STANDALONE is defined, which is used by the tools in the utils/ directory that use parts of Asterisk header files in hackish ways; also ensure that utils/extconf.c and utils/conf2ael.c are compiled with STANDALONE defined. (closes issue #13593) Reported by: pj ........ 2009-03-11 00:52 +0000 [r181034] Mark Michelson * /, channels/chan_sip.c: Merged revisions 181032-181033 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181032 | mmichelson | 2009-03-10 19:46:47 -0500 (Tue, 10 Mar 2009) | 19 lines Merged revisions 181029,181031 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181029 | mmichelson | 2009-03-10 19:30:26 -0500 (Tue, 10 Mar 2009) | 9 lines Fix incorrect tag checking on transfers when pedantic=yes is enabled. (closes issue #14611) Reported by: klaus3000 Patches: patch_chan_sip_attended_transfer_1.4.23.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 ........ r181031 | mmichelson | 2009-03-10 19:32:40 -0500 (Tue, 10 Mar 2009) | 3 lines Remove unused variables. ........ ................ r181033 | mmichelson | 2009-03-10 19:49:00 -0500 (Tue, 10 Mar 2009) | 3 lines Add missing comment that quotes RFC 3891 ................ 2009-03-10 22:05 +0000 [r180946] Jason Parker * /, configure, configure.ac, autoconf/ast_prog_sed.m4, autoconf/ast_check_gnu_make.m4: Merged revisions 180944 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r180944 | qwell | 2009-03-10 17:03:41 -0500 (Tue, 10 Mar 2009) | 9 lines Merged revisions 180941 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180941 | qwell | 2009-03-10 17:02:18 -0500 (Tue, 10 Mar 2009) | 1 line Make things happier when using autoconf 2.62+ ........ ................ 2009-03-10 14:41 +0000 [r180718-180801] Joshua Colp * main/manager.c, /: Merged revisions 180800 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r180800 | file | 2009-03-10 11:40:38 -0300 (Tue, 10 Mar 2009) | 5 lines Reset the thread local string buffer when handling the UserEvent action. (closes issue #14593) Reported by: JimDickenson ........ * channels/chan_sip.c: If a port is specified when dialing a peer then use it. (closes issue #14626) Reported by: acunningham * channels/chan_sip.c: Ensure that the new outgoing dialog to a peer is able to set the socket details, even if the default is present. (closes issue #14480) Reported by: jon 2009-03-06 18:26 +0000 [r180582] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 180579 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r180579 | mmichelson | 2009-03-06 12:25:44 -0600 (Fri, 06 Mar 2009) | 9 lines Merged revisions 180567 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180567 | mmichelson | 2009-03-06 12:23:09 -0600 (Fri, 06 Mar 2009) | 2 lines Make compilation succeed in dev-mode when IMAP storage is enabled. ........ ................ 2009-03-06 Leif Madsen * Release 1.6.0.7-rc1 2009-03-06 17:28 +0000 [r180535] David Vossel * main/enum.c, /: Merged revisions 180534 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r180534 | dvossel | 2009-03-06 11:26:38 -0600 (Fri, 06 Mar 2009) | 15 lines Merged revisions 180532 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180532 | dvossel | 2009-03-06 11:19:55 -0600 (Fri, 06 Mar 2009) | 9 lines Fix handling of backreferences for ENUM lookups enum.c did not handle regex backtraces correctly. The '\1' in the regex is a backreference that requires a pattern match to be inserted. The way the code used to work is that it would find the backreference and insert the entire input string minus the '+'. This is incorrect. The regexec() function takes in a variable called pmatch which is an array of structs containing the start and end indexes for each backreference substring. The original code actually passed the pmatch array pointer into regexec but never did anything with it. Now when a backtrace is found, the backtrace number is looked up in the pmatch array and the correct substring is inserted. (closes issue #14576) Reported by: chris-mac Review: http://reviewboard.digium.com/r/187/ ........ ................ 2009-03-05 23:28 +0000 [r180404-180466] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 180465 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r180465 | mmichelson | 2009-03-05 17:26:58 -0600 (Thu, 05 Mar 2009) | 22 lines Merged revisions 180464 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180464 | mmichelson | 2009-03-05 17:26:11 -0600 (Thu, 05 Mar 2009) | 16 lines [IMAP] Fix message retrieval issues when identical mailbox names were defined in separate contexts. There was a fix put in a while back so that an X-Asterisk-VM-Context message header was added to stored IMAP voicemails. This would allow for us to differentiate if the same mailbox name was used in multiple contexts. The problem still left was that not all places where messages were retrieved actually attempted to use this header for information when retrieving messages. This commit fixes that so that MWI and message retrieval from VoiceMailMain work as expected. (closes issue #13853) Reported by: vicks1 Patches: 13853_v2.patch uploaded by mmichelson (license 60) Tested by: lmadsen ........ ................ * apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged revisions 180383 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r180383 | mmichelson | 2009-03-05 13:14:14 -0600 (Thu, 05 Mar 2009) | 31 lines Merged revisions 180380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180380 | mmichelson | 2009-03-05 12:58:48 -0600 (Thu, 05 Mar 2009) | 25 lines Fix broken mailbox parsing when searchcontexts option is enabled. When using the searchcontexts option in voicemail.conf, the code made the assumption that all mailbox names defined were unique across all contexts. However, the code did nothing to actually enforce this assumption, nor did it do anything to alert a user that he may have created an ambiguity in his voicemail.conf file by defining the same mailbox name in multiple contexts. With this change, we now will issue a nice long warning if searchcontexts is on and we encounter the same mailbox name in multiple contexts and ignore any duplicates after the first box. Whether searchcontexts is enabled or not, if we come across a duplicate mailbox in the same context, then we will issue a warning and ignore the duplicated mailbox. I have also added a small note to voicemail.conf.sample in the explanation for searchcontexts explaining that you cannot define the same mailbox in multiple contexts if you have enabled the option. (closes issue #14599) Reported by: lmadsen Patches: 14599.patch uploaded by mmichelson (license 60) (with slight modification) Tested by: lmadsen ........ ................ 2009-03-05 18:36 +0000 [r180377] Kevin P. Fleming * main/rtp.c, main/frame.c, /, include/asterisk/frame.h: Merged revisions 180373 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r180373 | kpfleming | 2009-03-05 12:29:38 -0600 (Thu, 05 Mar 2009) | 15 lines Merged revisions 180372 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180372 | kpfleming | 2009-03-05 12:22:16 -0600 (Thu, 05 Mar 2009) | 9 lines Fix problems when RTP packet frame size is changed During some code analysis, I found that calling ast_rtp_codec_setpref() on an ast_rtp session does not work as expected; it does not adjust the smoother that may on the RTP session, in fact it summarily drops it, even if it has data in it, even if the current format's framing size has not changed. This is not good. This patch changes this behavior, so that if the packetization size for the current format changes, any existing smoother is safely updated to use the new size, and if no smoother was present, one is created. A new API call for smoothers, ast_smoother_reconfigure(), was required to implement these changes. Review: http://reviewboard.digium.com/r/184/ ........ ................ 2009-03-04 19:25 +0000 [r180121-180196] Joshua Colp * /, main/callerid.c: Merged revisions 180195 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r180195 | file | 2009-03-04 15:24:59 -0400 (Wed, 04 Mar 2009) | 11 lines Merged revisions 180194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180194 | file | 2009-03-04 15:22:50 -0400 (Wed, 04 Mar 2009) | 4 lines Look for the number in a callerid string starting from the end. This way a value using <> can exist in the name portion. (issue #AST-194) ........ ................ * apps/app_dial.c, /: Merged revisions 180120 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r180120 | file | 2009-03-04 10:39:28 -0400 (Wed, 04 Mar 2009) | 7 lines Remove duplicate 'k' and 'K' Dial options. (closes issue #14601) Reported by: alecdavis Patches: app_dial.optionk.diff.txt uploaded by alecdavis (license 585) ........ 2009-03-03 23:35 +0000 [r180078] David Vossel * main/channel.c, include/asterisk/app.h, apps/app_read.c, /, main/app.c: Merged revisions 180032 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r180032 | dvossel | 2009-03-03 17:21:18 -0600 (Tue, 03 Mar 2009) | 14 lines app_read does not break from prompt loop with user terminated empty string In app.c, ast_app_getdata is called to stream the prompts and receive DTMF input. If ast_app_getdata() receives an empty string caused by the user inputing the end of string character, in this case '#', it should break from the prompt loop and return to app_read, but instead it cycles through all the prompts. I've added a return value for this special case in ast_readstring() which uses an enum I've delcared in apps.h. This enum is now used as a return value for ast_app_getdata(). (closes issue #14279) Reported by: Marquis Patches: fix_app_read.patch uploaded by Marquis (license 32) read-ampersanmd.patch2 uploaded by dvossel (license 671) Tested by: Marquis, dvossel Review: http://reviewboard.digium.com/r/177/ ........ 2009-03-03 23:26 +0000 [r180058] Steve Murphy * main/ast_expr2.fl, main/ast_expr2.c, utils/Makefile, utils/expr2.testinput, /, main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2f.c: Merged revisions 179973 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179973 | murf | 2009-03-03 15:12:02 -0700 (Tue, 03 Mar 2009) | 33 lines Merged revisions 179807 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 I had some work to do to port these changes to trunk; the check_expr stuff hasn't been updated here for quite some time, it appears. I added some more tests to the check_expr2 suite. I had to play around with the makefile a bit, etc. I added STANDALONE2 #ifdefs to ast_expr2.y so as not to conflict structure with aelparse. ........ r179807 | murf | 2009-03-03 11:11:34 -0700 (Tue, 03 Mar 2009) | 19 lines These changes allow AEL to better check ${} constructs within $[...], that are concatenated with text. I modified and added rules in ast_expr2.fl to better handle the concatenations. I added some default routines to ast_expr2.y so the standalone would compile. It also looks like I haven't run this thru bison since 2.1, so it's good to get this updated. The Makefile has comments added now for check_expr2 and check_expr to explain what they are for, and how to run them. The testexpr2s stuff has been removed, in favor of check_expr2. expr2.testinput has been updated to include the two expressions that inspired these changes (from mcnobody on #asterisk this morning) The regression has been run and all looks well. ........ ................ 2009-03-03 22:49 +0000 [r179971-180008] Mark Michelson * /, configs/queues.conf.sample, apps/app_queue.c: Merged revisions 180007 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r180007 | mmichelson | 2009-03-03 16:49:07 -0600 (Tue, 03 Mar 2009) | 22 lines Merged revisions 180006 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180006 | mmichelson | 2009-03-03 16:48:18 -0600 (Tue, 03 Mar 2009) | 17 lines Clarify some documentation of queues.conf.sample It had always been possible to explicitly specify a "blank" value for a sound file in queues.conf and have no sound played back. The problem with this is that it would result in some ugly CLI warnings from file.c. This commit introduces a check when playing a file in app_queue to see if the name of the file is zero-length and return early if that is the case. Also, the ability to specify the blank sound files in queues.conf is now mentioned more clearly in queues.conf.sample (closes issue #14227) Reported by: caspy ........ ................ * apps/app_queue.c: Fix a memory leak when updating a realtime member field. This was discovered while looking at issue #14353 2009-03-03 18:29 +0000 [r179842] Joshua Colp * /, main/features.c: Merged revisions 179841 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179841 | file | 2009-03-03 14:28:46 -0400 (Tue, 03 Mar 2009) | 16 lines Merged revisions 179840 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179840 | file | 2009-03-03 14:27:09 -0400 (Tue, 03 Mar 2009) | 9 lines Do not assume that the bridge_cdr is still attached to the channel when the 'h' exten is finished executing. It is possible for a masquerade operation to occur when the 'h' exten is operating. This operation moves the CDR records around causing the bridge_cdr to no longer exist on the channel where it is expected to. We can not safely modify it afterwards because of this, so don't even try. (closes issue #14564) Reported by: meric ........ ................ 2009-03-03 16:48 +0000 [r179743] Russell Bryant * main/channel.c, /: Merged revisions 179742 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179742 | russell | 2009-03-03 10:47:28 -0600 (Tue, 03 Mar 2009) | 14 lines Merged revisions 179741 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179741 | russell | 2009-03-03 10:45:46 -0600 (Tue, 03 Mar 2009) | 6 lines Ensure chan->fdno always gets reset to -1 after handling a channel fd event. Since setting fdno to -1 had to be moved, a couple of other code paths that do process an fd event return early and do not pass through the code path where it was moved to. So, set it to -1 in a few other places, too. ........ ................ 2009-03-03 14:40 +0000 [r179673] Joshua Colp * main/channel.c, /: Merged revisions 179672 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179672 | file | 2009-03-03 10:40:04 -0400 (Tue, 03 Mar 2009) | 10 lines Merged revisions 179671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179671 | file | 2009-03-03 10:38:09 -0400 (Tue, 03 Mar 2009) | 3 lines Move where fdno is set to the default value to *after* the read callback of the channel driver is called. We have to do this as the underlying channel driver may need the fdno value to determine what to read. ........ ................ 2009-03-03 13:55 +0000 [r179610] Russell Bryant * main/channel.c, /: Merged revisions 179609 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179609 | russell | 2009-03-03 07:54:41 -0600 (Tue, 03 Mar 2009) | 17 lines Merged revisions 179608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179608 | russell | 2009-03-03 07:53:52 -0600 (Tue, 03 Mar 2009) | 9 lines Make it easier to detect an improper call to ast_read(). When you call ast_waitfor() on a channel, the index into the channel fds array that holds the file descriptor that poll() determines has input available is stored in fdno. This patch clears out this value after a call to ast_read() and also reports errors if ast_read() is called without an fdno set. From a discussion on the asterisk-dev list. ........ ................ 2009-03-03 00:03 +0000 [r179538] Jeff Peeler * main/channel.c, /: Merged revisions 179537 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179537 | jpeeler | 2009-03-02 18:01:51 -0600 (Mon, 02 Mar 2009) | 21 lines Merged revisions 179536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179536 | jpeeler | 2009-03-02 17:54:39 -0600 (Mon, 02 Mar 2009) | 15 lines Fix bridging regression from commit 176701 This fixes a bad regression where the bridge would exit after an attended transfer was made. The problem was due to nexteventts getting set after the masquerade which caused the bridge to return AST_BRIDGE_COMPLETE. (closes issue #14315) Reported by: tim_ringenbach ........ ................ 2009-03-02 23:38 +0000 [r179534] Russell Bryant * /, apps/app_meetme.c: Merged revisions 179533 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179533 | russell | 2009-03-02 17:36:38 -0600 (Mon, 02 Mar 2009) | 48 lines Merged revisions 179532 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179532 | russell | 2009-03-02 17:34:13 -0600 (Mon, 02 Mar 2009) | 40 lines Move ast_waitfor() down to avoid the results of the API call becoming stale. This call to ast_waitfor() was being done way too soon in this section of code. Specifically, there was code in between the call to waitfor and the code that uses the result that puts the channel in autoservice. By putting the channel in autoservice, the previous results of ast_waitfor() become meaningless, as the autoservice thread will do it's own ast_waitfor() and ast_read() on the channel. So, when we came back out of autoservice and eventually hit the block of code that calls ast_read() on the channel, there may not actually be any input on the channel available. Even though the previous call to ast_waitfor() in app_meetme said there was input, the autoservice thread has since serviced the channel for some period of time. This bug manifested itself while dvossel was doing some testing of MeetMe in Asterisk trunk. He was using the timerfd timing module. When the code hit ast_read() erroneously, it determined that it must have been called because of input on the timer fd, as chan->fdno was set to AST_TIMING_FD, since that was the cause of the last legitimate call to ast_read() done by autoservice. In this test, an IAX2 channel was calling into the MeetMe conference. It was _much_ more likely to be seen with an IAX2 channel because of the way audio is handled. Every audio frame that comes in results in a call to ast_queue_frame(), which then uses ast_timer_enable_continuous() to notify the channel thread that a frame is waiting to be handled. So, the chances of ast_waitfor() indicating that a channel needs servicing due to a timer event on an IAX2 event is very high. Finally, it is interesting to note that if a different timing interface was being used, this bug would probably not be noticed. When ast_read() is called and erroneously thinks that there is a timer event to handle, it calls the ast_timer_ack() function. The pthread and dahdi timing modules handle the ack() function being called when there is no event by simply ignoring it. In the case of the timerfd module, it results in a read() on the timer fd that will block forever, as there is no data to read. This caused Asterisk to lock up very quickly. Thanks to dvossel and mmichelson for the fun debugging session. :-) ........ ................ 2009-03-02 23:15 +0000 [r179473] Mark Michelson * /, channels/chan_sip.c: Merged revisions 151464 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r151464 | mmichelson | 2008-10-21 18:54:41 -0500 (Tue, 21 Oct 2008) | 11 lines Make the sip_standard_port function more granular by allowing separate type and port arguments. This is necessary because when building our From and Contact headers, we need to be absolutely sure that we are placing our source port there and not the peer's source port. (closes issue #12761) Reported by: asbestoshead Patches: patch-chan-sip-contact-port.txt uploaded by asbestoshead (license 455) ........ 2009-03-02 23:11 +0000 [r179470] Tilghman Lesher * /, main/app.c: Merged revisions 179469 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179469 | tilghman | 2009-03-02 17:10:18 -0600 (Mon, 02 Mar 2009) | 17 lines Merged revisions 179468 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179468 | tilghman | 2009-03-02 17:09:01 -0600 (Mon, 02 Mar 2009) | 10 lines When ending a recording with silence detection, remember to reduce the duration. The end of the recording is correspondingly trimmed, but the duration was not trimmed by the number of seconds trimmed, so the saved duration was necessarily longer than the actual soundfile duration. (closes issue #14406) Reported by: sasargen Patches: 20090226__bug14406.diff.txt uploaded by tilghman (license 14) Tested by: sasargen ........ ................ 2009-03-02 23:02 +0000 [r179463] Russell Bryant * main/channel.c, /: Merged revisions 179462 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179462 | russell | 2009-03-02 17:00:30 -0600 (Mon, 02 Mar 2009) | 16 lines Merged revisions 179461 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179461 | russell | 2009-03-02 16:58:18 -0600 (Mon, 02 Mar 2009) | 8 lines Ensure that only one thread is calling ast_settimeout() on a channel at a time. For example, with an IAX2 channel, you can have both the channel thread and the chan_iax2 processing threads calling this function, and doing so twice at the same time is a bad thing. (Found in a debugging session with dvossel and mmichelson) ........ ................ 2009-03-02 20:17 +0000 [r179402] Jason Parker * /, main/editline/configure, main/editline/np/unvis.c, main/editline/sys.h, main/editline/configure.in: Merged revisions 179396 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179396 | qwell | 2009-03-02 14:16:51 -0600 (Mon, 02 Mar 2009) | 9 lines Merged revisions 179395 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179395 | qwell | 2009-03-02 14:14:57 -0600 (Mon, 02 Mar 2009) | 1 line Remove several silly warnings in editline. One about a broken preprocessor directive, and another about strlcpy/strlcat. (closes issue #14264) Reported by: dimas ........ ................ 2009-03-02 17:58 +0000 [r179360-179363] Tilghman Lesher * apps/app_stack.c: KeepAlive application no longer exists, so fix gosub implementation to not use it. (closes issue #14571) Reported by: zktech Patches: 20090302__bug14571.diff.txt uploaded by tilghman (license 14) Tested by: tilghman * cdr/cdr_sqlite3_custom.c: If cdr registration somehow succeeds without a config file, don't crash. (closes issue #14563) Reported by: alerios 2009-03-01 22:07 +0000 [r179220-179222] Mark Michelson * apps/app_queue.c: Add error checking when updating the "paused" field of a realtime queue member. This code already existed in trunk and 1.6.1, but was not in 1.6.0 prior to this commit. (closes issue #14338) Reported by: fiddur Patches: 14338.patch uploaded by mmichelson (license 60) Tested by: fiddur * /, channels/chan_sip.c: Merged revisions 179219 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r179219 | mmichelson | 2009-03-01 15:45:08 -0600 (Sun, 01 Mar 2009) | 18 lines Properly free memory and remove scheduler entries when a transmission failure occurs. Previously, only the "data" field of the sip_pkt created during __sip_reliable_xmit was freed when XMIT_FAILURE was returned by __sip_xmit. When retrans_pkt was called, this inevitably resulted in the reading and writing of freed memory. XMIT_FAILURE is a condition meaning that we don't want to attempt resending the packet at all. The proper action to take is to remove the scheduler entry we just created, free the packet's data as well as the packet itself, and unlink it from the list of packets on the sip_pvt structure. (closes issue #14455) Reported by: Nick_Lewis Patches: 14455.patch uploaded by mmichelson (license 60) Tested by: Nick_Lewis ........ 2009-02-27 21:33 +0000 [r179162] Tilghman Lesher * cdr/cdr_sqlite3_custom.c, /: Merged revisions 179161 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r179161 | tilghman | 2009-02-27 15:32:13 -0600 (Fri, 27 Feb 2009) | 3 lines If config file is blank, don't load module. (Closes issue #14563) ........ 2009-02-27 19:05 +0000 [r179058] Jason Parker * /, doc/tex/channelvariables.tex: Merged revisions 179057 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r179057 | qwell | 2009-02-27 13:04:57 -0600 (Fri, 27 Feb 2009) | 8 lines Update documentation for DIALEDTIME and ANSWEREDTIME variables. (closes issue #14566) Reported by: klaus3000 Patches: ANSWEREDTIME-1.4-patch.txt uploaded by klaus3000 (license 65) ANSWEREDTIME-trunk-patch.txt uploaded by klaus3000 (license 65) ........ 2009-02-27 03:52 +0000 [r178987] Steve Murphy * configs/features.conf.sample, /, main/features.c: Merged revisions 178986 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r178986 | murf | 2009-02-26 20:45:58 -0700 (Thu, 26 Feb 2009) | 26 lines Merged revisions 178956 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 In this case, it's just a matter of reducing the default timeouts from 2000 to 1000 msec, as the max def feature digit timeout is no longer halved. ........ r178956 | murf | 2009-02-26 14:27:32 -0700 (Thu, 26 Feb 2009) | 18 lines This change moves the default feature digit timeout to 1000 ms from the previous default of 500. As per bug 14515, a dev discussion arrived at a "mediated concensus" of a default feature digit timeout of 1.0 sec. Some voted for 1300; ctooley thought 1500 for distracted phone users in phone booths; kpfleming put his foot down at 1.0 sec. Users who found the previous default max delay of 250 msec perfect, are welcome to override the new default. Notice that I said that 250 msec was the default; wait a minute, you might say, the config file said it was 500 msec!; well, because of the bug fix for 14515, we found that 500 msec was actually enforcing a max of 250. The bug fix would restore 500 msec, but we felt even that was a bit tight for most users... 2000 msec was pushed earlier by mmichelson, so that reduces to 1000 msec after the bug fix. Enjoy! ........ ................ 2009-02-26 17:50 +0000 [r178874] David Vossel * /, channels/chan_iax2.c: Merged revisions 178871 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r178871 | dvossel | 2009-02-26 11:46:12 -0600 (Thu, 26 Feb 2009) | 6 lines IAX2 prune realtime, minor tweak to last fix A return statement was missing which caused unexpected cli output. issue #14479 ........ 2009-02-26 17:29 +0000 [r178866] Steve Murphy * /, main/features.c: Merged revisions 178828 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r178828 | murf | 2009-02-26 10:22:11 -0700 (Thu, 26 Feb 2009) | 34 lines Merged revisions 178804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178804 | murf | 2009-02-26 10:09:03 -0700 (Thu, 26 Feb 2009) | 28 lines This patch prevents the feature detection timeout from being cut in half. Because the ast_channel_bridge() call will return 0 and pass a frame pointer for both DTMF_BEGIN and DTMF_END, the feature_timer field in hte config struct is getting decremented twice, which effectively cuts the digittimeout in half. I added conditions to the if statement to only let DTMF_END frames to flow thru, which solved the problem. Also, when the frame pointer is null, let control flow thru-- this usually happens on timeouts. I added a comment to the code to explain what's going on and why. Many thanks to sodom for reporting this problem. Personnally, it always seemed like something was wrong with the featuredigittimeout, but I never could quite decide what... and was too busy to investigate. This bug forced the issue, and now we know. Sodom had other issues in 14515, but I couldn't reproduce them. If he still has problems, and wants to get them solved, he is welcome to reopen 14515. (closes issue #14515) Reported by: sodom Patches: 14515.patch uploaded by murf (license 17) Tested by: murf, sodom ........ ................ 2009-02-26 16:01 +0000 [r178768] David Vossel * /, channels/chan_iax2.c: Merged revisions 178767 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r178767 | dvossel | 2009-02-26 09:50:22 -0600 (Thu, 26 Feb 2009) | 8 lines IAX2 prune realtime fix Iax2 prune realtime had issues. If "iax2 prune realtime all" was called, it would appear like the command was successful, but in reality nothing happened. This is because the reload that was supposed to take place checks the config files, sees no changes, and does nothing. If there had been a change in the the config file, the realtime users would have been marked for deletion and everything would have been fine. Now prune_users() and prune_peers() are called instead of reload_config() to prune all users/peers that are realtime. These functions remove all users/peers with the rtfriend and delme flags set. iax2_prune_realtime() also lacked the code to properly delete a single friend. For example. if iax2 prune realtime was called, only the peer instance would be removed. The user would still remain. (closes issue #14479) Reported by: mousepad99 Review: http://reviewboard.digium.com/r/176/ ........ 2009-02-25 12:46 +0000 [r178510] Russell Bryant * main/asterisk.c, /: Merged revisions 178509 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r178509 | russell | 2009-02-25 06:45:30 -0600 (Wed, 25 Feb 2009) | 10 lines Merged revisions 178508 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178508 | russell | 2009-02-25 06:43:36 -0600 (Wed, 25 Feb 2009) | 2 lines Update the copyright year for the main page of the doxygen documentation. ........ ................ 2009-02-24 23:28 +0000 [r178382-178447] Tilghman Lesher * configs/extensions.conf.sample, /: Merged revisions 178446 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r178446 | tilghman | 2009-02-24 17:27:23 -0600 (Tue, 24 Feb 2009) | 12 lines Merged revisions 178445 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178445 | tilghman | 2009-02-24 17:25:24 -0600 (Tue, 24 Feb 2009) | 5 lines Add section about the #exec command in configuration files. (closes issue #14540) Reported by: jtodd Patch by: jtodd, with additional notes by tilghman (license 14) ........ ................ * main/asterisk.c, /: Merged revisions 178381 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r178381 | tilghman | 2009-02-24 14:52:44 -0600 (Tue, 24 Feb 2009) | 2 lines Apparently, a void cast doesn't override warn_unused_result. ........ 2009-02-24 20:43 +0000 [r178378] Russell Bryant * main/rtp.c, /: Merged revisions 178374 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r178374 | russell | 2009-02-24 14:39:57 -0600 (Tue, 24 Feb 2009) | 14 lines Merged revisions 178373 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178373 | russell | 2009-02-24 14:36:19 -0600 (Tue, 24 Feb 2009) | 6 lines Only set dtmfcount on BEGIN, and ensure it gets reset to 0 properly. (issue #14460) Reported by: moliveras Tested by: russell ........ ................ 2009-02-24 20:40 +0000 [r178343-178376] Tilghman Lesher * main/asterisk.c, /: Merged revisions 178375 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r178375 | tilghman | 2009-02-24 14:40:02 -0600 (Tue, 24 Feb 2009) | 2 lines The 3 possible errors with pipe(2) are all impossible in this situation. ........ * main/asterisk.c, /, utils/astcanary.c: Merged revisions 178342 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r178342 | tilghman | 2009-02-24 14:06:48 -0600 (Tue, 24 Feb 2009) | 2 lines Use a SIGPIPE to kill the process, instead of depending upon the astcanary process being inherited by init. ........ 2009-02-24 18:05 +0000 [r178306] Terry Wilson * apps/app_dahdiras.c: Change include order to make compile on Centos 5 with DAHDI If BIT_TYPES_DEFINED gets defined before linux/types.h is included, the __s32 type doesn't get defined 2009-02-24 17:53 +0000 [r178304] Tilghman Lesher * /, utils/astcanary.c: Merged revisions 178303 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r178303 | tilghman | 2009-02-24 11:51:36 -0600 (Tue, 24 Feb 2009) | 7 lines Cause astcanary to exit if Asterisk exits abnormally and doesn't kill astcanary. Also, add some documentation supporting the use of astcanary. (closes issue #14538) Reported by: KNK Patches: asterisk-1.6.x-astcanary.diff uploaded by KNK (license 545) ........ 2009-02-24 15:20 +0000 [r178224] Joshua Colp * /, channels/chan_sip.c: Merged revisions 178213 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r178213 | file | 2009-02-24 11:18:38 -0400 (Tue, 24 Feb 2009) | 16 lines Merged revisions 178205 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178205 | file | 2009-02-24 11:16:07 -0400 (Tue, 24 Feb 2009) | 9 lines Skip check for extension when subscribing for MWI. Since the remote side is not actually subscribing to a specific extension when subscribing for MWI just skip the check to see if the extension exists. They can't use it to specify the mailbox either since we require configuration of that in sip.conf (closes issue #14531) Reported by: festr ........ ................ 2009-02-23 23:17 +0000 [r178145] Russell Bryant * main/rtp.c, /: Merged revisions 178142 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r178142 | russell | 2009-02-23 17:11:37 -0600 (Mon, 23 Feb 2009) | 22 lines Merged revisions 178141 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178141 | russell | 2009-02-23 17:09:01 -0600 (Mon, 23 Feb 2009) | 14 lines Fix infinite DTMF when a BEGIN is received without an END. This commit is related to rev 175124 of 1.4 where a previous attempt was made to fix this problem. The problem with the previous patch was that the inserted code needed to go _before_ setting the lastrxts to the current timestamp. Because those were the same, the dtmfcount variable was never decremented, and so the END was never sent. In passing, I removed the dtmfsamples variable which was completed unused. I also removed a redundant setting of the lastrxts variable. (closes issue #14460) Reported by: moliveras ........ ................ 2009-02-23 Leif Madsen * Released 1.6.0.6 2009-02-13 Leif Madsen * Released 1.6.0.6-rc1 2009-02-13 16:43 +0000 [r175550] Joshua Colp * /, apps/app_record.c: Merged revisions 175549 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175549 | file | 2009-02-13 12:41:15 -0400 (Fri, 13 Feb 2009) | 4 lines Add an option to keep the recorded file upon hangup. (closes issue #14341) Reported by: fnordian ........ 2009-02-12 21:41 +0000 [r175369] Russell Bryant * /, channels/chan_sip.c: Merged revisions 175368 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175368 | russell | 2009-02-12 15:41:01 -0600 (Thu, 12 Feb 2009) | 2 lines Remove useless string copy, and make sscanf safe again ........ 2009-02-12 21:27 +0000 [r175347] Tilghman Lesher * main/udptl.c, /: Merged revisions 175334 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r175334 | tilghman | 2009-02-12 15:25:14 -0600 (Thu, 12 Feb 2009) | 16 lines Merged revisions 175311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175311 | tilghman | 2009-02-12 15:19:40 -0600 (Thu, 12 Feb 2009) | 9 lines Fix crashes when receiving certain T.38 packets. Also, increase the maximum size of T.38 packets and warn users when they try to set the limits above those maximums. (closes issue #13050) Reported by: schern Patches: 20090212__bug13050.diff.txt uploaded by Corydon76 (license 14) Tested by: schern ........ ................ 2009-02-12 20:59 +0000 [r175299-175301] Jeff Peeler * main/features.c: Fix mistake in merging conflict from 175299. * /, main/features.c: Merged revisions 175298 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r175298 | jpeeler | 2009-02-12 14:48:56 -0600 (Thu, 12 Feb 2009) | 15 lines Merged revisions 175294 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175294 | jpeeler | 2009-02-12 14:34:36 -0600 (Thu, 12 Feb 2009) | 9 lines Fix ParkedCall event information for From field in the case of a blind transfer If the parker information can not be obtained from the peer, try and see if the BLINDTRANSFER channel variable has been set. Previously, a blind transfer to the ParkAndAnnounce app would return nothing for the From. Closes AST-189 ........ ................ 2009-02-12 20:46 +0000 [r175256-175296] Russell Bryant * /, channels/chan_sip.c: Merged revisions 175295 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175295 | russell | 2009-02-12 14:45:47 -0600 (Thu, 12 Feb 2009) | 2 lines Avoid using ast_strdupa() in a loop. ........ * build_tools/cflags.xml, /: Merged revisions 175255 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175255 | russell | 2009-02-12 13:11:08 -0600 (Thu, 12 Feb 2009) | 4 lines Don't enable something by default that has a dependency on something _not_ enabled by default. menuselect was not happy with this. ........ 2009-02-12 18:00 +0000 [r175189] Jeff Peeler * /, main/features.c: Merged revisions 175188 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r175188 | jpeeler | 2009-02-12 12:00:11 -0600 (Thu, 12 Feb 2009) | 12 lines Merged revisions 175187 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175187 | jpeeler | 2009-02-12 11:57:10 -0600 (Thu, 12 Feb 2009) | 6 lines Fix crash in event of failed attempt to transfer to parking The peer may not necessarily exist, such as in the case of a transfer to ParkAndAnnounce. In this case don't try to play a sound to it. ........ ................ 2009-02-12 17:03 +0000 [r175126] Russell Bryant * main/rtp.c, /: Merged revisions 175125 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r175125 | russell | 2009-02-12 10:57:25 -0600 (Thu, 12 Feb 2009) | 35 lines Merged revisions 175124 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175124 | russell | 2009-02-12 10:51:13 -0600 (Thu, 12 Feb 2009) | 27 lines Don't send DTMF for infinite time if we do not receive an END event. I thought that this was going to end up being a pretty gnarly fix, but it turns out that there was actually already a configuration option in rtp.conf, dtmftimeout, that was intended to handle this situation. However, in between Asterisk 1.2 and Asterisk 1.4, the code that processed the option got lost. So, this commit brings it back to life. The default timeout is 3 seconds. However, it is worth noting that having this be configurable at all is not really the recommended behavior in RFC 2833. From Section 3.5 of RFC 2833: Limiting the time period of extending the tone is necessary to avoid that a tone "gets stuck". Regardless of the algorithm used, the tone SHOULD NOT be extended by more than three packet interarrival times. A slight extension of tone durations and shortening of pauses is generally harmless. Three seconds will pretty much _always_ be far more than three packet interarrival times. However, that behavior is not required, so I'm going to leave it with our legacy behavior for now. Code from svn/asterisk/team/russell/issue_14460 (closes issue #14460) Reported by: moliveras ........ ................ 2009-02-12 16:33 +0000 [r175122] Mark Michelson * main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions 175121 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175121 | mmichelson | 2009-02-12 10:28:06 -0600 (Thu, 12 Feb 2009) | 11 lines Make lock information for ao2_trylock be more useful and gnarly Core show locks information involving an ao2_trylock did not show the function that called ao2_trylock, but would instead show ao2_trylock as the source of the lock. This is not useful when trying to debug locking issues. One bizarre note is that this logic is already in 1.4 but somehow did not get merged to trunk or the 1.6.X branches. ........ 2009-02-12 14:27 +0000 [r175059-175090] Philippe Sultan * /, channels/chan_gtalk.c: Merged revisions 175089 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175089 | phsultan | 2009-02-12 15:25:03 +0100 (Thu, 12 Feb 2009) | 6 lines Issue a warning message if our candidate's IP is the loopback address. (closes issue #13985) Reported by: jcovert Tested by: phsultan ........ * /, channels/chan_gtalk.c: Merged revisions 175058 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r175058 | phsultan | 2009-02-12 11:31:36 +0100 (Thu, 12 Feb 2009) | 20 lines Merged revisions 175029 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175029 | phsultan | 2009-02-12 11:16:21 +0100 (Thu, 12 Feb 2009) | 12 lines Set the initiator attribute to lowercase in our replies when receiving calls. This attribute contains a JID that identifies the initiator of the GoogleTalk voice session. The GoogleTalk client discards Asterisk's replies if the initiator attribute contains uppercase characters. (closes issue #13984) Reported by: jcovert Patches: chan_gtalk.2.patch uploaded by jcovert (license 551) Tested by: jcovert ........ ................ 2009-02-11 23:04 +0000 [r174765-174949] Mark Michelson * /, apps/app_queue.c: Merged revisions 174948 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r174948 | mmichelson | 2009-02-11 17:03:08 -0600 (Wed, 11 Feb 2009) | 35 lines Fix odd "thank you" sound playing behavior in app_queue.c If someone has configured the queue to play an position or holdtime announcement, then it is odd and potentially unexpected to hear a "Thank you for your patience" sound when no position or holdtime was actually announced. This fixes the announcement so that the "thanks" sound is only played in the case that a position or holdtime was actually announced. There is a way that the "thank you" sound can be played without a position or holdtime, and that is to set announce-frequency to a value but keep announce-position and announce-holdtime both turned off. (closes issue #14227) Reported by: caspy Patches: 14227_v3.patch uploaded by putnopvut (license 60) Tested by: caspy ................ * apps/app_dial.c, main/channel.c, main/pbx.c, /, apps/app_dictate.c, apps/app_waitforsilence.c, include/asterisk/channel.h: Merged revisions 174945 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174945 | mmichelson | 2009-02-11 16:41:01 -0600 (Wed, 11 Feb 2009) | 29 lines Fix 'd' option for app_dial and add new option to Answer application The 'd' option would not work for channel types which use RTP to transport DTMF digits. The only way to allow for this to work was to answer the channel if we saw that this option was enabled. I realized that this may cause issues with CDRs, specifically with giving false dispositions and answer times. I therefore modified ast_answer to take another parameter which would tell if the CDR should be marked answered. I also extended this to the Answer application so that the channel may be answered but not CDRified if desired. I also modified app_dictate and app_waitforsilence to only answer the channel if it is not already up, to help not allow for faulty CDR answer times. All of these changes are going into Asterisk trunk. For 1.6.0 and 1.6.1, however, all the changes except for the change to the Answer application will go in since we do not introduce new features into stable branches (closes issue #14164) Reported by: DennisD Patches: 14164.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/145 ........ * apps/app_chanspy.c, /: Merged revisions 174805 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174805 | mmichelson | 2009-02-10 17:17:03 -0600 (Tue, 10 Feb 2009) | 11 lines Fix potential for stack overflows in app_chanspy.c When using the 'g' or 'e' options, the stack allocations that were used could cause a stack overflow if a spyer stayed on the line long enough without actually successfully spying on anyone. The problem has been corrected by using static buffers and copying the contents of the appropriate strings into them instead of using functions like alloca or ast_strdupa ........ * main/manager.c, /: Merged revisions 174764 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174764 | mmichelson | 2009-02-10 15:45:14 -0600 (Tue, 10 Feb 2009) | 21 lines Fix an fd leak that would occur in HTTP AMI sessions The explanation behind this fix is a bit complicated, and I've already typed it up in the code as a huge comment inside of manager.c, so I'll give the abridged version here. We needed a way to separate action-specific data from session-specific data. Unfortunately, the only way to maintain API compatibility and to not have to change every single manager action was to rename the current mansession structure and wrap it inside a new mansession structure which actually contains action- specific data. (closes issue #14364) Reported by: awk Patches: 14364_better.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/148/ ........ 2009-02-10 20:16 +0000 [r174711] Joshua Colp * /, channels/chan_sip.c: Merged revisions 174710 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174710 | file | 2009-02-10 16:15:43 -0400 (Tue, 10 Feb 2009) | 4 lines Only decrease inringing count if above zero. (issue #13238) Reported by: kowalma ........ 2009-02-10 18:19 +0000 [r174596] Matthew Nicholson * /, main/jitterbuf.c: Merged revisions 174584 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r174584 | mnicholson | 2009-02-10 12:16:31 -0600 (Tue, 10 Feb 2009) | 25 lines Merged revisions 174583 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174583 | mnicholson | 2009-02-10 11:52:42 -0600 (Tue, 10 Feb 2009) | 18 lines Improve behavior of jitterbuffer when maxjitterbuffer is set. This change improves the way the jitterbuffer handles maxjitterbuffer and dramatically reduces the number of frames dropped when maxjitterbuffer is exceeded. In the previous jitterbuffer, when maxjitterbuffer was exceeded, all new frames were dropped until the jitterbuffer is empty. This change modifies the code to only drop frames until maxjitterbuffer is no longer exceeded. Also, previously when maxjitterbuffer was exceeded, dropped frames were not tracked causing stats for dropped frames to be incorrect, this change also addresses that problem. (closes issue #14044) Patches: bug14044-1.diff uploaded by mnicholson (license 96) Tested by: mnicholson Review: http://reviewboard.digium.com/r/144/ ........ ................ 2009-02-10 15:39 +0000 [r174544] Joshua Colp * /, channels/chan_sip.c: Merged revisions 174543 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174543 | file | 2009-02-10 11:37:07 -0400 (Tue, 10 Feb 2009) | 6 lines Make the logic for inuse and inringing manipluation match that of 1.4. The old broken logic would reset the values back to 0 during certain scenarios causing the wrong state to be reported. (closes issue #14399) Reported by: caspy (issue #13238) Reported by: kowalma ........ 2009-02-10 05:06 +0000 [r174439] Steve Murphy * apps/app_rpt.c: For some strange reason, I didn't think 1.6.0 needed this fix. I was wrong. Here it is. 2009-02-09 17:28 +0000 [r174322-174328] Mark Michelson * /, channels/chan_sip.c: Merged revisions 174327 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174327 | mmichelson | 2009-02-09 11:27:32 -0600 (Mon, 09 Feb 2009) | 3 lines Fix something I messed up in the merge I just did ........ * /, channels/chan_sip.c: Merged revisions 174301 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r174301 | mmichelson | 2009-02-09 11:20:55 -0600 (Mon, 09 Feb 2009) | 20 lines Merged revisions 174282 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174282 | mmichelson | 2009-02-09 11:11:05 -0600 (Mon, 09 Feb 2009) | 12 lines Don't do an SRV lookup if a port is specified RFC 3263 says to do A record lookups on a hostname if a port has been specified, so that's what we're going to do. See section 4.2. (closes issue #14419) Reported by: klaus3000 Patches: patch_chan_sip_nosrvifport_1.4.23.txt uploaded by klaus3000 (license 65) ........ ................ 2009-02-09 14:50 +0000 [r174220] Joshua Colp * /, res/res_musiconhold.c: Merged revisions 174219 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r174219 | file | 2009-02-09 10:49:24 -0400 (Mon, 09 Feb 2009) | 11 lines Merged revisions 174218 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174218 | file | 2009-02-09 10:48:21 -0400 (Mon, 09 Feb 2009) | 4 lines Don't overwrite our pointer to the music class when music on hold stops. We will use this if it starts again to see if we can resume the music where it left off. (closes issue #14407) Reported by: mostyn ........ ................ 2009-02-07 16:17 +0000 [r174151] Russell Bryant * /, res/snmp/agent.c: Merged revisions 174149 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r174149 | russell | 2009-02-07 10:16:50 -0600 (Sat, 07 Feb 2009) | 10 lines Merged revisions 174148 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174148 | russell | 2009-02-07 10:15:07 -0600 (Sat, 07 Feb 2009) | 2 lines Fix a race condition that could cause a crash. ........ ................ 2009-02-06 23:59 +0000 [r174085] Dwayne M. Hubbard * /, channels/chan_sip.c: Merged revisions 174084 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r174084 | dhubbard | 2009-02-06 17:51:56 -0600 (Fri, 06 Feb 2009) | 13 lines Merged revisions 174082 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174082 | dhubbard | 2009-02-06 17:36:03 -0600 (Fri, 06 Feb 2009) | 5 lines check ast_strlen_zero() before calling ast_strdupa() in sip_uri_headers_cmp() and sip_uri_params_cmp() The reporter didn't actually upload a properly-formed patch, instead a modified chan_sip.c file was uploaded. I created a patch to determine the changes, then modified the suggested changes to create a proper fix. The summary above is a complete description of the changes. (closes issue #13547) Reported by: tecnoxarxa Patches: chan_sip.c.gz uploaded by tecnoxarxa (license 258) Tested by: tecnoxarxa ........ ................ ------------------------------------------------------------------------ 2009-02-06 19:29 +0000 [r173986-174042] Joshua Colp * channels/chan_dahdi.c, /: Merged revisions 174041 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174041 | file | 2009-02-06 15:28:53 -0400 (Fri, 06 Feb 2009) | 4 lines Don't subscribe to a mailbox on pseudo channels. It is futile. This solves an issue where duplicated pseudo channels would cause a crash because the first one would unsubscribe and the next one would also try to unsubscribe the same subscription. (closes issue #14322) Reported by: amessina ........ * /, channels/chan_sip.c: Merged revisions 173974 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173974 | file | 2009-02-06 13:18:35 -0400 (Fri, 06 Feb 2009) | 15 lines Merged revisions 173967-173968 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173967 | file | 2009-02-06 13:14:15 -0400 (Fri, 06 Feb 2009) | 4 lines Some clients do not put the call-id for replaces at the beginning, so support it being anywhere in the string. (closes issue #14350) Reported by: fhackenberger ........ r173968 | file | 2009-02-06 13:15:01 -0400 (Fri, 06 Feb 2009) | 2 lines Remove a debug message I put in by accident. ........ ................ 2009-02-06 16:33 +0000 [r173963] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 173952 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173952 | mnicholson | 2009-02-06 10:28:19 -0600 (Fri, 06 Feb 2009) | 14 lines Merged revisions 173917 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173917 | mnicholson | 2009-02-06 10:20:23 -0600 (Fri, 06 Feb 2009) | 7 lines Limit the addition of the Contact header in SIP responses according to various SIP RFCs. (closes issue #13602) Reported by: hjourdain Tested by: mnicholson ........ ................ 2009-02-05 23:51 +0000 [r173774-173777] Mark Michelson * configs/extensions.conf.sample, /: Merged revisions 173776 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r173776 | mmichelson | 2009-02-05 17:48:48 -0600 (Thu, 05 Feb 2009) | 14 lines Update extensions.conf.sample to be correct. In trunk, the only necessary change pointed out was that the call to ChanIsAvail uses an option that has been removed. For the 1.6.1 branch, however, it appears that the sample file is badly in need of updating since there are |'s used all over the place there. My tentative plan is just to copy trunk's sample config file to those branches since the info there is most up-to-date and should be correct for use in 1.6.1 Thanks to macli in #asterisk-dev for bringing this up ........ * apps/app_voicemail.c, /: Merged revisions 173773 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r173773 | mmichelson | 2009-02-05 17:28:19 -0600 (Thu, 05 Feb 2009) | 7 lines Properly set "seen" and "unseen" flags when moving messages from the new to the old folder when using IMAP for voicemail storage (closes issue #13905) Reported by: jaroth Patches: foldermove_v2.patch uploaded by jaroth (license 50) ........ 2009-02-05 21:04 +0000 [r173698] Jeff Peeler * apps/app_voicemail.c, /: Merged revisions 173697 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173697 | jpeeler | 2009-02-05 15:00:26 -0600 (Thu, 05 Feb 2009) | 18 lines Merged revisions 173696 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173696 | jpeeler | 2009-02-05 14:47:51 -0600 (Thu, 05 Feb 2009) | 12 lines Add new configuration option to make shared IMAP mailboxes function as expected. The new option is "imapvmshareid" which is an ID to tag multiple mailboxes using the same IMAP storage location to function as one mailbox. This allows all messages to be retrieved for any user in the group. The patch alters the 'X-Asterisk-VM-Extension' header that is responsible for matching voicemails for a given user. (closes issue #13673) Reported by: howardwilkinson ........ ................ 2009-02-05 20:34 +0000 [r173590-173694] Mark Michelson * /, apps/app_queue.c: Merged revisions 173693 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173693 | mmichelson | 2009-02-05 14:30:45 -0600 (Thu, 05 Feb 2009) | 20 lines Merged revisions 173692 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173692 | mmichelson | 2009-02-05 14:29:09 -0600 (Thu, 05 Feb 2009) | 12 lines Fix situations where queue members could be autopaused unexpectedly Specifically, this patch prevents us from autopausing members when we receive a busy or congestion frame from them. (closes issue #14376) Reported by: fiddur Patches: 14376.patch uploaded by putnopvut (license 60) Tested by: fiddur ........ ................ * apps/app_mixmonitor.c, /: Merged revisions 173593 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173593 | mmichelson | 2009-02-05 12:48:55 -0600 (Thu, 05 Feb 2009) | 11 lines Merged revisions 173592 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173592 | mmichelson | 2009-02-05 12:47:24 -0600 (Thu, 05 Feb 2009) | 3 lines Add some missing cleanup to app_mixmonitor ........ ................ * apps/app_mixmonitor.c, /: Merged revisions 173589 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173589 | mmichelson | 2009-02-05 12:34:06 -0600 (Thu, 05 Feb 2009) | 33 lines Merged revisions 173559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173559 | mmichelson | 2009-02-05 11:34:33 -0600 (Thu, 05 Feb 2009) | 25 lines Fix a problem where a channel pointer becomes invalid due to masquerading or hanging up. app_mixmonitor runs its own thread to monitor the channel's activity and write the mixed audio to a file. Since this thread runs independently of the channel, it is possible that the mixmonitor thread's channel pointer will point to freed memory when the channel either is masqueraded or hangs up (technically, both cases are hangups, but we need to handle the cases slightly differently). The solution for this is to employ a datastore, which has the nice benefit of allowing us to hook into channel masquerades and hangups and update our pointer as necessary. If this looks familiar, this same technique is employed in app_chanspy. app_chanspy is a bit more involved since it does a lot more operations on the channel that is being spied upon. app_mixmonitor does have an extra touch that app_chanspy doesn't have, though. Since there is a thread race between the channel's thread and the mixmonitor thread on a hangup, we em- ploy a condition-and-boolean combination to ensure that the channel thread finishes with our structure before the mixmonitor thread attempts to free it. No crashes! (closes issue #14374) Reported by: aragon Patches: 14374.patch uploaded by putnopvut (license 60) Tested by: aragon, putnopvut ........ ................ 2009-02-05 16:23 +0000 [r173554] Jeff Peeler * build_tools/menuselect-deps.in: fix WORKING_FORK detection 2009-02-05 00:11 +0000 [r173548] Tilghman Lesher * build_tools/menuselect-deps.in: regenerate with bootstrap.sh 2009-02-04 23:44 +0000 [r173546-173547] Jeff Peeler * /: I messed up and accidentally reverted the trunk-merged prop before committing 173546. Added it manually. * main/features.c: Merged revisions 173500 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173500 | jpeeler | 2009-02-04 15:17:53 -0600 (Wed, 04 Feb 2009) | 23 lines Merged revisions 173211 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173211 | jpeeler | 2009-02-03 15:57:01 -0600 (Tue, 03 Feb 2009) | 17 lines Parking attempts made to one end of a bridge no longer will hang up due to a parking failure. Parking attempts made using either one-touch, or doing either a blind or assisted transfer to the parking extension now keep up the bridge instead of hanging up the attempted parked party. Normal causes for the parking attempt to fail includes the specific specified extension (via PARKINGEXTEN) not being available or if all the parking spaces are currently in use. To avoid having to reverse a masquerade park_space_reserve was made to provide foresight if a parking attempt will succeed and if so reserve the parking space. (closes issue #13494) Reported by: mdu113 Reviewed by Russell: http://reviewboard.digium.com/r/133/ ........ ................ 2009-02-04 22:23 +0000 [r173534] Mark Michelson * /, apps/app_queue.c: Merged revisions 173507 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r173507 | mmichelson | 2009-02-04 16:16:19 -0600 (Wed, 04 Feb 2009) | 7 lines Fix some areas where the incorrect interface was passed to ast_device_state I swear it feels like I already did this once... (closes issue #14359) Reported by: francesco_r ........ 2009-02-04 18:55 +0000 [r173460] Tilghman Lesher * main/tcptls.c, /: Merged revisions 173458 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r173458 | tilghman | 2009-02-04 12:48:06 -0600 (Wed, 04 Feb 2009) | 9 lines When using a socket as a FILE *, the stdio functions will sometimes try to do an fseek() on the stream, which is an invalid operation for a socket. Turning off buffering explicitly lets the stdio functions know they cannot do this, thus avoiding a potential error. (closes issue #14400) Reported by: fnordian Patches: tcptls.patch uploaded by fnordian (license 110) ........ 2009-02-04 17:46 +0000 [r173355-173398] Mark Michelson * apps/app_chanspy.c, /: Merged revisions 173397 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173397 | mmichelson | 2009-02-04 11:45:14 -0600 (Wed, 04 Feb 2009) | 11 lines Merged revisions 173396 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173396 | mmichelson | 2009-02-04 11:44:48 -0600 (Wed, 04 Feb 2009) | 3 lines Revert my previous change because it was stupid ........ ................ * apps/app_chanspy.c, /: Merged revisions 173393 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173393 | mmichelson | 2009-02-04 11:41:02 -0600 (Wed, 04 Feb 2009) | 11 lines Merged revisions 173392 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173392 | mmichelson | 2009-02-04 11:40:29 -0600 (Wed, 04 Feb 2009) | 3 lines Add a missing unlock. Extremely unlikely to ever matter, but it's needed. ........ ................ * /, main/file.c: Merged revisions 173354 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r173354 | mmichelson | 2009-02-04 09:30:12 -0600 (Wed, 04 Feb 2009) | 30 lines Fix a problem where file playback would cause fds to remain open forever The problem came from the fact that a frame read from a format interpreter was not freed. Adding a call to ast_frfree fixed this. The explanation for why this caused the problem is a bit complex, but here goes: There was a problem in all versions of Asterisk where the embedded frame of a filestream structure was referenced after the filestream was freed. This was fixed by adding reference counting to the filestream structure. The refcount would increase every time that a filestream's frame pointer was pointing to an actual frame of data. When the frame was freed, the refcount would decrease. Once the refcount reached 0, the filestream was freed, and as part of the operation, the open files were closed as well. Thus it becomes more clear why a missing ast_frfree would cause a reference leak and cause the files to not be closed. You may ask then if there was a frame leak before this patch. The answer to that is actually no! The filestream code was "smart" enough to know that since the frame we received came from a format interpreter, the frame had no malloced data and thus didn't need to be freed. Now, however, there is cleanup that needs to be done when we finish with the frame, so we do need to call ast_frfree on the frame to be sure that the refcount for the filestream is decremented appropriately. (closes issue #14384) Reported by: fiddur Patches: 14384.patch uploaded by putnopvut (license 60) Tested by: fiddur, putnopvut ........ 2009-02-04 00:45 +0000 [r173312] Tilghman Lesher * main/pbx.c, /: Merged revisions 173311 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r173311 | tilghman | 2009-02-03 18:43:52 -0600 (Tue, 03 Feb 2009) | 10 lines Ensure that commas placed in the middle of extension character classes do not interfere with correct parsing of the extension. Also, if an unterminated character class DOES make its way into the pbx core (through some other method), ensure that it does not crash Asterisk. (closes issue #14362) Reported by: Nick_Lewis Patches: 20090129__bug14362.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ 2009-02-03 23:41 +0000 [r173250] David Vossel * channels/chan_iax2.c: Fixes issue with IAX2 transfer not handing of calls. Fixes issue with IAX2 transfers not taking place. As it was, a call that was being transfered would never be handed off correctly to the call ends because of how call numbers were stored in a hash table. The hash table, "iax_peercallno_pvt", storing all the current call numbers did not take into account the complications associated with transferring a call, so a separate hash table was required. This second hash table "iax_transfercallno_pvt" handles calls being transfered, once the call transfer is complete the call is removed from the transfer hash table and added to the peer hash table resuming normal operations. Addition functions were created to handle storing, removing, and comparing items in the iax_transfercallno_pvt table. (issue #13468) Review: http://reviewboard.digium.com/r/140/ 2009-02-03 00:26 +0000 [r173111] Tilghman Lesher * configs/extensions.conf.sample, /: Merged revisions 173104 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173104 | tilghman | 2009-02-02 18:24:52 -0600 (Mon, 02 Feb 2009) | 12 lines Merged revisions 173070 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173070 | tilghman | 2009-02-02 18:15:59 -0600 (Mon, 02 Feb 2009) | 5 lines Add warning to standard config, that globals may be overridden by other dialplan configuration files. (closes issue #14388) Reported by: macli ........ ................ 2009-02-02 23:59 +0000 [r173068] Terry Wilson * /, main/features.c: Merged revisions 173067 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173067 | twilson | 2009-02-02 17:57:25 -0600 (Mon, 02 Feb 2009) | 9 lines Merged revisions 173066 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173066 | twilson | 2009-02-02 17:48:06 -0600 (Mon, 02 Feb 2009) | 2 lines Fix a feature inheritance bug I added after code review ........ ................ 2009-02-02 18:15 +0000 [r172896] Leif Madsen * /, configs/res_ldap.conf.sample: Merged revisions 172894 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r172894 | lmadsen | 2009-02-02 13:13:40 -0500 (Mon, 02 Feb 2009) | 7 lines Update the res_ldap.conf file with a better working example. (closes issue #13861) Reported by: scramatte Patches: __20080110-res_ldap.conf-2.patch uploaded by blitzrage (license 10) Tested by: jcovert ........ 2009-02-01 02:45 +0000 [r172707-172742] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 172741 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r172741 | tilghman | 2009-01-31 20:44:23 -0600 (Sat, 31 Jan 2009) | 4 lines Blank argument crashes Asterisk (closes issue #14377) Reported by: amorsen ........ * /, funcs/func_strings.c: Merged revisions 172706 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r172706 | tilghman | 2009-01-31 10:40:59 -0600 (Sat, 31 Jan 2009) | 7 lines Don't increment the loop, now that incrementing is taken care of by the decoder function. (closes issue #14363) Reported by: andrew53 Patches: func_strings_filter.patch uploaded by andrew53 (license 519) ........ 2009-01-31 00:06 +0000 [r172635-172637] Terry Wilson * configs/features.conf.sample, /: Merged revisions 172581 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r172581 | twilson | 2009-01-30 15:50:03 -0600 (Fri, 30 Jan 2009) | 2 lines Remove incorret line from sample config ........ * configs/features.conf.sample, apps/app_dial.c, main/global_datastores.c, /, main/features.c, include/asterisk/global_datastores.h, CHANGES: Merged revisions 172580 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r172580 | twilson | 2009-01-30 15:29:12 -0600 (Fri, 30 Jan 2009) | 44 lines Merged revisions 172517 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009) | 37 lines Fix feature inheritance with builtin features When using builtin features like parking and transfers, the AST_FEATURE_* flags would not be set correctly for all instances when either performing a builtin attended transfer, or parking a call and getting the timeout callback. Also, there was no way on a per-call basis to specify what features someone should have on picking up a parked call (since that doesn't involve the Dial() command). There was a global option for setting whether or not all users who pickup a parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT, AUTOMON, or PARKCALL. This patch: 1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the dialplan or with setvar in channels that support it. This variable can be set to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the equivalent dial options), to set what features should be activated on this channel. The patch moves the setting of the features datastores into the bridging code instead of app_dial to help facilitate this. 2) adds global options parkedcallparking, parkedcallhangup, and parkedcallrecording to be similar to the parkedcalltransfers option for globally setting features. 3) has builtin_atxfer call builtin_parkcall if being transfered to the parking extension since tracking everything through multiple masquerades, etc. is difficult and error-prone 4) attempts to fix all cases of return calls from parking and completed builtin transfers not having the correct permissions (closes issue #14274) Reported by: aragon Patches: fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396) Tested by: aragon, otherwiseguy Review http://reviewboard.digium.com/r/138/ ........ ................ 2009-01-30 22:23 +0000 [r172604] Mark Michelson * /, include/asterisk/channel.h: Merged revisions 172598 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r172598 | mmichelson | 2009-01-30 16:22:04 -0600 (Fri, 30 Jan 2009) | 3 lines Fix redefinition of flag in channel.h ........ 2009-01-29 23:47 +0000 [r172503] Tilghman Lesher * main/asterisk.c, apps/app_nbscat.c, /, autoconf/ast_func_fork.m4, apps/app_festival.c, build_tools/menuselect-deps.in, configure, apps/app_dahdiras.c, apps/app_mp3.c, res/res_agi.c, apps/app_externalivr.c, apps/app_ices.c, res/res_musiconhold.c: Merged revisions 172441 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r172441 | tilghman | 2009-01-29 17:15:40 -0600 (Thu, 29 Jan 2009) | 16 lines Merged revisions 172438 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172438 | tilghman | 2009-01-29 16:54:29 -0600 (Thu, 29 Jan 2009) | 9 lines Lose the CAP_NET_ADMIN at every fork, instead of at startup. Otherwise, if Asterisk runs as a non-root user and the administrator does a 'restart now', Asterisk loses the ability to set QOS on packets. (closes issue #14004) Reported by: nemo Patches: 20090105__bug14004.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ ................ 2009-01-29 21:35 +0000 [r172434] Richard Mudgett * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged revisions 172400 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r172400 | rmudgett | 2009-01-29 14:38:34 -0600 (Thu, 29 Jan 2009) | 12 lines channels/chan_dahdi.c * Added doxygen comments to the major dahdi structures. * Fixed PRI and SS7 using an incorrect string value if the extension delimiter is not present in the Dial() function. * Fixed SS7 not checking if the dialed extension is at least as long as the stripmsd option. * Fixed PRI not handling unknown TON/NPI prefix letters correctly. * Fixed some uninitialized string variables on FXS ports. configs/chan_dahdi.conf.sample * Updated some documentation. ........ 2009-01-29 16:49 +0000 [r172316] Tilghman Lesher * configs/func_odbc.conf.sample, /: Merged revisions 172315 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r172315 | tilghman | 2009-01-29 10:48:25 -0600 (Thu, 29 Jan 2009) | 2 lines Better document mode=multirow, based upon a conversation with Jared. ........ 2009-01-29 13:51 +0000 [r172273] Leif Madsen * contrib/scripts/realtime_pgsql.sql, /: Merged revisions 172271 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r172271 | lmadsen | 2009-01-29 08:47:27 -0500 (Thu, 29 Jan 2009) | 5 lines The realtime_pgsql.sql script is missing a couple of fields. closes issue #14339) Reported by: fiddur Patches: realtime_pgsql.sql.diff uploaded by fiddur (license 678) ........ 2009-01-29 09:56 +0000 [r172217] Olle Johansson * /, channels/chan_sip.c: Merged revisions 172173 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r172173 | oej | 2009-01-29 10:18:01 +0100 (Tor, 29 Jan 2009) | 24 lines Merged revisions 172169 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172169 | oej | 2009-01-29 09:48:18 +0100 (Tor, 29 Jan 2009) | 16 lines Make sure that we always add the hangupcause headers. In some cases, the owner was disconnected before we checked for the cause. This patch implements a temporary storage in the pvt and use that instead. The code is based on ideas from code from Adomjan in issue #13385 (Add support for Reason: header) Thanks to Klaus Darillion for testing! (closes issue #14294) related to issue #13385 Reported by: klaus3000 and adomjan Patches: bug14294b.diff uploaded by oej (license 306) Based on 20080829_chan_sip.c-q850reason_header.patch uploaded by adomjan (license 487) Tested by: oej, klaus3000 ........ ................ 2009-01-28 20:41 +0000 [r172065] Steve Murphy * apps/app_channelredirect.c, main/pbx.c, main/manager.c, /, main/features.c, include/asterisk/channel.h: Merged revisions 172063 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r172063 | murf | 2009-01-28 13:31:06 -0700 (Wed, 28 Jan 2009) | 52 lines Merged revisions 172030 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172030 | murf | 2009-01-28 11:51:16 -0700 (Wed, 28 Jan 2009) | 46 lines This patch fixes h-exten running misbehavior in manager-redirected situations. What it does: 1. A new Flag value is defined in include/asterisk/channel.h, AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the bridge hangup exten code not to run the h-exten there (nor publish the bridge cdr there). It will done at the pbx-loop level instead. 2. In the manager Redirect code, I set this flag on the channel if the channel has a non-null pbx pointer. I did the same for the second (chan2) channel, which gets run if name2 is set... and the first succeeds. 3. I restored the ending of the cdr for the pbx loop h-exten running code. Don't know why it was removed in the first place. 4. The first attempt at the fix for this bug was to place code directly in the async_goto routine, which was called from a large number of places, and could affect a large number of cases, so I tested that fix against a fair number of transfer scenarios, both with and without the patch. In the process, I saw that putting the fix in async_goto seemed not to affect any of the blind or attended scenarios, but still, I was was highly concerned that some other scenarios I had not tested might be negatively impacted, so I refined the patch to its current scope, and jmls tested both. In the process, tho, I saw that blind xfers in one situation, when the one-touch blind-xfer feature is used by the peer, we got strange h-exten behavior. So, I inserted code to swap CDRs and to set the HANGUP_DONT field, to get uniform behavior. 5. I added code to the bridge to obey the HANGUP_DONT flag, skipping both publishing the bridge CDR, and running the h-exten; they will be done at the pbx-loop (higher) level instead. 6. I removed all the debug logs from the patch before committing. 7. I moved the AUTOLOOP set/reset in the h-exten code in res_features so it's only done if the h-exten is going to be run. A very minor performance improvement, but technically correct. (closes issue #14241) Reported by: jmls Patches: 14241_redirect_no_bridgeCDR_or_h_exten_via_transfer uploaded by murf (license 17) Tested by: murf, jmls ........ ................ 2009-01-28 17:28 +0000 [r171965] Tilghman Lesher * channels/chan_dahdi.c, /: Merged revisions 171964 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r171964 | tilghman | 2009-01-28 11:27:40 -0600 (Wed, 28 Jan 2009) | 9 lines Merged revisions 171963 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171963 | tilghman | 2009-01-28 11:25:18 -0600 (Wed, 28 Jan 2009) | 2 lines Clarify log message (suggested by manxpower on #asterisk-dev) ........ ................ 2009-01-28 13:18 +0000 [r171846] Olle Johansson * /, configs/sip.conf.sample: Merged revisions 171838 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r171838 | oej | 2009-01-28 14:11:44 +0100 (Ons, 28 Jan 2009) | 10 lines Merged revisions 171837 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171837 | oej | 2009-01-28 14:07:27 +0100 (Ons, 28 Jan 2009) | 2 lines Add a better explanation of the difference between the device namespace and the dialplan for newbies. ........ ................ 2009-01-27 22:00 +0000 [r171619-171692] Mark Michelson * /, channels/chan_agent.c: Merged revisions 171691 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r171691 | mmichelson | 2009-01-27 15:58:39 -0600 (Tue, 27 Jan 2009) | 47 lines Merged revisions 171689 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171689 | mmichelson | 2009-01-27 15:55:08 -0600 (Tue, 27 Jan 2009) | 39 lines Fix devicestate problems for "always-on" agent channels A revision to chan_agent attempted to "inherit" the device state of the underlying channel in order to report the device state of an agent channel more accurately. The problem with the logic here is that it makes no sense to use this for always-on agents. If the agent is logged in, then to the underlying channel, the agent will always appear to be "in use," no matter if the agent is on a call or not. The reason is that to the underlying channel, the channel is currently in use on a call to the AgentLogin application. The most common cause that I found for this issue to occur was for a SIP channel to be the underlying channel type for an Agent channel. If the SIP phone re-registers, then the registration will cause the device state core to query the device state of the SIP channel. Since the SIP channel is in use, the Agent channel would also inherit this status. Once the agent channel was set to "in use" there was no way that the device state could change on that channel unless the agent logged out. The solution for this problem is a bit different in 1.4 than it is in the other branches. In 1.4, there will be a one-line fix to make sure that only callback agents will inherit device state from their underlying channel type. For the other branches of Asterisk, since callback support has been removed, there is also no need for device state inheritance in chan_agent, so I will simply be removing it from the code. In addition, the 1.4 source is getting a new comment to help the next person who edits chan_agent.c. I'm adding a comment that a agent_pvt's loginchan field may be used to determine if the agent is a callback agent or not. (closes issue #14173) Reported by: nathan Patches: 14173.patch uploaded by putnopvut (license 60) Tested by: nathan, aramirez ........ ................ * /, main/slinfactory.c: Merged revisions 171622 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r171622 | mmichelson | 2009-01-27 14:11:30 -0600 (Tue, 27 Jan 2009) | 26 lines Merged revisions 171621 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171621 | mmichelson | 2009-01-27 14:06:01 -0600 (Tue, 27 Jan 2009) | 18 lines Prevent a crash from occurring when a jitter buffer interpolated frame is removed from a slinfactory slinfactory used the "samples" field of an ast_frame in order to determine the amount of data contained within the frame. In certain cases, such as jitter buffer interpolated frames, the frame would have a non-zero value for "samples" but have NULL "data" This caused a problem when a memcpy call in ast_slinfactory_read would attempt to access invalid memory. The solution in use here is to never feed frames into the slinfactory if they have NULL "data" (closes issue #13116) Reported by: aragon Patches: 13116.diff uploaded by putnopvut (license 60) ........ ................ * /, apps/app_queue.c: Merged revisions 171618 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r171618 | mmichelson | 2009-01-27 13:30:54 -0600 (Tue, 27 Jan 2009) | 24 lines Fix queue crashes that would occur after the calling channel was masqueraded. The data passed to the end_bridge_callback was assumed to be data which was still stack'd. The problem was that with some call features, attended transfers in particular, a new bridge thread is started once the feature completes, meaning that when the end_bridge_callback is called, the end_bridge_callback_data was invalid. To fix this problem, there are two measures taken 1. Instead of pointing to stacked data, we now used heap-allocated data for passing to the end_bridge_callback in app_queue 2. Since bridges can end multiple times on a single logical call, we wait until the final bridge is broken to actually set any queue variables. This is accomplished through reference-counting and the use of an end_bridge_callback_data_fixup function in app_queue.c (closes issue #14260) Reported by: ccesario Patches: 14260.patch uploaded by putnopvut (license 60) Tested by: ccesario ........ 2009-01-27 16:15 +0000 [r171594-171595] Matthew Fredrickson * main/ast_expr2.c, main/ast_expr2.h: Revert some changes that shouldn't have made it in * main/ast_expr2.c, channels/chan_dahdi.c, main/ast_expr2.h: Make sure we do not go into alarm on PTMP links with non persistent D-channels 2009-01-27 15:13 +0000 [r171529] Olle Johansson * /, channels/chan_sip.c: Merged revisions 171528 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r171528 | oej | 2009-01-27 16:00:19 +0100 (Tis, 27 Jan 2009) | 23 lines Solving the same issue, but a bit different in trunk... Merged revisions 171527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171527 | oej | 2009-01-27 15:33:20 +0100 (Tis, 27 Jan 2009) | 13 lines Use the same branch tag in CANCEL as in INVITE Originally putnopvut implemented some changes in revision 142079 that according to the bug report seemed to have worked then, but somehow fails now. I guess code, as humans, get old and forget stuff. Anyway, this bug caused CANCEL not to work with picky systems. Thanks Fredrik for pointing out where the bug in the SIP messaging was. (closes issue #14346) Reported by: oej Patches: bug14346.diff uploaded by oej (license 306) Tested by: oej ........ ................ 2009-01-26 14:02 +0000 [r171327] Olle Johansson * /, channels/chan_sip.c: Merged revisions 171326 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r171326 | oej | 2009-01-26 14:44:40 +0100 (MÃ¥n, 26 Jan 2009) | 17 lines Merged revisions 171264 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171264 | oej | 2009-01-26 13:51:53 +0100 (MÃ¥n, 26 Jan 2009) | 9 lines Don't retransmit 401 on REGISTER requests when alwaysauthreject=yes (closes issue #14284) Reported by: klaus3000 Patches: patch_chan_sip_unreliable_1.4.23_14284.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 ........ ................ 2009-01-26 00:03 +0000 [r171189] Tilghman Lesher * channels/chan_oss.c, /: Merged revisions 171188 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r171188 | tilghman | 2009-01-25 17:58:00 -0600 (Sun, 25 Jan 2009) | 13 lines Merged revisions 171187 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171187 | tilghman | 2009-01-25 17:44:01 -0600 (Sun, 25 Jan 2009) | 6 lines Correctly track the hookstate (closes issue #13686) Reported by: itiliti Patches: 20081013__bug13686.diff.txt uploaded by Corydon76 (license 14) ........ ................ 2009-01-25 13:38 +0000 [r170981] Sean Bright * /, apps/app_page.c: Merged revisions 170980 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r170980 | seanbright | 2009-01-25 08:35:48 -0500 (Sun, 25 Jan 2009) | 16 lines Merged revisions 170979 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170979 | seanbright | 2009-01-25 08:33:20 -0500 (Sun, 25 Jan 2009) | 9 lines Resolve a logic error that was causing Page() to crash when more than one channel was specified. (closes issue #14308) Reported by: bluefox Patches: 20090124__bug14308.diff.txt uploaded by seanbright (license 71) Tested by: kc0bvu ........ ................ 2009-01-25 02:50 +0000 [r170944] Russell Bryant * include/asterisk/utils.h, /: Merged revisions 170943 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r170943 | russell | 2009-01-24 20:49:30 -0600 (Sat, 24 Jan 2009) | 6 lines Change ARRAY_LEN() to be more C++ safe. When the second part of this macro is written as 0[a] instead of a[0], it will force a failure if the macro is used on a C++ object that overloads the [] operator. ........ 2009-01-24 13:56 +0000 [r170838] Tilghman Lesher * configs/res_odbc.conf.sample, /: Merged revisions 170837 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r170837 | tilghman | 2009-01-24 07:55:53 -0600 (Sat, 24 Jan 2009) | 9 lines Merged revisions 170836 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170836 | tilghman | 2009-01-24 07:55:02 -0600 (Sat, 24 Jan 2009) | 2 lines Remove superfluous implementation note (closes issue #14319) ........ ................ 2009-01-23 23:52 +0000 [r170830] Richard Mudgett * /, doc/tex/Makefile: Merged revisions 170794 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r170794 | rmudgett | 2009-01-23 17:10:34 -0600 (Fri, 23 Jan 2009) | 1 line Fix asterisk.pdf generation if branch name has an underscore in it. ........ 2009-01-23 22:59 +0000 [r170791] Russell Bryant * /, doc/tex/Makefile: Merged revisions 170790 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r170790 | russell | 2009-01-23 16:58:37 -0600 (Fri, 23 Jan 2009) | 2 lines Don't blow up if a branch name has an underscore in it ........ 2009-01-23 20:56 +0000 [r170685-170721] Mark Michelson * configs/res_odbc.conf.sample, /: Merged revisions 170720 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r170720 | mmichelson | 2009-01-23 14:56:07 -0600 (Fri, 23 Jan 2009) | 16 lines Merged revisions 170719 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170719 | mmichelson | 2009-01-23 14:55:26 -0600 (Fri, 23 Jan 2009) | 8 lines Add notes to the idlecheck explanation in res_odbc.conf.sample (closes issue #14319) Reported by: klaus3000 Patches: patch_idlecheck_res_odbc.conf.sample.txt uploaded by klaus3000 (license 65) ........ ................ * contrib/i18n.testsuite.conf, /: Merged revisions 170677 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r170677 | mmichelson | 2009-01-23 14:23:00 -0600 (Fri, 23 Jan 2009) | 22 lines Merged revisions 170671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170671 | mmichelson | 2009-01-23 14:21:51 -0600 (Fri, 23 Jan 2009) | 14 lines Update contrib/i18n.testsuite.conf to not use deprecated syntax * Convert Wait,1 to Wait(1) * Convert SetLanguage to Set(CHANNEL(language)) * Use 'n' for all priorities beyond the first Also added test for Chinese numbers, too. (closes issue #14320) Reported by: dant Patches: i18n.testsuite.conf.issue14320.v2.diff uploaded by dant (license 670) ........ ................ 2009-01-23 20:19 +0000 [r170659] Joshua Colp * main/channel.c, /: Merged revisions 170652 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r170652 | file | 2009-01-23 16:18:05 -0400 (Fri, 23 Jan 2009) | 11 lines Merged revisions 170648 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170648 | file | 2009-01-23 16:16:39 -0400 (Fri, 23 Jan 2009) | 4 lines When a channel is answered make sure any indications currently playing stop. Usually the phone would do this but if the channel was already answered then they are being generated by Asterisk and we darn well need to stop them. (closes issue #14249) Reported by: RadicAlish ........ ................ 2009-01-23 Tilghman Lesher * Released 1.6.0.5 * channels/chan_iax2.c: Regression fixes for security fix AST-2009-001 2009-01-06 Tilghman Lesher * Released 1.6.0.3 * channels/chan_iax2.c: Security fix AST-2009-001 2008-12-03 Tilghman Lesher * Released 1.6.0.3-rc1 2008-12-03 14:13 +0000 [r160482] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 160481 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r160481 | tilghman | 2008-12-03 08:11:53 -0600 (Wed, 03 Dec 2008) | 14 lines Merged revisions 160480 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160480 | tilghman | 2008-12-03 08:09:35 -0600 (Wed, 03 Dec 2008) | 7 lines Jon Bonilla (Manwe) pointed out on the -dev list: "I guess that having only ip-phones in mind is not a good approach. Since it is possible to have a sip proxy connected to asterisk we could receive a 407 (unauthorized) or 483 (too many hops) as response and dialog ending would not be a good behavior." So modified. ........ ................ 2008-12-03 00:53 +0000 [r160427] Sean Bright * Makefile: Fix some 'make menuselect' breakage introduced by recent merges. 2008-12-02 23:22 +0000 [r160386-160393] Tilghman Lesher * apps/app_dial.c, /: Merged revisions 156388 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r156388 | tilghman | 2008-11-12 15:34:51 -0600 (Wed, 12 Nov 2008) | 12 lines Merged revisions 156386 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156386 | tilghman | 2008-11-12 15:18:57 -0600 (Wed, 12 Nov 2008) | 5 lines When using call limits under 1 second, infinite call lengths are allowed, instead. (closes issue #13851) Reported by: ruddy ........ ................ * /, apps/app_meetme.c: Merged revisions 156290 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r156290 | jpeeler | 2008-11-12 13:11:15 -0600 (Wed, 12 Nov 2008) | 11 lines Merged revisions 156289 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156289 | jpeeler | 2008-11-12 13:10:12 -0600 (Wed, 12 Nov 2008) | 3 lines For whatever reason, gcc only warned me about the possible use of an uninitialized variable when compiling 1.6.1. ........ ................ * /, apps/app_meetme.c: Merged revisions 156228 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r156228 | jpeeler | 2008-11-12 12:32:46 -0600 (Wed, 12 Nov 2008) | 16 lines Merged revisions 156178 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156178 | jpeeler | 2008-11-12 11:53:44 -0600 (Wed, 12 Nov 2008) | 8 lines (closes issue #13173) Reported by: pep This change adds an announce_thread responsible for playing announcements to an existing conference. This allows all announcing to be immediately stopped if necessary but more importantly allows other threads that need to play something to not block. There are multiple benefits to this, but the actual bug is for solving the scenario for a channel to be unusable after hang up for the entire duration of the parting announcement. The parting announcement can be extremely long depending on what the user recorded upon joining the conference. Reviewed by Russell on Review Board: http://reviewboard.digium.com/r/25/ ........ ................ * main/astobj2.c, main/asterisk.c, apps/app_while.c, apps/app_dial.c, main/pbx.c, channels/chan_misdn.c, main/manager.c, /, apps/app_meetme.c, channels/chan_sip.c, channels/chan_skinny.c, include/asterisk/astobj2.h, channels/chan_agent.c, channels/chan_h323.c, channels/chan_iax2.c: Merged revisions 152969,153122,154264,154268,154366,155399,155863,156166,156295,156690,156756,158066,158082,158540,158602,159276 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r152969 | tilghman | 2008-10-30 15:35:46 -0500 (Thu, 30 Oct 2008) | 10 lines Merged revisions 152958 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152958 | tilghman | 2008-10-30 15:33:28 -0500 (Thu, 30 Oct 2008) | 3 lines Cannot join detached threads. See http://www.opengroup.org/onlinepubs/000095399/functions/pthread_join.html (Closes issue #13400) ........ ................ r153122 | tilghman | 2008-10-31 11:35:21 -0500 (Fri, 31 Oct 2008) | 10 lines Merged revisions 153114 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r153114 | tilghman | 2008-10-31 11:30:32 -0500 (Fri, 31 Oct 2008) | 3 lines Turn off qualify on uncached realtime peers. (Closes issue #13383) ........ ................ r154264 | tilghman | 2008-11-04 12:59:48 -0600 (Tue, 04 Nov 2008) | 10 lines Recorded merge of revisions 154263 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154263 | tilghman | 2008-11-04 12:58:05 -0600 (Tue, 04 Nov 2008) | 3 lines Make the monitor thread non-detached, so it can be joined (suggested by Russell on -dev list). ........ ................ r154268 | rmudgett | 2008-11-04 13:07:26 -0600 (Tue, 04 Nov 2008) | 11 lines Merged revisions 154266 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154266 | rmudgett | 2008-11-04 13:01:08 -0600 (Tue, 04 Nov 2008) | 4 lines JIRA ABE-1703 mISDN sets the channel to the wrong state when it receives the indication AST_CONTROL_RINGING. ........ ................ r154366 | tilghman | 2008-11-04 14:51:18 -0600 (Tue, 04 Nov 2008) | 16 lines Merged revisions 154365 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154365 | tilghman | 2008-11-04 14:49:33 -0600 (Tue, 04 Nov 2008) | 9 lines On busy systems, it's possible for the values checked within a single line of code to change, unless the structure is locked to ensure a consistent state. (closes issue #13717) Reported by: kowalma Patches: 20081102__bug13717.diff.txt uploaded by Corydon76 (license 14) Tested by: kowalma ........ ................ r155399 | tilghman | 2008-11-07 16:28:58 -0600 (Fri, 07 Nov 2008) | 14 lines Merged revisions 155398 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r155398 | tilghman | 2008-11-07 16:27:32 -0600 (Fri, 07 Nov 2008) | 7 lines Clarify error message. (closes issue #13809) Reported by: denke Patches: 20081104__bug13809.diff.txt uploaded by Corydon76 (license 14) Tested by: denke ........ ................ r155863 | mmichelson | 2008-11-10 15:14:44 -0600 (Mon, 10 Nov 2008) | 22 lines Merged revisions 155861 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r155861 | mmichelson | 2008-11-10 15:07:39 -0600 (Mon, 10 Nov 2008) | 14 lines Channel drivers assume that when their indicate callback is invoked, that the channel on which the callback was called is locked. This patch corrects an instance in chan_agent where a channel's indicate callback is called directly without first locking the channel. This was leading to some observed locking issues in chan_local, but considering that all channel drivers operate under the same expectations, the generic fix in chan_agent is the right way to go. AST-126 ........ ................ r156166 | russell | 2008-11-12 11:38:20 -0600 (Wed, 12 Nov 2008) | 15 lines Merged revisions 156164 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156164 | russell | 2008-11-12 11:29:52 -0600 (Wed, 12 Nov 2008) | 7 lines Move the sanity check that makes sure "always fork" is not set along with the console option to be after the code that reads options from asterisk.conf. This resolves a situation where Asterisk can start taking up 100% when misconfigured. (Thanks to Bryce Porter (x86 on IRC) for letting me log in to his system to figure out what was causing the 100% CPU problem.) ........ ................ r156295 | tilghman | 2008-11-12 13:28:22 -0600 (Wed, 12 Nov 2008) | 13 lines Merged revisions 156294 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156294 | tilghman | 2008-11-12 13:26:45 -0600 (Wed, 12 Nov 2008) | 6 lines If the SLA thread is not started, then reload causes a memory leak. (closes issue #13889) Reported by: eliel Patches: app_meetme.c.patch uploaded by eliel (license 64) ........ ................ r156690 | tilghman | 2008-11-13 15:30:41 -0600 (Thu, 13 Nov 2008) | 14 lines Merged revisions 156688 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156688 | tilghman | 2008-11-13 15:24:00 -0600 (Thu, 13 Nov 2008) | 7 lines Provide more space for all the data which can appear in an originating channel name. (closes issue #13398) Reported by: bamby Patches: manager.c.diff uploaded by bamby (license 430) ........ ................ r156756 | tilghman | 2008-11-13 18:43:13 -0600 (Thu, 13 Nov 2008) | 13 lines Merged revisions 156755 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156755 | tilghman | 2008-11-13 18:41:37 -0600 (Thu, 13 Nov 2008) | 6 lines ast_waitfordigit() requires that the channel be up, for no good logical reason. This prevents While/EndWhile from working within the "h" extension. Reported by: jgalarneau (for ABE C.2) Fixed by: me ........ ................ r158066 | mmichelson | 2008-11-20 11:39:06 -0600 (Thu, 20 Nov 2008) | 20 lines Merged revisions 158053 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158053 | mmichelson | 2008-11-20 11:33:06 -0600 (Thu, 20 Nov 2008) | 12 lines Make sure to set the hangup cause on the calling channel in the case that ast_call() fails. For incoming SIP channels, this was causing us to send a 603 instead of a 486 when the call-limit was reached on the destination channel. (closes issue #13867) Reported by: still_nsk Patches: 13867.diff uploaded by putnopvut (license 60) Tested by: blitzrage ........ ................ r158082 | mmichelson | 2008-11-20 11:54:31 -0600 (Thu, 20 Nov 2008) | 24 lines Merged revisions 158071 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158071 | mmichelson | 2008-11-20 11:48:42 -0600 (Thu, 20 Nov 2008) | 16 lines We don't handle 4XX responses to BYE well. According to section 15 of RFC 3261, we should terminate a dialog if we receive a 481 or 408 in response to our BYE. Since I am aware of at least one phone manufacturer who may sometimes send a 404 as well, I am being liberal and saying that any 4XX response to a BYE should result in a terminated dialog. (closes issue #12994) Reported by: pabelanger Patches: 12994.patch uploaded by putnopvut (license 60) Closes AST-129 ........ ................ r158540 | russell | 2008-11-21 16:12:37 -0600 (Fri, 21 Nov 2008) | 10 lines Merged revisions 158539 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158539 | russell | 2008-11-21 16:05:55 -0600 (Fri, 21 Nov 2008) | 2 lines When compiling with DEBUG_THREADS, report the real file/func/line for ao2_lock/ao2_unlock ........ ................ r158602 | tilghman | 2008-11-21 17:14:11 -0600 (Fri, 21 Nov 2008) | 12 lines Merged revisions 158600 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158600 | tilghman | 2008-11-21 17:07:46 -0600 (Fri, 21 Nov 2008) | 5 lines The passed extension may not be the same in the list as the current entry, because we strip spaces when copying the extension into the structure. Therefore, use the copied item to place the item into the list. (found by lmadsen on -dev, fixed by me) ........ ................ r159276 | tilghman | 2008-11-25 15:57:59 -0600 (Tue, 25 Nov 2008) | 14 lines Merged revisions 159269 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r159269 | tilghman | 2008-11-25 15:56:48 -0600 (Tue, 25 Nov 2008) | 7 lines Don't try to send a response on a NULL pvt. (closes issue #13919) Reported by: barthpbx Patches: chan_iax2.c.patch uploaded by eliel (license 64) Tested by: barthpbx ........ ................ * configs/features.conf.sample, apps/app_voicemail.c, apps/app_dial.c, channels/chan_dahdi.c, channels/chan_local.c, /, channels/chan_sip.c, apps/app_queue.c: Merged revisions 152216,152287,152369,152467,152569,152605 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r152216 | tilghman | 2008-10-27 16:34:04 -0500 (Mon, 27 Oct 2008) | 13 lines Merged revisions 152215 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152215 | tilghman | 2008-10-27 16:32:00 -0500 (Mon, 27 Oct 2008) | 6 lines Inherit ALL elements of CallerID across a local channel. (closes issue #13368) Reported by: Peter Schlaile Patches: 20080826__bug13368.diff.txt uploaded by Corydon76 (license 14) ........ ................ r152287 | jpeeler | 2008-10-27 18:31:39 -0500 (Mon, 27 Oct 2008) | 10 lines Merged revisions 152286 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152286 | jpeeler | 2008-10-27 18:28:49 -0500 (Mon, 27 Oct 2008) | 2 lines Buffer policy setting for half is not needed. ........ ................ r152369 | tilghman | 2008-10-28 12:07:39 -0500 (Tue, 28 Oct 2008) | 15 lines Merged revisions 152368 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152368 | tilghman | 2008-10-28 12:04:56 -0500 (Tue, 28 Oct 2008) | 8 lines Reset all DIAL variables back to blank, in case Dial is called multiple times per call (which could otherwise lead to inconsistent status reports). (closes issue #13216) Reported by: ruddy Patches: 20081014__bug13216.diff.txt uploaded by Corydon76 (license 14) Tested by: ruddy ........ ................ r152467 | tilghman | 2008-10-28 17:33:40 -0500 (Tue, 28 Oct 2008) | 10 lines Merged revisions 152463 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152463 | tilghman | 2008-10-28 17:32:34 -0500 (Tue, 28 Oct 2008) | 3 lines Quoting in the wrong direction (Fixes AST-107) ........ ................ r152569 | russell | 2008-10-29 00:34:26 -0500 (Wed, 29 Oct 2008) | 15 lines Merged revisions 152539 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152539 | russell | 2008-10-29 00:23:51 -0500 (Wed, 29 Oct 2008) | 7 lines Fix an incorrect usage of sizeof() (closes issue #13795) Reported by: andrew53 Patches: chan_sip_sizeof.patch uploaded by andrew53 (license 519) ........ ................ r152605 | murf | 2008-10-29 00:47:13 -0500 (Wed, 29 Oct 2008) | 22 lines Merged revisions 152538 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152538 | murf | 2008-10-28 23:19:04 -0600 (Tue, 28 Oct 2008) | 14 lines A little documentation cross-ref between features and dial and queue... I wasted some time (stupidly) trying to get the one-touch parking stuff working, because it didn't occur to me that I had to also have the corresponding options in the dial command! Duh! (In all this time, I never set this up before!) So, to keep some poor fool from suffering the same fate, I made the features.conf.sample file mention the corresponding opts in dial/queue; and the docs for dial/app specifically mention the corresponding decls in the feature.conf file. I hope this doesn't spoil some vast, eternal plan... ........ ................ * apps/app_speech_utils.c, apps/app_voicemail.c, Makefile, channels/chan_dahdi.c, /, channels/chan_sip.c, include/asterisk/audiohook.h, apps/app_waitforsilence.c, main/features.c, main/audiohook.c, apps/app_queue.c: Merged revisions 147518,147689,148000,148112,148268,148917,148988,149062,149131,149201,149205,149208 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r147518 | file | 2008-10-08 09:53:51 -0500 (Wed, 08 Oct 2008) | 9 lines Merged revisions 147517 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r147517 | file | 2008-10-08 11:51:42 -0300 (Wed, 08 Oct 2008) | 2 lines If we receive DTMF make sure that the state of the speech structure goes back to being not ready. (issue #LUMENVOX-8) ........ ................ r147689 | kpfleming | 2008-10-08 17:26:55 -0500 (Wed, 08 Oct 2008) | 9 lines Merged revisions 147681 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r147681 | kpfleming | 2008-10-08 17:22:09 -0500 (Wed, 08 Oct 2008) | 3 lines when parsing a text configuration option, ensure that the buffer on the stack is actually large enough to hold the legal values of that option, and also ensure that sscanf() knows to stop parsing if it would overrun the buffer (without these changes, specifying "buffers=...,immediate" would overflow the buffer on the stack, and could not have worked as expected) ........ ................ r148000 | tilghman | 2008-10-09 14:39:34 -0500 (Thu, 09 Oct 2008) | 11 lines Merged revisions 147997 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r147997 | tilghman | 2008-10-09 14:38:33 -0500 (Thu, 09 Oct 2008) | 4 lines When blank, callerid name and number should display "unknown caller" in voicemail emails. (Closes issue #13643) ........ ................ r148112 | mmichelson | 2008-10-09 18:15:33 -0500 (Thu, 09 Oct 2008) | 26 lines Merged revisions 146026 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r146026 | murf | 2008-10-03 12:12:54 -0500 (Fri, 03 Oct 2008) | 18 lines (closes issue #13579) Reported by: dwagner (closes issue #13584) Reported by: dwagner Tested by: murf, putnopvut The thought occurred to me that the res= from the extension spawn was ending up being returned from the bridge. "Thou shalt not poison the return value". Made the change and it appears to allow blind xfers to work as normal. If I'm wrong, reopen the bugs. But it looks good to me! Many thanks to putnopvut for helping me reproduce this! ........ ................ r148268 | tilghman | 2008-10-10 11:31:31 -0500 (Fri, 10 Oct 2008) | 14 lines Merged revisions 148257 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r148257 | tilghman | 2008-10-10 11:25:31 -0500 (Fri, 10 Oct 2008) | 7 lines User not notified of temporary greeting, if ODBC storage is in use. (closes issue #13659) Reported by: moliveras Patches: 20081009__bug13659.diff.txt uploaded by Corydon76 (license 14) Tested by: moliveras ........ ................ r148917 | tilghman | 2008-10-14 12:46:48 -0500 (Tue, 14 Oct 2008) | 11 lines Merged revisions 148916 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r148916 | tilghman | 2008-10-14 12:41:08 -0500 (Tue, 14 Oct 2008) | 4 lines Ensure that mail headers are 7-bit clean, even when UTF-8 characters are used in headers like 'Subject' and 'To'. Closes AST-107. ........ ................ r148988 | tilghman | 2008-10-14 14:03:44 -0500 (Tue, 14 Oct 2008) | 9 lines Merged revisions 148987 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r148987 | tilghman | 2008-10-14 14:03:08 -0500 (Tue, 14 Oct 2008) | 2 lines Some compilers warn, some don't. Fixing. ........ ................ r149062 | tilghman | 2008-10-14 15:16:48 -0500 (Tue, 14 Oct 2008) | 13 lines Merged revisions 149061 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149061 | tilghman | 2008-10-14 15:09:06 -0500 (Tue, 14 Oct 2008) | 6 lines Check correct values in the return of ast_waitfor(); also, get rid of a possible memory leak. (closes issue #13658) Reported by: explidous Patch by: me ........ ................ r149131 | mmichelson | 2008-10-14 16:08:48 -0500 (Tue, 14 Oct 2008) | 15 lines Merged revisions 149130 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149130 | mmichelson | 2008-10-14 15:49:02 -0500 (Tue, 14 Oct 2008) | 7 lines Don't allow reserved characters to be used in register lines in sip.conf. (closes issue #13570) Reported by: putnopvut ........ ................ r149201 | mmichelson | 2008-10-14 17:41:13 -0500 (Tue, 14 Oct 2008) | 20 lines Merged revisions 149200 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149200 | mmichelson | 2008-10-14 17:40:42 -0500 (Tue, 14 Oct 2008) | 12 lines Update the queue with the correct number of calls and whether the call was completed within the service level when a transfer takes place. This way, we do not "break" the leastrecent and fewestcalls strategies by not logging a call until after the transferred call has ended. (closes issue #13395) Reported by: Marquis Patches: app_queue.c.transfer.patch uploaded by Marquis (license 32) ........ ................ r149205 | mmichelson | 2008-10-14 18:04:44 -0500 (Tue, 14 Oct 2008) | 20 lines Merged revisions 149204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149204 | mmichelson | 2008-10-14 18:00:01 -0500 (Tue, 14 Oct 2008) | 12 lines Add a tolerance period for sync-triggered audiohooks so that if packetization of audio is close (but not equal) we don't end up flushing the audiohooks over small inconsistencies in synchronization. Related to issue #13005, and solves the issue for most people who were experiencing the problem. However, a small number of people are still experiencing the problem on long calls, so I am not closing the issue yet ........ ................ r149208 | mmichelson | 2008-10-14 18:15:04 -0500 (Tue, 14 Oct 2008) | 17 lines Merged revisions 149207 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149207 | mmichelson | 2008-10-14 18:10:26 -0500 (Tue, 14 Oct 2008) | 9 lines Call register_peer_exten even in the case that the peer's IP/port does not change. (closes issue #13309) Reported by: dimas Patches: v2-13309.patch uploaded by dimas (license 88) ........ ................ * channels/misdn/isdn_lib.c, Makefile, channels/chan_dahdi.c, channels/chan_misdn.c, main/manager.c, /: Merged revisions 115313,121770,123272,139624,140205,144257 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115313 | tilghman | 2008-05-05 15:22:08 -0500 (Mon, 05 May 2008) | 10 lines Merged revisions 115312 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115312 | tilghman | 2008-05-05 15:17:55 -0500 (Mon, 05 May 2008) | 2 lines Reverse order, such that user configs override default selections ........ ................ r121770 | crichter | 2008-06-11 06:52:18 -0500 (Wed, 11 Jun 2008) | 9 lines Merged revisions 121751 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r121751 | crichter | 2008-06-11 11:28:04 +0200 (Mi, 11 Jun 2008) | 1 line fixed issue with previous commit, the find_free_channel test for channels which where inuse was broken. ........ ................ r123272 | russell | 2008-06-17 10:52:13 -0500 (Tue, 17 Jun 2008) | 12 lines Merged revisions 123271 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123271 | russell | 2008-06-17 10:48:31 -0500 (Tue, 17 Jun 2008) | 4 lines Fix a memory leak in astobj2 that was pointed out by seanbright. When a container got destroyed, the underlying bucket list entry for each object that was in the container at that time did not get free'd. ........ ................ r139624 | jpeeler | 2008-08-22 16:57:32 -0500 (Fri, 22 Aug 2008) | 13 lines Merged revisions 139621 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139621 | jpeeler | 2008-08-22 16:36:13 -0500 (Fri, 22 Aug 2008) | 5 lines (closes issue #13359) Reported by: Laureano Patches: originate_channel_check.patch uploaded by Laureano (license 265) ........ ................ r140205 | jpeeler | 2008-08-26 13:48:55 -0500 (Tue, 26 Aug 2008) | 17 lines Merged revisions 140056 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140056 | jpeeler | 2008-08-26 10:57:02 -0500 (Tue, 26 Aug 2008) | 9 lines (closes issue #12071) Reported by: tzafrir Patches: dahdi_close.diff uploaded by tzafrir (license 46) Tested by: tzafrir, jpeeler This patch fixes closing open file descriptors in the case of an error. ........ ................ r144257 | crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008) | 9 lines Merged revisions 144238 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r144238 | crichter | 2008-09-24 10:20:52 +0200 (Mi, 24 Sep 2008) | 1 line improved helptext of misdn_set_opt. ........ ................ 2008-12-02 18:05 +0000 [r160326-160337] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 160333 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r160333 | jpeeler | 2008-12-02 12:04:51 -0600 (Tue, 02 Dec 2008) | 1 line remove duplicate comment that I accidentally merged ........ * channels/chan_dahdi.c, /: Merged revisions 160319 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r160319 | jpeeler | 2008-12-02 12:00:24 -0600 (Tue, 02 Dec 2008) | 7 lines (closes issue #13786) Reported by: tzafrir Readding DAHDI_CHECK_HOOKSTATE define that was removed in r134260 which fixes not being able to make outgoing calls on some FXO adapters: http://lists.digium.com/pipermail/asterisk-users/2008-November/thread.html#221553 ........ 2008-12-02 18:01 +0000 [r160228-160322] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 160308 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r160308 | tilghman | 2008-12-02 11:56:24 -0600 (Tue, 02 Dec 2008) | 17 lines Merged revisions 160297 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160297 | tilghman | 2008-12-02 11:42:09 -0600 (Tue, 02 Dec 2008) | 10 lines When the text does not match exactly (e.g. RTP/SAVP), then the %n conversion fails, and the resulting integer is garbage. Thus, we must initialize the integer and check it afterwards for success. (closes issue #14000) Reported by: folke Patches: asterisk-sipbg-sscanf-1.4.22.diff uploaded by folke (license 626) asterisk-sipbg-sscanf-1.6.0.1.diff uploaded by folke (license 626) asterisk-sipbg-sscanf-trunk-r159896.diff uploaded by folke (license 626) ........ ................ * include/asterisk/stringfields.h, apps/app_voicemail.c, main/cli.c, main/pbx.c, main/frame.c, /, channels/chan_features.c: Merged revisions 160208 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r160208 | tilghman | 2008-12-01 18:37:21 -0600 (Mon, 01 Dec 2008) | 10 lines Merged revisions 160207 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160207 | tilghman | 2008-12-01 18:25:16 -0600 (Mon, 01 Dec 2008) | 3 lines Ensure that Asterisk builds with --enable-dev-mode, even on the latest gcc and glibc. ........ ................ 2008-12-01 23:41 +0000 [r160173] Sean Bright * channels/chan_phone.c, main/manager.c, /, utils/smsq.c: Merged revisions 160170-160172 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r160170 | seanbright | 2008-12-01 18:08:24 -0500 (Mon, 01 Dec 2008) | 1 line Pay attention to the return value of system(), even if we basically ignore it. ................ r160171 | seanbright | 2008-12-01 18:18:48 -0500 (Mon, 01 Dec 2008) | 1 line Silence a build warning. (chan_phone.c:810: warning: value computed is not used) ................ r160172 | seanbright | 2008-12-01 18:37:49 -0500 (Mon, 01 Dec 2008) | 10 lines Merged revisions 159976 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r159976 | mvanbaak | 2008-12-01 11:08:36 -0500 (Mon, 01 Dec 2008) | 3 lines Get rid of the useless format string and argument in the Bogus/ manager channelname. Noted by kpfleming and name Bogus/manager suggested by eliel ........ ................ 2008-12-01 Tilghman Lesher * Released 1.6.0.2 2008-12-01 21:45 +0000 [r160100] Tilghman Lesher * /, configure, configure.ac: Merged revisions 160097 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r160097 | tilghman | 2008-12-01 15:23:37 -0600 (Mon, 01 Dec 2008) | 2 lines Use AST_EXT_LIB_SETUP before using AST_EXT_LIB_CHECK or bad things happen. ........ 2008-12-01 21:07 +0000 [r160096] Sean Bright * include/asterisk.h, /: Merged revisions 154919 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r154919 | seanbright | 2008-11-05 17:01:22 -0500 (Wed, 05 Nov 2008) | 2 lines Fix a problem found while building res_snmp. ........ 2008-12-01 17:39 +0000 [r160005] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 160004 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r160004 | russell | 2008-12-01 11:34:31 -0600 (Mon, 01 Dec 2008) | 14 lines Merged revisions 160003 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160003 | russell | 2008-12-01 11:27:30 -0600 (Mon, 01 Dec 2008) | 6 lines Apply some logic used in iax2_indicate() to iax2_setoption(), as well, since they both have the potential to send control frames in the middle of call setup. We have to wait until we have received a message back from the remote end before we try to send any more frames. Otherwise, the remote end will consider it invalid, and we'll get stuck in an INVAL/VNAK storm. ........ ................ 2008-12-01 16:04 +0000 [r159974] Michiel van Baak * main/manager.c, /: Merged revisions 159898 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r159898 | mvanbaak | 2008-12-01 15:09:59 +0100 (Mon, 01 Dec 2008) | 11 lines Merged revisions 159897 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r159897 | mvanbaak | 2008-12-01 15:05:41 +0100 (Mon, 01 Dec 2008) | 4 lines make manager compile on OpenBSD. The last (10th) argument to ast_channel_alloc here should be a pointer and NULL is not really a pointer. ........ ................ 2008-12-01 14:56 +0000 [r159915] Russell Bryant * .cleancount, /: Merged revisions 159911 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r159911 | russell | 2008-12-01 08:56:10 -0600 (Mon, 01 Dec 2008) | 10 lines Merged revisions 159900 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r159900 | russell | 2008-12-01 08:52:56 -0600 (Mon, 01 Dec 2008) | 2 lines Force a "make clean" to avoid a bizarre build issue ... ........ ................ 2008-11-29 18:37 +0000 [r159855] Kevin P. Fleming * utils/conf2ael.c, cdr/cdr_tds.c, main/ast_expr2.c, Makefile, include/asterisk/logger.h, include/asterisk/res_odbc.h, main/srv.c, channels/chan_misdn.c, include/asterisk/linkedlists.h, main/event.c, include/asterisk/strings.h, utils/extconf.c, makeopts.in, include/asterisk/stringfields.h, utils/check_expr.c, channels/chan_vpb.cc, /, main/utils.c, res/res_config_sqlite.c, utils/frame.c, channels/misdn_config.c, include/asterisk/astmm.h, include/asterisk/compat.h, configure, channels/misdn/ie.c, include/asterisk/module.h, main/features.c, main/dns.c, funcs/Makefile, include/asterisk/devicestate.h, include/asterisk/utils.h, channels/chan_sip.c, main/Makefile, include/asterisk/dundi.h, include/asterisk/enum.h, configure.ac, channels/chan_agent.c, utils/astman.c, include/asterisk/cli.h, include/asterisk/channel.h, include/jitterbuf.h, include/asterisk/manager.h: Merged revisions 159818 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r159818 | kpfleming | 2008-11-29 11:57:39 -0600 (Sat, 29 Nov 2008) | 18 lines incorporates r159808 from branches/1.4: ------------------------------------------------------------------------ r159808 | kpfleming | 2008-11-29 10:58:29 -0600 (Sat, 29 Nov 2008) | 7 lines update dev-mode compiler flags to match the ones used by default on Ubuntu Intrepid, so all developers will see the same warnings and errors since this branch already had some printf format attributes, enable checking for them and tag functions that didn't have them format attributes in a consistent way ------------------------------------------------------------------------ in addition: move some format attributes from main/utils.c to the header files they belong in, and fix up references to the relevant functions based on new compiler warnings ........ 2008-11-26 19:58 +0000 [r159558] Mark Michelson * apps/app_dial.c, /: Merged revisions 159554 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r159554 | mmichelson | 2008-11-26 13:57:11 -0600 (Wed, 26 Nov 2008) | 19 lines Add some necessary hangup commands in the case that forwarding a call fails 1) Hang up the original destination if the local channel cannot be requested. 2) Hang up the local channel (in addition to the original destination) if ast_call fails when calling the newly created local channel. This prevents channels from sticking around forever in the case of a botched call forward (e.g. to an extension which does not exist). (closes issue #13764) Reported by: davidw Patches: 13764_v2.patch uploaded by putnopvut (license 60) Tested by: putnopvut, davidw ........ 2008-11-26 19:18 +0000 [r159536] Kevin P. Fleming * agi/Makefile, utils/Makefile, /, Makefile.moddir_rules, Makefile.rules: Merged revisions 159534 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r159534 | kpfleming | 2008-11-26 13:08:56 -0600 (Wed, 26 Nov 2008) | 11 lines Merged revisions 159476 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r159476 | kpfleming | 2008-11-26 12:36:24 -0600 (Wed, 26 Nov 2008) | 7 lines simplify (and slightly bug-fix) the recent developer-oriented COMPILE_DOUBLE mode ensure that 'make clean' removes dependency files for .i files that are created in COMPILE_DOUBLE mode ........ ................ 2008-11-26 18:40 +0000 [r159478] Tilghman Lesher * main/udptl.c, /: Merged revisions 159475 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r159475 | tilghman | 2008-11-26 12:33:04 -0600 (Wed, 26 Nov 2008) | 7 lines If the config file does not exist, then the first use crashes Asterisk. (closes issue #13848) Reported by: klaus3000 Patches: udptl.c.patch uploaded by eliel (license 64) Tested by: blitzrage ........ 2008-11-26 15:01 +0000 [r159439] Mark Michelson * channels/chan_agent.c: Merged revisions 159437 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r159437 | mmichelson | 2008-11-26 08:58:17 -0600 (Wed, 26 Nov 2008) | 10 lines Don't allow for configuration options to overwrite options set via channel variables on a reload. (closes issue #13921) Reported by: davidw Patches: 13921.patch uploaded by putnopvut (license 60) Tested by: davidw ........ 2008-11-25 23:09 +0000 [r159374] Steve Murphy * main/cdr.c, /, channels/chan_iax2.c: Merged revisions 159360 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r159360 | murf | 2008-11-25 16:03:01 -0700 (Tue, 25 Nov 2008) | 23 lines Merged revisions 159316 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r159316 | murf | 2008-11-25 15:41:10 -0700 (Tue, 25 Nov 2008) | 15 lines (closes issue #12694) Reported by: yraber Patches: 12694.2nd.diff uploaded by murf (license 17) Tested by: murf, laurav Thanks to file (Joshua Colp) for his IAX fix. the change to cdr.c allows no-answer to percolate up into CDR's, and feels like the right place to locate this fix; if BUSY is done here, no-answer should be, too. ........ ................ 2008-11-25 22:28 +0000 [r159314] Mark Michelson * main/channel.c: I don't care what anyone says, this change is going into 1.6.0. Otherwise, the simple act of logging an agent in spams the CLI with warning messages about failed reads of the alertpipe. 2008-11-25 21:43 +0000 [r159248] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 159247 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r159247 | tilghman | 2008-11-25 15:42:42 -0600 (Tue, 25 Nov 2008) | 21 lines Merged revisions 159246 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r159246 | tilghman | 2008-11-25 15:40:28 -0600 (Tue, 25 Nov 2008) | 14 lines Merged revisions 159245 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r159245 | tilghman | 2008-11-25 15:37:06 -0600 (Tue, 25 Nov 2008) | 7 lines Regression fix for last security fix. Set the iseqno correctly. (closes issue #13918) Reported by: ffloimair Patches: 20081119__bug13918.diff.txt uploaded by Corydon76 (license 14) Tested by: ffloimair ........ ................ ................ 2008-11-25 16:21 +0000 [r159024-159094] Terry Wilson * /, apps/app_festival.c: Merged revisions 159093 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r159093 | twilson | 2008-11-25 10:18:53 -0600 (Tue, 25 Nov 2008) | 2 lines Add missing variable declaration for PPC code ........ * channels/chan_usbradio.c, /: Merged revisions 158992 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158992 | twilson | 2008-11-24 21:49:30 -0600 (Mon, 24 Nov 2008) | 2 lines Make chan_usbradio compile under dev mode ........ 2008-11-21 22:40 +0000 [r158545] Steve Murphy * /, main/features.c: Merged revisions 158484 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r158484 | murf | 2008-11-21 14:47:16 -0700 (Fri, 21 Nov 2008) | 19 lines Merged revisions 158483 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158483 | murf | 2008-11-21 14:19:47 -0700 (Fri, 21 Nov 2008) | 11 lines (closes issue #13871) Reported by: mdu113 This one is totally my fault. The code doesn't even create a bridge CDR if the channel CDR has POST_DISABLED. I didn't check for that at the end of the bridge. Fixed with a few small insertions. Tested. Looks good. No cdr generated, no crash, no unnecc. data objects created either. ........ ................ 2008-11-21 22:13 +0000 [r158542] Russell Bryant * main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions 158540 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r158540 | russell | 2008-11-21 16:12:37 -0600 (Fri, 21 Nov 2008) | 10 lines Merged revisions 158539 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158539 | russell | 2008-11-21 16:05:55 -0600 (Fri, 21 Nov 2008) | 2 lines When compiling with DEBUG_THREADS, report the real file/func/line for ao2_lock/ao2_unlock ........ ................ 2008-11-21 20:44 +0000 [r158451] Kevin P. Fleming * /, UPGRADE-1.2.txt, UPGRADE-1.4.txt, UPGRADE.txt, UPGRADE-1.6.txt, CHANGES: Merged revisions 158449 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158449 | kpfleming | 2008-11-21 14:42:37 -0600 (Fri, 21 Nov 2008) | 3 lines as suggested by jtodd, document the purposes of the CHANGES and UPGRADE files ........ 2008-11-21 17:12 +0000 [r158376] Terry Wilson * cdr/cdr_csv.c, /: Merged revisions 158374 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158374 | twilson | 2008-11-21 11:08:16 -0600 (Fri, 21 Nov 2008) | 8 lines Reloading the config and having no changes still initialized some settings to 0. Initialize settings after doing all of the cfg checks. (closes issue #13942) Reported by: davidw Patches: cdr_diff.txt uploaded by otherwiseguy (license 396) Tested by: davidw ........ 2008-11-21 01:23 +0000 [r158231-158267] Mark Michelson * /, channels/chan_sip.c: Merged revisions 158265-158266 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158265 | mmichelson | 2008-11-20 19:14:20 -0600 (Thu, 20 Nov 2008) | 4 lines Use some magic constants to get the right size for this sscanf statement. Thanks Richard! ........ r158266 | mmichelson | 2008-11-20 19:22:18 -0600 (Thu, 20 Nov 2008) | 3 lines Use a more expressive constant for a 64-bit scanned int ........ * /, channels/chan_sip.c: Merged revisions 158262 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158262 | mmichelson | 2008-11-20 18:59:23 -0600 (Thu, 20 Nov 2008) | 6 lines Fix the build for 32-bit systems. %lu is only 32-bits on 32-bit systems, so we need to use %llu instead. Of course %llu is 128-bits on 64-bit systems, so we have to cast to unsigned long long. No harm, but it's sure annoying. ........ * /, channels/chan_sip.c: Merged revisions 158230 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158230 | mmichelson | 2008-11-20 17:12:50 -0600 (Thu, 20 Nov 2008) | 20 lines Change the remote user agent session version variable from an int to a uint64_t. This prevents potential comparison problems from happening if the version string exceeds INT_MAX. This was an apparent problem for one user who could not properly place a call on hold since the version in the SDP of the re-INVITE to place the call on hold greatly exceeded INT_MAX. This also aligns with RFC 2327 better since it recommends using an NTP timestamp for the version (which is a 64-bit number). (closes issue #13531) Reported by: sgofferj Patches: 13531.patch uploaded by putnopvut (license 60) Tested by: sgofferj ........ 2008-11-20 19:42 +0000 [r158190] Sean Bright * res/ael/pval.c, /: Merged revisions 158188 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158188 | seanbright | 2008-11-20 14:41:23 -0500 (Thu, 20 Nov 2008) | 10 lines Fix one case where the application argument was not converted from a pipe to a comma. This was causing problems with switch statements with empty expressions. (closes issue #13901) Reported by: smurfix Patches: 20081118_bug13901.diff uploaded by seanbright (license 71) Tested by: seanbright Reviewed by: murf ........ 2008-11-20 00:12 +0000 [r157738-157976] Kevin P. Fleming * main/stdtime/Makefile, codecs/gsm/src, main/db1-ast/btree, channels/misdn/Makefile, main/db1-ast/recno, pbx/ael, res/ael, channels, main/db1-ast/Makefile, main/stdtime, main/db1-ast/hash, codecs/gsm/Makefile, main/db1-ast/db, Makefile.moddir_rules, channels/misdn, main/db1-ast/mpool, Makefile.rules, res/snmp, pbx/Makefile, res/Makefile: Merged revisions 157974 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r157974 | kpfleming | 2008-11-19 18:08:12 -0600 (Wed, 19 Nov 2008) | 13 lines Merged revisions 157859 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157859 | kpfleming | 2008-11-19 15:34:47 -0600 (Wed, 19 Nov 2008) | 7 lines the gcc optimizer frequently finds broken code (use of uninitalized variables, unreachable code, etc.), which is good. however, developers usually compile with the optimizer turned off, because if they need to debug the resulting code, optimized code makes that process very difficult. this means that we get code changes committed that weren't adequately checked over for these sorts of problems. with this build system change, if (and only if) --enable-dev-mode was used and DONT_OPTIMIZE is turned on, when a source file is compiled it will actually be preprocessed (into a .i or .ii file), then compiled once with optimization (with the result sent to /dev/null) and again without optimization (but only if the first compile succeeded, of course). while making these changes, i did some cleanup work in Makefile.rules to move commonly-used combinations of flag variables into their own variables, to make the file easier to read and maintain ........ ................ * /, res/res_agi.c: Merged revisions 157743 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157743 | kpfleming | 2008-11-19 07:45:48 -0600 (Wed, 19 Nov 2008) | 1 line correct small bug introduced during API conversion ........ * apps/app_stack.c, include/asterisk/agi.h, /, channels/chan_sip.c, res/res_agi.c, UPGRADE.txt, UPGRADE-1.6.txt (added): Merged revisions 157706 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157706 | kpfleming | 2008-11-19 06:42:19 -0600 (Wed, 19 Nov 2008) | 5 lines make some corrections to the ast_agi_register_multiple(), ast_agi_unregister_multiple() and ast_agi_fdprintf() API calls to be consistent with API guidelines also, move UPGRADE.txt to UPGRADE-1.6.txt and make the new UPGRADE.txt contain information about upgrading between Asterisk 1.6 releases ........ 2008-11-19 00:33 +0000 [r157601] Sean Bright * Makefile, /, build_tools/make_version, configure, configure.ac, build_tools/make_buildopts_h, makeopts.in: Merged revisions 157600 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157600 | seanbright | 2008-11-18 19:27:45 -0500 (Tue, 18 Nov 2008) | 10 lines Fix a few build problems on Solaris (and check for an md5 utility in configure instead of the icky loop I was doing before). (closes issue #13842) Reported by: snuffy Patches: bug13842_20081106.diff uploaded by snuffy (license 35) 13842.diff uploaded by seanbright (license 71) Tested by: snuffy ........ 2008-11-18 22:59 +0000 [r157307-157541] Mark Michelson * channels/chan_sip.c: Merged revisions 157512 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r157512 | mmichelson | 2008-11-18 16:54:08 -0600 (Tue, 18 Nov 2008) | 21 lines Merged revisions 157503 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157503 | mmichelson | 2008-11-18 16:47:57 -0600 (Tue, 18 Nov 2008) | 13 lines Add some missing invite state changes necessary in the sip_write function. Not setting the invite state correctly on the call was resulting in the Record application leaving empty files. I also have updated the doxygen comment next to the declaration of the INV_EARLY_MEDIA constant to reflect that we also use this state when we *send* a 18X response to an INVITE. (closes issue #13878) Reported by: nahuelgreco Patches: sip-early-media-recording-1.4.22.patch uploaded by nahuelgreco (license 162) Tested by: putnopvut ........ ................ * channels/chan_sip.c: Once again, Russell to the rescue. Use the builtin astobj1 lock of the sip_peer and sip_user instead of adding a new one * /, channels/chan_sip.c: Merged revisions 157496 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157496 | mmichelson | 2008-11-18 15:59:24 -0600 (Tue, 18 Nov 2008) | 6 lines Based on Russell's advice on the asterisk-dev list, I have changed from using a global lock in update_call_counter to using the locks within the sip_pvt and sip_peer structures instead. ........ * /, channels/chan_sip.c: Merged revisions 157427 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157427 | mmichelson | 2008-11-18 14:23:58 -0600 (Tue, 18 Nov 2008) | 13 lines * Add a lock to be used in the update_call_counter function. * Revert logic to mirror 1.4's in the sense that it will not allow the call counter to dip below 0. These two measures prevent potential races that could cause a SIP peer to appear to be busy forever. (closes issue #13668) Reported by: mjc Patches: hintfix_trunk_rev152649.patch uploaded by wolfelectronic (license 586) ........ * apps/app_dial.c, channels/chan_local.c, /, main/features.c, include/asterisk/channel.h, apps/app_followme.c: Merged revisions 157306 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r157306 | mmichelson | 2008-11-18 12:31:08 -0600 (Tue, 18 Nov 2008) | 20 lines Merged revisions 157305 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157305 | mmichelson | 2008-11-18 12:25:55 -0600 (Tue, 18 Nov 2008) | 12 lines Fix a crash in the end_bridge_callback of app_dial and app_followme which would occur at the end of an attended transfer. The error occurred because we initially stored a pointer to an ast_channel which then was hung up due to a masquerade. This commit adds a "fixup" callback to the bridge_config structure to allow for end_bridge_callback_data to be changed in the case that a new channel pointer is needed for the end_bridge_callback. ........ ................ 2008-11-18 18:10 +0000 [r157303] Steve Murphy * main/config.c, /: Merged revisions 157302 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157302 | murf | 2008-11-18 11:07:55 -0700 (Tue, 18 Nov 2008) | 18 lines (closes issue #13420) Reported by: alex70 Patches: 13420.13539.patch uploaded by murf (license 17) Tested by: murf, awk This fixes two problems: a spurious linefeed insertion probably left over from pre-precomment times. Only generated when category had no previous comments. The other problem: Insertions could get the line-numbering out of whack and generate negative line numbers, causing chunks of line numbers to be emitted, on the scale of the number of lines up to that point in the file. In such cases, abort the looping, and all is well. ........ 2008-11-15 19:47 +0000 [r157107-157165] Kevin P. Fleming * Makefile, /, Makefile.moddir_rules, Makefile.rules: Merged revisions 157164 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r157164 | kpfleming | 2008-11-15 20:45:19 +0100 (Sat, 15 Nov 2008) | 13 lines Merged revisions 157162-157163 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157162 | kpfleming | 2008-11-15 20:24:24 +0100 (Sat, 15 Nov 2008) | 1 line dist-clean should remove dependency information files as well ........ r157163 | kpfleming | 2008-11-15 20:31:03 +0100 (Sat, 15 Nov 2008) | 1 line when an individual directory dist-clean is run, run clean in that directory first, and when running top-level dist-clean, do not run subdirectory clean operations twice ........ ................ * /, contrib/asterisk-ng-doxygen: Merged revisions 157105 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157105 | kpfleming | 2008-11-15 19:00:32 +0100 (Sat, 15 Nov 2008) | 13 lines major update to doxygen configuration file: 1) update to doxygen 1.5.x style file, as used in trunk 2) tell doxygen where are header files are, so include-file processing can be done 3) make all macros that are used to define variables/functions be expanded, so that doxygen will properly document the resulting variable/function 4) make all macros that are used to provide the contents of a variable (structure) be expanded, so that doxygen will be able to document the resulting fields 5) suppress compiler attributes (__attribute__(xxx)) from being seen by doxygen, so it will properly match up function definition and usage (for an example of th effect of this, look at the doxygen docs for ast_log() from before and afte this commit) ........ 2008-11-14 17:03 +0000 [r156912] Tilghman Lesher * main/manager.c, /: Merged revisions 156911 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r156911 | tilghman | 2008-11-14 11:02:00 -0600 (Fri, 14 Nov 2008) | 4 lines Ping is missing the standard double-newline after the event. (closes issue #13903) Reported by: kebl0155 ........ 2008-11-14 16:55 +0000 [r156818-156889] Mark Michelson * include/asterisk/strings.h, apps/app_queue.c: This is the 1.6.0 version of revision 156883 of trunk. This is different in that it preserves the case-sensitiveness of processing queues from configuration. closes issue #13703 * apps/app_voicemail.c, /: Merged revisions 156817 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r156817 | mmichelson | 2008-11-14 09:20:03 -0600 (Fri, 14 Nov 2008) | 18 lines Merged revisions 156816 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156816 | mmichelson | 2008-11-14 09:18:59 -0600 (Fri, 14 Nov 2008) | 10 lines If the prompt to reenter a voicemail password timed out, it resulted in the password not being saved, even if the input matched what you gave when first prompted to enter a new password. This is because the return value of ast_readstring was checked, but not checked properly. This bug was discovered by Jared Smith during an Asterisk training course. Thanks for reporting it! ........ ................ 2008-11-13 19:26 +0000 [r156652-156653] Brandon Kruse * main/manager.c: Update to Coding Guidelines * main/manager.c, /: Merged revisions 156017 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r156017 | pari | 2008-11-11 17:02:43 -0600 (Tue, 11 Nov 2008) | 5 lines Patch by Ryan Brindley -- Make sure that manager refuses any duplicate 'new category' requests in updateconfig (closes issue #13539) ........ 2008-11-12 19:56 +0000 [r156319] Steve Murphy * main/pbx.c, /: Merged revisions 156299 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r156299 | murf | 2008-11-12 12:47:29 -0700 (Wed, 12 Nov 2008) | 26 lines Merged revisions 156297 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156297 | murf | 2008-11-12 12:36:16 -0700 (Wed, 12 Nov 2008) | 18 lines It turns out that the 0x0XX00 codes being returned for N, X, and Z are off by one, as per conversation with jsmith on #asterisk-dev; he was teaching a class and disconcerted that this published rule was not being followed, with patterns _NXX, _[1-8]22 and _[2-9]22... and NXX was winning, but [1-8] should have been. This change, tested on these 3 patterns now picks the proper one. However, this change may surprise users who set up dialplans based on previous behavior, which has been there for what, 2 and half years or so now. ........ ................ 2008-11-12 18:57 +0000 [r156251] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 156243 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r156243 | tilghman | 2008-11-12 12:55:18 -0600 (Wed, 12 Nov 2008) | 18 lines Merged revisions 156229 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156229 | tilghman | 2008-11-12 12:39:21 -0600 (Wed, 12 Nov 2008) | 11 lines Revert revision 132506, since it occasionally caused IAX2 HANGUP packets not to be sent, and instead, schedule a task to destroy the iax2 pvt structure 10 seconds later. This allows the IAX2 HANGUP packet to be queued, transmitted, and ACKed before the pvt is destroyed. (closes issue #13645) Reported by: dzajro Patches: 20081111__bug13645__3.diff.txt uploaded by Corydon76 (license 14) Tested by: vazir Reviewed: http://reviewboard.digium.com/r/51/ ........ ................ 2008-11-12 17:47 +0000 [r156170] Mark Michelson * apps/app_dial.c, /: Merged revisions 156169 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r156169 | mmichelson | 2008-11-12 11:41:56 -0600 (Wed, 12 Nov 2008) | 15 lines Merged revisions 156167 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156167 | mmichelson | 2008-11-12 11:38:33 -0600 (Wed, 12 Nov 2008) | 7 lines When doing some tests, I was having a crash at the end of every call if an attended transfer occurred during the call. I traced the cause to the CDR on one of the channels being NULL. murf suggested a check in the end bridge callback to be sure the CDR is non-NULL before proceeding, so that's what I'm adding. ........ ................ 2008-11-11 21:28 +0000 [r156012] Russell Bryant * apps/app_directory.c: Don't blow up if we get NULL when trying to parse out the full name field (fixed for Jared in the training room) 2008-11-11 20:04 +0000 [r156007] Michiel van Baak * /: remove prop that shouldn't be here 2008-11-11 19:49 +0000 [r155815-156004] Tilghman Lesher * /, res/res_realtime.c: Merged revisions 155862 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r155862 | tilghman | 2008-11-10 15:12:28 -0600 (Mon, 10 Nov 2008) | 5 lines Make documentation of update method match documentation and update update2 method to match. Reported by: atis, via -dev mailing list. Fixed by: me ........ * doc/valgrind.txt, /: Merged revisions 155804 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r155803 | tilghman | 2008-11-10 14:49:59 -0600 (Mon, 10 Nov 2008) | 1 line I got tired of saying this in every single bugnote referring to this file. ........ 2008-11-09 01:34 +0000 [r155555] Sean Bright * apps/app_dial.c, /, main/features.c, include/asterisk/channel.h, apps/app_followme.c, apps/app_queue.c: Merged revisions 155554 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r155554 | seanbright | 2008-11-08 20:27:00 -0500 (Sat, 08 Nov 2008) | 14 lines Merged revisions 155553 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r155553 | seanbright | 2008-11-08 20:08:07 -0500 (Sat, 08 Nov 2008) | 6 lines Use static functions here instead of nested ones. This requires a small change to the ast_bridge_config struct as well. To understand the reason for this change, see the following post: http://gcc.gnu.org/ml/gcc-help/2008-11/msg00049.html ........ ................ 2008-11-07 23:42 +0000 [r155361-155468] Mark Michelson * /, channels/chan_sip.c: Merged revisions 155467 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r155467 | mmichelson | 2008-11-07 17:41:44 -0600 (Fri, 07 Nov 2008) | 12 lines Set the invite state to INV_CANCELLED in a place that makes more sense. Where it was set before, it was impossible to actually delay sending a CANCEL if we had not yet received a provisional response to an INVITE. (closes issue #13626) Reported by: atis Patches: 13626.patch uploaded by putnopvut (license 60) Tested by: atis ........ * /, configs/voicemail.conf.sample: Merged revisions 155360 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r155360 | mmichelson | 2008-11-07 15:14:49 -0600 (Fri, 07 Nov 2008) | 8 lines Remove one more instance of the sample configuration lying about what's possible. The tz cannot be set in a context like this. It can only be set in the general section or per-mailbox. Thanks to sasargen on #asterisk-dev for pointing this out ........ 2008-11-06 22:50 +0000 [r155123] Kevin P. Fleming * /, res/ael/ael_lex.c, utils/extconf.c, res/ael/ael.flex: Merged revisions 155121 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r155121 | kpfleming | 2008-11-06 16:49:19 -0600 (Thu, 06 Nov 2008) | 3 lines don't blindly assume that Darwin and Cygwin need GLOB_ABORTED defined; only define it if it is not already defined ........ 2008-11-06 19:47 +0000 [r155013] Mark Michelson * /, configs/voicemail.conf.sample: Merged revisions 155012 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r155012 | mmichelson | 2008-11-06 13:46:53 -0600 (Thu, 06 Nov 2008) | 16 lines Merged revisions 155011 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r155011 | mmichelson | 2008-11-06 13:45:52 -0600 (Thu, 06 Nov 2008) | 8 lines The documentation listed the ability to set 'maxmsg' per context. The truth is that you can only set this in the general section or per mailbox. Thus I am updating the sample config file to be more accurate. Thanks to sasargen on IRC for bringing up this issue. ........ ................ 2008-11-03 22:30 +0000 [r154062-154081] Tilghman Lesher * /: Recorded merge of revisions 154072 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r154072 | tilghman | 2008-11-03 16:28:12 -0600 (Mon, 03 Nov 2008) | 12 lines Merged revisions 154066 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154066 | tilghman | 2008-11-03 16:27:10 -0600 (Mon, 03 Nov 2008) | 5 lines Attempting to expunge a mailbox when the mailstream is NULL will crash Asterisk. (Closes issue #13829) Reported by: jaroth Patch by: me (modified jaroth's patch) ........ ................ * main/rtp.c, /: Merged revisions 154060 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154060 | tilghman | 2008-11-03 15:48:21 -0600 (Mon, 03 Nov 2008) | 3 lines Remove the potential for a division by zero error. (Closes issue #13810) ........ 2008-11-03 00:53 +0000 [r153743-153746] Kevin P. Fleming * /: record revisions that were manually merged * apps/app_stack.c, include/asterisk/agi.h, configure, include/asterisk/autoconfig.h.in, autoconf/ast_gcc_attribute.m4, configure.ac, include/asterisk/compiler.h: Merge revision 153709 from trunk ------------------------------------------------------------------------ r153709 | kpfleming | 2008-11-02 17:34:39 -0600 (Sun, 02 Nov 2008) | 3 lines instead of trying to forcibly load res_agi when app_stack is loaded (even if the administrator didn't want it loaded), use GCC weak symbols to determine whether it was loaded already or not; if it was loaded, then use it. ------------------------------------------------------------------------ * channels/chan_oss.c, agi/eagi-sphinx-test.c, res/ael/ael_lex.c, channels/chan_h323.c, main/file.c, apps/app_sms.c, pbx/pbx_dundi.c, res/ael/ael.flex, pbx/pbx_config.c, apps/app_chanspy.c, apps/app_stack.c, utils/streamplayer.c, main/asterisk.c, apps/app_voicemail.c, utils/muted.c, apps/app_authenticate.c, res/res_phoneprov.c, main/utils.c, res/res_musiconhold.c, formats/format_wav_gsm.c, res/res_jabber.c, channels/chan_iax2.c, utils/frame.c, utils/stereorize.c, main/channel.c, channels/chan_dahdi.c, main/manager.c, res/ael/ael.tab.c, funcs/func_odbc.c, main/ast_expr2f.c, res/res_agi.c, main/logger.c, main/http.c, formats/format_gsm.c, apps/app_adsiprog.c, apps/app_dial.c, channels/chan_sip.c, formats/format_wav.c, apps/app_festival.c, main/db1-ast/hash/hash_page.c, res/ael/ael.y, res/res_crypto.c, agi/eagi-test.c, utils/astman.c, pbx/pbx_lua.c, formats/format_ogg_vorbis.c, utils/astcanary.c, apps/app_queue.c: port gcc 4.3.x warning fixes from trunk to this branch 2008-10-31 21:49 +0000 [r153265] Terry Wilson * apps/app_dial.c, /, main/features.c, include/asterisk/channel.h, apps/app_followme.c, apps/app_queue.c: Merged revisions 153181 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r153181 | twilson | 2008-10-31 13:55:33 -0500 (Fri, 31 Oct 2008) | 5 lines Recent CDR fixes moved execution of the 'h' exten into the bridging code, so variables that were set after ast_bridge_call was called would not show up in the 'h' exten. Added a callback function to handle setting variables, etc. from w/in the bridging code. Calls back into a nested function within the function calling ast_bridge_call (closes issue #13793) Reported by: greenfieldtech ........ 2008-10-30 21:00 +0000 [r152994] Sean Bright * /, bootstrap.sh: Merged revisions 152993 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r152993 | seanbright | 2008-10-30 16:59:17 -0400 (Thu, 30 Oct 2008) | 10 lines Merged revisions 152992 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152992 | seanbright | 2008-10-30 16:58:24 -0400 (Thu, 30 Oct 2008) | 2 lines The -I argument to aclocal needs a space before the include directory name. ........ ................ 2008-10-30 16:54 +0000 [r152813] Kevin P. Fleming * main/cdr.c, /: Merged revisions 152812 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r152812 | kpfleming | 2008-10-30 11:54:29 -0500 (Thu, 30 Oct 2008) | 9 lines Merged revisions 152811 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152811 | kpfleming | 2008-10-30 11:53:48 -0500 (Thu, 30 Oct 2008) | 3 lines instead of comparing the string pointer to 0, let's compare the value that was actually parsed out of the string (found by sparse) ........ ................ 2008-10-30 04:28 +0000 [r152772] Tilghman Lesher * configs/extensions.conf.sample, /: Merged revisions 152765 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r152765 | tilghman | 2008-10-29 23:26:34 -0500 (Wed, 29 Oct 2008) | 5 lines Set up an example stdexten that preserves the original context and extension in the CDR. (Related to issue #13799) Reported by: davidw ........ 2008-10-29 20:54 +0000 [r152647] Mark Michelson * /, apps/app_directory.c: Merged revisions 152646 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r152646 | mmichelson | 2008-10-29 15:53:53 -0500 (Wed, 29 Oct 2008) | 9 lines If there was no named defined in a voicemail.conf mailbox entry, then app_directory would crash when attempting to read that entry from the file. We now check for the NULL or empty string properly so that there will be no crash. (closes issue #13804) Reported by: bluecrow76 ........ 2008-10-29 20:13 +0000 [r152644] Terry Wilson * apps/app_queue.c: Small modification to putnopvut's patch to fix this issue. Thanks for all the help, putnopvut! (closes issue #12884) Reported by: bcnit Patches: 12884v4-1.6.0-branch.patch uploaded by otherwiseguy (license 396) Tested by: otherwiseguy 2008-10-28 21:39 +0000 [r152443] Tilghman Lesher * channels/chan_mgcp.c, /: Merged revisions 152442 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r152442 | tilghman | 2008-10-28 16:38:26 -0500 (Tue, 28 Oct 2008) | 7 lines Only re-add the io port if it was closed, otherwise reload causes a memory leak. (closes issue #13785) Reported by: eliel Patches: chan_mgcp.c.patch uploaded by eliel (license 64) ........ 2008-10-27 16:33 +0000 [r152157] Tilghman Lesher * apps/app_stack.c, /: Merged revisions 152134 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r152134 | tilghman | 2008-10-27 11:24:11 -0500 (Mon, 27 Oct 2008) | 4 lines Oops, only delete the ARG variables once upon release. The following section would have removed them again (removing variables from 2 stack frames, instead of just one). ........ 2008-10-26 20:26 +0000 [r152062] Sean Bright * /, funcs/func_strings.c: Merged revisions 152060 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r152060 | seanbright | 2008-10-26 16:25:08 -0400 (Sun, 26 Oct 2008) | 15 lines Merged revisions 152059 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152059 | seanbright | 2008-10-26 16:23:36 -0400 (Sun, 26 Oct 2008) | 7 lines Since passing \0 as the second argument to strchr is valid (and will match the trailing \0 of a string) we need to check that first, otherwise we end up with incorrect results. Fix suggested by reporter. (closes issue #13787) Reported by: meitinger ........ ................ 2008-10-23 16:12 +0000 [r151765] Russell Bryant * channels/chan_sip.c: Fix some memory leaks. These issues are 1.6.0 specific. - Freeing the peer got accidentally removed from the peer's destructor. It is still needed for astobj, but not for astobj2. - Fix some places that called find_user or find_peer, but did not release the reference that was returned. (closes issue #13331) Reported by: sergee Patches: chan_sip-3leaks-16-r151244.diff uploaded by sergee (license 138) Tested by: sergee 2008-10-20 05:03 +0000 [r151244] Kevin P. Fleming * autoconf (added), autoconf/ast_check_pwlib.m4, autoconf/acx_pthread.m4, autoconf/ast_func_fork.m4, configure, autoconf/ast_gcc_attribute.m4, bootstrap.sh, autoconf/ast_check_gnu_make.m4, autoconf/ast_ext_lib.m4, autoconf/ast_prog_ld.m4, autoconf/ast_c_compile_check.m4, autoconf/ast_c_define_check.m4, autoconf/ast_prog_egrep.m4, autoconf/ast_ext_tool_check.m4, autoconf/ast_check_mandatory.m4, /, autoconf/ast_check_openh323.m4, autoconf/ast_prog_ld_gnu.m4, configure.ac, acinclude.m4 (removed), autoconf/ast_prog_sed.m4: Merged revisions 151242-151243 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r151242 | kpfleming | 2008-10-20 07:59:04 +0300 (Mon, 20 Oct 2008) | 9 lines Merged revisions 151240 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r151240 | kpfleming | 2008-10-20 07:45:56 +0300 (Mon, 20 Oct 2008) | 3 lines break up acinclude.m4 into individual files, which will make it easier to maintain, easier to add new macros (less patching) and will ease maintenance of these macros across Asterisk branches ........ ................ r151243 | kpfleming | 2008-10-20 08:00:56 +0300 (Mon, 20 Oct 2008) | 9 lines Merged revisions 151241 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r151241 | kpfleming | 2008-10-20 07:57:33 +0300 (Mon, 20 Oct 2008) | 2 lines rename this macro to properly reflect what it does ........ ................ 2008-10-18 02:35 +0000 [r150854] BJ Weschke * main/manager.c, /: Merged revisions 150817 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r150817 | bweschke | 2008-10-17 22:18:33 -0400 (Fri, 17 Oct 2008) | 8 lines Using the GetVar handler in AMI is potentially dangerous (insta-crash [tm]) when you use a dialplan function that requires a channel and then you don't provide one or provide an invalid one in the Channel: parameter. We'll handle this situation exactly the same way it was handled in pbx.c back on r61766. We'll create a bogus channel for the function call and destroy it when we're done. If we have trouble allocating the bogus channel then we're not going to try executing the function call at all and run the risk of crashing. (closes issue #13715) reported by: makoto patch by: bweschke ........ 2008-10-17 00:19 +0000 [r150308-150313] Mark Michelson * channels/chan_sip.c: Instead of merging commit 150307 to 1.6.0, I had meant to block it in 1.6.1...time to go home :) * /, channels/chan_sip.c: Merged revisions 150307 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r150307 | mmichelson | 2008-10-16 19:13:35 -0500 (Thu, 16 Oct 2008) | 14 lines After a long discussion on #asterisk-bugs, it seems kind of odd that a channel would be named after the port on which it came in on. For endpoints that always include ":5060" as part of the From: header, it will mean that you have a ton of channels with names like "SIP/5060-3ea38a8b." I am boldly moving forward with this change in trunk, but I'm not touching other branches with this one since this definitely would qualify as a behavior change. If there is a problem with this commit, and I haven't seen the obvious reason why you'd want to name the channel after the port from which the call originated, then please feel free to revert this ........ 2008-10-16 16:10 +0000 [r150126] Richard Mudgett * channels/chan_misdn.c, /: Merged revisions 150125 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r150125 | rmudgett | 2008-10-16 11:04:45 -0500 (Thu, 16 Oct 2008) | 9 lines Merged revisions 150124 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r150124 | rmudgett | 2008-10-16 10:56:06 -0500 (Thu, 16 Oct 2008) | 1 line Fix memory leak found by customer ........ ................ 2008-10-15 20:18 +0000 [r149757] BJ Weschke * configs/agents.conf.sample, /: Merged revisions 149756 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r149756 | bweschke | 2008-10-15 16:14:20 -0400 (Wed, 15 Oct 2008) | 10 lines Merged revisions 149683 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149683 | bweschke | 2008-10-15 14:28:54 -0400 (Wed, 15 Oct 2008) | 4 lines An update to the documentation/example of agents.conf.sample with the correct parameter for this feature as defined in chan_agent.c (closes issue #13709) ........ ................ 2008-10-15 11:29 +0000 [r149495] Kevin P. Fleming * /, channels/chan_sip.c: Merged revisions 149487 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r149487 | kpfleming | 2008-10-15 13:26:36 +0200 (Wed, 15 Oct 2008) | 9 lines Merged revisions 149452 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149452 | kpfleming | 2008-10-15 12:30:40 +0200 (Wed, 15 Oct 2008) | 3 lines fix some problems when parsing SIP messages that have the maximum number of headers or body lines that we support ........ ................ 2008-10-14 17:39 +0000 [r148914] Mark Michelson * channels/chan_local.c, /: Merged revisions 148913 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r148913 | mmichelson | 2008-10-14 12:38:06 -0500 (Tue, 14 Oct 2008) | 17 lines Merged revisions 148912 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r148912 | mmichelson | 2008-10-14 12:33:38 -0500 (Tue, 14 Oct 2008) | 9 lines Deadlock prevention in chan_local. (closes issue #13676) Reported by: tacvbo Patches: 13676.patch uploaded by putnopvut (license 60) Tested by: tacvbo ........ ................ 2008-10-14 10:34 +0000 [r148613-148739] Kevin P. Fleming * Makefile, /: Merged revisions 148738 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r148738 | kpfleming | 2008-10-14 12:33:14 +0200 (Tue, 14 Oct 2008) | 9 lines Merged revisions 148736 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r148736 | kpfleming | 2008-10-14 12:30:54 +0200 (Tue, 14 Oct 2008) | 3 lines on Ubuntu (at least), recent versions of ld in binutils delete all debugging symbols when -x is supplied; since the reasons why -x is being passed are lost in the mists of time, remove it so debugging will work properly ........ ................ * /, main/translate.c: Merged revisions 148612 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r148612 | kpfleming | 2008-10-14 03:06:45 -0500 (Tue, 14 Oct 2008) | 9 lines Merged revisions 148611 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r148611 | kpfleming | 2008-10-14 02:54:41 -0500 (Tue, 14 Oct 2008) | 3 lines it would be nice if this message printing code had actually been tested before it was committed... ........ ................ 2008-10-10 21:18 +0000 [r148374] Mark Michelson * /, channels/chan_sip.c: Merged revisions 148373 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r148373 | mmichelson | 2008-10-10 16:18:10 -0500 (Fri, 10 Oct 2008) | 8 lines Make sure that the inUse and inRinging fields for a sip peer cannot go below zero. This is a regression from 1.4 and so it will be applied to 1.6.0 as well. (closes issue #13668) Reported by: mjc ........ 2008-10-10 01:25 +0000 [r148201-148204] Sean Bright * res/res_config_sqlite.c, apps/app_voicemail.c, include/asterisk.h, /, main/tdd.c, main/cryptostub.c: Merged revisions 148200 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r148200 | seanbright | 2008-10-09 20:42:13 -0400 (Thu, 09 Oct 2008) | 12 lines Don't include logger.h in asterisk.h by default as it is causing problems building app_voicemail. Instead, include it where it is needed. This turned out to be a relatively minor issue because other headers include logger.h as well. Need to test -addons before merging this back to 1.6.0. (closes issue #13605) Reported by: tomo1657 Patches: 13605_seanbright.diff uploaded by seanbright (license 71) Tested by: mmichelson ........ * apps/app_rpt.c: Somehow we got conflict markers checked in! Might need a 1.6.0.1 sooner than we'd like. 2008-10-09 23:31 +0000 [r148147] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 148144 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r148144 | mmichelson | 2008-10-09 18:30:47 -0500 (Thu, 09 Oct 2008) | 10 lines Read the callerid in the correct order and make sure to read the Urgent flag value from the IMAP headers. (closes issue #13652) Reported by: jaroth Patches: imapheaders.patch uploaded by jaroth (license 50) ........ 2008-10-09 23:26 +0000 [r148124] Tilghman Lesher * /, configs/res_ldap.conf.sample: Merged revisions 148120 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r148120 | tilghman | 2008-10-09 18:25:53 -0500 (Thu, 09 Oct 2008) | 6 lines Fix example schema (closes issue #12860) Reported by: flyn Patches: res_ldap.conf.patch uploaded by flyn (license 503) ........ 2008-10-09 17:51 +0000 [r147900] Michiel van Baak * include/asterisk/endian.h, /: Merged revisions 147899 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r147899 | mvanbaak | 2008-10-09 19:48:53 +0200 (Thu, 09 Oct 2008) | 5 lines only include this for OpenBSD. At least FreeBSD is borked when including it (closes issue #13649) Reported by: ys ........ 2008-10-09 17:47 +0000 [r147897] Tilghman Lesher * configs/extensions.conf.sample, /: Merged revisions 147896 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r147896 | tilghman | 2008-10-09 12:46:15 -0500 (Thu, 09 Oct 2008) | 4 lines Remove "second form" of extensions, as it no longer applies. Also, cleanup the grammar, formatting, and introduce several clarifications to the text. (Closes issue #13654) ........ 2008-10-09 14:56 +0000 [r147809] Steve Murphy * main/astobj2.c, channels/chan_oss.c, main/config.c, main/rtp.c, main/cli.c, configure, channels/console_gui.c, utils/extconf.c, main/pbx.c, include/asterisk.h, doc/CODING-GUIDELINES, /, include/asterisk/autoconfig.h.in, main/translate.c, channels/vcodecs.c, configure.ac, channels/console_video.c, channels/chan_iax2.c: Merged revisions 147807 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r147807 | murf | 2008-10-09 08:17:33 -0600 (Thu, 09 Oct 2008) | 15 lines (closes issue #13557) Reported by: nickpeirson Patches: pbx.c.patch uploaded by nickpeirson (license 579) replace_bzero+bcopy.patch uploaded by nickpeirson (license 579) Tested by: nickpeirson, murf 1. replaced all refs to bzero and bcopy to memset and memmove instead. 2. added a note to the CODING-GUIDELINES 3. add two macros to asterisk.h to prevent bzero, bcopy from creeping back into the source 4. removed bzero from configure, configure.ac, autoconfig.h.in ........ 2008-10-08 12:16 +0000 [r147458] Russell Bryant * configs/chan_dahdi.conf.sample: Remove the sample configuration for configuration sections in chan_dahdi.conf. This code was not merged into 1.6.0. Reported by: angler (closes AST-119) 2008-10-08 Russell Bryant * Asterisk 1.6.0.1 released. * configs/chan_dahdi.conf.sample: Remove mention of configuration sections for defining channels in chan_dahdi.conf. This code is in 1.6.1, and was not merged into 1.6.0. 2008-10-01 Russell Bryant * Asterisk 1.6.0 released. 2008-09-09 Russell Bryant * Asterisk 1.6.0-rc6 released. 2008-09-09 15:44 +0000 [r142065] Russell Bryant * /, main/features.c: Merged revisions 142064 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r142064 | russell | 2008-09-09 10:44:10 -0500 (Tue, 09 Sep 2008) | 13 lines Merged revisions 142063 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142063 | russell | 2008-09-09 10:40:24 -0500 (Tue, 09 Sep 2008) | 5 lines Ensure that the stored CDR reference is still valid after the bridge before poking at it. Also, keep the channel locked while messing with this CDR. (fixes crashes reported in issue #13409) ........ ................ 2008-09-09 12:34 +0000 [r141996-141999] Mark Michelson * channels/chan_oss.c, /: Merged revisions 141995 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r141995 | mmichelson | 2008-09-09 05:20:58 -0500 (Tue, 09 Sep 2008) | 8 lines Fix a memory leak in chan_oss (closes issue #13311) Reported by: eliel Patches: chan_oss.c.patch uploaded by eliel (license 64) ........ 2008-09-09 01:49 +0000 [r141950] Russell Bryant * main/channel.c, /: Merged revisions 141949 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r141949 | russell | 2008-09-08 20:47:56 -0500 (Mon, 08 Sep 2008) | 9 lines Modify ast_answer() to not hold the channel lock while calling ast_safe_sleep() or when calling ast_waitfor(). These are inappropriate times to hold the channel lock. This is what has caused "could not get the channel lock" messages from chan_sip and has likely caused a negative impact on performance results of SIP in Asterisk 1.6. Thanks to file for pointing out this section of code. (closes issue #13287) (closes issue #13115) ........ 2008-09-08 21:07 +0000 [r141808] Russell Bryant * main/pbx.c, /: Merged revisions 141807 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r141807 | russell | 2008-09-08 16:05:01 -0500 (Mon, 08 Sep 2008) | 15 lines Merged revisions 141806 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141806 | russell | 2008-09-08 16:02:36 -0500 (Mon, 08 Sep 2008) | 7 lines When doing an async goto, detect if the channel is already in the middle of a masquerade. This can happen when chan_local is trying to optimize itself out. If this happens, fail the async goto instead of bursting into flames. (closes issue #13435) Reported by: geoff2010 ........ ................ 2008-09-08 Russell Bryant * Asterisk 1.6.0-rc5 released. 2008-09-08 20:19 +0000 [r141746] Jason Parker * Makefile, /, redhat (removed): Merged revisions 141745 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r141745 | qwell | 2008-09-08 15:18:17 -0500 (Mon, 08 Sep 2008) | 16 lines Merged revisions 141741 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141741 | qwell | 2008-09-08 15:15:42 -0500 (Mon, 08 Sep 2008) | 8 lines Remove RPM package targets from Makefile (and all associated parts). This has never worked in 1.4, and we decided that it makes no sense to be done here. There are many distros out there that already have "proper" spec files that can be (re)used. Closes issue #13113 Closes issue #10950 Closes issue #10952 ........ ................ 2008-09-08 17:14 +0000 [r141683] Sean Bright * /, build_tools/make_buildopts_h: Merged revisions 141682 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r141682 | seanbright | 2008-09-08 13:13:04 -0400 (Mon, 08 Sep 2008) | 9 lines Quote the arguments to grep so that sh on various platforms doesn't choke on the special characters (like ^). (closes issue #13417) Reported by: dougm Patches: 13417.make_buildopts_h.patch uploaded by seanbright (license 71) Tested by: dougm ........ 2008-09-06 20:21 +0000 [r141567] Steve Murphy * /, channels/chan_sip.c: Merged revisions 141566 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r141566 | murf | 2008-09-06 14:19:50 -0600 (Sat, 06 Sep 2008) | 9 lines Merged revisions 141565 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141565 | murf | 2008-09-06 14:13:16 -0600 (Sat, 06 Sep 2008) | 1 line This fix comes from Joshua Colp The Brilliant, who, given the trace, came up with a solution. This will most likely will close 13235 and 13409. I'll wait till Monday to verify, and then close these bugs. ........ ................ 2008-09-06 15:40 +0000 [r141505-141508] Tilghman Lesher * /, res/res_agi.c: Merged revisions 141504 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r141504 | tilghman | 2008-09-06 10:26:45 -0500 (Sat, 06 Sep 2008) | 12 lines Merged revisions 141503 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141503 | tilghman | 2008-09-06 10:23:42 -0500 (Sat, 06 Sep 2008) | 4 lines Reverting behavior change (AGI should not exit non-zero on SUCCESS) (closes issue #13434) Reported by: francesco_r ........ ................ 2008-09-05 22:06 +0000 [r141368-141426] Mark Michelson * /, channels/chan_agent.c: Merged revisions 141367 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r141367 | mmichelson | 2008-09-05 16:12:09 -0500 (Fri, 05 Sep 2008) | 15 lines Merged revisions 141366 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141366 | mmichelson | 2008-09-05 16:10:32 -0500 (Fri, 05 Sep 2008) | 7 lines Agent's should not try to call a channel's indicate callback if the channel has been hung up. It will likely crash otherwise ABE-1159 ........ ................ 2008-09-05 14:24 +0000 [r141116-141158] Steve Murphy * main/channel.c, /: Merged revisions 141157 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r141157 | murf | 2008-09-05 08:18:43 -0600 (Fri, 05 Sep 2008) | 9 lines Merged revisions 141156 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141156 | murf | 2008-09-05 08:15:43 -0600 (Fri, 05 Sep 2008) | 1 line A small change to prevent double-posting of CDR's; thanks to Daniel Ferrer for bringing it to our attention ........ ................ * pbx/ael/ael-test/ref.ael-vtest25 (added), /, pbx/ael/ael-test/ael-vtest25/extensions.ael, pbx/ael/ael-test/ael-vtest25 (added), res/ael/ael_lex.c, pbx/ael/ael-test/ref.ael-test6, res/ael/ael.flex: Merged revisions 141115 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r141115 | murf | 2008-09-04 17:31:41 -0600 (Thu, 04 Sep 2008) | 78 lines Merged revisions 141094 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141094 | murf | 2008-09-04 17:15:07 -0600 (Thu, 04 Sep 2008) | 70 lines (closes issue #13357) Reported by: pj Tested by: murf (closes issue #13416) Reported by: yarns Tested by: murf If you find this message overly verbose, relax, it's probably not meant for you. This message is meant for probably only two people in the whole world: me, or the poor schnook that has to maintain this code because I'm either dead or unavailable at the moment. This fix solves two reports, both having to do with embedding a function call in a ${} construct. It was tricky because the funccall syntax has parenthesis () in it. And up till now, the 'word' token in the flex stuff didn't allow that, because it would tend to steal the LP and RP tokens. To be truthful, the "word" token was the trickiest, most unstable thing in the whole lexer. I was lucky it made this long without complaints. I had to choose every character in the pattern with extreme care, and I knew that someday I'd have to revisit it. Well, the day has come. So, my brilliant idea (and I'm being modest), was to use the surrounding ${} construct to make a state machine and capture everything in it, no matter what it contains. But, I have to now treat the word token like I did with comments, in that I turn the whole thing into a state-machine sort of spec, with new contexts "curlystate", "wordstate", and "brackstate". Wait a minute, "brackstate"? Yes, well, it didn't take very many regression tests to point out if I do this for ${} constructs, I also have to do it with the $[] constructs, too. I had to create a separate pcbstack2 and pcbstack3 because these constructs can occur inside macro argument lists, and when we have two state machines operating on the same structures we'd get problems otherwise. I guess I could have stopped at pcbstack2 and had the brackstate stuff share it, but it doesn't hurt to be safe. So, the pcbpush and pcbpop routines also now have versions for "2" and "3". I had to add the {KEYWORD} construct to the initial pattern for "word", because previously word would match stuff like "default7", because it was a longer match than the keyword "default". But, not any more, because the word pattern only matches only one or two characters now, and it will always lose. So, I made it the winner again by making an optional match on any of the keywords before it's normal pattern. I added another regression test to make sure we don't lose this in future edits, and had to fix just one regression, where it no longer reports a 'cascaded' error, which I guess is a plus. I've given some thought as to whether to apply these fixes to 1.4 and the 1.6.x releases, vs trunk; I decided to put it in 1.4 because one of the bug reports was against 1.4; and it is unexpected that AEL cannot handle this situation. It actually reduced the amount of useless "cascade" error messages that appeared in the regressions (by one line, ehhem). There is a possible side-effect in that it does now do more careful checking of what's in those ${} constructs, as far as matching parens, and brackets are concerned. Some users may find a an insidious problem and correct it this way. This should be exceedingly rare, I hope. ........ ................ 2008-09-04 18:35 +0000 [r141086] Jeff Peeler * /, main/features.c, res/res_agi.c: Merged revisions 141039 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r141039 | jpeeler | 2008-09-04 12:27:56 -0500 (Thu, 04 Sep 2008) | 15 lines Merged revisions 141028 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141028 | jpeeler | 2008-09-04 12:00:29 -0500 (Thu, 04 Sep 2008) | 7 lines (closes issue #11979) Fixes multiple parking problems: Crash when executing a park on an extension dialed by AGI due to not returning the proper return code. Crash when using a builtin feature that was a subset of a enabled dynamic feature. Crash due to always hanging up the peer despite the fact that the peer was supposed to be parked. ........ ................ 2008-09-03 20:18 +0000 [r140976] Mark Michelson * /, apps/app_queue.c: Merged revisions 140975 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r140975 | mmichelson | 2008-09-03 15:16:12 -0500 (Wed, 03 Sep 2008) | 4 lines Fix some locking order issues in app_queue. This was brought up by atis on IRC a while ago. ........ 2008-09-03 Russell Bryant * Asterisk 1.6.0-rc4 released. 2008-09-03 14:17 +0000 [r140825-140827] Steve Murphy * main/cdr.c, /: Merged revisions 140749 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r140749 | murf | 2008-09-02 17:44:04 -0600 (Tue, 02 Sep 2008) | 11 lines Merged revisions 140747 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140747 | murf | 2008-09-02 17:36:56 -0600 (Tue, 02 Sep 2008) | 1 line I am turning the warnings generated in ast_cdr_free and post_cdr into verbose level 2 messages. Really, they matter little to end users. You either get the CDR's you wanted, or you don't, and it is a bug. For trunk, I am going one step further. These messages were pretty worthless even for debug, so I'm completely removing them. ........ ................ * main/channel.c, /: Merged revisions 140692 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r140692 | murf | 2008-09-02 16:55:12 -0600 (Tue, 02 Sep 2008) | 13 lines Merged revisions 140690 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140690 | murf | 2008-09-02 16:40:13 -0600 (Tue, 02 Sep 2008) | 1 line After reconsidering, with respect to 13409, ast_cdr_detach should be OK, better in fact, than ast_cdr_free, which generates lots of useless warnings that will undoubtably generate complaints. Hmmm. It doesn't hush the useless warnings, but it does allow control of posting via the detach and post routines, for those possible situations, where you'd want to post single-channel cdrs. ........ ................ * main/channel.c, main/pbx.c, /: Merged revisions 140691 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r140691 | murf | 2008-09-02 16:50:59 -0600 (Tue, 02 Sep 2008) | 22 lines Merged revisions 140670 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140670 | murf | 2008-09-02 16:15:57 -0600 (Tue, 02 Sep 2008) | 14 lines (closes issue #13409) Reported by: tomaso Patches: asterisk-1.6.0-rc2-cdrmemleak.patch uploaded by tomaso (license 564) I basically spent the day, verifying that this patch solves the problem, and doesn't hurt in non-problem cases. Why valgrind did not plainly reveal this leak absolutely mystifies and stuns me. Many, many thanks to tomaso for finding and providing the fix. ........ ................ 2008-09-03 13:27 +0000 [r140818] Russell Bryant * main/poll.c, /: Merged revisions 140817 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r140817 | russell | 2008-09-03 08:26:43 -0500 (Wed, 03 Sep 2008) | 12 lines Merged revisions 140816 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140816 | russell | 2008-09-03 08:24:35 -0500 (Wed, 03 Sep 2008) | 4 lines Don't freak out if the poll emulation receives NULL for the pollfds array (closes issue #13307) Reported by: jcovert ........ ................ 2008-09-02 18:17 +0000 [r140607] Sean Bright * /, channels/chan_iax2.c: Merged revisions 140606 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r140606 | seanbright | 2008-09-02 14:15:54 -0400 (Tue, 02 Sep 2008) | 16 lines Merged revisions 140605 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140605 | seanbright | 2008-09-02 14:14:57 -0400 (Tue, 02 Sep 2008) | 8 lines Make sure to use the correct length of the mohinterpret and mohsuggest buffers when copying configuration values. (closes issue #13336) Reported by: decryptus_proformatique Patches: chan_iax2_mohinterpret_mohsuggest_general_settings.patch uploaded by decryptus (license 555) ........ ................ 2008-09-02 15:12 +0000 [r140564-140567] Russell Bryant * apps/app_jack.c, /, codecs/codec_resample.c: Merged revisions 140566 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r140566 | russell | 2008-09-02 10:11:53 -0500 (Tue, 02 Sep 2008) | 2 lines Update instructions for getting libresample ........ 2008-08-27 20:15 +0000 [r140302-140304] Mark Michelson * channels/chan_sip.c: Revert commit 140302. Should not be merging changes like that into a release-candidate branch * channels/chan_sip.c: Merged revisions 140301 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r140301 | mmichelson | 2008-08-27 15:11:22 -0500 (Wed, 27 Aug 2008) | 19 lines Merged revisions 140299 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140299 | mmichelson | 2008-08-27 14:49:20 -0500 (Wed, 27 Aug 2008) | 11 lines Fix tag checking in get_sip_pvt_byid_locked when in pedantic mode. The problem was that the wrong tags would be compared depending on the direction of the call. (closes issue #13353) Reported by: flefoll Patches: chan_sip.c.br14.139015.patch-refer-pedantic uploaded by flefoll (license 244) ........ ................ 2008-08-26 18:12 +0000 [r140170] Russell Bryant * Makefile, /: Merged revisions 140169 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r140169 | russell | 2008-08-26 13:11:49 -0500 (Tue, 26 Aug 2008) | 4 lines Fix building menuselect-tree with PRINT_DIR set. We _must_ use the --quiet flag here, or else some arbitrary text will end up in the resulting menuselect-tree file and things will explode. ........ 2008-08-25 21:33 +0000 [r139918] Sean Bright * build_tools/get_moduleinfo, /, build_tools/get_makeopts: Merged revisions 139915 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139915 | seanbright | 2008-08-25 17:32:10 -0400 (Mon, 25 Aug 2008) | 17 lines Merged revisions 139909 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139909 | seanbright | 2008-08-25 17:31:03 -0400 (Mon, 25 Aug 2008) | 9 lines Some versions of awk (nawk, for example) don't like empty regular expressions so be slightly more verbose. (closes issue #13374) Reported by: dougm Patches: 13374.diff uploaded by seanbright (license 71) Tested by: dougm ........ ................ 2008-08-25 21:05 +0000 [r139872] Terry Wilson * /, channels/chan_sip.c: Merged revisions 139870 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139870 | twilson | 2008-08-25 15:59:58 -0500 (Mon, 25 Aug 2008) | 10 lines Merged revisions 139869 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139869 | twilson | 2008-08-25 15:46:10 -0500 (Mon, 25 Aug 2008) | 2 lines Make SIPADDHEADER() propagate indefinitely ........ ................ 2008-08-25 16:00 +0000 [r139774] Steve Murphy * main/pbx.c, /, main/features.c: Merged revisions 139770 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139770 | murf | 2008-08-25 09:54:18 -0600 (Mon, 25 Aug 2008) | 17 lines Merged revisions 139764 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139764 | murf | 2008-08-25 09:33:14 -0600 (Mon, 25 Aug 2008) | 9 lines This patch reverts the changes made via 139347, and 139635, as users are seeing adverse difference. I will un-close 13251. Back to the drawing board/ concept/ beginning/ whatever! ........ ................ 2008-08-24 16:30 +0000 [r139705-139708] Tilghman Lesher * /, cdr/cdr_pgsql.c: Merged revisions 139707 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r139707 | tilghman | 2008-08-24 11:26:48 -0500 (Sun, 24 Aug 2008) | 2 lines Memory leak ........ 2008-08-22 22:35 +0000 [r139628-139671] Steve Murphy * /, main/features.c: Merged revisions 139662 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139662 | murf | 2008-08-22 16:32:35 -0600 (Fri, 22 Aug 2008) | 14 lines Merged revisions 139635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139635 | murf | 2008-08-22 16:24:02 -0600 (Fri, 22 Aug 2008) | 6 lines I found some problems with the code I committed earlier, when I merged them into trunk, so I'm coming back to clean up. And, in the process, I found an error in the code I added to trunk and 1.6.x, that I'll fix using this patch also. ........ ................ * apps/app_dial.c, main/pbx.c, /, main/features.c: Merged revisions 139627 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139627 | murf | 2008-08-22 16:03:13 -0600 (Fri, 22 Aug 2008) | 59 lines Merged revisions 139347 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139347 | murf | 2008-08-21 17:03:50 -0600 (Thu, 21 Aug 2008) | 47 lines (closes issue #13251) Reported by: sergee Tested by: murf THis is a bold move for a static release fix, but I wouldn't have made it if I didn't feel confident (at least a *bit* confident) that it wouldn't mess everyone up. The reasoning goes something like this: 1. We simply cannot do anything with CDR's at the current point (in pbx.c, after the __ast_pbx_run loop). It's way too late to have any affect on the CDRs. The CDR is already posted and gone, and the remnants have been cleared. 2. I was very much afraid that moving the running of the 'h' extension down into the bridge code (where it would be now practical to do it), would result in a lot more calls to the 'h' exten, so I implemented it as another exten under another name, but found, to my pleasant surprise, that there was a 1:1 correspondence to the running of the 'h' exten in the pbx_run loop, and the new spot at the end of the bridge. So, I ifdef'd out the current 'h' loop, and moved it into the bridge code. The only difference I can see is the stuff about the AST_PBX_KEEPALIVE, and hopefully, if this is still an important decision point, I can replicate it if there are complaints. To be perfectly honest, the KEEPALIVE situation is not totally clear to me, and how it relates to a post-bridge situation is less clear. I suspect the users will point out everything in total clarity if this steps on anyone's toes! 3. I temporarily swap the bridge_cdr into the channel before running the 'h' exten, which makes it possible for users to edit the cdr before it goes out the door. And, of course, with the endbeforehexten config var set, the users can also get at the billsec/duration vals. After the h exten finishes, the cdr is swapped back and processing continues as normal. Please, all who deal with CDR's, please test this version of Asterisk, and file bug reports as appropriate! ........ I also made a little fix to the app_dial's 'e' option, that is related to my updates. ................ 2008-08-22 20:21 +0000 [r139458-139564] Mark Michelson * include/asterisk/threadstorage.h, /: Merged revisions 139554 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139554 | mmichelson | 2008-08-22 14:45:41 -0500 (Fri, 22 Aug 2008) | 16 lines Merged revisions 139553 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139553 | mmichelson | 2008-08-22 14:45:19 -0500 (Fri, 22 Aug 2008) | 8 lines Fix compilation when DEBUG_THREAD_LOCALS is selected (closes issue #13298) Reported by: snuffy Patches: bug13298_20080822.diff uploaded by snuffy (license 35) ........ ................ * /, channels/chan_iax2.c: Merged revisions 139469 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139469 | mmichelson | 2008-08-22 12:25:12 -0500 (Fri, 22 Aug 2008) | 11 lines Merged revisions 139466 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139466 | mmichelson | 2008-08-22 12:24:47 -0500 (Fri, 22 Aug 2008) | 3 lines Fix the build. Thanks, mvanbaak! ........ ................ * /, channels/chan_iax2.c: Merged revisions 139457 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139457 | mmichelson | 2008-08-22 11:58:21 -0500 (Fri, 22 Aug 2008) | 15 lines Merged revisions 139456 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139456 | mmichelson | 2008-08-22 11:57:38 -0500 (Fri, 22 Aug 2008) | 7 lines Prevent a deadlock in chan_iax2 resulting from incorrect locking order between iax2_pvt and ast_channel structures. AST-13 ........ ................ 2008-08-21 23:46 +0000 [r139400] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 139391 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139391 | jpeeler | 2008-08-21 18:41:50 -0500 (Thu, 21 Aug 2008) | 11 lines Merged revisions 139387 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139387 | jpeeler | 2008-08-21 18:39:31 -0500 (Thu, 21 Aug 2008) | 3 lines Fixes loop that could possibly never exit in the event of a channel never being able to be opened or specify after a restart. (closes issue #11017) ........ ................ 2008-08-21 10:02 +0000 [r139282] Philippe Sultan * /, channels/chan_gtalk.c: Merged revisions 139281 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r139281 | phsultan | 2008-08-21 11:55:31 +0200 (Thu, 21 Aug 2008) | 5 lines Fix two memory leaks in chan_gtalk, thanks Eliel! (closes issue #13310) Reported by: eliel Patches: chan_gtalk.c.patch uploaded by eliel (license 64) ........ 2008-08-20 Kevin P. Fleming * Asterisk 1.6.0-rc3 released. 2008-08-20 22:17 +0000 [r139216] Russell Bryant * apps/app_chanspy.c, /: Merged revisions 139215 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139215 | russell | 2008-08-20 17:16:36 -0500 (Wed, 20 Aug 2008) | 19 lines Merged revisions 139213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139213 | russell | 2008-08-20 17:14:35 -0500 (Wed, 20 Aug 2008) | 11 lines Fix a crash in the ChanSpy application. The issue here is that if you call ChanSpy and specify a spy group, and sit in the application long enough looping through the channel list, you will eventually run out of stack space and the application with exit with a seg fault. The backtrace was always inside of a harmless snprintf() call, so it was tricky to track down. However, it turned out that the call to snprintf() was just the biggest stack consumer in this code path, so it would always be the first one to hit the boundary. (closes issue #13338) Reported by: ruddy ........ ................ 2008-08-20 20:12 +0000 [r139155] Shaun Ruffell * codecs/codec_dahdi.c: Fix bug where the samples were not accurate when in G723 mode, which would cause the timestamp field of the RTP header to be invalid. 2008-08-20 17:30 +0000 [r139104] Steve Murphy * main/cdr.c, /: Merged revisions 139083 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139083 | murf | 2008-08-20 11:25:07 -0600 (Wed, 20 Aug 2008) | 20 lines Merged revisions 139074 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139074 | murf | 2008-08-20 11:14:55 -0600 (Wed, 20 Aug 2008) | 12 lines (closes issue #13263) Reported by: brainy Tested by: murf The specialized reset routine is tromping on the flags field of the CDR. I made a change to not reset the DISABLED bit. This should get rid of this problem. ........ ................ 2008-08-20 15:39 +0000 [r138889-139017] Mark Michelson * /, channels/chan_sip.c: Merged revisions 139016 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139016 | mmichelson | 2008-08-20 10:38:47 -0500 (Wed, 20 Aug 2008) | 14 lines Merged revisions 139015 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139015 | mmichelson | 2008-08-20 10:37:56 -0500 (Wed, 20 Aug 2008) | 6 lines sip_read should properly handle a NULL return from sip_rtp_read. (closes issue #13257) Reported by: travishein ........ ................ * apps/app_chanspy.c: Manually add revision 138887 from trunk to the 1.6.0 branch. I had misunderstood the policy for when to merge to 1.6.0 since it moved to rc status. 2008-08-19 16:38 +0000 [r138846-138847] Steve Murphy * utils/conf2ael.c, /, res/ael/ael.tab.c, res/ael/ael.y, res/ael/ael.tab.h, utils/ael_main.c: Merged revisions 138845 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138845 | murf | 2008-08-19 10:31:24 -0600 (Tue, 19 Aug 2008) | 1 line Oops. put a decl in a generated file. My bad, but fixed now. ........ * main/pbx.c, /, res/ael/ael.tab.c, res/ael/ael.y, res/ael/ael.tab.h: Merged revisions 138815 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138815 | murf | 2008-08-19 09:59:12 -0600 (Tue, 19 Aug 2008) | 19 lines These changes are in regards to bug 13249, where users are being surprised by the changes made to the Set app in trunk/1.6.x, as they come from the 1.4 world. They are only bitten if they write their AEL dialplan in the 1.4 world, and then carry it over to a trunk/1.6.x installation where a "make samples" was executed, or where they hand-edited the asterisk.conf file and added the [compat] category with app_set = 1.6 (or higher). (this commit does not totally solve 13249, at least not yet) The change involves issueing a single warning while the AEL file is loading, if: 1. app_set is present in the config file, and set to 1.6 or higher. 2. there are double quotes in an assignment statement (eg x = "hi there";) 3. the warning was not already issued. The standalone app, aelparse, does not (yet) issue this warning. I'd have to have it read in the asterisk.conf file, and that's a bit of hassle. I'll add it if users request it, tho. ........ 2008-08-19 00:15 +0000 [r138776-138781] Sean Bright * /, channels/chan_sip.c: Merged revisions 138778-138780 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138778 | seanbright | 2008-08-18 20:08:27 -0400 (Mon, 18 Aug 2008) | 1 line While we're at it, make this machine parseable too. ........ r138779 | seanbright | 2008-08-18 20:09:38 -0400 (Mon, 18 Aug 2008) | 1 line And remove code we don't need anymore. ........ r138780 | seanbright | 2008-08-18 20:10:56 -0400 (Mon, 18 Aug 2008) | 1 line Let it compile now, too (woops) ........ * /, channels/chan_sip.c: Merged revisions 138775 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138775 | seanbright | 2008-08-18 19:42:36 -0400 (Mon, 18 Aug 2008) | 3 lines Change event header to RegistrationTime to be more consistent (and avoid breaking existing frameworks). Pointed out by Laureano on #asterisk-dev. ........ 2008-08-18 20:23 +0000 [r138688-138695] Mark Michelson * /, apps/app_queue.c: Merged revisions 138687 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r138687 | mmichelson | 2008-08-18 15:04:10 -0500 (Mon, 18 Aug 2008) | 18 lines Merged revisions 138685 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r138685 | mmichelson | 2008-08-18 15:01:14 -0500 (Mon, 18 Aug 2008) | 10 lines Change the inequalities used in app_queue with regards to timeouts from being strict to non-strict for more accuracy. (closes issue #13239) Reported by: atis Patches: app_queue_timeouts_v2.patch uploaded by atis (license 242) ........ ................ 2008-08-18 15:54 +0000 [r138632] Jason Parker * Makefile, /: Merged revisions 138631 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138631 | qwell | 2008-08-18 10:54:07 -0500 (Mon, 18 Aug 2008) | 1 line Remove option that isn't valid here. ........ 2008-08-18 02:14 +0000 [r138519] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 138518 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138518 | jpeeler | 2008-08-17 21:13:04 -0500 (Sun, 17 Aug 2008) | 1 line add missing define for SS7 in dahdi_restart ........ 2008-08-17 14:14 +0000 [r138443-138483] Sean Bright * /, main/features.c: Merged revisions 138482 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138482 | seanbright | 2008-08-17 10:12:11 -0400 (Sun, 17 Aug 2008) | 6 lines Move Uniqueid to the end of the event for those that rely on the position of the name/value pairs, pointed out by snuffy-home on #asterisk-commits. For those of you who rely on the position of name/value pairs in manager events... stop... that is why associative arrays were invented. ........ * /, main/features.c: Merged revisions 138479 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138479 | seanbright | 2008-08-17 09:51:08 -0400 (Sun, 17 Aug 2008) | 7 lines Add Uniqueid header to ParkedCall manager event. (closes issue #13323) Reported by: srt Patches: 13323_unique_id_for_parkedcalls_event.diff uploaded by srt (license 378) ........ * main/rtp.c, /: Merged revisions 138476 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138476 | seanbright | 2008-08-17 09:40:36 -0400 (Sun, 17 Aug 2008) | 7 lines Add missing colons to RTCPReceived and RTCPSent manager events. (closes issue #13319) Reported by: srt Patches: 13319_rtcp_manager_event_headers.diff uploaded by srt (license 378) ........ * /, channels/chan_iax2.c: Merged revisions 138473 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138473 | seanbright | 2008-08-17 09:31:54 -0400 (Sun, 17 Aug 2008) | 7 lines Fix the output of the JitterBufStats manager event. (closes issue #13324) Reported by: srt Patches: 13324_missing_nl_in_jitterbufstats_event_2.diff uploaded by srt (license 378) ........ * configs/cdr_tds.conf.sample, /: Merged revisions 138442 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138442 | seanbright | 2008-08-16 12:40:43 -0400 (Sat, 16 Aug 2008) | 4 lines Since it's introduction in revision 3497, cdr_tds has *never* read the port configuration option from cdr_tds.conf. So go ahead and remove it from the sample config. ........ 2008-08-16 13:07 +0000 [r138410-138413] Tilghman Lesher * channels/chan_dahdi.c, /: Merged revisions 138412 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138412 | tilghman | 2008-08-16 08:07:08 -0500 (Sat, 16 Aug 2008) | 2 lines Fix compilation warnings (found with dev-mode) ........ 2008-08-16 01:14 +0000 [r138333-138362] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 138361 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r138361 | jpeeler | 2008-08-15 20:13:26 -0500 (Fri, 15 Aug 2008) | 9 lines Merged revisions 138360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r138360 | jpeeler | 2008-08-15 20:12:18 -0500 (Fri, 15 Aug 2008) | 1 line fixes use count to properly decrement if an active dahdi channel is destroyed allowing module to be unloaded ........ ................ * channels/chan_dahdi.c, /: Merged revisions 138311 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r138311 | jpeeler | 2008-08-15 18:46:09 -0500 (Fri, 15 Aug 2008) | 20 lines Merged revisions 138119,138151,138238 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r138119 | jpeeler | 2008-08-15 14:21:51 -0500 (Fri, 15 Aug 2008) | 4 lines Fixes the dahdi restart functionality. Dahdi restart allows one to restart all DAHDI channels, even if they are currently in use. This is different from unloading and then loading the module since unloading requires the use count to be zero. Reloading the module is different in that the signalling is not changed from what it was originally configured. Also, this fixes not closing all the file descriptors for D-channels upon module unload (which would prevent loading the module afterwards). (closes issue #11017) ........ r138151 | jpeeler | 2008-08-15 14:41:29 -0500 (Fri, 15 Aug 2008) | 1 line declared static mutexes using AST_MUTEX_DEFINE_STATIC macro ........ r138238 | jpeeler | 2008-08-15 16:28:26 -0500 (Fri, 15 Aug 2008) | 1 line initialize condition variable ss_thread_complete using ast_cond_init ........ ................ 2008-08-15 23:03 +0000 [r138207-138262] Tilghman Lesher * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 138260 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r138260 | tilghman | 2008-08-15 17:54:57 -0500 (Fri, 15 Aug 2008) | 16 lines Merged revisions 138258 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r138258 | tilghman | 2008-08-15 17:33:42 -0500 (Fri, 15 Aug 2008) | 8 lines More fixes for realtime peers. (closes issue #12921) Reported by: Nuitari Patches: 20080804__bug12921.diff.txt uploaded by Corydon76 (license 14) 20080815__bug12921.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ ................ * configs/extensions.conf.sample, main/pbx.c, /: Merged revisions 138206 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138206 | tilghman | 2008-08-15 15:35:24 -0500 (Fri, 15 Aug 2008) | 4 lines Remove deprecated syntax from sample config file (closes issue #13314) Reported by: kue ........ 2008-08-15 20:20 +0000 [r138156-138157] Jeff Peeler * channels/chan_dahdi.c: rename all zfd instances in chan_dahdi to dfd to match 1.4 (left over from DAHDI transition) 2008-08-15 15:12 +0000 [r138029] Russell Bryant * main/autoservice.c, /: Merged revisions 138028 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r138028 | russell | 2008-08-15 10:09:46 -0500 (Fri, 15 Aug 2008) | 17 lines Merged revisions 138027 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r138027 | russell | 2008-08-15 10:07:16 -0500 (Fri, 15 Aug 2008) | 9 lines Ensure that when a hangup occurs in autoservice, that a hangup frame gets properly deferred to be read from the channel owner when it gets taken out of autoservice. (closes issue #12874) Reported by: dimas Patches: v1-12874.patch uploaded by dimas (license 88) ........ ................ 2008-08-15 15:04 +0000 [r138025] Tilghman Lesher * /, funcs/func_strings.c: Merged revisions 138024 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r138024 | tilghman | 2008-08-15 10:03:32 -0500 (Fri, 15 Aug 2008) | 16 lines Merged revisions 138023 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r138023 | tilghman | 2008-08-15 09:51:12 -0500 (Fri, 15 Aug 2008) | 8 lines Additional check for more string specifiers than arguments. (closes issue #13299) Reported by: adomjan Patches: 20080813__bug13299.diff.txt uploaded by Corydon76 (license 14) func_strings.c-sprintf.patch uploaded by adomjan (license 487) Tested by: adomjan ........ ................ 2008-08-14 22:43 +0000 [r137988] Russell Bryant * /, doc/tex/Makefile: Merged revisions 137987 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137987 | russell | 2008-08-14 17:43:15 -0500 (Thu, 14 Aug 2008) | 2 lines Fix a bashism that causes an error when trying to build the pdf on ubuntu ........ 2008-08-14 18:48 +0000 [r137934] Sean Bright * cdr/cdr_sqlite3_custom.c, /: Merged revisions 137933 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137933 | seanbright | 2008-08-14 14:47:28 -0400 (Thu, 14 Aug 2008) | 8 lines Fix memory leak in cdr_sqlite3_custom. (closes issue #13304) Reported by: eliel Patches: sqlite.patch uploaded by eliel (license 64) (Slightly modified by me) ........ 2008-08-14 17:01 +0000 [r137849-137852] Tilghman Lesher * channels/chan_dahdi.c, /: Merged revisions 137848 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r137848 | tilghman | 2008-08-14 11:52:43 -0500 (Thu, 14 Aug 2008) | 17 lines Merged revisions 137847 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r137847 | tilghman | 2008-08-14 11:47:30 -0500 (Thu, 14 Aug 2008) | 9 lines When creating the secondary subchannel name, it is necessary to compare to the existing channel name without the "Zap/" or "DAHDI/" prefix, since our test string is also without that prefix. (closes issue #13027) Reported by: dferrer Patches: chan_zap-1.4.21.1_fix2.patch uploaded by dferrer (license 525) (Slightly modified by me, to compensate for both names) ........ ................ 2008-08-14 Jason Parker * Asterisk 1.6.0-rc2 released. 2008-08-14 15:37 +0000 [r137814] Jason Parker * /, channels/chan_sip.c: Merged revisions 137812 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137812 | qwell | 2008-08-14 10:32:16 -0500 (Thu, 14 Aug 2008) | 8 lines Make sure we set the socket port, so we don't try to use :0. (closes issue #13255) Reported by: falves11 Patches: 13255-socketport.diff uploaded by qwell (license 4) Tested by: falves11 ........ 2008-08-14 15:20 +0000 [r137783] Russell Bryant * /, configs/sip.conf.sample: Merged revisions 137732 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r137732 | russell | 2008-08-14 09:15:50 -0500 (Thu, 14 Aug 2008) | 12 lines Merged revisions 137731 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r137731 | russell | 2008-08-14 09:05:23 -0500 (Thu, 14 Aug 2008) | 4 lines Comments in this config file were aligned only if your tab size was set to 8. So, convert tabs to spaces so that things should be aligned regardless of what tab size you use in your editor. ........ ................ 2008-08-14 15:05 +0000 [r137781] Sean Bright * cdr/cdr_tds.c, /: Merged revisions 137780 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137780 | seanbright | 2008-08-14 11:03:03 -0400 (Thu, 14 Aug 2008) | 8 lines If we detect that we are no longer connected, try to reconnect a few times before giving up. This relies on the timeout settings in the freetds.conf file and, unfortunately, on a recent version of FreeTDS (0.82 or newer). I either need to change the current execs to be non-blocking (which I do not want to do) or we have to force people to run with the latest and greatest of FreeTDS. I'm on the fence... ........ 2008-08-14 02:04 +0000 [r137681] Kevin P. Fleming * /, Zaptel-to-DAHDI.txt: Merged revisions 137680 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r137680 | kpfleming | 2008-08-13 21:03:47 -0500 (Wed, 13 Aug 2008) | 9 lines Merged revisions 137679 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r137679 | kpfleming | 2008-08-13 21:03:04 -0500 (Wed, 13 Aug 2008) | 1 line forgot one module name that changed ........ ................ 2008-08-13 Kevin P. Fleming * Asterisk 1.6.0-rc1 released. 2008-08-13 23:00 +0000 [r137631-137641] Kevin P. Fleming * /, build_tools/prep_tarball: Merged revisions 137640 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137640 | kpfleming | 2008-08-13 18:00:37 -0500 (Wed, 13 Aug 2008) | 1 line make this script actually work ........ * /, Zaptel-to-DAHDI.txt (added), UPGRADE.txt: Merged revisions 137627 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r137627 | kpfleming | 2008-08-13 17:33:32 -0500 (Wed, 13 Aug 2008) | 9 lines Merged revisions 137530 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r137530 | kpfleming | 2008-08-13 15:46:19 -0500 (Wed, 13 Aug 2008) | 1 line add document describing what users will need to be aware of when upgrading to this version and using DAHDI ........ ................ 2008-08-13 21:09 +0000 [r137497-137533] Jason Parker * /, channels/chan_sip.c: Merged revisions 137532 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137532 | qwell | 2008-08-13 16:08:58 -0500 (Wed, 13 Aug 2008) | 8 lines Correctly end locally ended calls. (closes issue #12170) Reported by: pj Patches: 20080702__issue12170_clear_pendinginvite.diff uploaded by bbryant (license 36) Tested by: bbryant, pabelanger ........ * /, apps/app_fax.c: Merged revisions 137496 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137496 | qwell | 2008-08-13 15:05:50 -0500 (Wed, 13 Aug 2008) | 6 lines Add FAXMODE variable with what fax transport was used. (closes issue #13252) Patches: v1-13252.patch uploaded by dimas (license 88) ........ 2008-08-13 14:47 +0000 [r137350-137407] Sean Bright * /, doc/tex/cdrdriver.tex: Merged revisions 137406 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r137406 | seanbright | 2008-08-13 10:41:49 -0400 (Wed, 13 Aug 2008) | 9 lines Merged revisions 137405 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r137405 | seanbright | 2008-08-13 10:33:49 -0400 (Wed, 13 Aug 2008) | 1 line Update docs to reflect the change to cdr_tds ........ ................ * cdr/cdr_tds.c, /: Merged revisions 137403 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137403 | seanbright | 2008-08-13 10:22:47 -0400 (Wed, 13 Aug 2008) | 1 line Use the ast_vasprintf macro instead of vasprintf directly. ........ 2008-08-12 19:48 +0000 [r137300-137302] Russell Bryant * doc/tex/asterisk.tex, /: Merged revisions 137301 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137301 | russell | 2008-08-12 14:48:38 -0500 (Tue, 12 Aug 2008) | 2 lines Grammar hax from Qwell ........ * doc/tex/asterisk.tex, /: Merged revisions 137299 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137299 | russell | 2008-08-12 14:40:35 -0500 (Tue, 12 Aug 2008) | 3 lines Note that developer documentation belongs in doxygen, and not integrated with the user manual stuff in doc/tex/. ........ 2008-08-11 16:15 +0000 [r137240] Russell Bryant * Makefile, /: Merged revisions 137239 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137239 | russell | 2008-08-11 11:14:29 -0500 (Mon, 11 Aug 2008) | 2 lines Make PRINT_DIR work as advertised. ........ 2008-08-11 14:31 +0000 [r137217] Sean Bright * cdr/cdr_tds.c, /, UPGRADE.txt: Merged revisions 137203 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137203 | seanbright | 2008-08-11 10:25:15 -0400 (Mon, 11 Aug 2008) | 7 lines Log the userfield CDR variable like the other CDR backends, assuming the column is actually there. If it's not, we still log everything else as before. (closes issue #13281) Reported by: falves11 ........ 2008-08-11 00:27 +0000 [r137160] Tilghman Lesher * res/res_odbc.c, /: Merged revisions 137150 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r137150 | tilghman | 2008-08-10 19:25:28 -0500 (Sun, 10 Aug 2008) | 13 lines Merged revisions 137138 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r137138 | tilghman | 2008-08-10 19:20:38 -0500 (Sun, 10 Aug 2008) | 5 lines Deallocate database connection handle on disconnect, as we allocate another one on connect. (closes issue #13271) Reported by: dveiga ........ ................ 2008-08-09 15:27 +0000 [r136948] Tilghman Lesher * /, include/asterisk/compat.h, include/asterisk/astobj2.h: Merged revisions 136947 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r136947 | tilghman | 2008-08-09 10:26:27 -0500 (Sat, 09 Aug 2008) | 18 lines Merged revisions 136946 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r136946 | tilghman | 2008-08-09 10:25:36 -0500 (Sat, 09 Aug 2008) | 10 lines Merged revisions 136945 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r136945 | tilghman | 2008-08-09 10:24:36 -0500 (Sat, 09 Aug 2008) | 2 lines Regression fixes for Solaris ........ ................ ................ 2008-08-09 01:16 +0000 [r136860] Tilghman Lesher * /, res/res_agi.c: Merged revisions 136859 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r136859 | tilghman | 2008-08-08 20:15:38 -0500 (Fri, 08 Aug 2008) | 4 lines Update documentation as to the behavior of AGI in 1.6.0 and higher. Also, add an OOB message that answers the question of, if AGI no longer shuts down the connection on hangup, how will FastAGI know when to stop processing the call? ........ 2008-08-08 15:33 +0000 [r136785] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 136784 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r136784 | mmichelson | 2008-08-08 10:31:31 -0500 (Fri, 08 Aug 2008) | 3 lines Fix compilation for ODBC voicemail ........ 2008-08-08 06:45 +0000 [r136778] Steve Murphy * pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18, pbx/ael/ael-test/ref.ael-test19, pbx/ael/ael-test/ref.ael-vtest13, res/ael/pval.c, /, pbx/ael/ael-test/ref.ael-ntest10, include/asterisk/ael_structs.h, utils/ael_main.c: Merged revisions 136746 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r136746 | murf | 2008-08-07 18:48:35 -0600 (Thu, 07 Aug 2008) | 40 lines Merged revisions 136726 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r136726 | murf | 2008-08-07 18:15:34 -0600 (Thu, 07 Aug 2008) | 32 lines (closes issue #13236) Reported by: korihor Wow, this one was a challenge! I regrouped and ran a new strategy for setting the ~~MACRO~~ value; I set it once per extension, up near the top. It is only set if there is a switch in the extension. So, I had to put in a chunk of code to detect a switch in the pval tree. I moved the code to insert the set of ~~exten~~ up to the beginning of the gen_prios routine, instead of down in the switch code. I learned that I have to push the detection of the switches down into the code, so everywhere I create a new exten in gen_prios, I make sure to pass onto it the values of the mother_exten first, and the exten next. I had to add a couple fields to the exten struct to accomplish this, in the ael_structs.h file. The checked field makes it so we don't repeat the switch search if it's been done. I also updated the regressions. ........ ................ 2008-08-08 02:36 +0000 [r136753] Tilghman Lesher * /: Merged revisions 136751 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r136751 | tilghman | 2008-08-07 21:34:17 -0500 (Thu, 07 Aug 2008) | 2 lines Removing bad properties ........ 2008-08-07 23:42 +0000 [r136719-136724] Mark Michelson * apps/app_voicemail.c: This is weird. Either SVN or vim tabbed a bunch of functions over one level during a merge. * apps/app_voicemail.c, /: Merged revisions 136722 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r136722 | mmichelson | 2008-08-07 18:39:50 -0500 (Thu, 07 Aug 2008) | 3 lines Remove one last batch of debug messages ........ * apps/app_voicemail.c, /: Merged revisions 136715 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r136715 | mmichelson | 2008-08-07 17:25:50 -0500 (Thu, 07 Aug 2008) | 18 lines Merging the imap_consistency_trunk branch to trunk. For an explanation of what "imap_consistency" is, please see svn revision 134223 to the 1.4 branch. Coincidentally, this also fixes a recent bug report regarding the inability to save messages to the new folder when using IMAP storage since they will would be flagged as "seen" and not be recognized as new messages. (closes issue #13234) Reported by: jaroth ........ 2008-08-07 20:41 +0000 [r136672-136674] Shaun Ruffell * codecs/codec_dahdi.c: Removing code that was commented out. * codecs/codec_dahdi.c: Updated codec_dahdi to use the transcoder interface in the DAHDI. (Issue: DAHDI-42) 2008-08-07 20:26 +0000 [r136632-136663] Mark Michelson * /, main/features.c: Merged revisions 136660 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r136660 | mmichelson | 2008-08-07 15:25:43 -0500 (Thu, 07 Aug 2008) | 4 lines Bump a LOG_NOTICE message to LOG_DEBUG since it appears once for every bridged call ........ * main/pbx.c, /: Merged revisions 136635 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r136635 | mmichelson | 2008-08-07 14:58:32 -0500 (Thu, 07 Aug 2008) | 5 lines Don't allow Answer() to accept a negative argument. Negative argument means an infinite delay and we don't want that. ........ * main/channel.c, /: Merged revisions 136633 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r136633 | mmichelson | 2008-08-07 14:54:27 -0500 (Thu, 07 Aug 2008) | 7 lines Fix a calculation error I had made in the poll. The poll would reset to 500 ms every time a non-voice frame was received. The total time we poll should be 500 ms, so now we save the amount of time left after the poll returned and use that as our argument for the next call to poll ........ * main/channel.c, /: Merged revisions 136631 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r136631 | mmichelson | 2008-08-07 14:36:46 -0500 (Thu, 07 Aug 2008) | 13 lines Scrap the 500 ms delay when Asterisk auto-answers a channel. Instead, poll the channel until receiving a voice frame. The cap on this poll is 500 ms. The optional delay is still allowable in the Answer() application, but the delay has been moved back to its original position, after the call to the channel's answer callback. The poll for the voice frame will not happen if a delay is specified when calling Answer(). (closes issue #12708) Reported by: kactus ........ 2008-08-07 19:19 +0000 [r136598] Richard Mudgett * channels/misdn_config.c, channels/chan_misdn.c, /, configs/misdn.conf.sample: Merged revisions 136594 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r136594 | rmudgett | 2008-08-07 14:01:03 -0500 (Thu, 07 Aug 2008) | 13 lines Merged revisions 136241 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r136241 | rmudgett | 2008-08-06 16:18:53 -0500 (Wed, 06 Aug 2008) | 5 lines * The allowed_bearers setting in misdn.conf misspelled one of its options: digital_restricted. * Fixed some other spelling errors and typos. ........ ................ 2008-08-07 17:44 +0000 [r136506-136543] Kevin P. Fleming * include/asterisk/doxyref.h, /: Merged revisions 136542 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r136542 | kpfleming | 2008-08-07 12:44:20 -0500 (Thu, 07 Aug 2008) | 6 lines Merged revisions 136541 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ ........ ................ 2008-08-07 16:57 +0000 [r136490] Tilghman Lesher * /, apps/app_queue.c: Merged revisions 136489 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r136489 | tilghman | 2008-08-07 11:55:57 -0500 (Thu, 07 Aug 2008) | 15 lines Merged revisions 136488 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r136488 | tilghman | 2008-08-07 11:50:47 -0500 (Thu, 07 Aug 2008) | 7 lines Update persistent state on all exit conditions. (closes issue #12916) Reported by: sgenyuk Patches: app_queue.patch.txt uploaded by neutrino88 (license 297) Tested by: sgenyuk, aragon ........ ................ 2008-08-06 20:16 +0000 [r136113-136192] Tilghman Lesher * /, contrib/init.d/rc.redhat.asterisk: Merged revisions 136191 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r136191 | tilghman | 2008-08-06 15:15:34 -0500 (Wed, 06 Aug 2008) | 12 lines Merged revisions 136190 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r136190 | tilghman | 2008-08-06 15:14:54 -0500 (Wed, 06 Aug 2008) | 4 lines -C option takes a filename, not a directory path. (closes issue #13007) Reported by: klaus3000 ........ ................ * /, funcs/func_dialgroup.c: Merged revisions 136112 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r136112 | tilghman | 2008-08-06 11:58:42 -0500 (Wed, 06 Aug 2008) | 7 lines Persist DIALGROUP() values in astdb (closes issue #13138) Reported by: Corydon76 Patches: 20080725__bug13138.diff.txt uploaded by Corydon76 (license 14) Tested by: pj ........ 2008-08-06 16:00 +0000 [r136064] Mark Michelson * main/rtp.c, /, channels/chan_skinny.c: Merged revisions 136063 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r136063 | mmichelson | 2008-08-06 10:59:29 -0500 (Wed, 06 Aug 2008) | 24 lines Merged revisions 136062 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r136062 | mmichelson | 2008-08-06 10:58:40 -0500 (Wed, 06 Aug 2008) | 16 lines Since adding the AST_CONTROL_SRCUPDATE frame type, there are places where ast_rtp_new_source may be called where the tech_pvt of a channel may not yet have an rtp structure allocated. This caused a crash in chan_skinny, which was fixed earlier, but now the same crash has been reported against chan_h323 as well. It seems that the best solution is to modify ast_rtp_new_source to not attempt to set the marker bit if the rtp structure passed in is NULL. This change to ast_rtp_new_source also allows the removal of what is now a redundant pointer check from chan_skinny. (closes issue #13247) Reported by: pj ........ ................ 2008-08-06 13:59 +0000 [r136006] Olle Johansson * /, res/res_jabber.c: Merged revisions 136005 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r136005 | oej | 2008-08-06 15:34:08 +0200 (Ons, 06 Aug 2008) | 6 lines - Formatting - Changing debug messages from VERBOSE to DEBUG channel - Adding a few todo's - Adding a few more "XMPP"'s to compliment Jabber... ........ 2008-08-06 03:56 +0000 [r135901-135951] Tilghman Lesher * main/channel.c, /: Merged revisions 135950 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135950 | tilghman | 2008-08-05 22:55:49 -0500 (Tue, 05 Aug 2008) | 12 lines Merged revisions 135949 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135949 | tilghman | 2008-08-05 22:53:36 -0500 (Tue, 05 Aug 2008) | 4 lines Fix a longstanding bug in channel walking logic, and fix the explanation to make sense. (Closes issue #13124) ........ ................ * /, main/translate.c: Merged revisions 135938 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135938 | tilghman | 2008-08-05 22:29:42 -0500 (Tue, 05 Aug 2008) | 12 lines Merged revisions 135915 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135915 | tilghman | 2008-08-05 22:24:56 -0500 (Tue, 05 Aug 2008) | 4 lines Since powerof() can return an error condition, it's foolhardy not to detect and deal with that condition. (Related to issue #13240) ........ ................ * include/asterisk/threadstorage.h, include/asterisk/utils.h, /: Merged revisions 135900 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135900 | tilghman | 2008-08-05 22:04:01 -0500 (Tue, 05 Aug 2008) | 12 lines Merged revisions 135899 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135899 | tilghman | 2008-08-05 22:02:59 -0500 (Tue, 05 Aug 2008) | 4 lines 1) Bugfix for debugging code 2) Reduce compiler warnings for another section of debugging code (Closes issue #13237) ........ ................ 2008-08-06 00:31 +0000 [r135852] Mark Michelson * include/asterisk/abstract_jb.h, main/channel.c, /, main/abstract_jb.c, main/fixedjitterbuf.h: Merged revisions 135851 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135851 | mmichelson | 2008-08-05 19:30:53 -0500 (Tue, 05 Aug 2008) | 48 lines Merged revisions 135841,135847,135850 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135841 | mmichelson | 2008-08-05 19:25:10 -0500 (Tue, 05 Aug 2008) | 27 lines Merging the issue11259 branch. The purpose of this branch was to take into account "burps" which could cause jitterbuffers to misbehave. One such example is if the L option to Dial() were used to inject audio into a bridged conversation at regular intervals. Since the audio here was not passed through the jitterbuffer, it would cause a gap in the jitterbuffer's timestamps which would cause a frames to be dropped for a brief period. Now ast_generic_bridge will empty and reset the jitterbuffer each time it is called. This causes injected audio to be handled properly. ast_generic_bridge also will empty and reset the jitterbuffer if it receives an AST_CONTROL_SRCUPDATE frame since the change in audio source could negatively affect the jitterbuffer. All of this was made possible by adding a new public API call to the abstract_jb called ast_jb_empty_and_reset. (closes issue #11259) Reported by: plack Tested by: putnopvut ........ r135847 | mmichelson | 2008-08-05 19:27:54 -0500 (Tue, 05 Aug 2008) | 4 lines Revert inadvertent changes to app_skel that occurred when I was testing for a memory leak ........ r135850 | mmichelson | 2008-08-05 19:29:54 -0500 (Tue, 05 Aug 2008) | 3 lines Remove properties that should not be here ........ ................ 2008-08-05 23:52 +0000 [r135822] Steve Murphy * apps/app_dial.c, main/cdr.c, main/channel.c, /, main/features.c, include/asterisk/cdr.h: Merged revisions 135821 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135821 | murf | 2008-08-05 17:45:32 -0600 (Tue, 05 Aug 2008) | 42 lines Merged revisions 135799 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135799 | murf | 2008-08-05 17:13:20 -0600 (Tue, 05 Aug 2008) | 34 lines (closes issue #12982) Reported by: bcnit Tested by: murf I discovered that also, in the previous bug fixes and changes, the cdr.conf 'unanswered' option is not being obeyed, so I fixed this. And, yes, there are two 'answer' times involved in this scenario, and I would agree with you, that the first answer time is the time that should appear in the CDR. (the second 'answer' time is the time that the bridge was begun). I made the necessary adjustments, recording the first answer time into the peer cdr, and then using that to override the bridge cdr's value. To get the 'unanswered' CDRs to appear, I purposely output them, using the dial cmd to mark them as DIALED (with a new flag), and outputting them if they bear that flag, and you are in the right mode. I also corrected one small mention of the Zap device to equally consider the dahdi device. I heavily tested 10-sec-wait macros in dial, and without the macro call; I tested hangups while the macro was running vs. letting the macro complete and the bridge form. Looks OK. Removed all the instrumentation and debug. ........ ................ 2008-08-05 21:38 +0000 [r135749] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 135748 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135748 | tilghman | 2008-08-05 16:37:35 -0500 (Tue, 05 Aug 2008) | 17 lines Merged revisions 135747 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135747 | tilghman | 2008-08-05 16:34:46 -0500 (Tue, 05 Aug 2008) | 9 lines In a conversion to use ast_strlen_zero, the meaning of the flag IAX_HASCALLERID was perverted. This change reverts IAX2 to the original meaning, which was, that the callerid set on the client should be overridden on the server, even if that means the resulting callerid is blank. In other words, if you set "callerid=" in the IAX config, then the callerid should be overridden to blank, even if set on the client. Note that there's a distinction, even on realtime, between the field not existing (NULL in databases) and the field existing, but set to blank (override callerid to blank). ........ ................ 2008-08-05 13:27 +0000 [r135599] Sean Bright * main/cli.c, /: Merged revisions 135598 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135598 | seanbright | 2008-08-05 09:26:34 -0400 (Tue, 05 Aug 2008) | 9 lines Merged revisions 135597 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135597 | seanbright | 2008-08-05 09:25:00 -0400 (Tue, 05 Aug 2008) | 1 line Use PATH_MAX for filenames ........ ................ 2008-08-04 20:15 +0000 [r135538] Russell Bryant * configs/chan_dahdi.conf.sample, /: Merged revisions 135537 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135537 | russell | 2008-08-04 15:15:27 -0500 (Mon, 04 Aug 2008) | 10 lines Merged revisions 135536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135536 | russell | 2008-08-04 15:15:03 -0500 (Mon, 04 Aug 2008) | 2 lines fix a config sample typo ........ ................ 2008-08-04 17:12 +0000 [r135478-135486] Tilghman Lesher * contrib/init.d/rc.mandriva.asterisk (added), Makefile, contrib/init.d/rc.mandrake.asterisk (removed), /, contrib/init.d/rc.mandriva.zaptel (added), contrib/init.d/rc.mandrake.zaptel (removed): Merged revisions 135485 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r135485 | tilghman | 2008-08-04 12:12:15 -0500 (Mon, 04 Aug 2008) | 3 lines Rename Mandrake scripts to Mandriva (Closes issue #13221) ........ * contrib/init.d/rc.mandrake.asterisk, /: Merged revisions 135483 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135483 | tilghman | 2008-08-04 12:08:42 -0500 (Mon, 04 Aug 2008) | 11 lines Merged revisions 135482 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135482 | tilghman | 2008-08-04 12:07:52 -0500 (Mon, 04 Aug 2008) | 2 lines Define ASTSBINDIR for script (Closes issue #13221) ........ ................ * apps/app_voicemail.c, /: Merged revisions 135480 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135480 | tilghman | 2008-08-04 11:58:29 -0500 (Mon, 04 Aug 2008) | 14 lines Merged revisions 135479 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135479 | tilghman | 2008-08-04 11:56:19 -0500 (Mon, 04 Aug 2008) | 6 lines Memory leak on unload (closes issue #13231) Reported by: eliel Patches: app_voicemail.leak.patch uploaded by eliel (license 64) ........ ................ 2008-08-04 16:28 +0000 [r135440-135475] Russell Bryant * configs/chan_dahdi.conf.sample, /: Merged revisions 135474 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135474 | russell | 2008-08-04 11:28:07 -0500 (Mon, 04 Aug 2008) | 10 lines Merged revisions 135473 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135473 | russell | 2008-08-04 11:26:17 -0500 (Mon, 04 Aug 2008) | 2 lines Add a minor clarification to the documentation of mohinterpret and mohsuggest ........ ................ * /, channels/chan_console.c: Merged revisions 135439 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r135439 | russell | 2008-08-04 10:02:12 -0500 (Mon, 04 Aug 2008) | 4 lines Be explicit that we don't want a result from this callback. The callback would never indicate a match, so nothing would have been returned anyway, but it was still a poor example of proper usage. ........ 2008-08-02 05:15 +0000 [r135266] Steve Murphy * main/pbx.c, /: Merged revisions 135265 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r135265 | murf | 2008-08-01 22:51:29 -0600 (Fri, 01 Aug 2008) | 31 lines (closes issue #13202) Reported by: falves11 Tested by: murf falves11 == The changes I introduce here seem to clear up the problem for me. However, if they do not for you, please reopen this bug, and we'll keep digging. The root of this problem seems to be a subtle memory corruption introduced when creating an extension with an empty extension name. While valgrind cannot detect it outside of DEBUG_MALLOC mode, when compiled with DEBUG_MALLOC, this is certain death. The code in main/features.c is a puzzle to me. On the initial module load, the code is attempting to add the parking extension before the features.conf file has even been opened! I just wrapped the offending call with an if() that will not try to add the extension if the extension name is empty. THis seems to solve the corruption, and let the "memory show allocations" work as one would expect. But, really, adding an extension with an empty name is a seriously bad thing to allow, as it will mess up all the pattern matching algorithms, etc. So, I added a statement to the add_extension2 code to return a -1 if this is attempted. in 1.6.0, the changes to only main/pbx.c were applicable, as apparently the code added to main/features by jpeeler were not included in 1.6.0. ........ 2008-08-01 19:30 +0000 [r135198] Sean Bright * channels/chan_mgcp.c, /: Merged revisions 135197 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r135197 | seanbright | 2008-08-01 15:29:26 -0400 (Fri, 01 Aug 2008) | 6 lines Remove some code that used to do something but does not anymore, mainly to get rid of a shadow warning (but this seemed legitimate enough to fix here instead of in my branch). Thanks to putnopvut for taking a look as well. ........ 2008-08-01 17:10 +0000 [r135127-135129] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 135128 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r135128 | tilghman | 2008-08-01 12:09:50 -0500 (Fri, 01 Aug 2008) | 2 lines Picky, picky, buildbot ........ * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 135126 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r135126 | tilghman | 2008-08-01 11:39:51 -0500 (Fri, 01 Aug 2008) | 9 lines SIP should use the transport type set in the Moved Temporarily for the next invite. (closes issue #11843) Reported by: pestermann Patches: 20080723__issue11843_302_ignores_transport_16branch.diff uploaded by bbryant (license 36) 20080723__issue11843_302_ignores_transport_trunk.diff uploaded by bbryant (license 36) Tested by: pabelanger ........ 2008-08-01 14:43 +0000 [r135070] Mark Michelson * apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged revisions 135067-135068 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r135067 | mmichelson | 2008-08-01 09:29:48 -0500 (Fri, 01 Aug 2008) | 13 lines IMAP storage functioned under the assumption that folders such as "Work" and "Family" would be subfolders of the INBOX. This is an invalid assumption to make, but it could be desirable to set up folders in this manner, so a new option for voicemail.conf, "imapparentfolder" has been added to allow for this. (closes issue #13142) Reported by: jaroth Patches: parentfolder.patch uploaded by jaroth (license 50) ........ r135068 | mmichelson | 2008-08-01 09:42:24 -0500 (Fri, 01 Aug 2008) | 3 lines IMAP-specific items must go in IMAP_STORAGE defines... ........ 2008-08-01 12:18 +0000 [r135057-135062] Michiel van Baak * /, apps/app_ices.c: Merged revisions 135059 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135059 | mvanbaak | 2008-08-01 13:47:34 +0200 (Fri, 01 Aug 2008) | 10 lines Merged revisions 135058 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135058 | mvanbaak | 2008-08-01 13:43:46 +0200 (Fri, 01 Aug 2008) | 2 lines make app_ices compile on OpenBSD. ........ ................ * /, channels/chan_skinny.c: Merged revisions 135056 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135056 | mvanbaak | 2008-08-01 13:00:13 +0200 (Fri, 01 Aug 2008) | 16 lines Merged revisions 135055 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135055 | mvanbaak | 2008-08-01 12:55:27 +0200 (Fri, 01 Aug 2008) | 8 lines fix some potential deadlocks in chan_skinny (closes issue #13215) Reported by: qwell Patches: 2008080100_bug13215.diff.txt uploaded by mvanbaak (license 7) Tested by: mvanbaak ........ ................ 2008-07-31 22:34 +0000 [r135034] Kevin P. Fleming * /, main/http.c: Merged revisions 135016 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135016 | kpfleming | 2008-07-31 17:28:42 -0500 (Thu, 31 Jul 2008) | 11 lines Merged revisions 134983 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134983 | kpfleming | 2008-07-31 17:18:11 -0500 (Thu, 31 Jul 2008) | 3 lines accomodate users who seem to lack a sense of humor :-) ........ ................ 2008-07-31 21:58 +0000 [r134926-134981] Tilghman Lesher * sample.call, main/manager.c, pbx/pbx_spool.c, /: Merged revisions 134980 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134980 | tilghman | 2008-07-31 16:55:42 -0500 (Thu, 31 Jul 2008) | 16 lines Blocked revisions 134976 via svnmerge ........ r134976 | tilghman | 2008-07-31 16:53:19 -0500 (Thu, 31 Jul 2008) | 9 lines Specify codecs in callfiles and manager, to allow video calls to be set up from callfiles and AMI. (closes issue #9531) Reported by: Geisj Patches: 20080715__bug9531__1.4.diff.txt uploaded by Corydon76 (license 14) 20080715__bug9531__1.6.0.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ ................ * res/res_config_sqlite.c, /: Merged revisions 134977 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r134977 | tilghman | 2008-07-31 16:53:59 -0500 (Thu, 31 Jul 2008) | 2 lines Switch command order, to meet with current specs ........ 2008-07-31 19:54 +0000 [r134923] Steve Murphy * /, main/features.c: Merged revisions 134922 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134922 | murf | 2008-07-31 13:48:08 -0600 (Thu, 31 Jul 2008) | 63 lines Merged revisions 134883 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134883 | murf | 2008-07-31 13:23:42 -0600 (Thu, 31 Jul 2008) | 51 lines (closes issue #11849) Reported by: greyvoip Tested by: murf OK, a few days of debugging, a bunch of instrumentation in chan_sip, main/channel.c, main/pbx.c, etc. and 5 solid notebook pages of notes later, I have made the small tweek necc. to get the start time right on the second CDR when: A Calls B B answ. A hits Xfer button on sip phone, A dials C and hits the OK button, A hangs up C answers ringing phone B and C converse B and/or C hangs up But does not harm the scenario where: A Calls B B answ. B hits xfer button on sip phone, B dials C and hits the OK button, B hangs up C answers ringing phone A and C converse A and/or C hangs up The difference in start times on the second CDR is because of a Masquerade on the B channel when the xfer number is sent. It ends up replacing the CDR on the B channel with a duplicate, which ends up getting tossed out. We keep a pointer to the first CDR, and update *that* after the bridge closes. But, only if the CDR has changed. I hope this change is specific enough not to muck up any current CDR-based apps. In my defence, I assert that the previous information was wrong, and this change fixes it, and possibly other similar scenarios. I wonder if I should be doing the same thing for the channel, as I did for the peer, but I can't think of a scenario this might affect. I leave it, then, as an exersize for the users, to find the scenario where the chan's CDR changes and loses the proper start time. ........ ................ 2008-07-31 19:41 +0000 [r134918] Russell Bryant * /, apps/app_ices.c: Merged revisions 134917 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134917 | russell | 2008-07-31 14:39:50 -0500 (Thu, 31 Jul 2008) | 17 lines Merged revisions 134915 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134915 | russell | 2008-07-31 14:37:26 -0500 (Thu, 31 Jul 2008) | 9 lines Get app_ices working again (closes issue #12981) Reported by: dlogan Patches: 20080709__app_ices_v2_update_trunk.diff uploaded by bbryant (license 36) 20080709__app_ices_v2_update_14.diff uploaded by bbryant (license 36) Tested by: bbryant ........ ................ 2008-07-31 16:53 +0000 [r134816] Russell Bryant * channels/iax2-parser.c: Merged revisions 134815 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134815 | russell | 2008-07-31 11:50:10 -0500 (Thu, 31 Jul 2008) | 15 lines Merged revisions 134814 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134814 | russell | 2008-07-31 11:45:31 -0500 (Thu, 31 Jul 2008) | 7 lines In case we have some processing threads that free more frames than they allocate, do not let the frame cache grow forever. (closes issue #13160) Reported by: tavius Tested by: tavius, russell ........ ................ 2008-07-31 16:07 +0000 [r134760] Mark Michelson * /, apps/app_queue.c: Merged revisions 134759 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134759 | mmichelson | 2008-07-31 11:05:12 -0500 (Thu, 31 Jul 2008) | 24 lines Merged revisions 134758 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134758 | mmichelson | 2008-07-31 10:56:18 -0500 (Thu, 31 Jul 2008) | 16 lines Add more timeout checks into app_queue, specifically targeting areas where an unknown and potentially long time has just elapsed. Also added a check to try_calling() to return early if the timeout has elapsed instead of potentially setting a negative timeout for the call (thus making it have *no* timeout at all). (closes issue #13186) Reported by: miquel_cabrespina Patches: 13186.diff uploaded by putnopvut (license 60) Tested by: miquel_cabrespina ........ ................ 2008-07-30 22:41 +0000 [r134651-134707] Tilghman Lesher * main/sched.c, /, include/asterisk/sched.h: Merged revisions 134703 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r134703 | tilghman | 2008-07-30 17:38:58 -0500 (Wed, 30 Jul 2008) | 2 lines Oops, wrong define ........ * /, configure, configure.ac: Merged revisions 134650 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134650 | tilghman | 2008-07-30 16:40:08 -0500 (Wed, 30 Jul 2008) | 12 lines Merged revisions 134649 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134649 | tilghman | 2008-07-30 16:38:50 -0500 (Wed, 30 Jul 2008) | 4 lines Qwell pointed out, via IRC, that the previous fix only worked when explicitly set. When nothing is set, and the option is implied, it breaks, because configure sets the prefix to 'NONE'. Fixing. ........ ................ 2008-07-30 21:06 +0000 [r134599] Mark Michelson * /, apps/app_queue.c: Merged revisions 134598 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134598 | mmichelson | 2008-07-30 16:05:37 -0500 (Wed, 30 Jul 2008) | 15 lines Merged revisions 134556 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r134556 | mmichelson | 2008-07-30 15:24:40 -0500 (Wed, 30 Jul 2008) | 7 lines Fix the parsing of the "reason" parameter in the Diversion: header. (closes issue #13195) Reported by: woodsfsg ........ ................ 2008-07-30 20:39 +0000 [r134597] Russell Bryant * /, pbx/pbx_dundi.c: Merged revisions 134596 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134596 | russell | 2008-07-30 15:38:35 -0500 (Wed, 30 Jul 2008) | 14 lines Merged revisions 134595 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134595 | russell | 2008-07-30 15:37:17 -0500 (Wed, 30 Jul 2008) | 6 lines Reduce stack consumption by 12.5% of the max stack size to fix a crash when compiled with LOW_MEMORY. (closes issue #13154) Reported by: edantie ........ ................ 2008-07-30 20:25 +0000 [r134561] Mark Michelson * /, channels/chan_sip.c: Merged revisions 134556 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r134556 | mmichelson | 2008-07-30 15:24:40 -0500 (Wed, 30 Jul 2008) | 7 lines Fix the parsing of the "reason" parameter in the Diversion: header. (closes issue #13195) Reported by: woodsfsg ........ 2008-07-30 19:56 +0000 [r134542] Russell Bryant * funcs/func_curl.c, /: Merged revisions 134541 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134541 | russell | 2008-07-30 14:55:31 -0500 (Wed, 30 Jul 2008) | 12 lines Merged revisions 134540 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134540 | russell | 2008-07-30 14:52:53 -0500 (Wed, 30 Jul 2008) | 4 lines Fix a memory leak in func_curl. Every thread that used this function leaked an allocation the size of a pointer. (reported by jmls in #asterisk-dev) ........ ................ 2008-07-30 19:49 +0000 [r134482-134539] Tilghman Lesher * /, configure, configure.ac: Merged revisions 134538 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134538 | tilghman | 2008-07-30 14:48:37 -0500 (Wed, 30 Jul 2008) | 12 lines Merged revisions 134536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134536 | tilghman | 2008-07-30 14:47:16 -0500 (Wed, 30 Jul 2008) | 4 lines Only override sysconfdir and mandir when prefix=/usr (closes issue #13093) Reported by: pabelanger ........ ................ * /, apps/app_queue.c: Merged revisions 134483 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r134483 | tilghman | 2008-07-30 14:17:38 -0500 (Wed, 30 Jul 2008) | 4 lines Let "roundrobin" also reference rrmemory, for the 1.6 release (as described in UPGRADE-1.4.txt) (Closes issue #13181) ........ * /, res/res_agi.c: Merged revisions 134481 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134481 | tilghman | 2008-07-30 14:05:35 -0500 (Wed, 30 Jul 2008) | 13 lines Merged revisions 134480 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134480 | tilghman | 2008-07-30 14:03:44 -0500 (Wed, 30 Jul 2008) | 5 lines launch_netscript sometimes returns -1, which fails to set AGISTATUS. Map failure to -1, so that AGISTATUS is always set. (closes issue #13199) Reported by: smw1218 ........ ................ 2008-07-30 18:33 +0000 [r134477] Mark Michelson * /, main/app.c: Merged revisions 134476 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134476 | mmichelson | 2008-07-30 13:33:12 -0500 (Wed, 30 Jul 2008) | 12 lines Merged revisions 134475 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134475 | mmichelson | 2008-07-30 13:31:47 -0500 (Wed, 30 Jul 2008) | 4 lines Fix a spot where a function could return without bringing a channel out of autoservice. ........ ................ 2008-07-30 15:34 +0000 [r134356] Kevin P. Fleming * Makefile, /: Merged revisions 134355 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134355 | kpfleming | 2008-07-30 10:32:14 -0500 (Wed, 30 Jul 2008) | 10 lines Merged revisions 134352 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134352 | kpfleming | 2008-07-30 10:29:17 -0500 (Wed, 30 Jul 2008) | 2 lines use the proper method for building version.h ........ ................ 2008-07-29 22:29 +0000 [r134283] Kevin P. Fleming * apps/app_rpt.c, apps/app_dahdibarge.c, channels/chan_dahdi.c, /, apps/app_meetme.c, apps/app_dahdiscan.c, apps/app_dahdiras.c: Merged revisions 134260 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r134260 | kpfleming | 2008-07-29 17:22:13 -0500 (Tue, 29 Jul 2008) | 2 lines build against the now-typedef-free dahdi/user.h, and remove some #ifdefs for features that will always be present in DAHDI ........ 2008-07-28 22:16 +0000 [r134164] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 134163 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134163 | tilghman | 2008-07-28 17:07:12 -0500 (Mon, 28 Jul 2008) | 15 lines Merged revisions 134161 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134161 | tilghman | 2008-07-28 16:50:50 -0500 (Mon, 28 Jul 2008) | 7 lines Detect when sox fails to raise the volume, because sox can't read the file. (closes issue #12939) Reported by: rickbradley Patches: 20080728__bug12939.diff.txt uploaded by Corydon76 (license 14) Tested by: rickbradley ........ ................ 2008-07-28 19:55 +0000 [r134126] Mark Michelson * /, configure, main/Makefile, configure.ac, CHANGES: Merged revisions 134125 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r134125 | mmichelson | 2008-07-28 14:53:56 -0500 (Mon, 28 Jul 2008) | 27 lines This commit compensates for buggy poll(2) implementations. Asterisk has, for a long time, had its own implementation of poll(2) which just used the input arguments to call select(2). In 1.4, this internal implementation was used for Darwin systems. This was removed in Asterisk trunk at some point, but it seems as though this was not the right move to make. On Mac OS X, it appears as though the poll used to gather CLI input does not respond properly when connecting via a remote Asterisk console. Reverting to the use of Asterisk's poll fixed the issue. Also, there is now an option for the configure script, --enable-internal-poll, which will allow for anyone to use Asterisk's internal poll implementation in case they suspect that their system's poll implementation is buggy. closes issue #11928) Reported by: adriavidal Patches: 1.6.0-configurev2.patch uploaded by putnopvut (license 60) ........ 2008-07-28 16:49 +0000 [r134087] Kevin P. Fleming * apps/app_parkandannounce.c, main/loader.c, sample.call, contrib/scripts/autosupport, build_tools/cflags.xml, main/channel.c, apps/app_dahdibarge.c, channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, doc/ss7.txt, /, main/features.c, doc/osp.txt, main/file.c, pbx/pbx_config.c: Merged revisions 134086 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r134086 | kpfleming | 2008-07-28 11:42:00 -0500 (Mon, 28 Jul 2008) | 3 lines remove remaining Zaptel references in various places ........ 2008-07-28 16:13 +0000 [r134052] Mark Michelson * apps/app_chanspy.c, include/asterisk/options.h, main/asterisk.c, /, apps/app_meetme.c, apps/app_dahdiscan.c: Merged revisions 134050 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r134050 | mmichelson | 2008-07-28 11:00:19 -0500 (Mon, 28 Jul 2008) | 3 lines merging the zap_and_dahdi_trunk branch up to trunk ........ 2008-07-26 15:34 +0000 [r133942-133982] Russell Bryant * main/asterisk.c, include/asterisk/doxyref.h, /: Include the licensing page in 1.6.0 as well. Now, this page exists in 1.4, trunk, and 1.6.0. * /: unblock 133575 * /, main/devicestate.c: Merged revisions 133945-133946 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r133945 | russell | 2008-07-26 10:15:14 -0500 (Sat, 26 Jul 2008) | 6 lines ast_device_state() gets called in two different ways. The first way is when called from elsewhere in Asterisk to find the current state of a device. In that case, we want to use the cached value if it exists. The other way is when processing a device state change. In that case, we do not want to check the cache because returning the last known state is counter productive. ........ r133946 | russell | 2008-07-26 10:16:20 -0500 (Sat, 26 Jul 2008) | 1 line actually use the cache_cache argument ........ 2008-07-25 22:09 +0000 [r133863-133905] Tilghman Lesher * contrib/scripts/asterisk.ldif, /: Merged revisions 133902 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r133902 | tilghman | 2008-07-25 16:59:39 -0500 (Fri, 25 Jul 2008) | 6 lines Update version (closes issue #13163) Reported by: suretec Patches: asterisk.ldif uploaded by suretec (license 70) ........ 2008-07-25 19:37 +0000 [r133804-133806] Brandon Kruse * /: Blocking revert of code changes in r133770 * main/http.c: Include the http_decode function from trunk to replace the + with a space. 2008-07-25 17:33 +0000 [r133694] Brandon Kruse * /: Blocking a fix from trunk for the function http_decode. 1.6.0 does not have this function. 2008-07-25 17:26 +0000 [r133671] Tilghman Lesher * main/channel.c, /, channels/chan_agent.c, main/devicestate.c: Merged revisions 133665 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r133665 | tilghman | 2008-07-25 12:24:43 -0500 (Fri, 25 Jul 2008) | 16 lines Merged revisions 133649 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133649 | tilghman | 2008-07-25 12:19:39 -0500 (Fri, 25 Jul 2008) | 8 lines Fix some errant device states by making the devicestate API more strict in terms of the device argument (only without the unique identifier appended). (closes issue #12771) Reported by: davidw Patches: 20080717__bug12771.diff.txt uploaded by Corydon76 (license 14) Tested by: davidw, jvandal, murf ........ ................ 2008-07-25 15:01 +0000 [r133576-133580] Russell Bryant * /, LICENSE: Merged revisions 133579 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r133579 | russell | 2008-07-25 10:00:49 -0500 (Fri, 25 Jul 2008) | 18 lines Merged revisions 133578 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r133578 | russell | 2008-07-25 10:00:31 -0500 (Fri, 25 Jul 2008) | 10 lines Merged revisions 133577 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r133577 | russell | 2008-07-25 10:00:13 -0500 (Fri, 25 Jul 2008) | 2 lines Fix the IAX2 URI for calling Digium ........ ................ ................ 2008-07-25 14:41 +0000 [r133571-133574] Mark Michelson * /, channels/chan_sip.c: Merged revisions 133573 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r133573 | mmichelson | 2008-07-25 09:40:52 -0500 (Fri, 25 Jul 2008) | 15 lines Merged revisions 133572 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133572 | mmichelson | 2008-07-25 09:40:10 -0500 (Fri, 25 Jul 2008) | 7 lines We need to make sure to null-terminate the "name" portion of SIP URI parameters so that there are no bogus comparisons. Thanks to bbryant for pointing this out. ........ ................ 2008-07-25 13:25 +0000 [r133567-133569] Russell Bryant * /, channels/chan_sip.c: Merged revisions 133568 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r133568 | russell | 2008-07-25 08:01:59 -0500 (Fri, 25 Jul 2008) | 4 lines Minor coding guidelines tweaks ... - Use ast_strlen_zero in one place - check for successful string comparison the way most of Asterisk code does it ........ 2008-07-24 21:31 +0000 [r133524] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 133509 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r133509 | tilghman | 2008-07-24 16:27:06 -0500 (Thu, 24 Jul 2008) | 11 lines Merged revisions 133488 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133488 | tilghman | 2008-07-24 16:17:55 -0500 (Thu, 24 Jul 2008) | 3 lines Fix rtautoclear and rtcachefriends (Closes issue #12707) ........ ................ 2008-07-24 20:41 +0000 [r133487] Russell Bryant * /, channels/chan_agent.c: Merged revisions 133486 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r133486 | russell | 2008-07-24 15:40:15 -0500 (Thu, 24 Jul 2008) | 3 lines I made this change from DEVICE_STATE to DEVICE_STATE_CHANGE, but I had it backwards, this is the right event to subscribe to ... ........ 2008-07-24 19:55 +0000 [r133449] Mark Michelson * /, main/logger.c: Merged revisions 133448 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r133448 | mmichelson | 2008-07-24 14:53:37 -0500 (Thu, 24 Jul 2008) | 12 lines Print the correct PID in log messages. Prior to this commit, only the logger thread's PID would be printed. (closes issue #13150) Reported by: atis Patches: log_pid.diff uploaded by putnopvut (license 60) Tested by: eliel ........ 2008-07-24 05:21 +0000 [r133392-133405] Tilghman Lesher * contrib/scripts/asterisk.logrotate, Makefile, /: Merged revisions 133400 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r133400 | tilghman | 2008-07-24 00:21:00 -0500 (Thu, 24 Jul 2008) | 3 lines Build the logrotate script according to paths (Closes issue #13147) ........ * Makefile, /: Merged revisions 133391 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r133391 | tilghman | 2008-07-23 23:51:42 -0500 (Wed, 23 Jul 2008) | 3 lines Optionally install logrotate file (Closes issue #13148) ........ 2008-07-23 22:07 +0000 [r133300] Steve Murphy * main/pbx.c, /: Merged revisions 133299 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r133299 | murf | 2008-07-23 16:03:48 -0600 (Wed, 23 Jul 2008) | 27 lines (closes issue #13144) Reported by: murf Tested by: murf For: J. Geis The 'data' field in the ast_exten struct was being 'moved' from the current dialplan to the replacement dialplan. This was not good, as the current dialplan could have problems in the time between the change and when the new dialplan is swapped in. So, I modified the merge_and_delete code to strdup the 'data' field (the args to the app call), and then it's freed as normal. I improved a few messages; I added code to limit the number of calls to the context_merge_incls_swits_igps_other_registrars() to one per context. I don't think having it called multiple times per context was doing anything bad, but it was inefficient. I hope this fixes the problems Mr. Geiss was noting in asterisk-users, see http://lists.digium.com/pipermail/asterisk-users/2008-July/215634.html ........ 2008-07-23 21:50 +0000 [r133297] Jason Parker * channels/chan_dahdi.c, /: Merged revisions 133296 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r133296 | qwell | 2008-07-23 16:50:20 -0500 (Wed, 23 Jul 2008) | 9 lines Merged revisions 133295 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133295 | qwell | 2008-07-23 16:49:03 -0500 (Wed, 23 Jul 2008) | 1 line inbandrelease is gone - it's now inbanddisconnect ........ ................ 2008-07-23 20:39 +0000 [r133218] Brett Bryant * /, channels/chan_sip.c: Merged revisions 133197 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r133197 | bbryant | 2008-07-23 15:33:22 -0500 (Wed, 23 Jul 2008) | 2 lines Fix issue where tcp in sip is enabled by default, despite what it says in the config sample file. Also fix "sip show settings" for tcp connections. ........ 2008-07-23 19:50 +0000 [r133042-133172] Mark Michelson * apps/app_chanspy.c, include/asterisk/options.h, main/asterisk.c, /: Merged revisions 133171 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r133171 | mmichelson | 2008-07-23 14:48:03 -0500 (Wed, 23 Jul 2008) | 20 lines Merged revisions 133169 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133169 | mmichelson | 2008-07-23 14:39:47 -0500 (Wed, 23 Jul 2008) | 12 lines As suggested by seanbright, the PSEUDO_CHAN_LEN in app_chanspy should be set at load time, not at compile time, since dahdi_chan_name is determined at load time. Also changed the next_unique_id_to_use to have the static qualifier. Also added the dahdi_chan_name_len variable so that strlen(dahdi_chan_name) isn't necessary. Thanks to seanbright for the suggestion. ........ ................ * apps/app_chanspy.c, /: Merged revisions 133106 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r133106 | mmichelson | 2008-07-23 14:07:56 -0500 (Wed, 23 Jul 2008) | 13 lines Merged revisions 133104 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133104 | mmichelson | 2008-07-23 14:06:16 -0500 (Wed, 23 Jul 2008) | 5 lines Zap/pseudo is ten characters, but DAHDI/pseudo is twelve. The strncmp call in next_channel should account for this. ........ ................ * apps/app_chanspy.c, /: Merged revisions 133102 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r133102 | mmichelson | 2008-07-23 13:58:37 -0500 (Wed, 23 Jul 2008) | 14 lines Merged revisions 133101 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133101 | mmichelson | 2008-07-23 13:57:17 -0500 (Wed, 23 Jul 2008) | 6 lines Update the "last" channel in next_channel in app_chanspy so that the same pseudo channel isn't constantly returned. related to issue #13124 ........ ................ * channels/chan_dahdi.c, /: Merged revisions 133041 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r133041 | mmichelson | 2008-07-23 12:54:03 -0500 (Wed, 23 Jul 2008) | 15 lines Merged revisions 133038 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133038 | mmichelson | 2008-07-23 12:50:01 -0500 (Wed, 23 Jul 2008) | 7 lines Small cleanup. Move the declaration of the DAHDI_SPANINFO variable to the block where it is used. This allows one less #ifdef HAVE_PRI to clutter things up. Thanks to Tzafrir for pointing this out on #asterisk-dev ........ ................ 2008-07-23 17:21 +0000 [r132978-132983] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 132981 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132981 | tilghman | 2008-07-23 12:20:43 -0500 (Wed, 23 Jul 2008) | 6 lines Yet another conversion of '|' to ',' (closes issue #13137) Reported by: eliel Patches: chan_iax2trunk-IAXPEER.patch uploaded by eliel (license 64) ........ * contrib/scripts/asterisk.logrotate (added), /: Merged revisions 132977 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132977 | tilghman | 2008-07-23 12:14:56 -0500 (Wed, 23 Jul 2008) | 6 lines Add logrotate script for Asterisk (closes issue #13085) Reported by: pabelanger Patches: logrotate uploaded by pabelanger (license 224) ........ 2008-07-23 16:42 +0000 [r132965-132967] Kevin P. Fleming * channels/misdn/isdn_lib.c, /: Merged revisions 132883,132966 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132883 | crichter | 2008-07-23 07:07:15 -0500 (Wed, 23 Jul 2008) | 9 lines Merged revisions 132826 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132826 | crichter | 2008-07-23 13:37:50 +0200 (Mi, 23 Jul 2008) | 1 line another Fix because of r119585, this commit has broken high frequented BRI Ports, there was a possibility that a channel, that was marked as in_use would be reused later, the corresponding port could got stuck then. So it is recommended to upgrade for chan_misdn users. ........ ................ r132966 | kpfleming | 2008-07-23 11:38:28 -0500 (Wed, 23 Jul 2008) | 2 lines use correct function name... please compile with --enable-dev-mode ................ * include/asterisk/stringfields.h, /, main/utils.c: Merged revisions 132964 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132964 | kpfleming | 2008-07-23 11:30:18 -0500 (Wed, 23 Jul 2008) | 10 lines Merged revisions 132872 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132872 | kpfleming | 2008-07-23 06:52:18 -0500 (Wed, 23 Jul 2008) | 2 lines minor optimization for stringfields: when a field is being set to a larger value than it currently contains and it happens to be the most recent field allocated from the currentl pool, it is possible to 'grow' it without having to waste the space it is currently using (or potentially even allocate a new pool) ........ ................ 2008-07-23 08:18 +0000 [r132824] Olle Johansson * /, channels/chan_sip.c: Merged revisions 132823 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132823 | oej | 2008-07-23 10:13:07 +0200 (Ons, 23 Jul 2008) | 8 lines Well, the content of a channel variable may be longer than the size of a pointer... Thanks, eliel! Reported by: eliel Patches: chan_siptrunk.SIPPEER.patch uploaded by eliel (license 64) (closes issue #13135) ........ 2008-07-22 22:20 +0000 [r132797] Mark Michelson * /, channels/chan_sip.c: Merged revisions 132795 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132795 | mmichelson | 2008-07-22 17:17:09 -0500 (Tue, 22 Jul 2008) | 11 lines Merged revisions 132777 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ Allow Spiraled INVITEs to work correctly within Asterisk. Prior to this change, a spiraled INVITE would cause a 482 Loop Detected to be sent to the caller. With this change, if a potential loop is detected, the Request-URI is inspected to see if it has changed from what was originally received. If pedantic mode is on, then this inspection is fully RFC 3261 compliant. If pedantic mode is not on, then a string comparison is used to test the equality of the two R-URIs. This has been tested by using OpenSER to rewrite the R-URI and send the INVITE back to Asterisk. (closes issue #7403) Reported by: stephen_dredge Modified: branches/1.4/channels/chan_sip.c ........ ................ 2008-07-22 22:15 +0000 [r132793] Kevin P. Fleming * channels/chan_dahdi.c, /: Merged revisions 132791 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132791 | kpfleming | 2008-07-22 17:14:37 -0500 (Tue, 22 Jul 2008) | 2 lines correct fix made in r132777... the code *did* compile in dev-mode, as long as libpri was installed and enabled ........ 2008-07-22 21:59 +0000 [r132782] Olle Johansson * /, channels/chan_sip.c, doc/sip-retransmit.txt (added): Merged revisions 132703 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132703 | oej | 2008-07-22 22:46:11 +0200 (Tis, 22 Jul 2008) | 17 lines Merged revisions 132645 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132645 | oej | 2008-07-22 22:10:26 +0200 (Tis, 22 Jul 2008) | 9 lines The most common question on the #asterisk iRC channel and on mailing lists seems to be in regards to an error message when retransmit fails. This is frequently misunderstood as a failure of Asterisk, not a failure of the network to reach the other party. This document tries to assist the Asterisk user in sorting out these issues by explaining the logic and pointing at some possible causes. Hopefully, we will get other questions now :-) ........ ................ 2008-07-22 21:55 +0000 [r132780] Tilghman Lesher * configs/iax.conf.sample, /, channels/chan_iax2.c: Merged revisions 132778 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132778 | tilghman | 2008-07-22 16:53:40 -0500 (Tue, 22 Jul 2008) | 18 lines Merged revisions 132713 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r132713 | tilghman | 2008-07-22 16:19:39 -0500 (Tue, 22 Jul 2008) | 10 lines Merged revisions 132711 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r132711 | tilghman | 2008-07-22 16:14:10 -0500 (Tue, 22 Jul 2008) | 2 lines Fixes for AST-2008-010 and AST-2008-011 ........ ................ ................ 2008-07-22 21:54 +0000 [r132779] Mark Michelson * channels/chan_dahdi.c, /: Merged revisions 132777 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132777 | mmichelson | 2008-07-22 16:52:24 -0500 (Tue, 22 Jul 2008) | 3 lines Get chan_dahdi to compile in devmode ........ 2008-07-22 21:23 +0000 [r132574-132729] Kevin P. Fleming * channels/chan_dahdi.c, /: Merged revisions 132721 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132721 | kpfleming | 2008-07-22 16:21:56 -0500 (Tue, 22 Jul 2008) | 14 lines Merged revisions 132712 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132712 | kpfleming | 2008-07-22 16:17:23 -0500 (Tue, 22 Jul 2008) | 6 lines ensure that if any alarms exist at channel creation time, they are handled identically to if they occurred later, so that later alarm clearing will work properly and 'make sense' (closes issue #12160) Reported by: tzafrir ........ ................ * /, configure, configure.ac, acinclude.m4: Merged revisions 132705 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132705 | kpfleming | 2008-07-22 15:54:07 -0500 (Tue, 22 Jul 2008) | 10 lines Merged revisions 132704 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132704 | kpfleming | 2008-07-22 15:49:41 -0500 (Tue, 22 Jul 2008) | 2 lines make AST_C_COMPILE_CHECK able to print a 'pretty' description of what it is doing ........ ................ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 132643 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132643 | kpfleming | 2008-07-22 14:59:10 -0500 (Tue, 22 Jul 2008) | 10 lines Merged revisions 132641 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132641 | kpfleming | 2008-07-22 14:49:11 -0500 (Tue, 22 Jul 2008) | 2 lines use renamed libpri API call for controlling this feature (was improperly named before) ........ ................ * channels/chan_dahdi.c, /: Merged revisions 132573 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132573 | kpfleming | 2008-07-21 17:51:16 -0500 (Mon, 21 Jul 2008) | 10 lines Merged revisions 132571 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132571 | kpfleming | 2008-07-21 17:45:16 -0500 (Mon, 21 Jul 2008) | 2 lines teach chan_dahdi how to find the D-channel on BRI spans, and don't attempt to use channel 24 as a D-channel on spans of unexpected sizes ........ ................ 2008-07-21 21:13 +0000 [r132515] Brett Bryant * configs/features.conf.sample, configs/gtalk.conf.sample, /, configs/jingle.conf.sample, configs/manager.conf.sample: Merged revisions 132514 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132514 | bbryant | 2008-07-21 16:12:51 -0500 (Mon, 21 Jul 2008) | 8 lines Update configuration files to add missing options for jingle, gtalk, manager.conf, and features.conf. (closes issue #13128) Reported by: caio1982 Patches: missing_options1.diff uploaded by caio1982 (license 22) ........ 2008-07-21 21:02 +0000 [r132512-132513] Tilghman Lesher * main/fskmodem.c (added), /, include/asterisk/fskmodem.h (added): Merged revisions 132511 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132511 | tilghman | 2008-07-21 16:00:47 -0500 (Mon, 21 Jul 2008) | 2 lines (Step 2 of 2) ........ * main/fskmodem.c (removed), include/asterisk/fskmodem_int.h (added), build_tools/cflags.xml, main/fskmodem_float.c (added), /, main/tdd.c, include/asterisk/fskmodem.h (removed), main/fskmodem_int.c (added), main/callerid.c, include/asterisk/fskmodem_float.h (added): Merged revisions 132510 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132510 | tilghman | 2008-07-21 15:59:03 -0500 (Mon, 21 Jul 2008) | 5 lines Optionally build integer-based routines for FSK tone decoding (but default to the more accurate float-based routines). (Closes issue #11679) (Step 1 of 2) ........ 2008-07-21 20:55 +0000 [r132467-132509] Brett Bryant * /, apps/app_sendtext.c: Merged revisions 132508 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132508 | bbryant | 2008-07-21 15:54:09 -0500 (Mon, 21 Jul 2008) | 9 lines Fix a bug where SENDTEXTSTATUS isn't set properly when it isn't supported on a channel (yet _another_ useful patch by eliel). (closes issue #13081) Reported by: eliel Patches: app_sendtext.c.patch uploaded by eliel (license 64) Tested by: eliel ........ * /, channels/chan_sip.c: Merged revisions 132468 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132468 | bbryant | 2008-07-21 12:42:45 -0500 (Mon, 21 Jul 2008) | 5 lines Fix bug where ast_parse_arg would inadvertantly enable sip tcp when parsing a tcpbindaddr if it was disabled. (closes issue #13117) Reported by: pj ........ * /, channels/chan_iax2.c: Merged revisions 132466 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132466 | bbryant | 2008-07-21 12:22:02 -0500 (Mon, 21 Jul 2008) | 3 lines Fix an issue in iax2 where a call that's been rejected still kept an open channel on the side that attempted to make the call (not the side of the call that rejected the call). Changes were load tested and also approved by Russell. ........ 2008-07-21 15:34 +0000 [r132426] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 132425 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132425 | jpeeler | 2008-07-21 10:33:13 -0500 (Mon, 21 Jul 2008) | 2 lines make buffers config option (chan_dahdi.conf) parsing safer and added logging in case of failure ........ 2008-07-21 14:48 +0000 [r132389-132391] Russell Bryant * apps/app_jack.c, include/asterisk/libresample.h (removed), /, build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, main/Makefile, main/libresample (removed), configure.ac, codecs/codec_resample.c, makeopts.in: Merged revisions 132390 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132390 | russell | 2008-07-21 09:47:41 -0500 (Mon, 21 Jul 2008) | 16 lines Remove libresample from the Asterisk source tree. It is now available in its own repository, and must be installed like any other library for Asterisk to use. The two modules that require it are codec_resample and app_jack. To install libresample: $ svn co http://svn.digium.com/svn/libresample/trunk libresample $ cd libresample $ ./configure $ make $ sudo make install This code is currently in our own repository because the build system did not include the appropriate targets for building a dynamic library or for installing the library. ........ * apps/app_jack.c, /, codecs/codec_resample.c: Merged revisions 132388 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132388 | russell | 2008-07-21 08:51:05 -0500 (Mon, 21 Jul 2008) | 3 lines Enable higher quality resampling, as it doesn't have a noticeable performance impact on my machine .. ........ 2008-07-19 16:47 +0000 [r132313] Kevin P. Fleming * /, LICENSE: Merged revisions 132312 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132312 | kpfleming | 2008-07-19 11:46:33 -0500 (Sat, 19 Jul 2008) | 10 lines Merged revisions 132311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132311 | kpfleming | 2008-07-19 11:45:52 -0500 (Sat, 19 Jul 2008) | 2 lines grant a license exception to allow distribution of Asterisk binaries that use the UW IMAP Toolkit (which is licensed under a non-GPL-compatible license) ........ ................ 2008-07-19 10:47 +0000 [r132278] Michiel van Baak * res/res_config_sqlite.c, /: Merged revisions 132277 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132277 | mvanbaak | 2008-07-19 12:46:12 +0200 (Sat, 19 Jul 2008) | 7 lines fix a couple of comments in sqlite resource driver. (closes issue #13110) Reported by: gknispel_proformatique Patches: res_config_sqlite_comments.patch uploaded by gknispel (license 261) ........ 2008-07-18 22:20 +0000 [r132245] Brett Bryant * main/manager.c, /: Merged revisions 132242 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132242 | bbryant | 2008-07-18 17:19:56 -0500 (Fri, 18 Jul 2008) | 4 lines Fixes problem where manager users loaded from users.conf would be removed early (before the routine to load the configuration was finished) because a variable wasn't initialized. ........ 2008-07-18 20:58 +0000 [r132114-132207] Tilghman Lesher * /, main/say.c: Merged revisions 132113 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132113 | tilghman | 2008-07-18 14:09:39 -0500 (Fri, 18 Jul 2008) | 14 lines Merged revisions 132112 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132112 | tilghman | 2008-07-18 14:06:10 -0500 (Fri, 18 Jul 2008) | 6 lines Fix for Taiwanese number syntax (closes issue #12319) Reported by: CharlesWang Patches: saynumber-tw-1.4.18.1.patch uploaded by CharlesWang (license 444) ........ ................ 2008-07-18 18:53 +0000 [r132111] Matthew Fredrickson * channels/chan_dahdi.c, /: Merged revisions 132108 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132108 | mattf | 2008-07-18 13:50:00 -0500 (Fri, 18 Jul 2008) | 1 line Make sure we break the poll so that messages queued will be sent on the SS7 when using CLI commands for blocking and blocking of CICs and linksets. ........ 2008-07-18 18:51 +0000 [r132110] Tilghman Lesher * main/config.c, /: Merged revisions 132109 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132109 | tilghman | 2008-07-18 13:50:37 -0500 (Fri, 18 Jul 2008) | 14 lines Merged revisions 132107 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132107 | tilghman | 2008-07-18 13:47:50 -0500 (Fri, 18 Jul 2008) | 6 lines Textual clarification (closes issue #13106) Reported by: flefoll Patches: config.c.br14.120173.patch-unknown-directive uploaded by flefoll (license 244) ........ ................ 2008-07-18 17:56 +0000 [r132051] Brett Bryant * main/hashtab.c, /, cdr/cdr_radius.c: Merged revisions 132050 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132050 | bbryant | 2008-07-18 12:55:41 -0500 (Fri, 18 Jul 2008) | 8 lines Fix magic Revision keywords in hashtab.c and change cdr_radius.c to use the same keyword as the other files (patch by eliel). (closes issue #13104) Reported by: eliel Patches: revision.patch uploaded by eliel (license 64) ........ 2008-07-18 17:40 +0000 [r131984-132047] Tilghman Lesher * main/sched.c, /: Merged revisions 131989 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131989 | tilghman | 2008-07-18 12:10:34 -0500 (Fri, 18 Jul 2008) | 10 lines Merged revisions 131988 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131988 | tilghman | 2008-07-18 12:10:01 -0500 (Fri, 18 Jul 2008) | 2 lines Oops ........ ................ * main/sched.c, /, include/asterisk/sched.h: Merged revisions 131986 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131986 | tilghman | 2008-07-18 11:48:18 -0500 (Fri, 18 Jul 2008) | 10 lines Merged revisions 131985 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131985 | tilghman | 2008-07-18 11:46:23 -0500 (Fri, 18 Jul 2008) | 2 lines Preserve ABI compatibility with last change ........ ................ * main/sched.c, /, include/asterisk/sched.h, channels/chan_iax2.c: Merged revisions 131982 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131982 | tilghman | 2008-07-18 11:33:56 -0500 (Fri, 18 Jul 2008) | 10 lines Merged revisions 131970 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131970 | tilghman | 2008-07-18 11:30:31 -0500 (Fri, 18 Jul 2008) | 2 lines Make the ast_assert call within ast_sched_del report something useful. ........ ................ 2008-07-18 16:16 +0000 [r131924] Kevin P. Fleming * main/dlfcn.c (removed), main/loader.c, /, main/Makefile, include/asterisk/dlfcn-compat.h (removed): Merged revisions 131923 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131923 | kpfleming | 2008-07-18 11:16:12 -0500 (Fri, 18 Jul 2008) | 10 lines Merged revisions 131921 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131921 | kpfleming | 2008-07-18 11:15:41 -0500 (Fri, 18 Jul 2008) | 2 lines remove the dlfcn compatibility stuff, because no platforms that Asterisk currently runs on it use it, and it doesn't build anyway ........ ................ 2008-07-18 15:39 +0000 [r131917] Brett Bryant * /, main/features.c: Merged revisions 131916 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131916 | bbryant | 2008-07-18 10:38:22 -0500 (Fri, 18 Jul 2008) | 12 lines Merged revisions 131915 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131915 | bbryant | 2008-07-18 10:34:42 -0500 (Fri, 18 Jul 2008) | 4 lines Fix a bug in blind transfers where the BLINDTRANSFER variable isn't always set to the other end of the blind transfer. (closes issue #12586) ........ ................ 2008-07-17 22:45 +0000 [r131869] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 131868 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r131868 | jpeeler | 2008-07-17 17:40:00 -0500 (Thu, 17 Jul 2008) | 6 lines Add configuration option to chan_dahdi.conf to allow buffering policy and number of buffers to be configured per channel. Syntax: buffers=, Where the number of buffers is some non-negative integer and the policy is either "full", "half", or "immediate". ........ 2008-07-17 21:27 +0000 [r131830] Mark Michelson * /, apps/app_senddtmf.c: Merged revisions 131824 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r131824 | mmichelson | 2008-07-17 16:26:41 -0500 (Thu, 17 Jul 2008) | 10 lines Document that the duration of dtmf may be passed to the SendDTMF application. Also correct the default pause between digits. (closes issue #13102) Reported by: eliel Patches: app_senddtmf.c.patch uploaded by eliel (license 64) ........ 2008-07-17 20:38 +0000 [r131754-131792] Tilghman Lesher * channels/chan_dahdi.c, /: Merged revisions 131791 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131791 | tilghman | 2008-07-17 15:37:14 -0500 (Thu, 17 Jul 2008) | 15 lines Merged revisions 131790 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131790 | tilghman | 2008-07-17 15:35:44 -0500 (Thu, 17 Jul 2008) | 7 lines Revert part of issue #5620 (revision 6965) as it appears that it was in error. This should fix talk call progress on analog lines. (closes issue #12178) Reported by: michael-fig Patches: 20080717__bug12178.diff.txt uploaded by Corydon76 (license 14) ........ ................ * res/res_config_sqlite.c, /: Merged revisions 131753 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r131753 | tilghman | 2008-07-17 13:36:34 -0500 (Thu, 17 Jul 2008) | 6 lines Fix memory leaks (closes issue #13099) Reported by: gknispel_proformatique Patches: res_config_sqlite_leak_on_error.patch uploaded by gknispel (license 261) ........ 2008-07-17 18:15 +0000 [r131718] Brett Bryant * /, main/features.c: Merged revisions 131717 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r131717 | bbryant | 2008-07-17 13:14:42 -0500 (Thu, 17 Jul 2008) | 8 lines Fix a memory leak in register_group_feature when attempting to register a feature without specifying a group or feature to register. (closes issue #13101) Reported by: eliel Patches: features.c.patch uploaded by eliel (license 64) ........ 2008-07-17 15:46 +0000 [r131682] Tilghman Lesher * res/res_config_sqlite.c, /: Merged revisions 131681 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r131681 | tilghman | 2008-07-17 10:45:25 -0500 (Thu, 17 Jul 2008) | 4 lines Fix memory leak. (Closes issue #13096) Reported by gknispel_proformatique ........ 2008-07-16 23:56 +0000 [r131571] Steve Murphy * /: The commit from 131570 should not be applied to 1.6.0, as it is not as necessary, because log_show_lock in trunk is not available in 1.6.0, and is estimated to be the only function that might care about the lock_addr values. 2008-07-16 22:18 +0000 [r131493] Brett Bryant * /, channels/chan_iax2.c: Merged revisions 131492 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131492 | bbryant | 2008-07-16 17:17:36 -0500 (Wed, 16 Jul 2008) | 14 lines Merged revisions 131491 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131491 | bbryant | 2008-07-16 17:17:07 -0500 (Wed, 16 Jul 2008) | 6 lines Fix a bug in iax2 registration that allowed peers to register with case-insensitive names (user_cmp_cb and peer_cmp_cb are now both case-sensitive). (closes issue #13091) ........ ................ 2008-07-16 21:54 +0000 [r131455-131486] Brett Bryant * /, funcs/func_sysinfo.c: Merged revisions 131484 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r131484 | bbryant | 2008-07-16 16:54:08 -0500 (Wed, 16 Jul 2008) | 4 lines Fixes sysinfo operator issue also fixed elsewhere in r131445. (issue #13057) ........ * main/asterisk.c, /: Merged revisions 131445 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r131445 | bbryant | 2008-07-16 16:24:18 -0500 (Wed, 16 Jul 2008) | 9 lines Fixes an issue with "core show sysinfo" that used the wrong operator to calculate the number of bytes from a sysinfo structure. unsigned long. (closes issue #13057) Reported by: eliel Patches: asterisk.c.patch uploaded by eliel (license 64) ........ 2008-07-16 20:48 +0000 [r131423] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 131422 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131422 | russell | 2008-07-16 15:48:27 -0500 (Wed, 16 Jul 2008) | 15 lines Merged revisions 131421 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131421 | russell | 2008-07-16 15:47:53 -0500 (Wed, 16 Jul 2008) | 7 lines Always ensure that the channel's tech_pvt reference is NULL after calling the destroy callback. (closes issue #13060) Reported by: jpgrayson Patches: chan_iax2_tech_pvt_crash.patch uploaded by jpgrayson (license 492) ........ ................ 2008-07-16 20:24 +0000 [r131301-131378] Mark Michelson * /, apps/app_queue.c: Merged revisions 131375 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131375 | mmichelson | 2008-07-16 15:24:12 -0500 (Wed, 16 Jul 2008) | 22 lines Merged revisions 131369 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131369 | mmichelson | 2008-07-16 15:23:02 -0500 (Wed, 16 Jul 2008) | 14 lines Move the init_queue call back to where it used to be (changed Sept 12 last year). It was moved then to prevent a memory leak. Since then, the same memory leak recurred and was fixed in a better way. Now it has been found that the placement of this init_queue call can cause problems if a realtime queue has values changed to an empty string. The problem is that the default value for that queue parameter would not be set. (closes issue #13084) Reported by: elbriga ........ ................ * res/res_config_sqlite.c, /: Merged revisions 131361 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r131361 | mmichelson | 2008-07-16 14:57:02 -0500 (Wed, 16 Jul 2008) | 9 lines Don't try to dereference the dbfile pointer if we know that it's NULL. (closes issue #13092) Reported by: gknispel_proformatique Patches: trunk_sqlite_check_vars_null.patch uploaded by gknispel (license 261) ........ * /, apps/app_queue.c: Merged revisions 131358 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131358 | mmichelson | 2008-07-16 14:37:42 -0500 (Wed, 16 Jul 2008) | 14 lines Merged revisions 131357 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131357 | mmichelson | 2008-07-16 14:37:08 -0500 (Wed, 16 Jul 2008) | 6 lines Apparently, "thread safety" is important, whatever that means. :P (Thanks Russell!) ........ ................ * /, apps/app_queue.c: Merged revisions 131300 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131300 | mmichelson | 2008-07-16 13:59:27 -0500 (Wed, 16 Jul 2008) | 21 lines Merged revisions 131299 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131299 | mmichelson | 2008-07-16 13:57:34 -0500 (Wed, 16 Jul 2008) | 13 lines Make absolutely certain that the transfer datastore is removed from the calling channel once the caller is finished in the queue. This could have weird con- sequences when dialing local queue members when multiple transfers occur on a single call. Also fixed a memory leak that would occur when an attended transfer occurred from a queue member. (closes issue #13047) Reported by: festr ........ ................ 2008-07-16 18:20 +0000 [r131248] Steve Murphy * res/ael/pval.c, /: Merged revisions 131243 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131243 | murf | 2008-07-16 11:59:33 -0600 (Wed, 16 Jul 2008) | 27 lines Merged revisions 131242 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131242 | murf | 2008-07-16 11:53:43 -0600 (Wed, 16 Jul 2008) | 19 lines (closes issue #13090) Reported by: murf The problem was that, esoteric as it is, because the hangerupper context immediately preceded the std-priv-extent macro, that the checking code accidentally would fall from traversing hangerupper into the std-priv-exten macro, where it would hit the hangerupper in the 'includes', and proceed into an infinite recursion. A small fix to traverse into the statements of the context instead of the context solves this issue. I also added some commented out printfs for debug, which were pretty handy in the face of a dorky gdb. This was a problem around since the package was first written; but evidently pretty rare in turning up in the field. ........ ................ 2008-07-16 15:04 +0000 [r131206] Luigi Rizzo * channels/chan_agent.c: add missing terminator argument for ast_event_subscribe(). Without it the function will randomly walk on the stack possibly causing a panic 2008-07-16 00:54 +0000 [r131168] Tilghman Lesher * /, main/logger.c: Merged revisions 131166 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r131166 | tilghman | 2008-07-15 19:52:48 -0500 (Tue, 15 Jul 2008) | 3 lines Fix rotate strategy (Closes issue #13086) ........ 2008-07-15 23:41 +0000 [r131131] Steve Murphy * main/pbx.c, /: Merged revisions 131129 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r131129 | murf | 2008-07-15 17:36:19 -0600 (Tue, 15 Jul 2008) | 21 lines (closes issue #12960) Reported by: mnicholson Spent most of the day on this bug, and the solution was so simple. Just had to find and understand the problem. The problem was, that the routine to copy the existing switches, includes, and ignorepats from the old context to the new one, wasn't getting called when the context is already existent. (In other words, if AEL is adding a new context to the mix, they get copied, but if pbx_config already defined a context, then the copy wasn't happening. This made no sense, so I moved the call to copy the includes & etc, no matter the case. ........ 2008-07-15 18:47 +0000 [r131073] Russell Bryant * /, res/res_agi.c: Merged revisions 131072 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r131072 | russell | 2008-07-15 13:46:40 -0500 (Tue, 15 Jul 2008) | 5 lines Fix a couple of places in res_agi where the agi_commands lock would not be released, causing a deadlock. (Reported by mvanbaak in #asterisk-dev, discovered by bbryant's change to the lock tracking code to yell at you if a thread exits with a lock still held) ........ 2008-07-15 18:29 +0000 [r131060] Tilghman Lesher * main/pbx.c, main/manager.c, /, channels/chan_sip.c: Merged revisions 131044 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131044 | tilghman | 2008-07-15 13:25:34 -0500 (Tue, 15 Jul 2008) | 16 lines Merged revisions 130959 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130959 | tilghman | 2008-07-15 12:19:13 -0500 (Tue, 15 Jul 2008) | 8 lines astman_send_error does not need a newline appended -- the API takes care of that for us. (closes issue #13068) Reported by: gknispel_proformatique Patches: asterisk_1_4_astman_send.patch uploaded by gknispel (license 261) asterisk_trunk_astman_send.patch uploaded by gknispel (license 261) ........ ................ 2008-07-15 18:00 +0000 [r131014] Michiel van Baak * main/cdr.c, /: Merged revisions 131013 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131013 | mvanbaak | 2008-07-15 19:49:48 +0200 (Tue, 15 Jul 2008) | 15 lines Merged revisions 131012 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131012 | mvanbaak | 2008-07-15 19:47:15 +0200 (Tue, 15 Jul 2008) | 7 lines remove 4 lines of redundant code. (closes issue #13080) Reported by: gknispel_proformatique Patches: trunk_ast_cdr_setapp.patch uploaded by gknispel (license 261) ........ ................ 2008-07-15 13:14 +0000 [r130946] Steve Murphy * utils/conf2ael.c, utils/Makefile, res/ael/pval.c, channels/chan_skinny.c, res/ael/ael.tab.c, main/features.c, pbx/pbx_ael.c, res/ael/ael_lex.c, res/ael/ael.tab.h, utils/ael_main.c, include/asterisk/pbx.h, utils/extconf.c, res/ael/ael.flex, pbx/pbx_config.c, apps/app_stack.c, apps/app_dial.c, main/pbx.c, include/asterisk/pval.h, /, channels/chan_sip.c, apps/app_meetme.c, res/ael/ael.y, channels/chan_iax2.c, apps/app_queue.c: Merged revisions 130145 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk Merging this rev from trunk to 1.6.0 was not simple. Why? Because we've enhanced trunk to do a [fast] merge-and-delete operation which also solved problems with contexts having entries from different registrars. Fast as in the amount of time the contexts are locked down. That *is* fast, but traversing the entire dialplan looking for priorities to delete takes more time overall. This particular fix involved pulling in those enhancements from trunk, along with all the various fixes and refinements made along the way. Merging all this from trunk into 1.6 involved: a. mergetrunk6 in the stuff from 130145; b. revert all but the prop changes c. catalog all revisions to pbx.c since 1.6.0 was forked (at rev 105596). d. catalog all revisions to pbx.c in trunk since 1.6.0 was forked, making special note of all revs that were not merged into 1.6.0. e. study each rev in trunk not applied to 1.6.0, and determine if it was involved in the merge_and_delete enhancements in trunk. 25 commits were done in 1.6.0, all but one (106306) was a merge from trunk. Trunk had 22 additional changes, of which 7 were involved in the merge_and_delete enhancements: 106757 108894 109169 116461 123358 130145 130297 f. Go to trunk and collect patches, one by one, of the changes made by each rev across the entire source tree, using svn diff -c > pfile g. Apply each patch in order to 1.6.0, and resolve all failures and compilation problems before proceding to the next patch. h. test the stuff. i. profit! ........ r130145 | murf | 2008-07-11 12:24:31 -0600 (Fri, 11 Jul 2008) | 40 lines (closes issue #13041) Reported by: eliel Tested by: murf (closes issue #12960) Reported by: mnicholson In this 'omnibus' fix, I **think** I solved both the problem in 13041, where unloading pbx_ael.so caused crashes, or incomplete removal of previous registrar'ed entries. And I added code to completely remove all includes, switches, and ignorepats that had a matching registrar entry, which should appease 12960. I also added a lot of seemingly useless brackets around single statement if's, which helped debug so much that I'm leaving them there. I added a routine to check the correlation between the extension tree lists and the hashtab tables. It can be amazingly helpful when you have lots of dialplan stuff, and need to narrow down where a problem is occurring. It's ifdef'd out by default. I cleaned up the code around the new CIDmatch code. It was leaving hanging extens with bad ptrs, getting confused over which objects to remove, etc. I tightened up the code and changed the call to remove_exten in the merge_and_delete code. I added more conditions to check for empty context worthy of deletion. It's not empty if there are any includes, switches, or ignorepats present. If I've missed anything, please re-open this bug, and be prepared to supply example dialplan code. ........ 2008-07-15 00:00 +0000 [r130891] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 130890 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r130890 | tilghman | 2008-07-14 18:59:54 -0500 (Mon, 14 Jul 2008) | 16 lines Merged revisions 130889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130889 | tilghman | 2008-07-14 18:59:13 -0500 (Mon, 14 Jul 2008) | 8 lines Override the callerid in all cases when the callerid is set in the user, not just when a remote callerid is set. Also, if not set in the user, allow the remote CallerID to pass through. (closes issue #12875) Reported by: dimas Patches: 20080714__bug12875.diff.txt uploaded by Corydon76 (license 14) ........ ................ 2008-07-14 22:24 +0000 [r130795-130855] Mark Michelson * main/asterisk.c, /: Merged revisions 130854 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r130854 | mmichelson | 2008-07-14 17:22:57 -0500 (Mon, 14 Jul 2008) | 9 lines Fix a memory leak in the case that /dev/null cannot be opened when running startup commands from cli.conf (closes issue #13066) Reported by: eliel Patches: asterisk.c.patch uploaded by eliel (license 64) ........ * apps/app_dial.c, /: Merged revisions 130794 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r130794 | mmichelson | 2008-07-14 12:54:11 -0500 (Mon, 14 Jul 2008) | 16 lines Merged revisions 130792 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130792 | mmichelson | 2008-07-14 12:50:21 -0500 (Mon, 14 Jul 2008) | 8 lines Add a check to the CAN_EARLY_BRIDGE macro in app_dial to be sure there are no audiohooks present on the channels involved. This fixed a one-way audio situation I had in my test setup. I couldn't find any open issues that suggested one-way audio with regards to mixmonitor (or other audiohook) usage, though. ........ ................ 2008-07-14 17:22 +0000 [r130752] Michiel van Baak * main/dnsmgr.c, /: Merged revisions 130744 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r130744 | mvanbaak | 2008-07-14 19:21:18 +0200 (Mon, 14 Jul 2008) | 18 lines Merged revisions 130735 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130735 | mvanbaak | 2008-07-14 19:10:21 +0200 (Mon, 14 Jul 2008) | 10 lines notify the user that dnsmgr refresh wont work when dnsmgr is not enabled. Previously this command would automagically appear and disappear. This was confusing. (closes issue #12796) Reported by: chappell Patches: dnsmgr_refresh_3.diff uploaded by chappell (license 8) Tested by: russell, chappell, mvanbaak ........ ................ 2008-07-14 10:40 +0000 [r130636-130637] Russell Bryant * /, include/asterisk/astobj.h: Merged revisions 129987 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r129987 | russell | 2008-07-11 09:22:44 -0500 (Fri, 11 Jul 2008) | 10 lines Merged revisions 129970 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129970 | russell | 2008-07-11 09:18:43 -0500 (Fri, 11 Jul 2008) | 2 lines add a simple ASTOBJ_TRYWRLOCK macro ... ........ ................ * /, main/audiohook.c: Merged revisions 130635 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r130635 | russell | 2008-07-14 05:39:23 -0500 (Mon, 14 Jul 2008) | 10 lines Merged revisions 130634 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130634 | russell | 2008-07-14 05:38:14 -0500 (Mon, 14 Jul 2008) | 2 lines Bump up the debug level for a message. ........ ................ 2008-07-13 23:20 +0000 [r130575-130582] Michiel van Baak * /, doc/tex/Makefile, build_tools/prep_tarball, res/Makefile: Merged revisions 130578 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r130578 | mvanbaak | 2008-07-14 01:14:03 +0200 (Mon, 14 Jul 2008) | 15 lines Make all sed calls Posix sed compatible. To make sure nobody commits script-modified files we first make a backup of asterisk.tex, run the script, generate the pdf and / or html, and put the original asterisk.tex back. This will guard us for the stuff that happened before that someone committed a locally modified asterisk.tex, with changes done by this script. (closes issue #13062) Reported by: mvanbaak Patches: sed_without-i-v3.diff uploaded by mvanbaak (license 7) Tested by: mvanbaak Feedback from Corydon. Thanks for taking the time to go through this. ........ * main/manager.c, /: Merged revisions 130574 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r130574 | mvanbaak | 2008-07-14 00:50:31 +0200 (Mon, 14 Jul 2008) | 16 lines Merged revisions 130573 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130573 | mvanbaak | 2008-07-14 00:48:51 +0200 (Mon, 14 Jul 2008) | 8 lines fix memory leak when originate from manager cannot create a thread (closes issue #13069) Reported by: gknispel_proformatique Patches: asterisk_trunk_action_originate.patch uploaded by gknispel (license 261) Tested by: gknispel_proformatique, mvanbaak ........ ................ 2008-07-13 17:59 +0000 [r130516] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 130515 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r130515 | tilghman | 2008-07-13 12:58:47 -0500 (Sun, 13 Jul 2008) | 12 lines Merged revisions 130514 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130514 | tilghman | 2008-07-13 12:56:10 -0500 (Sun, 13 Jul 2008) | 4 lines Reverting 2 changesets, as it breaks incoming IAX2 calls (Related to issue #12963) Reported by: mvanbaak ........ ................ 2008-07-13 15:00 +0000 [r130480] Michiel van Baak * doc/tex/asterisk.tex, /: Merged revisions 130479 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r130479 | mvanbaak | 2008-07-13 16:58:40 +0200 (Sun, 13 Jul 2008) | 3 lines restore ASTERISKVERSION marker to asterisk.tex. This got lost in commit 97634 ........ 2008-07-13 02:35 +0000 [r130445] Tilghman Lesher * /, channels/chan_agent.c: Merged revisions 130444 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r130444 | tilghman | 2008-07-12 21:34:32 -0500 (Sat, 12 Jul 2008) | 2 lines Unlock list before returning ........ 2008-07-11 21:39 +0000 [r130294] Matthew Fredrickson * channels/chan_dahdi.c, /: Merged revisions 130293 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r130293 | mattf | 2008-07-11 16:36:26 -0500 (Fri, 11 Jul 2008) | 1 line Support new TRANSPORT definitions in libss7 ........ 2008-07-11 20:04 +0000 [r130238] Mark Michelson * /, main/audiohook.c: Merged revisions 130237 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r130237 | mmichelson | 2008-07-11 15:03:55 -0500 (Fri, 11 Jul 2008) | 11 lines Merged revisions 130236 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130236 | mmichelson | 2008-07-11 15:03:23 -0500 (Fri, 11 Jul 2008) | 3 lines Remove redundant logic ........ ................ 2008-07-11 19:57 +0000 [r130231-130235] Tilghman Lesher * channels/chan_dahdi.c, /, channels/chan_agent.c, utils/astman.c: Merged revisions 130230 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r130230 | tilghman | 2008-07-11 14:40:55 -0500 (Fri, 11 Jul 2008) | 2 lines Fix trunk breakage ........ 2008-07-11 19:14 +0000 [r130175] Mark Michelson * /, main/audiohook.c: Merged revisions 130174 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r130174 | mmichelson | 2008-07-11 14:14:15 -0500 (Fri, 11 Jul 2008) | 15 lines Merged revisions 130173 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130173 | mmichelson | 2008-07-11 14:13:29 -0500 (Fri, 11 Jul 2008) | 7 lines Fix a typo in audiohook_read_frame_both. While this change has not been proven to fix any specific issue, it is incorrect and could cause unforeseen problems. ........ ................ 2008-07-11 18:53 +0000 [r130171] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 130170 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r130170 | tilghman | 2008-07-11 13:52:42 -0500 (Fri, 11 Jul 2008) | 15 lines Merged revisions 130169 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130169 | tilghman | 2008-07-11 13:51:56 -0500 (Fri, 11 Jul 2008) | 7 lines Ensure that a destination callno of 0 will not match for frames that do not start a dialog (new, lagrq, and ping). (closes issue #12963) Reported by: russellb Patches: chan_iax2_dup_new_fix4.patch uploaded by jpgrayson (license 492) ........ ................ 2008-07-11 18:33 +0000 [r130168] Sean Bright * /, channels/chan_sip.c: Merged revisions 130167 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r130167 | seanbright | 2008-07-11 14:32:26 -0400 (Fri, 11 Jul 2008) | 1 line Missed one. Formatting only. ........ 2008-07-11 18:14 +0000 [r130130] Brett Bryant * main/cli.c, channels/chan_jingle.c, channels/chan_dahdi.c, channels/chan_skinny.c, main/abstract_jb.c, apps/app_minivm.c, codecs/codec_resample.c, codecs/codec_dahdi.c, apps/app_chanspy.c, main/asterisk.c, apps/app_milliwatt.c, main/dsp.c, codecs/codec_g722.c, /, channels/chan_sip.c, main/threadstorage.c, utils/astman.c, main/utils.c, channels/chan_gtalk.c, pbx/dundi-parser.c: Merged revisions 130129 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r130129 | bbryant | 2008-07-11 13:09:35 -0500 (Fri, 11 Jul 2008) | 8 lines Janitor patch to change uses of sizeof to ARRAY_LEN (closes issue #13054) Reported by: pabelanger Patches: ARRAY_LEN.patch2 uploaded by pabelanger (license 224) Tested by: seanbright ........ 2008-07-11 17:30 +0000 [r130127] Tilghman Lesher * /, channels/chan_agent.c: Merged revisions 130126 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r130126 | tilghman | 2008-07-11 12:29:24 -0500 (Fri, 11 Jul 2008) | 17 lines Merged revisions 130102 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130102 | tilghman | 2008-07-11 11:50:42 -0500 (Fri, 11 Jul 2008) | 9 lines Pass the devicestate from an underlying channel up through the Agent channel. This should make the Agent always report the correct device state, even when the underlying channel is used for other purposes. (closes issue #12773) Reported by: davidw Patches: 20080710__bug12773.diff.txt uploaded by Corydon76 (license 14) Tested by: davidw ........ ................ 2008-07-11 16:18 +0000 [r129936-130045] Kevin P. Fleming * doc/ss7.txt, /, contrib/utils/zones2indications.c, CHANGES: Merged revisions 130044 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r130044 | kpfleming | 2008-07-11 11:18:01 -0500 (Fri, 11 Jul 2008) | 2 lines clean up a bunch more Zaptel-related references ........ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 130040 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r130040 | kpfleming | 2008-07-11 10:57:17 -0500 (Fri, 11 Jul 2008) | 12 lines Merged revisions 130039 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130039 | kpfleming | 2008-07-11 10:41:56 -0500 (Fri, 11 Jul 2008) | 4 lines add support for a configuration parameter for 'inband audio during RELEASE', which is currently mandatory in libpri-1.4.4 but will become configurable in libpri-1.4.5 later today (related to issue #13042) ........ ................ * /, main/astmm.c: Merged revisions 129968 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r129968 | kpfleming | 2008-07-11 09:16:15 -0500 (Fri, 11 Jul 2008) | 18 lines Merged revisions 129966 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129966 | kpfleming | 2008-07-11 09:03:52 -0500 (Fri, 11 Jul 2008) | 5 lines fix a flaw found while experimenting with structure alignment and padding; low-fence checking would not work properly on 64-bit platforms, because the compiler was putting 4 bytes of padding between the fence field and the allocation memory block added a very obvious runtime warning if this condition reoccurs, so the developer who broke it can be chastised into fixing it :-) ........ r129967 | kpfleming | 2008-07-11 09:03:52 -0500 (Fri, 11 Jul 2008) | 5 lines simplify calculation ........ ................ * /, sounds/Makefile: Merged revisions 129916 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r129916 | kpfleming | 2008-07-11 07:21:29 -0500 (Fri, 11 Jul 2008) | 10 lines Merged revisions 129907 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129907 | kpfleming | 2008-07-11 07:15:42 -0500 (Fri, 11 Jul 2008) | 2 lines don't attempt to set user/group ownership of extracted sound files (reported on asterisk-users) ........ ................ 2008-07-11 01:01 +0000 [r129865] Sean Bright * res/res_config_pgsql.c, /, res/res_config_ldap.c: Merged revisions 129864 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r129864 | seanbright | 2008-07-10 20:55:06 -0400 (Thu, 10 Jul 2008) | 1 line Fix some usages of snprintf, and clarify a couple variable names. ........ 2008-07-10 22:07 +0000 [r129764-129805] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 129804 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r129804 | tilghman | 2008-07-10 17:06:07 -0500 (Thu, 10 Jul 2008) | 16 lines Merged revisions 129803 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129803 | tilghman | 2008-07-10 16:57:05 -0500 (Thu, 10 Jul 2008) | 8 lines Correctly deal with duplicate NEW frames (due to retransmission). Also, fixup the destination call number matching to be more strict and reliable. (closes issue #12963) Reported by: jpgrayson Patches: chan_iax2_dup_new_fix3.patch uploaded by jpgrayson (license 492) Tested by: jpgrayson, Corydon76 ........ ................ * res/res_config_odbc.c, /: Merged revisions 129758 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r129758 | tilghman | 2008-07-10 16:23:23 -0500 (Thu, 10 Jul 2008) | 10 lines Merged revisions 129741 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129741 | tilghman | 2008-07-10 16:19:48 -0500 (Thu, 10 Jul 2008) | 2 lines Oops ........ ................ 2008-07-10 21:05 +0000 [r129739] Terry Wilson * Makefile, /: Merged revisions 129738 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r129738 | twilson | 2008-07-10 15:56:20 -0500 (Thu, 10 Jul 2008) | 2 lines Move phoneprov config files to be installed with 'make samples' so changes aren't inadvertently lost on a 'make install' ........ 2008-07-10 19:14 +0000 [r129685] Brett Bryant * /, apps/app_queue.c: Merged revisions 129684 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r129684 | bbryant | 2008-07-10 14:13:12 -0500 (Thu, 10 Jul 2008) | 8 lines Fixes a bug where the interface for a queue member gets reloaded as the state_interface, if a state_interface was set, on reload because the state_interface isn't stored in the ast_db. (closes issue #13043) Reported by: jvandal Patches: app_queue.patch uploaded by jvandal (license 413) ........ 2008-07-10 18:20 +0000 [r129640-129647] Sean Bright * /, channels/chan_sip.c: Merged revisions 129642 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r129642 | seanbright | 2008-07-10 14:19:17 -0400 (Thu, 10 Jul 2008) | 1 line A couple more minor text changes ........ * /, channels/chan_sip.c: Merged revisions 129638 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r129638 | seanbright | 2008-07-10 14:16:21 -0400 (Thu, 10 Jul 2008) | 1 line Remove extraneous \n. Pointed out by eliel on #asterisk-dev. ........ 2008-07-10 16:13 +0000 [r129570] Russell Bryant * sample.call, /: Merged revisions 129569 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r129569 | russell | 2008-07-10 11:12:51 -0500 (Thu, 10 Jul 2008) | 11 lines Merged revisions 129567 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129567 | russell | 2008-07-10 11:03:59 -0500 (Thu, 10 Jul 2008) | 3 lines Note that pbx_spool.so is the module used for call files (inspired by a question in #asterisk) ........ ................ 2008-07-10 14:09 +0000 [r129504-129507] Sean Bright * /, main/editline: Merged revisions 129503 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r129503 | seanbright | 2008-07-10 09:54:29 -0400 (Thu, 10 Jul 2008) | 2 lines Update svn:ignore ........ 2008-07-09 19:41 +0000 [r129438] Mark Michelson * main/rtp.c, /: Merged revisions 129437 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r129437 | mmichelson | 2008-07-09 14:40:30 -0500 (Wed, 09 Jul 2008) | 21 lines Merged revisions 129436 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129436 | mmichelson | 2008-07-09 14:32:20 -0500 (Wed, 09 Jul 2008) | 13 lines Fix a problem where inbound rfc2833 audio would be sent to the core instead of being P2P bridged. When the core regenerated the rfc2833 packet for the outbound leg, the SSRC would be different than the RTP audio on the call leg causing DTMF detection issues on the far end. (closes issue #12955) Reported by: tonyredstone Patches: dynamic_rtp.patch uploaded by tsearle (license 373) Tested by: tonyredstone ........ ................ 2008-07-09 16:01 +0000 [r129400] Matthew Fredrickson * main/pbx.c, /: Merged revisions 129399 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r129399 | mattf | 2008-07-09 10:57:06 -0500 (Wed, 09 Jul 2008) | 1 line Add Proceeding() application (#13025) ........ 2008-07-09 13:46 +0000 [r129345] Sean Bright * main/editline/makelist (removed), main/editline/makelist.in (added), /, main/editline/configure, main/editline/Makefile.in, main/editline/configure.in: Merged revisions 129344 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r129344 | seanbright | 2008-07-09 09:44:43 -0400 (Wed, 09 Jul 2008) | 12 lines Merged revisions 129343 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129343 | seanbright | 2008-07-09 09:41:21 -0400 (Wed, 09 Jul 2008) | 4 lines Look for the system installed awk instead of assuming it's at /usr/bin/awk. Pointed out by jmls via #asterisk-dev. ........ ................ 2008-07-08 22:56 +0000 [r129160-129271] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 129270 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r129270 | mmichelson | 2008-07-08 17:56:12 -0500 (Tue, 08 Jul 2008) | 3 lines Fix compilation error when IMAP storage is enabled ........ 2008-07-08 21:04 +0000 [r129157] Brett Bryant * main/dns.c, main/srv.c, /: Merged revisions 129156 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r129156 | bbryant | 2008-07-08 16:00:01 -0500 (Tue, 08 Jul 2008) | 6 lines Fix a bug in SRV lookups where dnsmgr would discard everything but the first SRV result from DNS before processing weights and priorities and dns_parse_answer wouldn't report that there were no records in DNS unless a failure occured. Also fixed a bug where dnsmgr_refresh would report that a entry was being changed when ast_gethostbyname had failed. ........ 2008-07-08 20:31 +0000 [r129049-129153] Tilghman Lesher * apps/app_dial.c, /, channels/chan_sip.c, include/asterisk/causes.h: Merged revisions 129152 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r129152 | tilghman | 2008-07-08 15:30:29 -0500 (Tue, 08 Jul 2008) | 16 lines Merged revisions 129149 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129149 | tilghman | 2008-07-08 15:27:47 -0500 (Tue, 08 Jul 2008) | 8 lines Cause SIP to return a 480 instead of a 404 when a sip peer exists, but is not registered. (closes issue #12885) Reported by: ibc Patches: 20080701__bug12885__2.diff.txt uploaded by Corydon76 (license 14) Tested by: ibc ........ ................ * /, channels/chan_iax2.c: Merged revisions 129048 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r129048 | tilghman | 2008-07-08 11:49:01 -0500 (Tue, 08 Jul 2008) | 15 lines Merged revisions 129047 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129047 | tilghman | 2008-07-08 11:45:23 -0500 (Tue, 08 Jul 2008) | 7 lines Timestamp decoding for video mini-frames is bogus, because the timestamp only includes 15 bits, unlike voice frames, which contain a 16-bit timestamp. (closes issue #13013) Reported by: jpgrayson Patches: chan_iax2_unwrap_ts.patch uploaded by jpgrayson (license 492) ........ ................ 2008-07-08 16:41 +0000 [r129041-129046] Brett Bryant * main/rtp.c, main/channel.c, channels/chan_dahdi.c, main/manager.c, formats/format_pcm.c, main/logger.c, main/callerid.c, apps/app_parkandannounce.c, apps/app_adsiprog.c, main/pbx.c, main/frame.c, /, channels/chan_sip.c, apps/app_meetme.c, channels/h323/ast_h323.cxx, res/res_limit.c, main/acl.c, channels/iax2-provision.c, pbx/dundi-parser.c, channels/chan_iax2.c: Merged revisions 129045 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r129045 | bbryant | 2008-07-08 11:40:28 -0500 (Tue, 08 Jul 2008) | 7 lines Janitor project to convert sizeof to ARRAY_LEN macro. (closes issue #13002) Reported by: caio1982 Patches: janitor_arraylen5.diff uploaded by caio1982 (license 22) ........ * /, channels/chan_sip.c: Merged revisions 127621 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127621 | bbryant | 2008-07-02 17:16:29 -0500 (Wed, 02 Jul 2008) | 1 line Update transport= in sip so that the option is not broken from a recent commit. ........ * /, channels/chan_sip.c: Merged revisions 127434 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127434 | bbryant | 2008-07-02 12:27:36 -0500 (Wed, 02 Jul 2008) | 1 line Fix to sip_parse_host so that it passes the correct information to sip_registry. ........ 2008-07-08 14:18 +0000 [r129007] Russell Bryant * /, apps/app_fax.c: Merged revisions 129006 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r129006 | russell | 2008-07-08 09:17:37 -0500 (Tue, 08 Jul 2008) | 9 lines Update app_fax for better compatibility with spandsp 0.0.5. Add a call to t38_terminal_release, and make sure that the phase E handler gets called with proper status. (closes issue #13020) Reported by: dimas Patches: v1-appfax.patch uploaded by dimas (license 88) ........ 2008-07-08 10:06 +0000 [r128913-128952] Olle Johansson * /, channels/chan_sip.c: Merged revisions 128951 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r128951 | oej | 2008-07-08 12:02:12 +0200 (Tis, 08 Jul 2008) | 19 lines Merged revisions 128950 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r128950 | oej | 2008-07-08 11:52:21 +0200 (Tis, 08 Jul 2008) | 11 lines Don't hangup the call if we can't resolve the Contact if there's a proxy route set for the call. ---- This comment was added a while ago and today it hit me badly. /* OEJ: Possible issue that may need a check: If we have a proxy route between us and the device, should we care about resolving the contact or should we just send it? */ ........ ................ * /, channels/chan_sip.c: Merged revisions 128927 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r128927 | oej | 2008-07-08 11:26:37 +0200 (Tis, 08 Jul 2008) | 15 lines Merged revisions 128912 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r128912 | oej | 2008-07-08 11:06:08 +0200 (Tis, 08 Jul 2008) | 7 lines Fix issues where repeated messages where ignored, but retransmitted reliably instead of unreliably. Reported by: johan Patches: 12746.txt uploaded by oej (license 306) Tested by: johan (issue #12746) ........ ................ 2008-07-08 00:03 +0000 [r128855-128858] Tilghman Lesher * /: Merged revisions 128857 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r128857 | tilghman | 2008-07-07 19:02:11 -0500 (Mon, 07 Jul 2008) | 15 lines Merged revisions 128856 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r128856 | tilghman | 2008-07-07 19:01:30 -0500 (Mon, 07 Jul 2008) | 7 lines Check for non-NULL before stripping characters. (closes issue #12954) Reported by: bfsworks Patches: 20080701__bug12954.diff.txt uploaded by Corydon76 (license 14) Tested by: deti ........ ................ * apps/app_voicemail.c, /: Merged revisions 128830 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r128830 | tilghman | 2008-07-07 18:25:39 -0500 (Mon, 07 Jul 2008) | 10 lines Merged revisions 128812 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r128812 | tilghman | 2008-07-07 18:21:52 -0500 (Mon, 07 Jul 2008) | 2 lines Stop using deprecated method, as requested by Kevin. ........ ................ 2008-07-07 22:44 +0000 [r128797] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 128796 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r128796 | russell | 2008-07-07 17:42:30 -0500 (Mon, 07 Jul 2008) | 16 lines Merged revisions 128795 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r128795 | russell | 2008-07-07 17:41:48 -0500 (Mon, 07 Jul 2008) | 8 lines Fix handling of when a pvt disappears. Properly return the pvt locked and don't hold the pvt lock while destroying the ast_channel. (closes issue #13014) Reported by: jpgrayson Patches: chan_iax2_ast_iax2_new2.patch uploaded by jpgrayson (license 492) ........ ................ 2008-07-07 20:51 +0000 [r128739] Sean Bright * /, channels/chan_iax2.c: Merged revisions 128738 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r128738 | seanbright | 2008-07-07 16:50:29 -0400 (Mon, 07 Jul 2008) | 17 lines Merged revisions 128737 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r128737 | seanbright | 2008-07-07 16:47:56 -0400 (Mon, 07 Jul 2008) | 9 lines Remove spurious trailing whitespace from log messages and fix a spelling error in a log message. (closes issue #13017) Reported by: jpgrayson Patches: chan_iax2_space_after_newline.patch uploaded by jpgrayson (license 492) chan_iax2_spelling.patch uploaded by jpgrayson (license 492) ........ ................ 2008-07-07 20:31 +0000 [r128601-128735] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 128733 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128733 | mmichelson | 2008-07-07 15:30:46 -0500 (Mon, 07 Jul 2008) | 3 lines Crap ........ * apps/app_voicemail.c, /: Merged revisions 128731 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128731 | mmichelson | 2008-07-07 15:28:33 -0500 (Mon, 07 Jul 2008) | 7 lines If imapfolder=foo were set in voicemail.conf, then when calling VoiceMailMain, app_voicemail would attempt to play a file called vm-foo instead of playing vm-INBOX to play the "new" sound file. This commit fixes that issue. This may fix one of the problems reported in issue #12987 ........ * /, channels/chan_iax2.c: Merged revisions 128640 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r128640 | mmichelson | 2008-07-07 12:09:11 -0500 (Mon, 07 Jul 2008) | 18 lines Merged revisions 128639 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r128639 | mmichelson | 2008-07-07 12:02:28 -0500 (Mon, 07 Jul 2008) | 10 lines By using the iaxdynamicthreadcount to identify a thread, it was possible for thread identifiers to be duplicated. By using a globally-unique monotonically- increasing integer, this is now avoided. (closes issue #13009) Reported by: jpgrayson Patches: chan_iax2_dyn_threadnum.patch uploaded by jpgrayson (license 492) ........ ................ * configs/extensions.conf.sample, /, doc/tex/extensions.tex: Merged revisions 128599 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128599 | mmichelson | 2008-07-07 09:35:27 -0500 (Mon, 07 Jul 2008) | 6 lines Update a few instances of "extensions reload" to "dialplan reload" in the documentation. Patch provided by caio1982 (license 22) ........ 2008-07-06 20:22 +0000 [r128288-128543] Olle Johansson * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 128524 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128524 | oej | 2008-07-06 22:11:37 +0200 (Sön, 06 Jul 2008) | 5 lines - Fixing issues with "sip show settings" - Adding IP address for TCP and/or TLS too if auto-domain is enabled and binding to a different IP address - Fixing documentation in sip.conf.sample ........ * /, channels/chan_sip.c: Merged revisions 128491 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128491 | oej | 2008-07-06 21:14:06 +0200 (Sön, 06 Jul 2008) | 3 lines - Remove unused variable "expiry" - Set global_outboundproxy.force at reload. ........ * doc/realtimetext.txt (added), /: The following patch with references to t140red removed, since it only exists in trunk. Merged revisions 128417 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128417 | oej | 2008-07-06 12:13:45 +0200 (Sön, 06 Jul 2008) | 3 lines Adding documentation on the T.140 support in Asterisk. This is a function that we're the reference implementation on now. :-) ........ * /: Merged revisions 128343 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128343 | oej | 2008-07-06 10:10:27 +0200 (Sön, 06 Jul 2008) | 2 lines Removing the CLI dumpdb command (see asterisk-dev discussion and decision) ........ * /, channels/chan_sip.c: Merged revisions 128290 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128290 | oej | 2008-07-05 23:55:57 +0200 (Lör, 05 Jul 2008) | 5 lines Adding doxygen comments to missing parts, moving some #define ...trying to get my head around the thoughts behind the TCP/TLS stuff and figure out what needs to be done to make it useful... ........ * /, channels/chan_sip.c: Merged revisions 128287 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128287 | oej | 2008-07-05 23:37:57 +0200 (Lör, 05 Jul 2008) | 3 lines Adding TCP and TLS to "sip show settings". TLS needs to have one configuration per configured domain at some point. ........ * /: Blocking changes in trunk. 2008-07-05 21:02 +0000 [r128238-128243] Olle Johansson * /: Keep the "sip-user" structure in 1.6.0, while testing new funky stuff in trunk. * /: Blocking the AGi changes from 1.6.0. Let's test them for a while in trunk before a release. * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 128237 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128237 | oej | 2008-07-05 22:39:54 +0200 (Lör, 05 Jul 2008) | 2 lines Make TCP disabled by default (it's considered experimental) ........ * /, configs/sip.conf.sample: Merged revisions 128236 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128236 | oej | 2008-07-05 22:37:53 +0200 (Lör, 05 Jul 2008) | 2 lines Reformatting the config sample ........ 2008-07-05 15:19 +0000 [r128161] Tilghman Lesher * contrib/scripts/asterisk.ldap-schema, contrib/scripts/asterisk.ldif, /: Merged revisions 128160 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128160 | tilghman | 2008-07-05 10:17:51 -0500 (Sat, 05 Jul 2008) | 7 lines LDAP schema updates (closes issue #12860) Reported by: flyn Patches: asterisk.ldif uploaded by suretec (license 70) asterisk.schema uploaded by suretec (license 70) ........ 2008-07-05 03:40 +0000 [r128124-128127] Matthew Fredrickson * channels/chan_dahdi.c, /: Merged revisions 128125 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128125 | mattf | 2008-07-04 22:39:07 -0500 (Fri, 04 Jul 2008) | 1 line It would help if we actually parsed the ss7_explicitacm option in the config file... ........ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged revisions 128122 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128122 | mattf | 2008-07-04 22:26:42 -0500 (Fri, 04 Jul 2008) | 1 line Add option to wait to be able to explicitly send ACM via the Proceeding() application in the dialplan. Also minor documentation update explaining how to setup multiple signalling links within a linkset ........ 2008-07-04 16:12 +0000 [r128028-128031] Tilghman Lesher * main/pbx.c, /, include/asterisk/pbx.h, pbx/pbx_config.c: Merged revisions 128027 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r128027 | tilghman | 2008-07-04 11:06:34 -0500 (Fri, 04 Jul 2008) | 16 lines Merged revisions 127973 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r127973 | tilghman | 2008-07-03 22:30:30 -0500 (Thu, 03 Jul 2008) | 8 lines Fix the 'dialplan remove extension' logic, so that it a) works with cidmatch, and b) completes contexts correctly when the extension is ambiguous. (closes issue #12980) Reported by: licedey Patches: 20080703__bug12980.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ ................ 2008-07-03 22:23 +0000 [r127905] Kevin P. Fleming * Makefile, /, apps/Makefile, main/editline/np/vis.c: Merged revisions 127903 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r127903 | kpfleming | 2008-07-03 17:23:04 -0500 (Thu, 03 Jul 2008) | 20 lines Merged revisions 127892,127895 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r127892 | kpfleming | 2008-07-03 17:18:38 -0500 (Thu, 03 Jul 2008) | 6 lines a couple of small Solaris-related fixes (closes issue #11885) Reported by: snuffy, asgaroth ........ r127895 | kpfleming | 2008-07-03 17:20:16 -0500 (Thu, 03 Jul 2008) | 3 lines remove this, it has been moved to the main Makefile ........ ................ 2008-07-03 19:12 +0000 [r127830] Steve Murphy * main/cdr.c, main/channel.c, channels/chan_dahdi.c, main/pbx.c, /, channels/chan_sip.c, main/features.c, include/asterisk/cdr.h: Merged revisions 127793 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r127793 | murf | 2008-07-03 11:16:44 -0600 (Thu, 03 Jul 2008) | 38 lines Merged revisions 127663 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r127663 | murf | 2008-07-02 18:16:25 -0600 (Wed, 02 Jul 2008) | 30 lines The CDRfix4/5/6 omnibus cdr fixes. (closes issue #10927) Reported by: murf Tested by: murf, deeperror (closes issue #12907) Reported by: falves11 Tested by: murf, falves11 (closes issue #11849) Reported by: greyvoip As to 11849, I think these changes fix the core problems brought up in that bug, but perhaps not the more global problems created by the limitations of CDR's themselves not being oriented around transfers. Reopen if necc, but bug reports are not the best medium for enhancement discussions. We need to start a second-generation CDR standardization effort to cover transfers. (closes issue #11093) Reported by: rossbeer Tested by: greyvoip, murf ........ ................ 2008-07-03 16:50 +0000 [r127790-127792] Olle Johansson * /, channels/chan_sip.c: Merged revisions 127791 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127791 | oej | 2008-07-03 18:48:23 +0200 (Tor, 03 Jul 2008) | 5 lines Make sure we stop session timers as soon as we start hanging up an active call. May fix issue 12919. ........ * /, channels/chan_sip.c: Merged revisions 127779 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127779 | oej | 2008-07-03 18:25:59 +0200 (Tor, 03 Jul 2008) | 4 lines Revert some logic for session timers. We do send in-dialog requests that should not have session-timer require headers, like MESSAGE and REFER. So in the future, only add them on requests and responses that are related to INVITEs and re-INVITEs. ........ 2008-07-03 16:24 +0000 [r127778] Kevin P. Fleming * /, configure, include/asterisk/autoconfig.h.in, configure.ac, acinclude.m4: Merged revisions 127767 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127767 | kpfleming | 2008-07-03 11:22:02 -0500 (Thu, 03 Jul 2008) | 2 lines some minor fixes found while working on issue #12911 (and block the rev from 1.4 since the equivalent is already here) ........ 2008-07-02 21:10 +0000 [r127567] Mark Michelson * /, doc/janitor-projects.txt: Merged revisions 127566 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127566 | mmichelson | 2008-07-02 16:09:18 -0500 (Wed, 02 Jul 2008) | 4 lines Add a janitor project to use ARRAY_LEN instead of in-line sizeof() and division. ........ 2008-07-02 20:49 +0000 [r127559-127563] Mark Michelson * /, channels/chan_agent.c: Merged revisions 127562 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r127562 | mmichelson | 2008-07-02 15:49:08 -0500 (Wed, 02 Jul 2008) | 11 lines Merged revisions 127560 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r127560 | mmichelson | 2008-07-02 15:47:38 -0500 (Wed, 02 Jul 2008) | 3 lines Fix thread-safety of some of the pbx_builtin_getvar_helper calls ........ ................ 2008-07-02 19:48 +0000 [r127467-127503] Tilghman Lesher * /, main/acl.c: Merged revisions 127466 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127466 | tilghman | 2008-07-02 13:31:11 -0500 (Wed, 02 Jul 2008) | 6 lines Solaris fix (closes issue #12949) Reported by: snuffy Patches: bug_12949.diff uploaded by snuffy (license 35) ........ 2008-07-02 14:30 +0000 [r127396-127399] Sean Bright * cdr/cdr_tds.c, /: Merged revisions 127398 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127398 | seanbright | 2008-07-02 10:30:09 -0400 (Wed, 02 Jul 2008) | 1 line Fix a bug I noticed while doing the previous merge ........ * cdr/cdr_tds.c, /, doc/tex/freetds.tex, configure, include/asterisk/autoconfig.h.in, configure.ac, UPGRADE.txt: Merged revisions 126226,126513 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r126226 | seanbright | 2008-06-28 17:28:16 -0400 (Sat, 28 Jun 2008) | 8 lines Merge in changes from my cdr-tds-conversion branch. This changes the internal implementation from using the volatile libtds, to using the db-lib front end. The unintended side effect of this is that we support (at least) versions 0.62 through 0.82 of the FreeTDS distribution without any #ifdef ugliness. (closes issue #12844) Reported by: jcollie ........ r126513 | seanbright | 2008-06-30 07:57:42 -0400 (Mon, 30 Jun 2008) | 4 lines Cast a few more strings to char *, so that we can compile cleanly against FreeTDS 0.60. Update the docs to reflect that we can now compile and run against all modern releases of FreeTDS (0.60 through 0.82) ........ * /: Unblock some revisions so I can merge the cdr_tds changes from trunk 2008-07-02 12:09 +0000 [r127364] Russell Bryant * doc/CODING-GUIDELINES, /: Merged revisions 127363 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127363 | russell | 2008-07-02 07:08:33 -0500 (Wed, 02 Jul 2008) | 13 lines Add a locking section to the coding guidelines document. This section covers some locking fundamentals, as well as some information on locking as it is used in Asterisk. It describes some of the ways that are used and could be used to achieve deadlock avoidance. It also demonstrates the unfortunate conclusion that with the use of recursive locks, none of the constructs in use today are failsafe from deadlocks. Finally, it makes some recommendations for new code being written. As proper locking strategies is a complex subject, this section still has room for expansion and improvement. This is a result of collaboration between Luigi Rizzo and myself on the asterisk-dev mailing list. ........ 2008-07-02 02:49 +0000 [r127298] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 127297 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127297 | tilghman | 2008-07-01 21:48:43 -0500 (Tue, 01 Jul 2008) | 12 lines Change the global timer B to be dependent on the value of the T1 timer, as recommended in RFC 3261, instead of being hardcoded to 32 seconds. This is important for LANs, as it allows autocongestion to occur much more quickly, if desired by the local PBX administrator. It also corrects a bug: if the T1 timer was increased beyond 500ms, then timer B would have been set at a much lower value than recommended. (closes issue #12544) Reported by: kactus Patches: 20080616__bug12544.diff.txt uploaded by Corydon76 (license 14) Tested by: kactus ........ 2008-07-01 23:39 +0000 [r127246] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 127245 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r127245 | mmichelson | 2008-07-01 18:38:12 -0500 (Tue, 01 Jul 2008) | 13 lines Merged revisions 127244 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r127244 | mmichelson | 2008-07-01 18:36:40 -0500 (Tue, 01 Jul 2008) | 5 lines Add error message to failed open(2) calls inside the copy() function of app_voicemail. This idea came as part of my work in helping to resolve issue #12764. ........ ................ 2008-07-01 21:19 +0000 [r127163] Brett Bryant * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 127154 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127154 | bbryant | 2008-07-01 16:03:52 -0500 (Tue, 01 Jul 2008) | 2 lines Add a configuration option so the global outboundproxy can use tcptls without it being defined by each sip user. ........ 2008-07-01 21:16 +0000 [r127156-127158] Mark Michelson * main/channel.c, /: Merged revisions 127157 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127157 | mmichelson | 2008-07-01 16:16:00 -0500 (Tue, 01 Jul 2008) | 8 lines Place the delay in __ast_answer prior to the channel-specific answer callback. This change differs from commit 127113 in that now the channel is not set to AST_STATE_UP until after the answer callback. (closes issue #12924) Reported by: snyfer ........ * main/channel.c, /: Merging Revision 127113 from trunk 2008-07-01 20:52 +0000 [r127153] Jason Parker * Makefile, /: Merged revisions 127152 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127152 | qwell | 2008-07-01 15:51:43 -0500 (Tue, 01 Jul 2008) | 7 lines Fix a typo that caused this asterisk.conf to not get correctly generated. (closes issue #12966) Reported by: ibc Patches: 12966.patch uploaded by bkruse (license 132) ........ 2008-07-01 20:29 +0000 [r127085-127149] Tilghman Lesher * build_tools/cflags.xml, /, channels/chan_iax2.c: Merged revisions 127143 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r127143 | tilghman | 2008-07-01 15:28:54 -0500 (Tue, 01 Jul 2008) | 10 lines Merged revisions 127133 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r127133 | tilghman | 2008-07-01 15:25:37 -0500 (Tue, 01 Jul 2008) | 2 lines Disable the old, slow search for matching callno in chan_iax2 (but allow it to be reenabled for debugging) ........ ................ * /, channels/chan_iax2.c: Merged revisions 127074 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r127074 | tilghman | 2008-07-01 14:20:25 -0500 (Tue, 01 Jul 2008) | 16 lines Merged revisions 127068 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r127068 | tilghman | 2008-07-01 13:52:53 -0500 (Tue, 01 Jul 2008) | 8 lines Change around how we schedule pings and lagrqs, and fix a reason why the jobs were not getting properly cancelled. (closes issue #12903) Reported by: stevedavies Patches: 20080620__bug12903__2.diff.txt uploaded by Corydon76 (license 14) Tested by: stevedavies ........ ................ 2008-07-01 16:53 +0000 [r127001] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 127000 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r127000 | tilghman | 2008-07-01 11:52:29 -0500 (Tue, 01 Jul 2008) | 10 lines Merged revisions 126999 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r126999 | tilghman | 2008-07-01 11:50:46 -0500 (Tue, 01 Jul 2008) | 2 lines Suppress annoying warning by finding the remaining cases where the callno is not in the hash. ........ ................ 2008-07-01 15:05 +0000 [r126756-126904] Olle Johansson * /, channels/chan_sip.c: Merged revisions 126903 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r126903 | oej | 2008-07-01 17:03:59 +0200 (Tis, 01 Jul 2008) | 15 lines Merged revisions 126902 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r126902 | oej | 2008-07-01 16:59:31 +0200 (Tis, 01 Jul 2008) | 7 lines Use domain part of SIP uri in register= configuration as fromdomain. Reported by: one47 Patches: sip-reg-fromdom2.dpatch uploaded by one47 (license 23) (closes issue #12474) ........ ................ * /, channels/chan_sip.c: Merged revisions 126900 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r126900 | oej | 2008-07-01 16:32:15 +0200 (Tis, 01 Jul 2008) | 16 lines Merged revisions 126899 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r126899 | oej | 2008-07-01 16:27:33 +0200 (Tis, 01 Jul 2008) | 8 lines Handle escaped URI's in call pickups. Patch by oej and IgorG. Reported by: IgorG Patches: bug12299-11062-v2.patch uploaded by IgorG (license 20) Tested by: IgorG, oej (closes issue #12299) ........ ................ * /, configs/sip.conf.sample: Merged revisions 126845 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r126845 | oej | 2008-07-01 14:54:57 +0200 (Tis, 01 Jul 2008) | 14 lines Merged revisions 126844 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r126844 | oej | 2008-07-01 14:53:01 +0200 (Tis, 01 Jul 2008) | 5 lines Clear up documentation on "domain=" setting in sip.conf Reported by: davidw (closes issue #12413) ........ ................ * /, channels/chan_sip.c: Merged revisions 126790 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r126790 | oej | 2008-07-01 13:58:17 +0200 (Tis, 01 Jul 2008) | 14 lines Merged revisions 126789 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r126789 | oej | 2008-07-01 13:51:38 +0200 (Tis, 01 Jul 2008) | 6 lines Report 200 OK to all in-dialog OPTIONs requests (to confirm that the dialog exist). Don't bother checking the request URI. (closes issue #11264) Reported by: ibc ........ ................ * /, channels/chan_sip.c: Merged revisions 126755 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r126755 | oej | 2008-07-01 11:51:22 +0200 (Tis, 01 Jul 2008) | 15 lines Merged revisions 126735 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r126735 | oej | 2008-07-01 09:49:15 +0200 (Tis, 01 Jul 2008) | 7 lines Fix bad XML for hold notification. Reported by: gowen72 Patches: hold.patch uploaded by gowen72 (license 432) (closes issue #12942) ........ ................ 2008-06-30 22:34 +0000 [r126676] Jeff Peeler * configs/zapata.conf.sample (removed), configs/chan_dahdi.conf.sample (added), /: Merged revisions 126675 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r126675 | jpeeler | 2008-06-30 17:34:08 -0500 (Mon, 30 Jun 2008) | 1 line rename zapata.conf.sample to chan_dahdi.conf.sample ........ 2008-06-30 20:32 +0000 [r126638] Matthew Fredrickson * channels/chan_dahdi.c, /: Merged revisions 126637 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r126637 | mattf | 2008-06-30 15:25:46 -0500 (Mon, 30 Jun 2008) | 1 line Add support to see MTP2 down events when the link layer drops in SS7 ........ 2008-06-30 16:09 +0000 [r126575] Russell Bryant * /, include/asterisk/lock.h: Merged revisions 126574 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r126574 | russell | 2008-06-30 11:07:25 -0500 (Mon, 30 Jun 2008) | 18 lines Merged revisions 126573 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r126573 | russell | 2008-06-30 11:05:08 -0500 (Mon, 30 Jun 2008) | 10 lines Fix a typo in the non-DEBUG_THREADS version of the recently added DEADLOCK_AVOIDANCE() macro. This caused the lock to not actually be released, and as a result, not avoid deadlocks at all. This resolves the issues reported in the last while about Asterisk locking up all over the place (and most commonly, in chan_iax2). (closes issue #12927) (closes issue #12940) (closes issue #12925) (potentially closes others ...) ........ ................ 2008-06-30 13:07 +0000 [r126518] Olle Johansson * /, channels/chan_sip.c: Merged revisions 126517 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r126517 | oej | 2008-06-30 15:03:53 +0200 (MÃ¥n, 30 Jun 2008) | 20 lines The following patch with some changes for trunk... Merged revisions 126516 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r126516 | oej | 2008-06-30 14:50:55 +0200 (MÃ¥n, 30 Jun 2008) | 10 lines Send all responses to an INVITE reliably, so that we retransmit if we don't get an ACK and also fail if we don't get the very same precious ACK. Based on patch by tsearle, with my own additions. (closes issue #12951) Reported by: tsearle Patches: busy_retransmit.patch uploaded by tsearle (license 373) ........ ................ 2008-06-29 17:02 +0000 [r126362-126364] Kevin P. Fleming * apps/app_zapbarge.c (removed): finish converting this module * pbx/pbx_gtkconsole.c, /, configure, configure.ac, pbx/pbx_lua.c, pbx/Makefile: Merged revisions 126356 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r126356 | kpfleming | 2008-06-29 09:19:29 -0700 (Sun, 29 Jun 2008) | 9 lines various minor fixes created while i worked on getting *every* Asterisk module to build on laptop in dev mode: remove weird pre-setting of LUA paths; they are not necessary; also use the proper path for including the files in pbx_lua.c make the compiler shut up about some ignored function results in pbx_gtkconsole; this module is badly coded anyway ........ * apps/app_dahdibarge.c (added): don't know how this got missed in the DAHDI conversion of this branch 2008-06-29 13:20 +0000 [r126227-126322] Sean Bright * /, cdr/cdr_pgsql.c: Merged revisions 126274 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r126274 | seanbright | 2008-06-29 08:06:46 -0400 (Sun, 29 Jun 2008) | 6 lines Quote column names when inserting CDRs into postgres to avoid conflicts with reserved words. (closes issue #12947) Reported by: panolex ........ 2008-06-28 15:58 +0000 [r126155-126188] Kevin P. Fleming * Makefile, /: update this branch to use the trunk goodness version of menuselect 2008-06-27 22:43 +0000 [r126058-126112] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 126057 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r126057 | tilghman | 2008-06-27 17:10:34 -0500 (Fri, 27 Jun 2008) | 12 lines Merged revisions 126056 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r126056 | tilghman | 2008-06-27 17:01:09 -0500 (Fri, 27 Jun 2008) | 4 lines When we get a 408 Timeout, don't stop trying to re-register. (closes issue #12863) Reported by: ricvil ........ ................ 2008-06-27 21:00 +0000 [r126023] Mark Michelson * apps/app_queue.c: Port revisions 124661 and 123650 from trunk to 1.6.0 Thanks to Atis Lezdins for pointing this out on the asterisk-dev mailing list 2008-06-27 19:20 +0000 [r125994] Russell Bryant * /, doc/siptls.txt: Merged revisions 125988 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125988 | russell | 2008-06-27 14:19:08 -0500 (Fri, 27 Jun 2008) | 3 lines Fix a typo. Someone on IRC copied this literally and then wondered why it wasn't working. :) ........ 2008-06-27 19:06 +0000 [r125981-125985] Matthew Fredrickson * channels/chan_dahdi.c, /: Merged revisions 125984 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125984 | mattf | 2008-06-27 14:05:40 -0500 (Fri, 27 Jun 2008) | 1 line Revert this part of the fix. We'll fix it in libss7 ........ * channels/chan_dahdi.c, /: Merged revisions 125982 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125982 | mattf | 2008-06-27 14:00:44 -0500 (Fri, 27 Jun 2008) | 1 line Obviously somebody didn't compile with libss7 support when doing the DAHDI conversion. ........ * channels/chan_dahdi.c, /: Merged revisions 125980 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125980 | mattf | 2008-06-27 13:32:17 -0500 (Fri, 27 Jun 2008) | 1 line Add support for new commands to block/unblock all CICs on a linkset ........ 2008-06-27 17:36 +0000 [r125948] Brett Bryant * /, channels/chan_sip.c: Merged revisions 125947 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125947 | bbryant | 2008-06-27 12:35:41 -0500 (Fri, 27 Jun 2008) | 1 line Small error in the function that converts peer transports to a string. ........ 2008-06-27 16:29 +0000 [r125892] Brett Bryant * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 125891 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125891 | bbryant | 2008-06-27 11:28:06 -0500 (Fri, 27 Jun 2008) | 6 lines Change the way that the transport option works for sip users. transport will now take multiple arguments, the first one listed will be the one used for new dialogs, and the rest listed will be acceptable ways for that peer to contact us. This fixes a minor bug where, because SIP TCP/UDP run on the same port, could cause a TCP peer to be saved in the ast_db. There will also be warnings when a transport is changed for an unexpected reason. (issue #12799) ........ 2008-06-27 16:19 +0000 [r125859-125863] Mark Michelson * /, apps/app_queue.c: Merged revisions 125855 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125855 | mmichelson | 2008-06-27 11:16:13 -0500 (Fri, 27 Jun 2008) | 5 lines Ensure the thread-safety of the monexec variable in app_queue. Thanks to Russell for pointing out the problem ........ 2008-06-27 16:01 +0000 [r125854] Tilghman Lesher * channels/chan_local.c, /: Merged revisions 125853 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125853 | tilghman | 2008-06-27 11:00:05 -0500 (Fri, 27 Jun 2008) | 3 lines Revert half of the fix, as this part may have been unnecessary (related to issue #12914) Requested here: http://lists.digium.com/pipermail/asterisk-dev/2008-June/033658.html ........ 2008-06-27 14:57 +0000 [r125800-125852] Mark Michelson * main/asterisk.c, main/channel.c, channels/chan_iax2.c: Make sure to only include dahdi/user.h if we have installed DAHDI. * channels/chan_iax2.c: I accidentally committed a change to chan_iax2.c in addition to a change to app_queue.c. Reverting the change to chan_iax2.c, even though it may turn out that this change is necessary. * utils/Makefile, /: Merged revisions 125799 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125799 | mmichelson | 2008-06-27 09:14:09 -0500 (Fri, 27 Jun 2008) | 3 lines Remove an unneeded target from the Makefile ........ 2008-06-27 14:09 +0000 [r125742-125797] Tilghman Lesher * /, main/utils.c, include/asterisk/lock.h: Merged revisions 125794 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r125794 | tilghman | 2008-06-27 08:54:13 -0500 (Fri, 27 Jun 2008) | 10 lines Merged revisions 125793 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125793 | tilghman | 2008-06-27 08:45:03 -0500 (Fri, 27 Jun 2008) | 2 lines In this debugging function, copy to a buffer instead of using potentially unsafe pointers. ........ ................ * channels/chan_local.c, /: Merged revisions 125741 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r125741 | tilghman | 2008-06-27 07:28:38 -0500 (Fri, 27 Jun 2008) | 15 lines Merged revisions 125740 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125740 | tilghman | 2008-06-27 07:19:39 -0500 (Fri, 27 Jun 2008) | 7 lines Add proper deadlock avoidance. (closes issue #12914) Reported by: ozan Patches: 20080625__bug12914.diff.txt uploaded by Corydon76 (license 14) Tested by: ozan ........ ................ 2008-06-27 07:41 +0000 [r125704] Philippe Sultan * /, include/asterisk/jabber.h, res/res_jabber.c: Merged revisions 125703 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125703 | phsultan | 2008-06-27 09:28:17 +0200 (Fri, 27 Jun 2008) | 1 line Fix a compile time error that occurs if OpenSSL is not installed. Reported by Noel Morais on the users mailing list ........ 2008-06-27 01:09 +0000 [r125648-125684] Mark Michelson * apps/app_queue.c, channels/chan_iax2.c: SENTINEL is not defined in 1.6.0 * /, apps/app_queue.c: Merged revisions 125666 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125666 | mmichelson | 2008-06-26 19:22:03 -0500 (Thu, 26 Jun 2008) | 3 lines Make this compile with dev-mode on ........ * /, apps/app_queue.c: Merged revisions 125649 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125649 | mmichelson | 2008-06-26 19:15:54 -0500 (Thu, 26 Jun 2008) | 15 lines The monitor-join option for queues was deprecated in favor of using MixMonitor to mix audio. However, it was pointed out to me that because of this, the command set for the MONITOR_EXEC variable is ignored as well. This means that people can't do their own custom mixing commands at the end of recordings in order to make, for instance, stereo recordings of calls. With this patch, app_queue will set the "joinfiles" variable for the channel's monitor if MONITOR_EXEC is not zero-length. This means that for normal audio mixing, MixMonitor is still the preferred choice, but we allow custom mixing to be done with the two Monitor streams if desired. (closes issue #12923) Reported by: snyfer ........ 2008-06-26 23:06 +0000 [r125592] Mark Michelson * /, apps/app_queue.c: Merged revisions 125591 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125591 | mmichelson | 2008-06-26 18:06:18 -0500 (Thu, 26 Jun 2008) | 3 lines Fix a really stupid mistake ........ 2008-06-26 23:05 +0000 [r125590] Jason Parker * /, main/utils.c: Merged revisions 125589 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r125589 | qwell | 2008-06-26 18:04:18 -0500 (Thu, 26 Jun 2008) | 9 lines Merged revisions 125587 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125587 | qwell | 2008-06-26 18:03:15 -0500 (Thu, 26 Jun 2008) | 1 line Make sure to unlock the lock_info lock (huh?). Possible deadlock? ........ ................ 2008-06-26 23:04 +0000 [r125588] Mark Michelson * /, apps/app_queue.c: Merged revisions 125586 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r125586 | mmichelson | 2008-06-26 18:01:02 -0500 (Thu, 26 Jun 2008) | 19 lines Merged revisions 125585 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125585 | mmichelson | 2008-06-26 17:52:39 -0500 (Thu, 26 Jun 2008) | 11 lines Add the interface of a queue member to the output of the "queue show" command so that it can easily be associated with a queue member's name. This helps so that the appropriate queue member can be removed or paused since the interface is required, not the member's name. (closes issue #12783) Reported by: davevg Patches: app_queue.diff uploaded by davevg (license 209) with small mod from me ........ ................ 2008-06-26 22:50 +0000 [r125584] Tilghman Lesher * /, contrib/scripts/astcli: Merged revisions 125583 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125583 | tilghman | 2008-06-26 17:49:16 -0500 (Thu, 26 Jun 2008) | 2 lines Don't hang if the command is blank ........ 2008-06-26 22:06 +0000 [r125478-125532] Mark Michelson * /, apps/app_queue.c: Merged revisions 125477 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r125477 | mmichelson | 2008-06-26 15:57:41 -0500 (Thu, 26 Jun 2008) | 19 lines Merged revisions 125476 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125476 | mmichelson | 2008-06-26 15:56:01 -0500 (Thu, 26 Jun 2008) | 11 lines Prior to this patch, the "queue show" command used cached information for realtime queues instead of giving up-to-date info. Now realtime is queried for the latest and greatest in queue info. (closes issue #12858) Reported by: bcnit Patches: queue_show.patch uploaded by putnopvut (license 60) ........ ................ 2008-06-26 17:07 +0000 [r125388] Olle Johansson * /, channels/chan_sip.c: Merged revisions 125385 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r125385 | oej | 2008-06-26 18:54:22 +0200 (Tor, 26 Jun 2008) | 12 lines Merged revisions 125384 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125384 | oej | 2008-06-26 18:32:08 +0200 (Tor, 26 Jun 2008) | 3 lines Add support for peer realm based auth (a few missing lines, the rest is well documented but never worked) ........ ................ 2008-06-26 15:52 +0000 [r125280-125334] Kevin P. Fleming * channels/chan_dahdi.c, /: Merged revisions 125333 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r125333 | kpfleming | 2008-06-26 10:50:07 -0500 (Thu, 26 Jun 2008) | 13 lines Merged revisions 125327 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125327 | kpfleming | 2008-06-26 10:30:33 -0500 (Thu, 26 Jun 2008) | 5 lines ensure that (whenever possible) if we generate a log message because an ioctl() call to DAHDI/Zaptel failed, that we include the reason it failed by including the stringified error number (issue AST-80) ........ ................ * /, res/res_musiconhold.c: Merged revisions 125279 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125279 | kpfleming | 2008-06-26 07:09:24 -0500 (Thu, 26 Jun 2008) | 2 lines fix compile failure found by buildbot (go, buildbot!) ........ 2008-06-26 11:08 +0000 [r125192-125278] Tilghman Lesher * main/rtp.c, /: Merged revisions 125277 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r125277 | tilghman | 2008-06-26 06:02:11 -0500 (Thu, 26 Jun 2008) | 15 lines Merged revisions 125276 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125276 | tilghman | 2008-06-26 06:01:21 -0500 (Thu, 26 Jun 2008) | 7 lines Check for rtcp structure before trying to delete schedule. (closes issue #12872) Reported by: destiny6628 Patches: 20080621__bug12872.diff.txt uploaded by Corydon76 (license 14) Tested by: destiny6628 ........ ................ * configs/agents.conf.sample, /: Merged revisions 125223 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r125223 | tilghman | 2008-06-25 20:25:16 -0500 (Wed, 25 Jun 2008) | 12 lines Merged revisions 125218 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125218 | tilghman | 2008-06-25 20:24:26 -0500 (Wed, 25 Jun 2008) | 4 lines Document ackcall=always. (closes issue #12852) Reported by: davidw ........ ................ * configs/http.conf.sample, /: Merged revisions 125191 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125191 | tilghman | 2008-06-25 20:11:43 -0500 (Wed, 25 Jun 2008) | 6 lines Update sample configuration to match what are now the defaults for the prefix. (closes issue #12838, related to issue #12198) Reported by: pabelanger Patches: http.conf.diff2 uploaded by pabelanger (license 224) ........ 2008-06-25 23:20 +0000 [r125146] Kevin P. Fleming * main/channel.c, channels/chan_dahdi.c, apps/app_flash.c, configure, codecs/codec_dahdi.c, apps/app_rpt.c, main/asterisk.c, /, apps/app_meetme.c, main/Makefile, apps/app_dahdiscan.c, apps/app_dahdiras.c, configure.ac, include/asterisk/dahdi.h (removed), res/res_musiconhold.c, channels/chan_iax2.c: Merged revisions 125138 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r125138 | kpfleming | 2008-06-25 18:05:28 -0500 (Wed, 25 Jun 2008) | 18 lines Merged revisions 125132 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125132 | kpfleming | 2008-06-25 17:21:30 -0500 (Wed, 25 Jun 2008) | 10 lines allow tonezone to live in a different place than DAHDI/Zaptel, since dahdi-tools and dahdi-linux are now separate packages and can be installed in different places don't include tonezone.h in dahdi_compat.h, because only a couple of modules need it get app_rpt building again after the DAHDI changes (closes issue #12911) Reported by: tzafrir ........ ................ 2008-06-25 01:13 +0000 [r124964-124967] Tilghman Lesher * channels/chan_dahdi.c, /, include/asterisk/lock.h: Merged revisions 124966 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r124966 | tilghman | 2008-06-24 20:08:37 -0500 (Tue, 24 Jun 2008) | 15 lines Merged revisions 124965 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r124965 | tilghman | 2008-06-24 19:46:24 -0500 (Tue, 24 Jun 2008) | 7 lines Pvt deadlock causes some channels to get stuck in Reserved status. (closes issue #12621) Reported by: fabianoheringer Patches: 20080612__bug12621.diff.txt uploaded by Corydon76 (license 14) Tested by: fabianoheringer ........ ................ * apps/app_voicemail.c, /: Merged revisions 124912 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r124912 | tilghman | 2008-06-24 16:18:52 -0500 (Tue, 24 Jun 2008) | 16 lines Merged revisions 124910 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r124910 | tilghman | 2008-06-24 16:08:52 -0500 (Tue, 24 Jun 2008) | 8 lines Occasionally control characters find their way into CallerID. These need to be stripped prior to placing CallerID in the headers of an email. (closes issue #12759) Reported by: RobH Patches: 20080602__bug12759__2.diff.txt uploaded by Corydon76 (license 14) Tested by: RobH ........ ................ 2008-06-24 17:52 +0000 [r124871-124873] Philippe Sultan * /, res/res_jabber.c: Merged revisions 124872 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r124872 | phsultan | 2008-06-24 19:50:22 +0200 (Tue, 24 Jun 2008) | 6 lines Subscribe to buddy's presence only if we really need to. That is, if the corresponding roster item has a subscription value set to "none" or "from". Make the code more readable. ........ * /, res/res_jabber.c: Merged revisions 124870 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r124870 | phsultan | 2008-06-24 19:28:39 +0200 (Tue, 24 Jun 2008) | 1 line Code simplification ........ 2008-06-23 15:44 +0000 [r124708] Dwayne M. Hubbard * /: blocked revision 124707, taskprocessors are not in 1.6.0 2008-06-22 03:18 +0000 [r124542] Steve Murphy * apps/app_forkcdr.c, /: Merged revisions 124541 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r124541 | murf | 2008-06-21 20:58:06 -0600 (Sat, 21 Jun 2008) | 17 lines Merged revisions 124540 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r124540 | murf | 2008-06-21 20:54:52 -0600 (Sat, 21 Jun 2008) | 9 lines (closes issue #12910) Reported by: chris-mac Sorry, my testing did not contain the simple case of forkCDR(v), I am much embarrassed to admit. If I had, I would have more solidly initialized the opts element for varset. ........ ................ 2008-06-21 12:54 +0000 [r124397-124506] Tilghman Lesher * /, res/res_config_ldap.c: Merged revisions 124505 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r124505 | tilghman | 2008-06-21 07:53:48 -0500 (Sat, 21 Jun 2008) | 4 lines Reduce warning to debug, otherwise we flood the log when we (legitimately) can't find a record. (Closes issue #12908) ........ * apps/app_rpt.c, /: Merged revisions 124451 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r124451 | tilghman | 2008-06-20 18:13:21 -0500 (Fri, 20 Jun 2008) | 14 lines Merged revisions 124450 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r124450 | tilghman | 2008-06-20 18:12:33 -0500 (Fri, 20 Jun 2008) | 6 lines usleep with a value over 1,000,000 is nonportable. Changing to use sleep() instead. (closes issue #12814) Reported by: pputman Patches: app_rtp_sleep.patch uploaded by pputman (license 81) ........ ................ * /, main/app.c: Merged revisions 124396 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r124396 | tilghman | 2008-06-20 17:04:37 -0500 (Fri, 20 Jun 2008) | 11 lines Merged revisions 124395 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r124395 | tilghman | 2008-06-20 17:02:55 -0500 (Fri, 20 Jun 2008) | 3 lines If the last character in a string to be parsed is the delimiter, then we should count that final empty string as an additional argument. ........ ................ 2008-06-20 21:48 +0000 [r124394] Jeff Gehlbach * doc/asterisk-mib.txt, /, doc/digium-mib.txt: Merged revisions 124392-124393 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r124392 | jeffg | 2008-06-20 17:36:01 -0400 (Fri, 20 Jun 2008) | 9 lines Merged revisions 124372 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r124372 | jeffg | 2008-06-20 17:14:40 -0400 (Fri, 20 Jun 2008) | 1 line Fix issues in digium-mib.txt and asterisk-mib.txt to placate smilint - bug 12905 ........ ................ r124393 | jeffg | 2008-06-20 17:43:18 -0400 (Fri, 20 Jun 2008) | 12 lines (Missed committing . on previous commit.....) Merged revisions 124372 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r124372 | jeffg | 2008-06-20 17:14:40 -0400 (Fri, 20 Jun 2008) | 1 line Fix issues in digium-mib.txt and asterisk-mib.txt to placate smilint - bug 12905 ........ ................ ................ 2008-06-20 20:18 +0000 [r124317] Tilghman Lesher * channels/chan_local.c, /: Merged revisions 124316 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r124316 | tilghman | 2008-06-20 15:17:04 -0500 (Fri, 20 Jun 2008) | 16 lines Merged revisions 124315 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r124315 | tilghman | 2008-06-20 15:16:02 -0500 (Fri, 20 Jun 2008) | 8 lines When using a Local channel, started by a call file, with a destination of an AGI script, the AGI script does not always get notified of a hangup if the underlying channel hangs up early. (closes issue #11833) Reported by: IgorG Patches: local_hangup-v1.diff uploaded by IgorG (license 20) ........ ................ 2008-06-20 16:31 +0000 [r124244-124279] Mark Michelson * main/ast_expr2.fl, include/asterisk/doxyref.h, /, main/ast_expr2f.c: Merged revisions 124278 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r124278 | mmichelson | 2008-06-20 11:30:18 -0500 (Fri, 20 Jun 2008) | 6 lines Change references to doc/channelvariables.txt to doc/tex/channelvariables.tex. This issue came up on the asterisk-dev mailing list. ........ * /, channels/chan_sip.c: Merged revisions 124243 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r124243 | mmichelson | 2008-06-20 10:20:11 -0500 (Fri, 20 Jun 2008) | 9 lines Add a missing "ChannelType" header to one of the "PeerStatus" manager events in chan_sip (closes issue #12904) Reported by: eliel Patches: chan_sip.c.patch uploaded by eliel (license 64) ........ 2008-06-19 23:02 +0000 [r124184] Tilghman Lesher * channels/chan_dahdi.c, /: Merged revisions 124183 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r124183 | tilghman | 2008-06-19 17:59:41 -0500 (Thu, 19 Jun 2008) | 15 lines Merged revisions 124182 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r124182 | tilghman | 2008-06-19 17:53:22 -0500 (Thu, 19 Jun 2008) | 7 lines It's possible for a hangup to be received, even just after the initial cid spill. (closes issue #12453) Reported by: Alex728 Patches: 20080604__bug12453.diff.txt uploaded by Corydon76 (license 14) ........ ................ 2008-06-19 20:32 +0000 [r124124] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 124121 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r124121 | mmichelson | 2008-06-19 15:30:23 -0500 (Thu, 19 Jun 2008) | 16 lines Merged revisions 124112 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r124112 | mmichelson | 2008-06-19 15:28:41 -0500 (Thu, 19 Jun 2008) | 8 lines Fix IMAP forwarding so that messages are sent to the proper mailbox. (closes issue #12897) Reported by: jaroth Patches: destination_forward.patch uploaded by jaroth (license 50) ........ ................ 2008-06-19 19:49 +0000 [r124065] Brett Bryant * /, main/utils.c: Merged revisions 124064 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r124064 | bbryant | 2008-06-19 14:48:26 -0500 (Thu, 19 Jun 2008) | 2 lines Add errors that report any locks held by threads when they are being closed. ........ 2008-06-19 18:57 +0000 [r124026] Brett Bryant * /, channels/chan_sip.c: Merged revisions 124024 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r124024 | bbryant | 2008-06-19 13:57:04 -0500 (Thu, 19 Jun 2008) | 2 lines Fix bug in sip registration that sets the default port to 5060 for tls. ........ 2008-06-19 17:58 +0000 [r123871-123989] Tilghman Lesher * /, res/res_config_ldap.c: Merged revisions 123952 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r123952 | tilghman | 2008-06-19 12:22:27 -0500 (Thu, 19 Jun 2008) | 6 lines Don't change pointers that need to be later passed back for deallocation. (closes issue #12572) Reported by: flyn Patches: 20080613__bug12572.diff.txt uploaded by Corydon76 (license 14) ........ * main/channel.c, /: Merged revisions 123931 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123931 | tilghman | 2008-06-19 12:02:54 -0500 (Thu, 19 Jun 2008) | 13 lines Merged revisions 123930 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123930 | tilghman | 2008-06-19 11:58:19 -0500 (Thu, 19 Jun 2008) | 5 lines Change informative messages to use the _multiple variant when multiple formats are possible. (Closes issue #12848) Reported by klaus3000 ........ ................ * /, build_tools/strip_nonapi: Merged revisions 123913 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123913 | tilghman | 2008-06-19 11:26:50 -0500 (Thu, 19 Jun 2008) | 13 lines Merged revisions 123909 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123909 | tilghman | 2008-06-19 11:26:03 -0500 (Thu, 19 Jun 2008) | 5 lines Only process 40 arguments (20 files) at once with xargs, because some older shells may force xargs to separate on an odd boundary. (Closes issue #12883) Reported by Nik Soggia ........ ................ * /, configs/sip.conf.sample: Merged revisions 123887 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123887 | tilghman | 2008-06-19 11:21:32 -0500 (Thu, 19 Jun 2008) | 12 lines Merged revisions 123883 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123883 | tilghman | 2008-06-19 11:20:41 -0500 (Thu, 19 Jun 2008) | 4 lines Correct description of notifyringing option. (Closes issue #12890) Reported by gminet ........ ................ * main/asterisk.c, /: Merged revisions 123870 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123870 | tilghman | 2008-06-19 11:08:29 -0500 (Thu, 19 Jun 2008) | 14 lines Merged revisions 123869 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123869 | tilghman | 2008-06-19 11:07:23 -0500 (Thu, 19 Jun 2008) | 6 lines The RDTSC instruction was introduced on the Pentium line of microprocessors, and is not compatible with certain 586 clones, like Cyrix. Hence, asking for i386 compatibility was always incorrect. See http://en.wikipedia.org/wiki/RDTSC (Closes issue #12886) Reported by tecnoxarxa ........ ................ 2008-06-18 22:18 +0000 [r123718-123772] Tilghman Lesher * /, main/say.c, doc/lang (added), doc/lang/hebrew.ods: Merged revisions 123770 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123770 | tilghman | 2008-06-18 17:17:17 -0500 (Wed, 18 Jun 2008) | 16 lines Merged revisions 123769 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123769 | tilghman | 2008-06-18 17:08:30 -0500 (Wed, 18 Jun 2008) | 8 lines Add support for saying numbers in Hebrew. (closes issue #11662) Reported by: greenfieldtech Patches: say.c.patch-12042008 uploaded by greenfieldtech (license 369) Hebrew-Sounds.ods uploaded by greenfieldtech (with signficant changes to the spreadsheet by me) ........ ................ * pbx/pbx_spool.c, /: Merged revisions 123715 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123715 | tilghman | 2008-06-18 15:23:58 -0500 (Wed, 18 Jun 2008) | 15 lines Merged revisions 123710 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123710 | tilghman | 2008-06-18 15:22:42 -0500 (Wed, 18 Jun 2008) | 7 lines Set the variables top-down, so that if a script sets a variable more than once, the last one will take precedence. (closes issue #12673) Reported by: phber Patches: 20080519__bug12673.diff.txt uploaded by Corydon76 (license 14) ........ ................ 2008-06-18 20:08 +0000 [r123693] Brett Bryant * main/tcptls.c, /: Merged revisions 123692 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r123692 | bbryant | 2008-06-18 15:07:56 -0500 (Wed, 18 Jun 2008) | 2 lines Fix a crash in tcp and tls connections related to reference counts. ........ 2008-06-18 15:09 +0000 [r123651-123653] Mark Michelson * /, apps/app_queue.c: Merged revisions 123652 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r123652 | mmichelson | 2008-06-18 10:08:56 -0500 (Wed, 18 Jun 2008) | 7 lines A portion of the code which handled the 'c' queue option had been removed. No telling when it happened. Anyway, it's back in now and works properly. (Based on issue reported on mailing list) ........ 2008-06-18 12:34 +0000 [r123646-123647] Russell Bryant * apps/app_fax.c: don't use trunk only API for frame data (closes issue #12881) * apps/app_fax.c (added): re-add app_fax ... it got accidentally removed (closes issue #12881) 2008-06-17 21:57 +0000 [r123547] Brett Bryant * main/tcptls.c, main/manager.c, /, channels/chan_sip.c, main/http.c, include/asterisk/tcptls.h: Merged revisions 123546 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r123546 | bbryant | 2008-06-17 16:46:57 -0500 (Tue, 17 Jun 2008) | 5 lines Updates all usages of ast_tcptls_session_instance to be managed by reference counts so that they only get destroyed when all threads are done using them, and memory does not get free'd causing strange issues with SIP. This code was originally written by russellb in the team/group/issue_11972/ branch. ........ 2008-06-17 21:34 +0000 [r123487-123542] Mark Michelson * /, channels/chan_sip.c: Merged revisions 123486 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123486 | mmichelson | 2008-06-17 15:28:47 -0500 (Tue, 17 Jun 2008) | 12 lines Merged revisions 123485 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123485 | mmichelson | 2008-06-17 15:26:38 -0500 (Tue, 17 Jun 2008) | 4 lines Make chan_sip build under dev mode with compilers >= GCC 4.2 Thanks to jpeeler for alerting me of this ........ ................ 2008-06-17 20:23 +0000 [r123473] Steve Murphy * /: block 123448 from trunk; it doesn't apply here. 2008-06-17 19:01 +0000 [r123394] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 123392 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123392 | tilghman | 2008-06-17 13:57:45 -0500 (Tue, 17 Jun 2008) | 11 lines Merged revisions 123391 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123391 | tilghman | 2008-06-17 13:56:53 -0500 (Tue, 17 Jun 2008) | 3 lines Fix 3 more places where failure to lock the structure could cause the wrong lock to be unlocked. (Closes issue #12795) ........ ................ 2008-06-17 18:28 +0000 [r123382-123387] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 123238 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r123238 | jpeeler | 2008-06-16 18:05:18 -0500 (Mon, 16 Jun 2008) | 1 line Fix some (more) variables that were forgotten to be renamed, related to 117658 ........ 2008-06-17 18:10 +0000 [r123335] Mark Michelson * /, channels/chan_sip.c: Merged revisions 123334 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123334 | mmichelson | 2008-06-17 13:09:54 -0500 (Tue, 17 Jun 2008) | 19 lines Merged revisions 123333 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123333 | mmichelson | 2008-06-17 13:09:16 -0500 (Tue, 17 Jun 2008) | 11 lines Cisco BTS sends SIP responses with a tab between the Cseq number and SIP request method in the Cseq: header. Asterisk did not handle this properly, but with this patch, all is well. (closes issue #12834) Reported by: tobias_e Patches: 12834.patch uploaded by putnopvut (license 60) Tested by: tobias_e ........ ................ 2008-06-17 18:08 +0000 [r123332] Jeff Peeler * doc/tex/configuration.tex, configs/zapata.conf.sample, Makefile, doc/janitor-projects.txt, configs/vpb.conf.sample, doc/sms.txt, contrib/scripts/loadtest.tcl, codecs/codec_dahdi.c (added), configs/smdi.conf.sample, pbx/pbx_config.c, apps/app_chanspy.c, main/asterisk.c, configs/users.conf.sample, doc/ss7.txt, apps/app_meetme.c, configs/rpt.conf.sample, doc/backtrace.txt, doc/tex/queues-with-callback-members.tex, include/asterisk/dahdi.h (added), configs/extensions.ael.sample, res/res_musiconhold.c, configs/meetme.conf.sample, codecs/codec_zap.c (removed), contrib/init.d/rc.mandrake.zaptel, cdr/cdr_csv.c, main/channel.c, doc/tex/manager.tex, doc/tex/sla.tex, include/asterisk/dsp.h, doc/tex/localchannel.tex, apps/app_rpt.c, channels/chan_mgcp.c, contrib/scripts/autosupport, doc/manager_1_1.txt, channels/chan_zap.c (removed), doc/asterisk.8, doc/tex/ael.tex, doc/tex/channelvariables.tex, apps/app_getcpeid.c, doc/tex/enum.tex, apps/app_queue.c, configs/sla.conf.sample, doc/tex/security.tex, include/asterisk/zapata.h (removed), doc/tex/privacy.tex, build_tools/menuselect-deps.in, apps/app_flash.c, main/file.c, doc/osp.txt, contrib/utils/zones2indications.c, utils/extconf.c, makeopts.in, configs/extensions.conf.sample, doc/asterisk.sgml, README, contrib/init.d/rc.mandrake.asterisk, /, include/asterisk/autoconfig.h.in, apps/app_dahdiscan.c (added), apps/app_chanisavail.c, channels/chan_iax2.c, configs/muted.conf.sample, main/loader.c, channels/chan_dahdi.c (added), include/asterisk/doxyref.h, configure, doc/tex/backtrace.tex, apps/app_zapscan.c (removed), doc/tex/app-sms.tex, apps/app_zapras.c (removed), configs/extensions.lua.sample, include/asterisk/options.h, contrib/init.d/rc.suse.asterisk, apps/app_dial.c, apps/app_page.c, doc/tex/hardware.tex, apps/app_fax.c (removed), apps/app_dahdiras.c (added), configure.ac, configs/queues.conf.sample, include/asterisk/channel.h: Goodbye Zaptel, hello DAHDI. Removes Zaptel driver support with DAHDI. Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow. 2008-06-17 15:58 +0000 [r123276] Mark Michelson * /, apps/app_queue.c: Merged revisions 123275 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123275 | mmichelson | 2008-06-17 10:57:43 -0500 (Tue, 17 Jun 2008) | 20 lines Merged revisions 123274 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123274 | mmichelson | 2008-06-17 10:56:55 -0500 (Tue, 17 Jun 2008) | 12 lines davidw pointed out that the holdtime calculation used by app_queue does not use "boxcar" filtering as the comments say. The term "boxcar" means that the number of samples used to calculate stays constant, with new samples replacing the oldest ones. The queue holdtime calculation uses all holdtime samples collected since the queue was loaded, so the comment has been changed to be accurate. (closes issue #12781) Reported by: davidw ........ ................ 2008-06-17 15:52 +0000 [r123273] Russell Bryant * main/astobj2.c, /: Merged revisions 123272 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123272 | russell | 2008-06-17 10:52:13 -0500 (Tue, 17 Jun 2008) | 12 lines Merged revisions 123271 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123271 | russell | 2008-06-17 10:48:31 -0500 (Tue, 17 Jun 2008) | 4 lines Fix a memory leak in astobj2 that was pointed out by seanbright. When a container got destroyed, the underlying bucket list entry for each object that was in the container at that time did not get free'd. ........ ................ 2008-06-16 21:20 +0000 [r123178] Jeff Peeler * channels/chan_zap.c: Fix some variables that were forgotten to be renamed, related to 117658. Couldn't merge from trunk since the chan_dahdi transition has not occurred here yet 2008-06-16 21:19 +0000 [r123173] Steve Murphy * apps/app_stack.c, apps/app_dial.c, main/pbx.c, /, main/features.c, include/asterisk/pbx.h, apps/app_queue.c: Merged revisions 123165 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r123165 | murf | 2008-06-16 14:43:46 -0600 (Mon, 16 Jun 2008) | 19 lines (closes issue #12689) Reported by: ys Many thanks to ys for doing the research on this problem. I didn't think it would be best to unlock the contexts and then relock them after the remove_extension2() call, so I added an extra arg to remove_extension2() and set it appropriately in each call. There were not that many. I considered forcing the code to lock the contexts before the call to remove_extension2(), but that would require a slightly greater degree of changes, especially since the find_context_locked is local to pbx.c I did a simple sanity test to make sure the code doesn't mess things up in general. ........ 2008-06-16 20:03 +0000 [r123112-123116] Tilghman Lesher * channels/chan_mgcp.c, /, channels/chan_sip.c, channels/chan_skinny.c, channels/chan_h323.c, channels/chan_iax2.c: Merged revisions 123114 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123114 | tilghman | 2008-06-16 14:57:05 -0500 (Mon, 16 Jun 2008) | 10 lines Merged revisions 123113 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123113 | tilghman | 2008-06-16 14:50:12 -0500 (Mon, 16 Jun 2008) | 2 lines Port "hasvoicemail" change from SIP to other channel drivers ........ ................ * /, channels/chan_sip.c: Merged revisions 123111 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123111 | tilghman | 2008-06-16 14:23:51 -0500 (Mon, 16 Jun 2008) | 16 lines Merged revisions 123110 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123110 | tilghman | 2008-06-16 14:21:58 -0500 (Mon, 16 Jun 2008) | 8 lines People expect that if "hasvoicemail" is set in users.conf, even if "mailbox" isn't set, that SIP will detect a mailbox. (closes issue #12855) Reported by: PLL Patches: 20080614__bug12855__2.diff.txt uploaded by Corydon76 (license 14) Tested by: PLL ........ ................ 2008-06-16 17:29 +0000 [r123075] Chris Tooley * apps/app_externalivr.c: Fixes and closes bug number 12804 2008-06-16 12:32 +0000 [r122871-122921] Joshua Colp * /, channels/chan_sip.c: Merged revisions 122920 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r122920 | file | 2008-06-16 09:32:02 -0300 (Mon, 16 Jun 2008) | 14 lines Merged revisions 122919 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r122919 | file | 2008-06-16 09:31:09 -0300 (Mon, 16 Jun 2008) | 6 lines Only compare the first 15 characters so that even if the charset is specified we still accept it as SDP. (closes issue #12803) Reported by: lanzaandrea Patches: chan_sip.c.diff uploaded by lanzaandrea (license 496) ........ ................ * /, channels/chan_sip.c: Merged revisions 122870 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r122870 | file | 2008-06-16 09:09:54 -0300 (Mon, 16 Jun 2008) | 14 lines Merged revisions 122869 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r122869 | file | 2008-06-16 09:08:28 -0300 (Mon, 16 Jun 2008) | 6 lines Don't send a BYE on a dialog that is already gone during a REFER. (closes issue #12865) Reported by: flefoll Patches: chan_sip.c.br14.121495.patch-ALREADYGONE uploaded by flefoll (license 244) ........ ................ 2008-06-13 21:47 +0000 [r122715] Mark Michelson * main/autoservice.c, /: Merged revisions 122714 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r122714 | mmichelson | 2008-06-13 16:45:21 -0500 (Fri, 13 Jun 2008) | 17 lines Merged revisions 122713 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r122713 | mmichelson | 2008-06-13 16:44:53 -0500 (Fri, 13 Jun 2008) | 9 lines Short circuit the loop in autoservice_run if there are no channels to poll. If we continued, then the result would be calling poll() with a NULL pollfd array. While this is fine with POSIX's poll(2) system call, those who use Asterisk's internal poll mechanism (Darwin systems) would have a failed assertion occur when poll is called. (related to issue #10342) ........ ................ 2008-06-13 14:15 +0000 [r122558] Tilghman Lesher * main/dial.c, /: Merged revisions 122557 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r122557 | tilghman | 2008-06-13 09:15:07 -0500 (Fri, 13 Jun 2008) | 7 lines Convert one more delimiter to use comma. (closes issue #12850) Reported by: bcnit Patches: 20080613__bug12850.diff.txt uploaded by Corydon76 (license 14) Tested by: bcnit ........ 2008-06-13 00:18 +0000 [r122467] Jeff Peeler * apps/app_parkandannounce.c, /, main/features.c: Merged revisions 122433 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r122433 | jpeeler | 2008-06-12 18:08:37 -0500 (Thu, 12 Jun 2008) | 4 lines (closes issue 0012193) Reported by: davidw Patch by: Corydon76, modified by me to work properly with ParkAndAnnounce app ........ 2008-06-12 18:54 +0000 [r122313] Mark Michelson * /, apps/app_queue.c: Merged revisions 122312 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r122312 | mmichelson | 2008-06-12 13:53:17 -0500 (Thu, 12 Jun 2008) | 17 lines Merged revisions 122311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r122311 | mmichelson | 2008-06-12 13:50:58 -0500 (Thu, 12 Jun 2008) | 9 lines Properly play a holdtime message if the announce-holdtime option is set to "once." (closes issue #12842) Reported by: ramonpeek Patches: patch001.diff uploaded by ramonpeek (license 266) ........ ................ 2008-06-12 18:24 +0000 [r122242-122266] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 122262 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r122262 | russell | 2008-06-12 13:23:54 -0500 (Thu, 12 Jun 2008) | 11 lines Merged revisions 122259 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r122259 | russell | 2008-06-12 13:22:44 -0500 (Thu, 12 Jun 2008) | 3 lines Fix some race conditions that cause ast_assert() to report that chan_iax2 tried to remove an entry that wasn't in the scheduler ........ ................ 2008-06-12 15:27 +0000 [r122132-122180] Tilghman Lesher * /, apps/app_meetme.c: Merged revisions 122174 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r122174 | tilghman | 2008-06-12 10:26:07 -0500 (Thu, 12 Jun 2008) | 16 lines Merged revisions 122137 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r122137 | tilghman | 2008-06-12 10:18:39 -0500 (Thu, 12 Jun 2008) | 8 lines Flipflop the sections for two options, since the section for 'X' (exit context) may otherwise absorb keypresses meant for 's' (admin/user menu). (closes issue #12836) Reported by: blitzrage Patches: 20080611__bug12836.diff.txt uploaded by Corydon76 (license 14) Tested by: blitzrage ........ ................ * main/channel.c, /: Merged revisions 122131 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r122131 | tilghman | 2008-06-12 10:14:37 -0500 (Thu, 12 Jun 2008) | 12 lines Merged revisions 122130 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r122130 | tilghman | 2008-06-12 10:11:30 -0500 (Thu, 12 Jun 2008) | 4 lines Occasionally, the alertpipe loses its nonblocking status, so detect and correct that situation before it causes a deadlock. (Reported and tested by ctooley via #asterisk-dev) ........ ................ 2008-06-12 15:01 +0000 [r122126-122129] Steve Murphy * main/cdr.c, apps/app_forkcdr.c, /, CHANGES: Merged revisions 122128 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r122128 | murf | 2008-06-12 08:56:26 -0600 (Thu, 12 Jun 2008) | 9 lines Merged revisions 122127 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r122127 | murf | 2008-06-12 08:51:44 -0600 (Thu, 12 Jun 2008) | 1 line Arkadia tried to warn me, but the code added to ast_cdr_busy, _failed, and _noanswer was redundant. Didn't spot it until I was resolving conflicts in trunk. Ugh. Redundant code removed. It wasn't harmful. Just dumb. ........ ................ * main/cdr.c, apps/app_forkcdr.c, /, funcs/func_cdr.c, include/asterisk/cdr.h, CHANGES: Merged revisions 122091 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r122091 | murf | 2008-06-12 08:28:01 -0600 (Thu, 12 Jun 2008) | 45 lines Merged revisions 122046 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r122046 | murf | 2008-06-12 07:47:34 -0600 (Thu, 12 Jun 2008) | 37 lines (closes issue #10668) Reported by: arkadia Tested by: murf, arkadia Options added to forkCDR() app and the CDR() func to remove some roadblocks for CDR applications. The "show application ForkCDR" output was upgraded to more fully explain the inner workings of forkCDR. The A option was added to forkCDR to force the CDR system to NOT change the disposition on the original CDR, after the fork. This involves ast_cdr_answer, _busy, _failed, and so on. The T option was added to forkCDR to force obedience of the cdr LOCKED flag in the ast_cdr_end, all the disposition changing funcs (ast_cdr_answer, etc), and in the ast_cdr_setvar func. The CHANGES file was updated to explain ALL the new options added to satisfy this bug report (and some requests made verbally and via email, irc, etc, over the past months/year) The 's' option was added to the CDR() func, to force it to skip LOCKED cdr's in the chain. Again, the new options should be totally transparent to existing apps! Current behavior of CDR, forkCDR, and the rest of the CDR system should not change one little bit. Until you add the new options, at least! ........ ................ 2008-06-11 18:57 +0000 [r121915] Matthew Fredrickson * channels/chan_zap.c, /: Merged revisions 121914 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r121914 | mattf | 2008-06-11 13:53:10 -0500 (Wed, 11 Jun 2008) | 1 line Fix pseudo channel allocation errors on startup when using SS7 ........ 2008-06-11 18:20 +0000 [r121872] Tilghman Lesher * main/sched.c, main/channel.c, /, channels/chan_agent.c, main/abstract_jb.c: Merged revisions 121867 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r121867 | tilghman | 2008-06-11 13:19:24 -0500 (Wed, 11 Jun 2008) | 11 lines Merged revisions 121861 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r121861 | tilghman | 2008-06-11 13:18:16 -0500 (Wed, 11 Jun 2008) | 3 lines Make calls to ast_assert() actually test something, so that the error message printed is not nonsensical (reported by mvanbaak via #asterisk-bugs). ........ ................ 2008-06-11 17:59 +0000 [r121858] Matthew Fredrickson * channels/chan_zap.c, /: Merged revisions 121857 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r121857 | mattf | 2008-06-11 12:50:17 -0500 (Wed, 11 Jun 2008) | 1 line Make sure we hangup any calls we have and NULL out the ss7call value when we get a reset circuit message. Fixes crash bug ........ 2008-06-11 17:45 +0000 [r121856] Tilghman Lesher * contrib/scripts/realtime_pgsql.sql, /, UPGRADE.txt, include/asterisk/cdr.h: Merged revisions 121855 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r121855 | tilghman | 2008-06-11 12:44:39 -0500 (Wed, 11 Jun 2008) | 3 lines Expand CDR uniqueid field to 150 chars, to account for maximum systemname. (Closes issue #12831) ........ 2008-06-11 16:13 +0000 [r121806] Jeff Peeler * /, doc/backtrace.txt: Merged revisions 121805 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r121805 | jpeeler | 2008-06-11 11:11:40 -0500 (Wed, 11 Jun 2008) | 9 lines Merged revisions 121804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r121804 | jpeeler | 2008-06-11 11:11:09 -0500 (Wed, 11 Jun 2008) | 1 line add instructions for logging gdb output via set logging on ........ ................ 2008-06-10 18:36 +0000 [r121598] Sean Bright * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 121597 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r121597 | seanbright | 2008-06-10 14:35:37 -0400 (Tue, 10 Jun 2008) | 14 lines Merged revisions 121596 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r121596 | seanbright | 2008-06-10 14:34:45 -0400 (Tue, 10 Jun 2008) | 6 lines Fixes a problem with some buggy versions of GNU awk (3.1.3) not liking carriage returns in scripts. (closes issue #12749) Reported by: alinux Tested by: Laureano (on #asterisk-dev), juggie ........ ................ 2008-06-10 12:55 +0000 [r121445] Joshua Colp * main/channel.c, /: Merged revisions 121444 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r121444 | file | 2008-06-10 09:54:39 -0300 (Tue, 10 Jun 2008) | 12 lines Merged revisions 121442 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r121442 | file | 2008-06-10 09:52:06 -0300 (Tue, 10 Jun 2008) | 4 lines Update BRIDGEPEER variable before we do a generic bridge in case we just broke out of a native bridge and fell through to generic. (closes issue #12815) Reported by: ramonpeek ........ ................ 2008-06-10 00:53 +0000 [r121404-121408] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 121407 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r121407 | russell | 2008-06-09 19:52:46 -0500 (Mon, 09 Jun 2008) | 2 lines Bump up the debug level of a couple of messages ........ 2008-06-09 16:37 +0000 [r121283] Russell Bryant * main/channel.c, /: Merged revisions 121282 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r121282 | russell | 2008-06-09 11:37:08 -0500 (Mon, 09 Jun 2008) | 18 lines Merged revisions 121280 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r121280 | russell | 2008-06-09 11:35:40 -0500 (Mon, 09 Jun 2008) | 10 lines Do not attempt to do emulation if an END digit is received and the length is less than the defined minimum digit length, and the other end only wants END digits (SIP INFO, for example). (closes issue #12778) Reported by: tsearle Patches: 12778.rev1.txt uploaded by russell (license 2) Tested by: tsearle ........ ................ 2008-06-09 16:36 +0000 [r121281] Tilghman Lesher * main/pbx.c, /: Merged revisions 121279 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r121279 | tilghman | 2008-06-09 11:35:06 -0500 (Mon, 09 Jun 2008) | 6 lines Implement FINDLABEL matching for the new extension matching engine. (closes issue #12800) Reported by: chris-mac Patches: 20080608__bug12800.diff.txt uploaded by Corydon76 (license 14) ........ 2008-06-09 15:10 +0000 [r121231] Mark Michelson * /, channels/chan_agent.c: Merged revisions 121230 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r121230 | mmichelson | 2008-06-09 10:08:58 -0500 (Mon, 09 Jun 2008) | 27 lines Merged revisions 121229 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (Note that this is being merged to trunk/1.6.0 because it may affect non-callback agents with ackcall set) ........ r121229 | mmichelson | 2008-06-09 10:02:37 -0500 (Mon, 09 Jun 2008) | 16 lines A unique situation of timeouts brought forth a failure situation for autologoff in chan_agent. If using AgentCallbackLogin-style agents, then if the timeout specified by the Dial() to reach the agent's phone was shorter than the timeout specified in queues.conf, then autologoff would only work if the caller hung up while the agent's phone was ringing. This patch allows autologoff to work in this situation when the call in queue transfers to the next available agent (as it would have if the timeout in queues.conf were less than the timeout in the Dial()). (closes issue #12754) Reported by: Rodrigo Patches: 12754.patch uploaded by putnopvut (license 60) Tested by: Rodrigo ........ ................ 2008-06-08 01:43 +0000 [r121138-121164] Jeff Peeler * /, channels/chan_console.c: Merged revisions 121163 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r121163 | jpeeler | 2008-06-07 20:41:59 -0500 (Sat, 07 Jun 2008) | 4 lines This was accidentally reverted. Fixes a bug where if a stream monitor thread was not created (caused from failure of opening or starting the stream) pthread_cancel was called with an invalid thread ID. ........ * apps/app_parkandannounce.c, /: Merged revisions 121131 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r121131 | jpeeler | 2008-06-07 20:16:25 -0500 (Sat, 07 Jun 2008) | 2 lines Fixes segfault when using ParkAndAnnounce. Also, loop made more efficient as announce template only needs to be checked until the number of colon separated arguments run out, not the entire pointer storage array. Was done in a similiar fashion in 1.4, but here we're using less variables. ........ 2008-06-07 14:19 +0000 [r121080] Russell Bryant * channels/chan_local.c, /, channels/chan_agent.c: Merged revisions 121079 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r121079 | russell | 2008-06-07 09:18:44 -0500 (Sat, 07 Jun 2008) | 15 lines Merged revisions 121078 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r121078 | russell | 2008-06-07 09:10:56 -0500 (Sat, 07 Jun 2008) | 7 lines Don't run LIST_HEAD_DESTROY on a STATIC list (closes issue #12807) Reported by: ys Patches: chan_agent_local.diff uploaded by ys (license 281) ........ ................ 2008-06-06 20:25 +0000 [r121011-121047] Tilghman Lesher * main/pbx.c, /: Merged revisions 121010 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r121010 | tilghman | 2008-06-06 14:55:08 -0500 (Fri, 06 Jun 2008) | 6 lines Make extension match characters case-insensitive. (closes issue #12777) Reported by: jsmith Patches: lower_case_patterns-trunk-v1.patch uploaded by jsmith (license 15) ........ 2008-06-06 18:31 +0000 [r120907-120961] Jeff Peeler * /, channels/chan_sip.c: Merged revisions 120960 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r120960 | jpeeler | 2008-06-06 13:30:17 -0500 (Fri, 06 Jun 2008) | 9 lines Merged revisions 120959 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120959 | jpeeler | 2008-06-06 13:29:14 -0500 (Fri, 06 Jun 2008) | 1 line add another LOW_MEMORY define I forgot ........ ................ * /, channels/chan_sip.c: Merged revisions 120909 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r120909 | jpeeler | 2008-06-06 13:06:06 -0500 (Fri, 06 Jun 2008) | 9 lines Merged revisions 120908 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120908 | jpeeler | 2008-06-06 13:05:15 -0500 (Fri, 06 Jun 2008) | 1 line only define thread storage variable if necessary for LOW_MEMORY ........ ................ * channels/chan_sip.c: Merged revisions 120906 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r120906 | jpeeler | 2008-06-06 12:50:05 -0500 (Fri, 06 Jun 2008) | 16 lines Merged revisions 120863,120885 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120863 | jpeeler | 2008-06-06 10:33:15 -0500 (Fri, 06 Jun 2008) | 3 lines This fixes a crash when LOW_MEMORY is turned on. Two allocations of the ast_rtp struct that were previously allocated on the stack have been modified to use thread local storage instead. ........ r120885 | jpeeler | 2008-06-06 11:39:20 -0500 (Fri, 06 Jun 2008) | 2 lines Correction to commmit 120863, make sure proper destructor function is called as well define two thread storage local variables. ........ ................ 2008-06-06 17:35 +0000 [r120864-120905] Tilghman Lesher * /, apps/app_exec.c: Merged revisions 120904 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r120904 | tilghman | 2008-06-06 12:34:21 -0500 (Fri, 06 Jun 2008) | 3 lines For the purpose of making the changed syntax to ExecIf easier to transition, allow the deprecated syntax (fixed for jmls on -dev). ........ 2008-06-05 21:39 +0000 [r120829] Steve Murphy * main/pbx.c, /: Merged revisions 120828 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r120828 | murf | 2008-06-05 15:34:42 -0600 (Thu, 05 Jun 2008) | 1 line a small fix for a crash that occurs when compiling AEL with global vars ........ 2008-06-05 17:17 +0000 [r120677] Philippe Sultan * /, res/res_jabber.c: Merged revisions 120676 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r120676 | phsultan | 2008-06-05 19:02:39 +0200 (Thu, 05 Jun 2008) | 10 lines Merged revisions 120675 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120675 | phsultan | 2008-06-05 18:56:15 +0200 (Thu, 05 Jun 2008) | 2 lines Ignore appended resource when comparing JIDs. ........ ................ 2008-06-05 16:42 +0000 [r120643-120674] Brett Bryant 2008-06-05 16:01 +0000 [r120566-120603] Tilghman Lesher * apps/app_stack.c, main/loader.c, /, res/res_agi.c: Merged revisions 120602 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r120602 | tilghman | 2008-06-05 10:58:11 -0500 (Thu, 05 Jun 2008) | 4 lines Conditionally load the AGI command gosub, depending on whether or not res_agi has been loaded, fix a return value in the loader, and ensure that the help workhorse header does not print on load. ........ * /, UPGRADE.txt: Merged revisions 120567 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r120567 | tilghman | 2008-06-05 09:35:47 -0500 (Thu, 05 Jun 2008) | 2 lines Add info on the [compat] section of asterisk.conf. ........ * apps/app_fax.c: Fix frame API for 1.6.0 (Closes issue #12793) 2008-06-04 22:08 +0000 [r120515] Mark Michelson * /, apps/app_queue.c: Merged revisions 120514 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r120514 | mmichelson | 2008-06-04 17:07:37 -0500 (Wed, 04 Jun 2008) | 14 lines Merged revisions 120513 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120513 | mmichelson | 2008-06-04 17:05:33 -0500 (Wed, 04 Jun 2008) | 6 lines Make sure that the string we set will survive the unref of the queue member. Thanks to Russell, who pointed this out. ........ ................ 2008-06-04 20:35 +0000 [r120478] Tilghman Lesher * main/pbx.c, /: Merged revisions 120477 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r120477 | tilghman | 2008-06-04 15:34:52 -0500 (Wed, 04 Jun 2008) | 2 lines MSet doesn't necessarily need chan to be set ........ 2008-06-04 15:38 +0000 [r120338] Joshua Colp * /, pbx/pbx_config.c: Merged revisions 120337 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r120337 | file | 2008-06-04 12:38:00 -0300 (Wed, 04 Jun 2008) | 2 lines We like tabs. ........ 2008-06-04 14:13 +0000 [r120287] Mark Michelson * /, apps/app_queue.c: Merged revisions 120286 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r120286 | mmichelson | 2008-06-04 09:12:45 -0500 (Wed, 04 Jun 2008) | 15 lines Merged revisions 120285 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120285 | mmichelson | 2008-06-04 09:11:12 -0500 (Wed, 04 Jun 2008) | 7 lines Tab completion when removing a member should give the member's interface, not the name, since the interface is what is expected for the command. (closes issue #12783) Reported by: davevg ........ ................ 2008-06-04 13:34 +0000 [r120284] Joshua Colp * /, pbx/pbx_config.c: Merged revisions 120283 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r120283 | file | 2008-06-04 10:33:59 -0300 (Wed, 04 Jun 2008) | 14 lines Merged revisions 120282 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120282 | file | 2008-06-04 10:31:09 -0300 (Wed, 04 Jun 2008) | 6 lines Fix a log message and add a message for when the dialplan is done reloading. (closes issue #12716) Reported by: chappell Patches: dialplan_reload_2.diff uploaded by chappell (license 8) ........ ................ 2008-06-03 23:18 +0000 [r120228-120234] Tilghman Lesher * pbx/pbx_loopback.c, /: Merged revisions 120227 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r120227 | tilghman | 2008-06-03 17:42:03 -0500 (Tue, 03 Jun 2008) | 16 lines Merged revisions 120226 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120226 | tilghman | 2008-06-03 17:41:04 -0500 (Tue, 03 Jun 2008) | 8 lines Due to incorrect use of the AST_LIST_INSERT_HEAD() macro the loopback switch cannot perform any translation on the extension number before searching for it in the target context. (closes issue #12473) Reported by: chappell Patches: pbx_loopback.c.diff uploaded by chappell (license 8) ........ ................ 2008-06-03 22:18 +0000 [r120178] Jeff Peeler * main/config.c: Merged revisions 120174 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r120174 | jpeeler | 2008-06-03 17:17:07 -0500 (Tue, 03 Jun 2008) | 14 lines Merged revisions 120173 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120173 | jpeeler | 2008-06-03 17:15:33 -0500 (Tue, 03 Jun 2008) | 6 lines (closes issue #11594) Reported by: yem Tested by: yem This change decreases the buffer size allocated on the stack substantially in config_text_file_load when LOW_MEMORY is turned on. This change combined with the fix from revision 117462 (making mkintf not copy the zt_chan_conf structure) was enough to prevent the crash. ........ ................ 2008-06-03 22:08 +0000 [r120172] Tilghman Lesher * include/asterisk/options.h, main/asterisk.c, Makefile, main/pbx.c, /, res/res_agi.c, pbx/pbx_realtime.c, configs/pbx_realtime.conf (removed): Merged revisions 120171 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r120171 | tilghman | 2008-06-03 17:05:16 -0500 (Tue, 03 Jun 2008) | 5 lines Move compatibility options into asterisk.conf, default them to on for upgrades, and off for new installations. This includes the translation from pipes to commas for pbx_realtime and the EXEC command for AGI, as well as the change to the Set application not to support multiple variables at once. ........ 2008-06-03 21:35 +0000 [r120170] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 120169 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r120169 | russell | 2008-06-03 16:35:11 -0500 (Tue, 03 Jun 2008) | 12 lines Merged revisions 120168 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120168 | russell | 2008-06-03 16:34:55 -0500 (Tue, 03 Jun 2008) | 4 lines Fix another place where peer->callno could change at a very bad time, and also fix a place where a peer was used after the reference was released. (inspired by rev 120001) ........ ................ 2008-06-03 16:24 +0000 [r120034] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 120012 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r120012 | tilghman | 2008-06-03 11:19:35 -0500 (Tue, 03 Jun 2008) | 17 lines Merged revisions 120001 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120001 | tilghman | 2008-06-03 11:10:53 -0500 (Tue, 03 Jun 2008) | 9 lines Save the callno when we're poking, because our peer structure could change during destruction (and thus we unlock the wrong callno, causing a cascade failure). (closes issue #12717) Reported by: gewfie Patches: 20080525__bug12717.diff.txt uploaded by Corydon76 (license 14) Tested by: gewfie ........ ................ 2008-06-03 15:57 +0000 [r119931-120000] Steve Murphy * pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18, pbx/ael/ael-test/ref.ael-vtest21, pbx/ael/ael-test/ref.ael-test19, pbx/ael/ael-test/ref.ael-vtest13, pbx/ael/ael-test/ref.ael-vtest17, pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1, pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-test5, pbx/ael/ael-test/ref.ael-test15: Merged revisions 119998 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119998 | murf | 2008-06-03 09:49:34 -0600 (Tue, 03 Jun 2008) | 16 lines Merged revisions 119966 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119966 | murf | 2008-06-03 09:26:56 -0600 (Tue, 03 Jun 2008) | 8 lines Updated the regressions on AEL. Hadn't updated this for the changes I made to preserve ${EXTEN} in switches, which affected several tests because it adds extra priorities, and at least one needed to be updated because of the removal of the empty extension warning message. ........ ................ * res/ael/pval.c, /: Merged revisions 119930 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119930 | murf | 2008-06-03 09:07:20 -0600 (Tue, 03 Jun 2008) | 24 lines Merged revisions 119929 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119929 | murf | 2008-06-03 08:49:46 -0600 (Tue, 03 Jun 2008) | 16 lines as per http://lists.digium.com/pipermail/asterisk-users/2008-June/212934.html, which is a message from Philipp Kempgen, requesting that the WARNING that an extension is empty be reduced to a NOTICE or less, as empty extensions are syntactically possible, and no big deal. With which I agree, and have removed that WARNING message entirely. I think it is not necessary to see this message. It didn't state that a NoOp() was inserted automatically on your behalf, and really, as users, who cares? Why freak out dialplan writers with unnecessary warnings? The details of the machinations a compiler goes thru to produce working assembly code is of little interest to most programmers-- we will follow the unix principal of doing our work silently. ........ ................ 2008-06-03 14:48 +0000 [r119928] Joshua Colp * /, channels/chan_sip.c: Merged revisions 119927 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119927 | file | 2008-06-03 11:47:54 -0300 (Tue, 03 Jun 2008) | 10 lines Merged revisions 119926 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119926 | file | 2008-06-03 11:46:24 -0300 (Tue, 03 Jun 2008) | 2 lines Treat ECONNREFUSED as an error that will stop further retransmissions. (issue #AST-58, patch from Switchvox) ........ ................ 2008-06-03 13:30 +0000 [r119745-119893] Russell Bryant * /, main/logger.c: Merged revisions 119892 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r119892 | russell | 2008-06-03 08:29:16 -0500 (Tue, 03 Jun 2008) | 9 lines Do a deep copy of file and function strings to avoid a potential crash when modules are unloaded. (closes issue #12780) Reported by: ys Patches: logger.diff uploaded by ys (license 281) -- modified by me for coding guidelines ........ * /, channels/chan_iax2.c: Merged revisions 119839 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119839 | russell | 2008-06-02 15:08:24 -0500 (Mon, 02 Jun 2008) | 15 lines Merged revisions 119838 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119838 | russell | 2008-06-02 15:08:04 -0500 (Mon, 02 Jun 2008) | 7 lines Revert a change made for issue #12479. This change caused a regression such that a dial string such as (IAX2/foo) did not automatically fall back to dialing the 's' extension anymore. (closes issue #12770) Reported by: dagmoller ........ ................ * /, apps/app_fax.c (added): Merged revisions 119801 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r119801 | russell | 2008-06-02 11:14:15 -0500 (Mon, 02 Jun 2008) | 4 lines Add app_fax from asterisk-addons, with some additional changes to resolve compiler warnings, as well as update to the APIs in spandsp 0.0.5. Spandsp 0.0.5 is being distributed under the LGPL, so we can move this module into the main tree. ........ * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 119799 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r119799 | russell | 2008-06-02 10:57:43 -0500 (Mon, 02 Jun 2008) | 4 lines After determining that the version of spandsp installed is an acceptable version, do a build and link test to ensure that the library is usable, and that libtiff is also available ........ * /, build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Merged revisions 119795 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r119795 | russell | 2008-06-02 10:43:40 -0500 (Mon, 02 Jun 2008) | 2 lines Add a configure script check for spandsp ........ * main/manager.c, /: Merged revisions 119744 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119744 | russell | 2008-06-02 09:41:55 -0500 (Mon, 02 Jun 2008) | 13 lines Merged revisions 119742 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119742 | russell | 2008-06-02 09:39:45 -0500 (Mon, 02 Jun 2008) | 5 lines Improve CLI command blacklist checking for the command manager action. Previously, it did not handle case or whitespace properly. This made it possible for blacklisted commands to get executed anyway. (closes issue #12765) ........ ................ 2008-06-02 14:40 +0000 [r119743] Philippe Sultan * channels/chan_jingle.c, /, channels/chan_gtalk.c, res/res_jabber.c: Merged revisions 119741 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r119741 | phsultan | 2008-06-02 16:35:24 +0200 (Mon, 02 Jun 2008) | 13 lines Do not link the guest account with any configured XMPP client (in jabber.conf). The actual connection is made when a call comes in Asterisk. Apply this fix to Jingle too. Fix the ast_aji_get_client function that was not able to retrieve an XMPP client from its JID. (closes issue #12085) Reported by: junky Tested by: phsultan ........ 2008-06-02 12:32 +0000 [r119532-119690] Russell Bryant * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, /: Merged revisions 119586,119637 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119586 | crichter | 2008-06-02 03:46:23 -0500 (Mon, 02 Jun 2008) | 9 lines Merged revisions 119585 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119585 | crichter | 2008-06-02 10:35:28 +0200 (Mo, 02 Jun 2008) | 1 line Added counter for unhandled_bmsg Print, this prevents the logs to be flooded to fast and save CPU in this error scenario. Added 'last_used' element to bc structure, when a bchannel changes from used to free this exact time will be marked in last_used. When a new channel is requested the find_free_chan function will check if the new empty channel was used within the last second, if yes it will search for the next channel, if no it will return this channel. This simple mechanism has prooven to prevent race conditions where the NT and TE tried to allocate the exact same channel at the same time (RELEASE cause: 44). ........ ................ r119637 | crichter | 2008-06-02 04:35:04 -0500 (Mon, 02 Jun 2008) | 9 lines Merged revisions 119636 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119636 | crichter | 2008-06-02 11:29:21 +0200 (Mo, 02 Jun 2008) | 1 line fixed compile issue when dev-mode is enabled ........ ................ * /, channels/chan_iax2.c: Merged revisions 119688 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119688 | russell | 2008-06-02 07:30:42 -0500 (Mon, 02 Jun 2008) | 11 lines Merged revisions 119687 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119687 | russell | 2008-06-02 07:30:17 -0500 (Mon, 02 Jun 2008) | 3 lines Even of the first PING or LAGRQ doesn't get sent because it comes up too soon, make sure to reschedule so it gets sent later. ........ ................ * /, channels/chan_iax2.c: Merged revisions 119534 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119534 | russell | 2008-06-01 20:08:16 -0500 (Sun, 01 Jun 2008) | 10 lines Merged revisions 119533 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119533 | russell | 2008-06-01 20:06:09 -0500 (Sun, 01 Jun 2008) | 2 lines Change a debug message to an actual debug message ........ ................ * apps/app_dial.c, /: Merged revisions 119531 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119531 | russell | 2008-06-01 20:04:01 -0500 (Sun, 01 Jun 2008) | 10 lines Merged revisions 119530 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119530 | russell | 2008-06-01 20:03:22 -0500 (Sun, 01 Jun 2008) | 2 lines Fix another typo in documentation ........ ................ 2008-06-01 21:59 +0000 [r119529] Michiel van Baak * apps/app_dial.c, /: Merged revisions 119479 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119479 | mvanbaak | 2008-06-01 23:06:27 +0200 (Sun, 01 Jun 2008) | 10 lines Merged revisions 119478 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119478 | mvanbaak | 2008-06-01 22:47:55 +0200 (Sun, 01 Jun 2008) | 2 lines small typo fix 'retires' => 'retries' ........ ................ 2008-05-30 21:24 +0000 [r119420] Tilghman Lesher * /, apps/app_queue.c: Merged revisions 119419 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119419 | tilghman | 2008-05-30 16:23:14 -0500 (Fri, 30 May 2008) | 14 lines Merged revisions 119404 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119404 | tilghman | 2008-05-30 16:17:45 -0500 (Fri, 30 May 2008) | 6 lines When joinempty=strict, it only failed on join if there were busy members. If all members were logged out OR paused, then it (incorrectly) let callers join the queue. (closes issue #12451) Reported by: davidw ........ ................ 2008-05-30 19:48 +0000 [r119356] Joshua Colp * main/autoservice.c, /: Merged revisions 119355 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119355 | file | 2008-05-30 16:47:30 -0300 (Fri, 30 May 2008) | 10 lines Merged revisions 119354 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119354 | file | 2008-05-30 16:46:37 -0300 (Fri, 30 May 2008) | 2 lines Fix a bug I found while testing for another issue. ........ ................ 2008-05-30 17:13 +0000 [r119304] Tilghman Lesher * apps/app_stack.c: Oops, broke 1.6 (thanks MattF) 2008-05-30 16:57 +0000 [r119303] Michiel van Baak * contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk, contrib/init.d/rc.debian.asterisk, contrib/init.d/rc.mandrake.asterisk, /, contrib/init.d/rc.redhat.asterisk, contrib/init.d/rc.gentoo.asterisk, contrib/init.d/rc.slackware.asterisk: Merged revisions 119302 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119302 | mvanbaak | 2008-05-30 18:47:24 +0200 (Fri, 30 May 2008) | 22 lines Merged revisions 119301 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119301 | mvanbaak | 2008-05-30 18:44:39 +0200 (Fri, 30 May 2008) | 14 lines dont use a bashism way to check the $VERSION variable. The rc/init.d scripts, and safe_asterisk work on normal sh now again. Tested on: OpenBSD 4.2 (me) Debian etch (me) Ubuntu Hardy (me and loloski) FC9 (loloski) (closes issue #12687) Reported by: loloski Patches: 20080529-12687-safe_asterisk-fixversion.diff.txt uploaded by mvanbaak (license 7) Tested by: loloski, mvanbaak ........ ................ 2008-05-30 16:40 +0000 [r119297-119300] Tilghman Lesher * apps/app_stack.c, /: Merged revisions 119299 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r119299 | tilghman | 2008-05-30 11:40:13 -0500 (Fri, 30 May 2008) | 2 lines Suppress warning about pbx structure already existing ........ * apps/app_stack.c, apps/app_dial.c, include/asterisk/agi.h, /, CHANGES: Merged revisions 119296 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r119296 | tilghman | 2008-05-30 11:10:46 -0500 (Fri, 30 May 2008) | 8 lines Add native AGI command GOSUB, as invoking Gosub with EXEC does not work properly. (closes issue #12760) Reported by: Corydon76 Patches: 20080530__bug12760.diff.txt uploaded by Corydon76 (license 14) Tested by: tim_ringenbach, Corydon76 ........ 2008-05-30 13:01 +0000 [r119158-119240] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 119239 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119239 | russell | 2008-05-30 07:59:11 -0500 (Fri, 30 May 2008) | 23 lines Merged revisions 119238 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r119238 | russell | 2008-05-30 07:55:36 -0500 (Fri, 30 May 2008) | 15 lines Merged revisions 119237 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r119237 | russell | 2008-05-30 07:49:39 -0500 (Fri, 30 May 2008) | 7 lines - Instead of only enforcing destination call number checking on an ACK, check all full frames except for PING and LAGRQ, which may be sent by older versions too quickly to contain the destination call number. (As suggested by Tim Panton on the asterisk-dev list) - Merge changes from team/russell/iax2-frame-race, which prevents PING and LAGRQ from being sent before the destination call number is known. ........ ................ ................ * main/autoservice.c, /: Merged revisions 119157 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119157 | russell | 2008-05-29 17:28:50 -0500 (Thu, 29 May 2008) | 18 lines Merged revisions 119156 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119156 | russell | 2008-05-29 17:24:29 -0500 (Thu, 29 May 2008) | 10 lines Fix a race condition in channel autoservice. There was still a small window of opportunity for a DTMF frame, or some other deferred frame type, to come in and get dropped. (closes issue #12656) (closes issue #12656) Reported by: dimas Patches: v3-12656.patch uploaded by dimas (license 88) -- with some modifications by me ........ ................ 2008-05-29 20:26 +0000 [r119073] Tilghman Lesher * channels/chan_zap.c, /: Merged revisions 119072 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119072 | tilghman | 2008-05-29 15:25:33 -0500 (Thu, 29 May 2008) | 15 lines Merged revisions 119071 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119071 | tilghman | 2008-05-29 15:24:11 -0500 (Thu, 29 May 2008) | 7 lines Call waiting tone occurs too often, because it's getting serviced by both subchannels. (closes issue #11354) Reported by: cahen Patches: 20080512__bug11354.diff.txt uploaded by Corydon76 (license 14) ........ ................ 2008-05-29 19:06 +0000 [r118960-119014] Russell Bryant * apps/app_milliwatt.c, /: Merged revisions 119013 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119013 | russell | 2008-05-29 14:05:33 -0500 (Thu, 29 May 2008) | 12 lines Merged revisions 119012 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119012 | russell | 2008-05-29 14:04:52 -0500 (Thu, 29 May 2008) | 4 lines - Fix a typo in the argument to Playtones - use ast_safe_sleep() instead of calling the wait application (thanks to tilghman for pointing these out!) ........ ................ * /, channels/chan_iax2.c: Merged revisions 119010 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119010 | russell | 2008-05-29 13:54:11 -0500 (Thu, 29 May 2008) | 24 lines Merged revisions 119009 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r119009 | russell | 2008-05-29 13:49:12 -0500 (Thu, 29 May 2008) | 16 lines Merged revisions 119008 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r119008 | russell | 2008-05-29 13:45:21 -0500 (Thu, 29 May 2008) | 7 lines Merge changes from team/russell/iax2-another-fix-to-the-fix As described in the following post to the asterisk-dev mailing list, only enforce destination call numbers when processing an ACK. http://lists.digium.com/pipermail/asterisk-dev/2008-May/033217.html (closes issue #12631) ........ ................ ................ * apps/app_milliwatt.c, /: Merged revisions 118962 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118962 | russell | 2008-05-29 12:52:00 -0500 (Thu, 29 May 2008) | 11 lines Merged revisions 118961 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118961 | russell | 2008-05-29 12:51:29 -0500 (Thu, 29 May 2008) | 3 lines - Mark app_milliwatt dependent on res_indications (thanks to jsmith) - fix a typo in a log message (thanks to qwell) ........ ................ * apps/app_milliwatt.c, /: Merged revisions 118959 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118959 | russell | 2008-05-29 12:46:04 -0500 (Thu, 29 May 2008) | 11 lines Merged revisions 118956 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118956 | russell | 2008-05-29 12:38:38 -0500 (Thu, 29 May 2008) | 3 lines Change milliwatt to use the proper tone by default (1004 Hz) instead of 1000 Hz. An option is there to use 1000 Hz for anyone that might want it. ........ ................ 2008-05-29 17:42 +0000 [r118958] Tilghman Lesher * channels/chan_mgcp.c, channels/chan_zap.c, /, channels/chan_agent.c, channels/chan_alsa.c, main/utils.c, include/asterisk/lock.h, channels/chan_iax2.c: Merged revisions 118955,118957 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118955 | tilghman | 2008-05-29 12:35:19 -0500 (Thu, 29 May 2008) | 11 lines Merged revisions 118953 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118953 | tilghman | 2008-05-29 12:20:16 -0500 (Thu, 29 May 2008) | 3 lines Add some debugging code that ensures that when we do deadlock avoidance, we don't lose the information about how a lock was originally acquired. ........ ................ r118957 | tilghman | 2008-05-29 12:39:50 -0500 (Thu, 29 May 2008) | 10 lines Merged revisions 118954 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118954 | tilghman | 2008-05-29 12:33:01 -0500 (Thu, 29 May 2008) | 2 lines Define also when not DEBUG_THREADS ........ ................ 2008-05-29 04:11 +0000 [r118909] Steve Murphy * main/cdr.c, apps/app_forkcdr.c, /: Merged revisions 118880 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118880 | murf | 2008-05-28 19:29:09 -0600 (Wed, 28 May 2008) | 54 lines Merged revisions 118858 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118858 | murf | 2008-05-28 18:25:28 -0600 (Wed, 28 May 2008) | 46 lines (closes issue #10668) (closes issue #11721) (closes issue #12726) Reported by: arkadia Tested by: murf These changes: 1. revert the changes made via bug 10668; I should have known that such changes, even tho they made sense at the time, seemed like an omission, etc, were actually integral to the CDR system via forkCDR. It makes sense to me now that forkCDR didn't natively end any CDR's, but rather depended on natively closing them all at hangup time via traversing and closing them all, whether locked or not. I still don't completely understand the benefits of setvar and answer operating on locked cdrs, but I've seen enough to revert those changes also, and stop messing up users who depended on that behavior. bug 12726 found reverting the changes fixed his changes, and after a long review and working on forkCDR, I can see why. 2. Apply the suggested enhancements proposed in 10668, but in a completely compatible way. ForkCDR will behave exactly as before, but now has new options that will allow some actions to be taken that will slightly modify the outcome and side-effects of forkCDR. Based on conversations I've had with various people, these small tweaks will allow some users to get the behavior they need. For instance, users executing forkCDR in an AGI script will find the answer time set, and DISPOSITION set, a situation not covered when the routines were first written. 3. A small problem in the cdr serializer would output answer and end times even when they were not set. This is now fixed. ........ ................ 2008-05-28 18:07 +0000 [r118781] Michiel van Baak * /, channels/chan_skinny.c: Merged revisions 118750 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r118750 | mvanbaak | 2008-05-28 19:58:21 +0200 (Wed, 28 May 2008) | 2 lines remove unused astobj.h header file from chan_skinny.c ........ 2008-05-28 14:31 +0000 [r118648] Joshua Colp * /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Merged revisions 118647 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118647 | file | 2008-05-28 11:29:01 -0300 (Wed, 28 May 2008) | 12 lines Merged revisions 118646 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118646 | file | 2008-05-28 11:23:34 -0300 (Wed, 28 May 2008) | 4 lines Add an option to use the source IP address of RTP as the destination IP address of UDPTL when a specific option is enabled. If the remote side is properly configured (ports forwarded) then UDPTL will flow. (closes issue #10417) Reported by: cstadlmann ........ ................ 2008-05-28 14:13 +0000 [r118615-118645] Philippe Sultan * channels/chan_jingle.c, /, include/asterisk/jingle.h: Merged revisions 118644 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r118644 | phsultan | 2008-05-28 16:10:48 +0200 (Wed, 28 May 2008) | 10 lines Changed to temporary namespaces to match with latest XEPs. As soon as Jingle is completely standardized, we can set those namespaces to their final values. Added two attributes to the jingle_pvt struct to store the content name attributes. Reported by Robert McQueen on Telepathy's framework mailing list : http://lists.freedesktop.org/archives/telepathy/2008-May/001971.html Keeping working on our Jingle stack! ........ * channels/chan_jingle.c, /: Merged revisions 118614 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r118614 | phsultan | 2008-05-28 10:39:10 +0200 (Wed, 28 May 2008) | 1 line Code simplification ........ 2008-05-27 19:35 +0000 [r118561] Joshua Colp * /, channels/chan_sip.c: Merged revisions 118560 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118560 | file | 2008-05-27 16:34:14 -0300 (Tue, 27 May 2008) | 12 lines Merged revisions 118558 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118558 | file | 2008-05-27 16:32:38 -0300 (Tue, 27 May 2008) | 4 lines Fix an issue where codec preferences were not set on dialogs that were not authenticated via a user or peer and allow framing to work without rtpmap in the SDP. (closes issue #12501) Reported by: slimey ........ ................ 2008-05-27 19:28 +0000 [r118557] Russell Bryant * /, include/asterisk/compat.h: Merged revisions 118556 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r118556 | russell | 2008-05-27 14:27:48 -0500 (Tue, 27 May 2008) | 6 lines Add printf format attribute for vasprintf(). (closes issue #12729) Reported by: snuffy Patches: bug_12729.diff uploaded by snuffy (license 35) ........ 2008-05-27 19:22 +0000 [r118555] Tilghman Lesher * main/cli.c, /: Merged revisions 118554 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118554 | tilghman | 2008-05-27 14:21:03 -0500 (Tue, 27 May 2008) | 14 lines Merged revisions 118551 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118551 | tilghman | 2008-05-27 14:15:27 -0500 (Tue, 27 May 2008) | 6 lines When showing an error message for a command, don't shorten the command output, as it tends to confuse the user (it's fine for suggesting other commands, however). Reported by: seanbright (on #asterisk-dev) Fixed by: me ........ ................ 2008-05-27 19:09 +0000 [r118518] Mark Michelson * apps/app_chanspy.c, /: Merged revisions 118514 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118514 | mmichelson | 2008-05-27 14:08:24 -0500 (Tue, 27 May 2008) | 19 lines Merged revisions 118509 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118509 | mmichelson | 2008-05-27 14:07:26 -0500 (Tue, 27 May 2008) | 11 lines Russell noted to me that in the case that separate threads use their own addressing system, the fix I made for issue 12376 does not guarantee uniqueness to the datastores' uids. Though I know of no system that works this way, I am going to change this right now to prevent trying to track down some future bug that may occur and cause untold hours of debugging time to track down. The change involves using a global counter which increases with each new chanspy_ds which is created. This guarantees uniqueness. ........ ................ 2008-05-27 18:59 +0000 [r118471] Tilghman Lesher * main/asterisk.c, /: Merged revisions 118466 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118466 | tilghman | 2008-05-27 13:59:06 -0500 (Tue, 27 May 2008) | 16 lines Merged revisions 118465 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118465 | tilghman | 2008-05-27 13:58:09 -0500 (Tue, 27 May 2008) | 8 lines NULL character should terminate only commands back to the core, not log messages to the console. (closes issue #12731) Reported by: seanbright Patches: 20080527__bug12731.diff.txt uploaded by Corydon76 (license 14) Tested by: seanbright ........ ................ 2008-05-27 17:25 +0000 [r118418] Michiel van Baak * apps/app_voicemail.c: small update to the g() option of app_voicemail to note that gain changes only work on zap channels right now. issue #12578 shows it's not clear right now. 2008-05-27 16:48 +0000 [r118378-118382] Mark Michelson * apps/app_chanspy.c, /: Merged revisions 118371 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118371 | mmichelson | 2008-05-27 11:43:36 -0500 (Tue, 27 May 2008) | 22 lines Merged revisions 118365 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118365 | mmichelson | 2008-05-27 11:38:38 -0500 (Tue, 27 May 2008) | 14 lines Add a unique id to the datastore allocated in app_chanspy since it is possible that multiple spies may be listening to the same channel. (closes issue #12376) Reported by: DougUDI Patches: 12376_chanspy_uid.diff uploaded by putnopvut (license 60) Tested by: destiny6628 (closes issue #12243) Reported by: atis ........ ................ * /: Hmm, I apparently forgot to commit the block of revision 118175. Now I'm doing it. 2008-05-27 15:47 +0000 [r118360] Tilghman Lesher * /, configs/queues.conf.sample: Merged revisions 118359 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118359 | tilghman | 2008-05-27 10:46:58 -0500 (Tue, 27 May 2008) | 11 lines Merged revisions 118358 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118358 | tilghman | 2008-05-27 10:45:37 -0500 (Tue, 27 May 2008) | 3 lines Add a note that pbx_config.so is needed for Local channels. (Closes issue #12671) ........ ................ 2008-05-27 14:51 +0000 [r118331] Russell Bryant * /, include/asterisk/compat.h: Merged revisions 118328 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r118328 | russell | 2008-05-27 09:51:13 -0500 (Tue, 27 May 2008) | 2 lines Add printf attribute to asprintf ........ 2008-05-27 13:30 +0000 [r118301-118303] Tilghman Lesher * /, res/res_config_ldap.c: Merged revisions 118302 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r118302 | tilghman | 2008-05-27 08:30:10 -0500 (Tue, 27 May 2008) | 6 lines When binding anonymously, credentials are still needed. (closes issue #12601) Reported by: suretec Patches: res_config_ldap.c.patch uploaded by suretec (license 70) ........ * /, pbx/pbx_realtime.c: Merged revisions 118300 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r118300 | tilghman | 2008-05-27 08:13:17 -0500 (Tue, 27 May 2008) | 4 lines In compat14 mode, don't translate pipes inside expressions, as they aren't argument delimiters, but rather 'or' symbols. (Closes issue #12723) ........ 2008-05-25 16:20 +0000 [r118253] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 118252 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118252 | tilghman | 2008-05-25 11:17:05 -0500 (Sun, 25 May 2008) | 20 lines Merged revisions 118251 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118251 | tilghman | 2008-05-25 11:02:04 -0500 (Sun, 25 May 2008) | 12 lines Realtime flag affects construction in multiple ways, so consulting whether rtcachefriends was set was done too soon (needed to be done inside build_peer, not just as a flag to build_peer). Also, fullcontact needed to be reconstructed, because realtime separates the embedded ';' into multiple fields. (closes issue #12722) Reported by: barthpbx Patches: 20080525__bug12722.diff.txt uploaded by Corydon76 (license 14) Tested by: barthpbx (Much of the discussion happened on #asterisk-dev for diagnosing this issue) ........ ................ 2008-05-24 01:15 +0000 [r118177-118179] Jeff Peeler * doc/api-1.6.0-changes.odt (added), /: Merged revisions 118178 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r118178 | jpeeler | 2008-05-23 20:14:41 -0500 (Fri, 23 May 2008) | 1 line add document describing API changes from 1.4.0 to 1.6.0 ........ 2008-05-23 21:37 +0000 [r118168] Brett Bryant * main/manager.c, /, main/http.c, include/asterisk/manager.h: Merged revisions 118161 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r118161 | bbryant | 2008-05-23 16:19:42 -0500 (Fri, 23 May 2008) | 3 lines Add new functionality to http server that requires manager authentication for any path that includes a directory named 'private'. This patch also requires manager authentication for any POST's being sent to the server as well to help secure uploads. ........ 2008-05-23 21:31 +0000 [r118165] Jeff Peeler * channels/chan_zap.c: Merged revisions 118164 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118164 | jpeeler | 2008-05-23 16:26:39 -0500 (Fri, 23 May 2008) | 9 lines Merged revisions 118163 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118163 | jpeeler | 2008-05-23 16:21:35 -0500 (Fri, 23 May 2008) | 1 line Fix a few things I missed to ensure zt_chan_conf structure is not modified in mkintf ........ ................ 2008-05-23 18:15 +0000 [r118130] Tilghman Lesher * res/res_odbc.c, /: Merged revisions 118129 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r118129 | tilghman | 2008-05-23 13:09:14 -0500 (Fri, 23 May 2008) | 3 lines Protect the object from changing while the 'odbc show' CLI command is running (Closes issue #12704) ........ 2008-05-23 13:00 +0000 [r118054] Tilghman Lesher * doc/cli.txt (added), /: Merged revisions 118053 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118053 | tilghman | 2008-05-23 08:00:10 -0500 (Fri, 23 May 2008) | 11 lines Merged revisions 118052 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118052 | tilghman | 2008-05-23 07:59:16 -0500 (Fri, 23 May 2008) | 3 lines Add information on using the Asterisk console, including tab command line completion. (Closes issue #12681) ........ ................ 2008-05-23 12:37 +0000 [r118050] Russell Bryant * include/asterisk/utils.h, /, main/utils.c: Merged revisions 118049 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118049 | russell | 2008-05-23 07:37:31 -0500 (Fri, 23 May 2008) | 17 lines Merged revisions 118048 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118048 | russell | 2008-05-23 07:30:53 -0500 (Fri, 23 May 2008) | 9 lines Don't declare a function that takes variable arguments as inline, because it's not valid, and on some compilers, will emit a warning. http://gcc.gnu.org/onlinedocs/gcc/Inline.html#Inline (closes issue #12289) Reported by: francesco_r Patches by Tilghman, final patch by me ........ ................ 2008-05-23 11:02 +0000 [r118021] Philippe Sultan * /, channels/chan_gtalk.c, res/res_jabber.c: Merged revisions 118020 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r118020 | phsultan | 2008-05-23 12:33:21 +0200 (Fri, 23 May 2008) | 15 lines - remove whitespaces between tags in received XML packets before giving them to the parser ; - report Gtalk error messages from a buddy to the console. This patch makes Asterisk "Google Jingle" (chan_gtalk) implementation work with Empathy. Note that this is only true for audio streams, not video. Thank you to PH for his great help! (closes issue #12647) Reported by: PH Patches: trunk-12647-1.diff uploaded by phsultan (license 73) Tested by: phsultan, PH ........ 2008-05-22 21:43 +0000 [r117984-117987] Tilghman Lesher * /, pbx/pbx_realtime.c, configs/pbx_realtime.conf (added): Merged revisions 117986 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r117986 | tilghman | 2008-05-22 16:42:50 -0500 (Thu, 22 May 2008) | 2 lines Add a compatibility option for upgrading realtime extensions ........ 2008-05-22 18:55 +0000 [r117901] Tilghman Lesher * main/asterisk.c, /: Merged revisions 117900 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r117900 | tilghman | 2008-05-22 13:54:41 -0500 (Thu, 22 May 2008) | 10 lines Merged revisions 117899 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117899 | tilghman | 2008-05-22 13:53:53 -0500 (Thu, 22 May 2008) | 2 lines Also remove preamble from asynchronous events (reported by jsmith on #asterisk-dev) ........ ................ 2008-05-22 15:51 +0000 [r117793] Sean Bright * /, configs/jabber.conf.sample: Merged revisions 117792 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r117792 | seanbright | 2008-05-22 11:49:17 -0400 (Thu, 22 May 2008) | 1 line Minor text fix. roster -> resource. ........ 2008-05-22 13:41 +0000 [r117757] Russell Bryant * main/asterisk.c, /, build_tools/make_buildopts_h: Merged revisions 117756 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r117756 | russell | 2008-05-22 08:40:52 -0500 (Thu, 22 May 2008) | 5 lines Store build-time options as a string in AST_BUILDOPTS in buildopts.h. Also, display this information in the "core show settings" CLI command. This is useful if you want to verify that you're running a build with DONT_OPTIMIZE, DEBUG_THREADS, etc. ........ 2008-05-21 22:01 +0000 [r117659-117660] Jeff Peeler * channels/chan_zap.c: Merged revisions 117658 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r117658 | jpeeler | 2008-05-21 16:31:17 -0500 (Wed, 21 May 2008) | 10 lines Merged revisions 117582 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117582 | jpeeler | 2008-05-21 15:11:14 -0500 (Wed, 21 May 2008) | 2 lines Ensure that passed in zt_chan_conf structure is not modified in mkintf. ........ ................ * channels/chan_zap.c, /: Merged revisions 117628 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r117628 | jpeeler | 2008-05-21 15:44:04 -0500 (Wed, 21 May 2008) | 12 lines Merged revisions 117462 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117462 | jpeeler | 2008-05-21 11:58:40 -0500 (Wed, 21 May 2008) | 3 lines Pass a pointer for the conf parameter to the function mkintf rather than the whole zt_chan_conf structure. Another commit is following to make sure the zt_chan_conf structure is not modified. ........ ................ 2008-05-21 19:45 +0000 [r117576] Joshua Colp * /, channels/chan_sip.c: Merged revisions 117575 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r117575 | file | 2008-05-21 16:39:42 -0300 (Wed, 21 May 2008) | 10 lines Merged revisions 117574 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117574 | file | 2008-05-21 16:38:28 -0300 (Wed, 21 May 2008) | 2 lines Apply the autoframing setting to dialogs that do not get matched against a user or peer. ........ ................ 2008-05-21 18:44 +0000 [r117522] Tilghman Lesher * main/asterisk.c, /: Merged revisions 117520 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r117520 | tilghman | 2008-05-21 13:43:26 -0500 (Wed, 21 May 2008) | 11 lines Merged revisions 117519 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117519 | tilghman | 2008-05-21 13:40:14 -0500 (Wed, 21 May 2008) | 3 lines Strip the preamble from the output also when -rx is not being used (Related to issue #12702) ........ ................ 2008-05-21 18:29 +0000 [r117486-117516] Russell Bryant * main/asterisk.c, /: Merged revisions 117515 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r117515 | russell | 2008-05-21 13:29:05 -0500 (Wed, 21 May 2008) | 12 lines Merged revisions 117514 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117514 | russell | 2008-05-21 13:28:46 -0500 (Wed, 21 May 2008) | 4 lines Don't filter the magic character in the network verboser. It gets filtered once it reaches the client. (related to issue #12702, pointed out by tilghman) ........ ................ * main/asterisk.c, pbx/pbx_gtkconsole.c, /: Merged revisions 117508 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r117508 | russell | 2008-05-21 13:20:11 -0500 (Wed, 21 May 2008) | 15 lines Merged revisions 117507 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117507 | russell | 2008-05-21 13:19:34 -0500 (Wed, 21 May 2008) | 7 lines 1) Don't print the verbose marker in front of every message from ast_verbose() being sent to remote consoles. 2) Fix pbx_gtkconsole to filter out the verbose marker. (related to issue #12702) ........ ................ * main/asterisk.c, /: Merged revisions 117481 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r117481 | russell | 2008-05-21 13:12:19 -0500 (Wed, 21 May 2008) | 14 lines Merged revisions 117479 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117479 | russell | 2008-05-21 13:11:51 -0500 (Wed, 21 May 2008) | 6 lines Don't display the verbose marker for calls to ast_verbose() that do not include a VERBOSE_PREFIX in front of the message. (closes issue #12702) Reported by: johnlange Patched by me ........ ................ 2008-05-21 02:21 +0000 [r117368] Mark Michelson * main/config.c, /: Merged revisions 117367 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r117367 | mmichelson | 2008-05-20 21:20:31 -0500 (Tue, 20 May 2008) | 19 lines Be sure that we cache included files for each source file which loads a configuration file. As it was, only the first did so. This led to a problem if the included file was changed (but not the configuration file which includes it) and the second source file attempted to reload the configuration. It would not see that the included file had changed. In this particular example, res_phoneprov and chan_sip both loaded sip.conf, which included a file call sip.peers.conf. Since res_phoneprov was the first to load sip.conf, only it cached the fact that sip.conf included sip.peers.conf. If sip.peers.conf were changed and sip.conf were not and a sip reload were issued (meaning that chan_sip attempts to reload sip.conf only if it and its included files have changed) the changes made to sip.peers.conf would not be seen and therefore no action would be taken. (closes issue #12693) Reported by: marsosa ........ 2008-05-21 01:20 +0000 [r117365] Steve Murphy * /, utils/ael_main.c: Merged revisions 117335 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r117335 | murf | 2008-05-20 19:00:28 -0600 (Tue, 20 May 2008) | 10 lines These changes were made via the comments atis_work made at 4:30am (Mountain Time zone- US) in #asterisk-dev on 20 May 2008. He noted that a backslash was being inserted before commas in app call arguments in the extensions.conf.aeldump file that you get from aelparse with the -w arg. This was being generated from code left over from 1.4, where commas were substituted with '|', and any remaining commas needed to be escaped. Many thanks to atis for his comment; please let us know if these changes break anything! ........ 2008-05-19 16:58 +0000 [r117134-117137] Joshua Colp * res/res_smdi.c, /: Merged revisions 117136 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r117136 | file | 2008-05-19 13:53:33 -0300 (Mon, 19 May 2008) | 14 lines Merged revisions 117135 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117135 | file | 2008-05-19 13:50:52 -0300 (Mon, 19 May 2008) | 6 lines Use the right pthread lock and condition when waiting. (closes issue #12664) Reported by: tomo1657 Patches: res_smdi.c.patch uploaded by tomo1657 (license 484) ........ ................ 2008-05-19 16:07 +0000 [r117089] Tilghman Lesher * include/asterisk/utils.h, /: Merged revisions 117088 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r117088 | tilghman | 2008-05-19 11:07:09 -0500 (Mon, 19 May 2008) | 10 lines Merged revisions 117086 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117086 | tilghman | 2008-05-19 11:05:05 -0500 (Mon, 19 May 2008) | 2 lines The addition of usleep(2) within ast_assert requires the inclusion of the unistd.h header ........ ................ 2008-05-19 16:05 +0000 [r117083-117087] Joshua Colp * /, main/logger.c: Merged revisions 117085 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r117085 | file | 2008-05-19 13:03:33 -0300 (Mon, 19 May 2008) | 4 lines The logger closes the files it is logging to when reloading so we have to read in the logger configuration even if it has not changed so that the logs get opened again. (closes issue #12665) Reported by: DennisD ........ * /, channels/h323/ast_h323.cxx: Merged revisions 117082 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r117082 | file | 2008-05-19 12:24:44 -0300 (Mon, 19 May 2008) | 14 lines Merged revisions 117081 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117081 | file | 2008-05-19 12:22:10 -0300 (Mon, 19 May 2008) | 6 lines Make chan_h323 work with pwlib 1.12.0 (closes issue #12682) Reported by: bamby Patches: pwlib_nopipe.diff uploaded by bamby (license 430) ........ ................ 2008-05-19 03:44 +0000 [r116980] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 116979 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r116979 | russell | 2008-05-18 22:44:28 -0500 (Sun, 18 May 2008) | 12 lines Merged revisions 116978 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116978 | russell | 2008-05-18 22:44:04 -0500 (Sun, 18 May 2008) | 4 lines Avoid access of uninitialized memory. This caused a bunch of crashes for me while doing load testing of development branch where I'm working on some performance improvements. ........ ................ 2008-05-18 21:18 +0000 [r116949] Tilghman Lesher * /, utils/astcanary.c: Merged revisions 116948 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r116948 | tilghman | 2008-05-18 16:15:58 -0500 (Sun, 18 May 2008) | 4 lines Add a set of text to the file astcanary uses to communicate back the main Asterisk process, which explains the purpose for the file being there. This should assist people who find the file and wonder why it exists. ........ 2008-05-18 19:59 +0000 [r116922] Russell Bryant * /, channels/chan_sip.c: Merged revisions 116919 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r116919 | russell | 2008-05-18 14:58:10 -0500 (Sun, 18 May 2008) | 3 lines Remove duplicate colon on Reason header (closes issue #12678) ........ 2008-05-17 19:40 +0000 [r116849-116885] Joshua Colp * /, channels/chan_skinny.c: Merged revisions 116800 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r116800 | file | 2008-05-16 17:30:24 -0300 (Fri, 16 May 2008) | 12 lines Merged revisions 116799 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116799 | file | 2008-05-16 17:28:11 -0300 (Fri, 16 May 2008) | 4 lines Check to make sure an RTP structure exists before calling ast_rtp_new_source on it. (closes issue #12669) Reported by: sbisker ........ ................ 2008-05-16 20:03 +0000 [r116798] Matthew Fredrickson * channels/chan_zap.c, /: Merged revisions 116797 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r116797 | mattf | 2008-05-16 15:00:04 -0500 (Fri, 16 May 2008) | 1 line Try to see if we can make our ringback situation a little better ........ 2008-05-15 22:07 +0000 [r116636-116695] Tilghman Lesher * include/asterisk/utils.h, /, include/asterisk/strings.h: Merged revisions 116694 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r116694 | tilghman | 2008-05-15 17:05:47 -0500 (Thu, 15 May 2008) | 4 lines Add an extra check in ast_strlen_zero, and make ast_assert() not print the file, line, and function name twice. (Closes issue #12650) ........ * cdr/cdr_csv.c, /: Merged revisions 116631 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r116631 | tilghman | 2008-05-15 12:58:22 -0500 (Thu, 15 May 2008) | 3 lines Don't unload config on reload, when config has not changed. (Closes issue #12652) ........ 2008-05-14 21:41 +0000 [r116470] Russell Bryant * main/rtp.c, main/sched.c, main/channel.c, main/udptl.c, include/asterisk/utils.h, /, channels/chan_agent.c, main/abstract_jb.c, include/asterisk/channel.h: Merged revisions 116469 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r116469 | russell | 2008-05-14 16:40:43 -0500 (Wed, 14 May 2008) | 12 lines Merged revisions 116463 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116463 | russell | 2008-05-14 16:32:00 -0500 (Wed, 14 May 2008) | 4 lines Add ast_assert(), which can be used to handle fatal errors. It is only compiled in if dev-mode is enabled, and only aborts if DO_CRASH is defined. (inspired by issue #12650) ........ ................ 2008-05-14 21:39 +0000 [r116468] Tilghman Lesher * /, res/res_agi.c: Merged revisions 116467 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r116467 | tilghman | 2008-05-14 16:39:06 -0500 (Wed, 14 May 2008) | 15 lines Merged revisions 116466 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116466 | tilghman | 2008-05-14 16:38:09 -0500 (Wed, 14 May 2008) | 7 lines Avoid zombies when the channel exits before the AGI. (closes issue #12648) Reported by: gkloepfer Patches: 20080514__bug12648.diff.txt uploaded by Corydon76 (license 14) Tested by: gkloepfer ........ ................ 2008-05-14 20:43 +0000 [r116408-116411] Jason Parker * /, configs/voicemail.conf.sample: Merged revisions 116410 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r116410 | qwell | 2008-05-14 15:43:26 -0500 (Wed, 14 May 2008) | 9 lines Merged revisions 116409 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116409 | qwell | 2008-05-14 15:43:08 -0500 (Wed, 14 May 2008) | 1 line Document exitcontext in app_voicemail sample config ........ ................ * apps/app_voicemail.c, /: Merged revisions 116407 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r116407 | qwell | 2008-05-14 15:36:55 -0500 (Wed, 14 May 2008) | 9 lines Voicemail "* exit" should not require an exitcontext to be specified. The behavior in 1.4 was that it would use the current context if an exitcontext existed. (closes issue #12605) Reported by: kenjreno Patches: 12605-starexit.diff uploaded by qwell (license 4) Tested by: file ........ 2008-05-14 18:54 +0000 [r116351-116354] Joshua Colp * /, main/Makefile: Merged revisions 116353 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r116353 | file | 2008-05-14 15:54:16 -0300 (Wed, 14 May 2008) | 12 lines Merged revisions 116352 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116352 | file | 2008-05-14 15:53:39 -0300 (Wed, 14 May 2008) | 4 lines Add linux-gnueabi in. (closes issue #12529) Reported by: tzafrir ........ ................ * /, res/res_config_ldap.c: Merged revisions 116350 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r116350 | file | 2008-05-14 15:25:54 -0300 (Wed, 14 May 2008) | 4 lines Make the ldap version setting work without having both version and protocol set. (closes issue #12613) Reported by: suretec ........ 2008-05-14 17:01 +0000 [r116319] Tilghman Lesher * /, apps/app_externalivr.c: Merged revisions 116298 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r116298 | tilghman | 2008-05-14 11:53:23 -0500 (Wed, 14 May 2008) | 15 lines Merged revisions 116296 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116296 | tilghman | 2008-05-14 11:46:48 -0500 (Wed, 14 May 2008) | 2 lines Detect another way for a connection to have gone away. (closes issue #12618) Reported by: ctooley Patches: 1.4-externalivr-test_fd.diff uploaded by ctooley (license 136) trunk-externalivr-test_fd.diff uploaded by ctooley (license 136) ........ ................ 2008-05-14 Russell Bryant * Asterisk 1.6.0-beta9 released. 2008-05-14 13:13 +0000 [r116236] Olle Johansson * /, channels/chan_sip.c: Merged revisions 116234 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r116234 | oej | 2008-05-14 15:05:15 +0200 (Ons, 14 Maj 2008) | 11 lines Merged revisions 116230 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116230 | oej | 2008-05-14 14:51:06 +0200 (Ons, 14 Maj 2008) | 3 lines Accept text messages even with Content-Type: text/plain;charset=Södermanländska ........ ................ 2008-05-14 00:20 +0000 [r116096-116139] Mark Michelson * main/channel.c, /, include/asterisk/lock.h: Merged revisions 116089 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r116089 | mmichelson | 2008-05-13 18:54:01 -0500 (Tue, 13 May 2008) | 20 lines Merged revisions 116088 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116088 | mmichelson | 2008-05-13 18:47:49 -0500 (Tue, 13 May 2008) | 12 lines A change to the way channel locks are handled when DEBUG_CHANNEL_LOCKS is defined. After debugging a deadlock, it was noticed that when DEBUG_CHANNEL_LOCKS is enabled in menuselect, the actual origin of channel locks is obscured by the fact that all channel locks appear to happen in the function ast_channel_lock(). This code change redefines ast_channel_lock to be a macro which maps to __ast_channel_lock(), which then relays the proper file name, line number, and function name information to the core lock functions so that this information will be displayed in the case that there is some sort of locking error or core show locks is issued. ........ ................ 2008-05-13 21:19 +0000 [r116020-116040] Russell Bryant * channels/chan_local.c, /: Merged revisions 116039 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r116039 | russell | 2008-05-13 16:18:55 -0500 (Tue, 13 May 2008) | 32 lines Merged revisions 116038 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116038 | russell | 2008-05-13 16:17:23 -0500 (Tue, 13 May 2008) | 24 lines Fix a deadlock involving channel autoservice and chan_local that was debugged and fixed by mmichelson and me. We observed a system that had a bunch of threads stuck in ast_autoservice_stop(). The reason these threads were waiting around is because this function waits to ensure that the channel list in the autoservice thread gets rebuilt before the stop() function returns. However, the autoservice thread was also locked, so the autoservice channel list was never getting rebuilt. The autoservice thread was stuck waiting for the channel lock on a local channel. However, the local channel was locked by a thread that was stuck in the autoservice stop function. It turned out that the issue came down to the local_queue_frame() function in chan_local. This function assumed that one of the channels passed in as an argument was locked when called. However, that was not always the case. There were multiple cases in which this channel was not locked when the function was called. We fixed up chan_local to indicate to this function whether this channel was locked or not. The previous assumption had caused local_queue_frame() to improperly return with the channel locked, where it would then never get unlocked. (closes issue #12584) (related to issue #12603) ........ ................ * main/autoservice.c, /: Merged revisions 116001 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r116001 | russell | 2008-05-13 16:07:59 -0500 (Tue, 13 May 2008) | 13 lines Merged revisions 115990 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115990 | russell | 2008-05-13 16:05:57 -0500 (Tue, 13 May 2008) | 5 lines Fix an issue that I noticed in autoservice while mmichelson and I were debugging a different problem. I noticed that it was theoretically possible for two threads to attempt to start the autoservice thread at the same time. This change makes the process of starting the autoservice thread, thread-safe. ........ ................ 2008-05-13 20:30 +0000 [r115946] Joshua Colp * /, channels/chan_alsa.c: Merged revisions 115945 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115945 | file | 2008-05-13 17:29:27 -0300 (Tue, 13 May 2008) | 12 lines Merged revisions 115944 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115944 | file | 2008-05-13 17:28:23 -0300 (Tue, 13 May 2008) | 4 lines Use the right flag to open the audio in non-blocking. (closes issue #12616) Reported by: nicklewisdigiumuser ........ ................ 2008-05-13 20:19 +0000 [r115940-115942] Matthew Fredrickson * channels/chan_zap.c, /: Merged revisions 115941 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115941 | mattf | 2008-05-13 15:18:04 -0500 (Tue, 13 May 2008) | 1 line Need to clear calling_party_cat variable after we retrieve it ........ * channels/chan_zap.c, /: Merged revisions 115939 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115939 | mattf | 2008-05-13 15:11:20 -0500 (Tue, 13 May 2008) | 1 line Add support for receiving calling party category ........ 2008-05-13 18:38 +0000 [r115887] Tilghman Lesher * main/asterisk.c, /: Merged revisions 115886 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115886 | tilghman | 2008-05-13 13:38:11 -0500 (Tue, 13 May 2008) | 11 lines Merged revisions 115884 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115884 | tilghman | 2008-05-13 13:36:13 -0500 (Tue, 13 May 2008) | 3 lines If the socket dies (read returns 0=EOF), return immediately. (Closes issue #12637) ........ ................ 2008-05-13 17:48 +0000 [r115848-115851] Russell Bryant * res/res_smdi.c, /: Merged revisions 115847 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115847 | russell | 2008-05-13 12:14:22 -0500 (Tue, 13 May 2008) | 2 lines Initialize the start time in smdi_msg_wait. Somehow this code got lost in trunk. ........ 2008-05-12 17:57 +0000 [r115738] Mark Michelson * main/utils.c: Merged revisions 115737 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115737 | mmichelson | 2008-05-12 12:55:08 -0500 (Mon, 12 May 2008) | 15 lines Merged revisions 115735 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115735 | mmichelson | 2008-05-12 12:51:14 -0500 (Mon, 12 May 2008) | 7 lines If a thread holds no locks, do not print any information on the thread when issuing a core show locks command. This will help to de-clutter output somewhat. Russell said it would be fine to place this improvement in the 1.4 branch, so that's why it's going here too. ........ ................ 2008-05-12 16:36 +0000 [r115706] Jason Parker * /, apps/app_queue.c: Merged revisions 115705 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115705 | qwell | 2008-05-12 11:35:50 -0500 (Mon, 12 May 2008) | 1 line Correctly document state interface for AddQueueMember. Discovered while looking at issue #12626. ........ 2008-05-12 15:18 +0000 [r115672] Brett Bryant * /, channels/chan_iax2.c: Merged revisions 115669 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115669 | bbryant | 2008-05-12 10:17:32 -0500 (Mon, 12 May 2008) | 3 lines A small change to fix iax2 native bridging. ........ 2008-05-11 03:27 +0000 [r115599-115601] Matthew Fredrickson * channels/chan_zap.c, /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 115600 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115600 | mattf | 2008-05-10 22:23:05 -0500 (Sat, 10 May 2008) | 1 line Add Zap MTP2 support to chan_zap ........ * channels/chan_zap.c, /: Merged revisions 115598 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115598 | mattf | 2008-05-10 21:19:21 -0500 (Sat, 10 May 2008) | 1 line Open up audio channel when we get ACM on SS7 event ........ 2008-05-10 14:22 +0000 [r115597] Tilghman Lesher * /, cdr/cdr_pgsql.c: Merged revisions 115596 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115596 | tilghman | 2008-05-10 09:19:41 -0500 (Sat, 10 May 2008) | 2 lines Ensure that "calldate" is acceptable for a column name. ........ 2008-05-09 16:38 +0000 [r115581] Joshua Colp * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 115580 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115580 | file | 2008-05-09 13:36:58 -0300 (Fri, 09 May 2008) | 10 lines Merged revisions 115579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115579 | file | 2008-05-09 13:34:08 -0300 (Fri, 09 May 2008) | 2 lines Improve res_ninit and res_ndestroy autoconf logic on the Darwin platform. ........ ................ 2008-05-08 19:21 +0000 [r115553-115570] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 115569 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115569 | russell | 2008-05-08 14:20:35 -0500 (Thu, 08 May 2008) | 10 lines Merged revisions 115568 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115568 | russell | 2008-05-08 14:19:50 -0500 (Thu, 08 May 2008) | 2 lines Remove debug output. ........ ................ * /, channels/chan_iax2.c: Merged revisions 115566 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115566 | russell | 2008-05-08 14:17:04 -0500 (Thu, 08 May 2008) | 41 lines Merged revisions 115565 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r115565 | russell | 2008-05-08 14:15:25 -0500 (Thu, 08 May 2008) | 33 lines Merged revisions 115564 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r115564 | russell | 2008-05-08 14:14:04 -0500 (Thu, 08 May 2008) | 25 lines Fix a race condition that bbryant just found while doing some IAX2 testing. He was running Asterisk trunk running IAX2 calls through a few Asterisk boxes, however, the audio was extremely choppy. We looked at a packet trace and saw a storm of INVAL and VNAK frames being sent from one box to another. It turned out that what had happened was that one box tried to send a CONTROL frame before the 3 way handshake had completed. So, that frame did not include the destination call number, because it didn't have it yet. Part of our recent work for security issues included an additional check to ensure that frames that are supposed to include the destination call number have the correct one. This caused the frame to be rejected with an INVAL. The frame would get retransmitted for forever, rejected every time ... This race condition exists in all versions that got the security changes, in theory. However, it is really only likely that this would cause a problem in Asterisk trunk. There was a control frame being sent (SRCUPDATE) at the _very_ beginning of the call, which does not exist in 1.2 or 1.4. However, I am fixing all versions that could potentially be affected by the introduced race condition. These changes are what bbryant and I came up with to fix the issue. Instead of simply dropping control frames that get sent before the handshake is complete, the code attempts to wait a little while, since in most cases, the handshake will complete very quickly. If it doesn't complete after yielding for a little while, then the frame gets dropped. ........ ................ ................ * /, channels/chan_sip.c: Merged revisions 115562 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115562 | russell | 2008-05-08 11:14:08 -0500 (Thu, 08 May 2008) | 11 lines Merged revisions 115561 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115561 | russell | 2008-05-08 11:11:33 -0500 (Thu, 08 May 2008) | 3 lines Don't give up on attempting an outbound registration if we receive a 408 Timeout. (closes issue #12323) ........ ................ * /, contrib/scripts/postgres_cdr.sql (removed): Merged revisions 115558 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115558 | russell | 2008-05-08 10:38:27 -0500 (Thu, 08 May 2008) | 11 lines Merged revisions 115557 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115557 | russell | 2008-05-08 10:37:49 -0500 (Thu, 08 May 2008) | 3 lines remove postgres_cdr.sql, as the CDR schema is in realtime_pgsql.sql, as well (closes issue #9676) ........ ................ * contrib/init.d/rc.debian.asterisk, /: Merged revisions 115555 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115555 | russell | 2008-05-08 10:32:48 -0500 (Thu, 08 May 2008) | 11 lines Merged revisions 115554 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115554 | russell | 2008-05-08 10:32:08 -0500 (Thu, 08 May 2008) | 3 lines Don't exit the script if Asterisk is not running. (closes issue #12611) ........ ................ * main/pbx.c, /: Merged revisions 115552 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115552 | russell | 2008-05-08 10:26:49 -0500 (Thu, 08 May 2008) | 12 lines Merged revisions 115551 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115551 | russell | 2008-05-08 10:24:54 -0500 (Thu, 08 May 2008) | 4 lines Don't use a channel before checking for channel allocation failure. (closes issue #12609) Reported by: edantie ........ ................ 2008-05-08 15:08 +0000 [r115549] Matthew Fredrickson * channels/chan_zap.c, /: Merged revisions 115548 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115548 | mattf | 2008-05-08 10:04:45 -0500 (Thu, 08 May 2008) | 1 line Remove unused code as well as demote an error message to a debug message ........ 2008-05-08 14:41 +0000 [r115538-115547] Russell Bryant * contrib/init.d/rc.debian.asterisk, /: Merged revisions 115546 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115546 | russell | 2008-05-08 09:41:12 -0500 (Thu, 08 May 2008) | 12 lines Merged revisions 115545 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115545 | russell | 2008-05-08 09:40:53 -0500 (Thu, 08 May 2008) | 4 lines Use the same method for executing Asterisk as the rest of the script. (closes issue #12611) Reported by: b_plessis ........ ................ 2008-05-07 18:35 +0000 [r115514-115524] Russell Bryant * /, res/res_config_ldap.c: Merged revisions 115523 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115523 | russell | 2008-05-07 13:33:50 -0500 (Wed, 07 May 2008) | 6 lines Only save a password if a username exists. (closes issue #12600) Reported By: suretec Patch by me ........ * /, res/res_config_ldap.c: Merged revisions 115521 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115521 | russell | 2008-05-07 13:30:12 -0500 (Wed, 07 May 2008) | 7 lines Use the default that the log output claims will be used for the basedn (closes issue #12599) Reported by: suretec Patches: 12599.patch uploaded by juggie (license 24) ........ * /, channels/chan_h323.c: Merged revisions 115519 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115519 | russell | 2008-05-07 13:24:51 -0500 (Wed, 07 May 2008) | 2 lines Let chan_h323 build in dev mode ........ * /, include/asterisk/dlinkedlists.h (removed), channels/chan_iax2.c: Merged revisions 115513 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115513 | russell | 2008-05-07 12:28:19 -0500 (Wed, 07 May 2008) | 19 lines Merged revisions 115512 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r115512 | russell | 2008-05-07 11:24:09 -0500 (Wed, 07 May 2008) | 11 lines Merged revisions 115511 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r115511 | russell | 2008-05-07 11:22:49 -0500 (Wed, 07 May 2008) | 3 lines Remove remnants of dlinkedlists. I didn't actually use them in the final version of my IAX2 improvements. ........ ................ ................ 2008-05-07 13:49 +0000 [r115510] Tilghman Lesher * contrib/scripts/asterisk.ldap-schema, contrib/scripts/asterisk.ldif, /: Merged revisions 115509 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115509 | tilghman | 2008-05-07 08:49:15 -0500 (Wed, 07 May 2008) | 2 lines Update typos in description fields (closes issue #12598) Reported by: suretec Patches: asterisk_schema_changes.patch uploaded by suretec (license 70) ........ 2008-05-06 19:56 +0000 [r115420-115424] Jason Parker * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 115423 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115423 | qwell | 2008-05-06 14:55:45 -0500 (Tue, 06 May 2008) | 23 lines Merged revisions 115422 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r115422 | qwell | 2008-05-06 14:55:29 -0500 (Tue, 06 May 2008) | 15 lines Merged revisions 115421 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r115421 | qwell | 2008-05-06 14:54:57 -0500 (Tue, 06 May 2008) | 7 lines read requires an argument on some non-bash shells (closes issue #12593) Reported by: bkruse Patches: getilbc.sh_12593_v1.diff uploaded by bkruse (license 132) ........ ................ ................ * /, res/res_musiconhold.c: Merged revisions 115419 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115419 | qwell | 2008-05-06 14:38:44 -0500 (Tue, 06 May 2008) | 15 lines Merged revisions 115418 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115418 | qwell | 2008-05-06 14:34:58 -0500 (Tue, 06 May 2008) | 7 lines Switch to using ast_random() rather than just rand(). This does not fix the bug reported, but I believe it is correct. (from issue #12446) Patches: bug_12446.diff uploaded by snuffy (license 35) ........ ................ 2008-05-06 19:33 +0000 [r115417] Tilghman Lesher * main/asterisk.c, /: Merged revisions 115416 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115416 | tilghman | 2008-05-06 14:32:29 -0500 (Tue, 06 May 2008) | 10 lines Merged revisions 115415 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115415 | tilghman | 2008-05-06 14:31:39 -0500 (Tue, 06 May 2008) | 2 lines Don't print the terminating NUL. (Closes issue #12589) ........ ................ 2008-05-06 13:57 +0000 [r115343] Joshua Colp * /, configure, configure.ac: Merged revisions 115342 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115342 | file | 2008-05-06 10:55:44 -0300 (Tue, 06 May 2008) | 10 lines Merged revisions 115341 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115341 | file | 2008-05-06 10:54:15 -0300 (Tue, 06 May 2008) | 2 lines Add in missing argument. ........ ................ 2008-05-05 23:01 +0000 [r115335] Tilghman Lesher * main/asterisk.c, /, main/logger.c: Merged revisions 115334 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115334 | tilghman | 2008-05-05 18:00:31 -0500 (Mon, 05 May 2008) | 15 lines Merged revisions 115333 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115333 | tilghman | 2008-05-05 17:50:31 -0500 (Mon, 05 May 2008) | 7 lines Separate verbose output from CLI output, by using a preamble. (closes issue #12402) Reported by: Corydon76 Patches: 20080410__no_verbose_in_rx_output.diff.txt uploaded by Corydon76 (license 14) 20080501__no_verbose_in_rx_output__1.4.diff.txt uploaded by Corydon76 (license 14) ........ ................ 2008-05-05 22:17 +0000 [r115331] Joshua Colp * /, build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, codecs/codec_speex.c, configure.ac: Merged revisions 115328 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115328 | file | 2008-05-05 19:13:57 -0300 (Mon, 05 May 2008) | 10 lines Merged revisions 115327 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115327 | file | 2008-05-05 19:10:05 -0300 (Mon, 05 May 2008) | 2 lines Make sure that either the main speex library contains preprocess functions or that speexdsp does. If both fail then speex stuff can not be built. ........ ................ 2008-05-05 22:14 +0000 [r115330] Mark Michelson * main/config.c, /: Merged revisions 115329 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115329 | mmichelson | 2008-05-05 17:14:06 -0500 (Mon, 05 May 2008) | 15 lines #execing the same file multiple times led to warning messages saying that the same file was being #included twice. This was due to the fact that #exec created a temporary file which was then #included. The name of the temporary file was the name of the #exec'd file, with the Unix timestamp and thread ID concatenated. The issue was that if multiple #exec statements of the same file were reached in the same second, then the result was that the temporary files would have duplicate names. To resolve this, the temporary file now has microsecond resolution for the timestamp portion. (closes issue #12574) Reported by: jmls Patches: 12574.patch uploaded by putnopvut (license 60) Tested by: jmls, putnopvut ........ 2008-05-05 21:44 +0000 [r115322] Mark Michelson * /, apps/app_queue.c: Merged revisions 115321 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115321 | mmichelson | 2008-05-05 16:43:21 -0500 (Mon, 05 May 2008) | 21 lines Merged revisions 115320 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115320 | mmichelson | 2008-05-05 16:41:34 -0500 (Mon, 05 May 2008) | 13 lines Don't consider a caller "handled" until the caller is bridged with a queue member. There was too much of an opportunity for the member to hang up (either during a delay, announcement, or overly long agi) between the time that he answered the phone and the time when he actually was bridged with the caller. The consequence of this was that if the member hung up in that interval, then proper abandonment details would not be noted in the queue log if the caller were to hang up at any point after the member hangup. (closes issue #12561) Reported by: ablackthorn ........ ................ 2008-05-05 20:28 +0000 [r115316] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 115315 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115315 | russell | 2008-05-05 15:28:17 -0500 (Mon, 05 May 2008) | 2 lines Remove my rant, since I have now replaced the rant with code. ........ 2008-05-05 19:58 +0000 [r115310] Tilghman Lesher * include/asterisk/res_odbc.h, /: Merged revisions 115309 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115309 | tilghman | 2008-05-05 14:57:28 -0500 (Mon, 05 May 2008) | 10 lines Merged revisions 115308 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115308 | tilghman | 2008-05-05 14:55:55 -0500 (Mon, 05 May 2008) | 2 lines Err, the documentation on the return value of ast_odbc_backslash_is_escape is exactly backwards. ........ ................ 2008-05-05 19:50 +0000 [r115306] Russell Bryant * /, channels/chan_sip.c: Merged revisions 115305 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115305 | russell | 2008-05-05 14:50:24 -0500 (Mon, 05 May 2008) | 13 lines Merged revisions 115304 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115304 | russell | 2008-05-05 14:49:25 -0500 (Mon, 05 May 2008) | 5 lines Avoid putting opaque="" in Digest authentication. This patch came from switchvox. It fixes authentication with Primus in Canada, and has been in use for a very long time without causing problems with any other providers. (closes issue AST-36) ........ ................ 2008-05-05 19:43 +0000 [r115303] Tilghman Lesher * /, UPGRADE.txt: Merged revisions 115302 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115302 | tilghman | 2008-05-05 14:42:36 -0500 (Mon, 05 May 2008) | 2 lines Note change for ExecIf syntax (caught by jmls on IRC) ........ 2008-05-05 10:55 +0000 [r115289] Kevin P. Fleming * /, UPGRADE.txt: Merged revisions 115288 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115288 | kpfleming | 2008-05-05 05:55:09 -0500 (Mon, 05 May 2008) | 2 lines clarify wording ........ 2008-05-05 03:26 +0000 [r115287] Tilghman Lesher * contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk, contrib/init.d/rc.debian.asterisk, contrib/init.d/rc.mandrake.asterisk, /, contrib/init.d/rc.redhat.asterisk, contrib/init.d/rc.gentoo.asterisk, contrib/init.d/rc.slackware.asterisk: Merged revisions 115286 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115286 | tilghman | 2008-05-04 22:25:35 -0500 (Sun, 04 May 2008) | 15 lines Merged revisions 115285 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115285 | tilghman | 2008-05-04 22:22:25 -0500 (Sun, 04 May 2008) | 7 lines When starting Asterisk, bug out if Asterisk is already running. (closes issue #12525) Reported by: explidous Patches: 20080428__bug12525.diff.txt uploaded by Corydon76 (license 14) Tested by: mvanbaak ........ ................ 2008-05-04 02:12 +0000 [r115278-115284] Joshua Colp * /, configure, acinclude.m4: Merged revisions 115283 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115283 | file | 2008-05-03 23:11:01 -0300 (Sat, 03 May 2008) | 10 lines Merged revisions 115282 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115282 | file | 2008-05-03 23:09:44 -0300 (Sat, 03 May 2008) | 2 lines Expand the test function for GCC attributes so that more complex attributes are properly recognized. ........ ................ * /, include/asterisk/compiler.h: Merged revisions 115280 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115280 | file | 2008-05-03 22:52:00 -0300 (Sat, 03 May 2008) | 10 lines Merged revisions 115279 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115279 | file | 2008-05-03 22:50:59 -0300 (Sat, 03 May 2008) | 2 lines For my next trick I will make these work with what our autoconf header file gives us. ........ ................ * /, configure, acinclude.m4: Merged revisions 115277 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115277 | file | 2008-05-03 22:45:21 -0300 (Sat, 03 May 2008) | 10 lines Merged revisions 115276 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115276 | file | 2008-05-03 22:43:26 -0300 (Sat, 03 May 2008) | 2 lines Treat warnings as errors when checking if a GCC attribute exists. We have to do this as GCC will just ignore the attribute and pop up a warning, it won't actually fail to compile. ........ ................ 2008-05-03 04:25 +0000 [r115269-115275] Dwayne M. Hubbard * /: block voicemail mwi notification subscriptions taskprocessor * /: block pbx taskprocessor * /: block app_queue taskprocessor * /: blocked taskprocessors 2008-05-02 14:55 +0000 [r115198-115200] Mark Michelson * /, include/asterisk/sched.h: Merged revisions 115197 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115197 | mmichelson | 2008-05-02 09:28:55 -0500 (Fri, 02 May 2008) | 14 lines Merged revisions 115196 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115196 | mmichelson | 2008-05-02 09:28:19 -0500 (Fri, 02 May 2008) | 6 lines Clarify a comment that was, well, just wrong. It turns out that ignoring the way that macros expand. Instead, I have clarified in the comment why the macro will work even if the scheduler id for the task to be deleted changes during the execution of the macro. ........ ................ 2008-05-02 02:57 +0000 [r115107-115160] Tilghman Lesher * include/asterisk/res_odbc.h, /: Merged revisions 115104 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115104 | tilghman | 2008-05-01 18:21:13 -0500 (Thu, 01 May 2008) | 10 lines Merged revisions 115102 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115102 | tilghman | 2008-05-01 18:20:25 -0500 (Thu, 01 May 2008) | 2 lines Change the comment of deprecated to an actual compiler deprecation ........ ................ 2008-05-01 19:01 +0000 [r115020] Tilghman Lesher * /, main/utils.c: Merged revisions 115018 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115018 | tilghman | 2008-05-01 14:00:18 -0500 (Thu, 01 May 2008) | 14 lines Merged revisions 115017 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115017 | tilghman | 2008-05-01 13:59:08 -0500 (Thu, 01 May 2008) | 6 lines '#' is another reserved character for URIs that also needs to be escaped. (closes issue #10543) Reported by: blitzrage Patches: 20080418__bug10543.diff.txt uploaded by Corydon76 (license 14) ........ ................ 2008-05-01 17:28 +0000 [r114932] Russell Bryant * /, UPGRADE.txt: Merged revisions 114931 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114931 | russell | 2008-05-01 12:28:25 -0500 (Thu, 01 May 2008) | 4 lines Clarify the deprecation notice about Macro() to note that it will not be removed for the sake of backwards compatibility, since it is a non-trivial task to convert existing large dialplans that depend on Macro() to use GoSub(), instead. ........ 2008-05-01 16:52 +0000 [r114923] Jason Parker * channels/chan_zap.c, /: Merged revisions 114922 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114922 | qwell | 2008-05-01 11:49:24 -0500 (Thu, 01 May 2008) | 10 lines Allow dringXrange to properly default to 10, as was done in 1.4. dringXrange is a new feature that was added, and it attempted to default, but only when the option was specified. (closes issue #12536) Reported by: bjm Patches: 12536-dringXrange.diff uploaded by qwell (license 4) Tested by: bjm ........ 2008-04-30 20:20 +0000 [r114909] Russell Bryant * include/asterisk/dlinkedlists.h (added): Add the dlinkedlists implementation from trunk 2008-04-30 20:17 +0000 [r114907-114908] Mark Michelson * channels/chan_sip.c: Make 1.6.0 compile 2008-04-30 17:06 +0000 [r114900] Olle Johansson * /, channels/chan_sip.c: Merged revisions 114899 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114899 | oej | 2008-04-30 18:55:49 +0200 (Ons, 30 Apr 2008) | 15 lines Merged revisions 114890 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114890 | oej | 2008-04-30 18:23:17 +0200 (Ons, 30 Apr 2008) | 7 lines Don't crash on bad SIP replys. Fix created in Huntsville together with Mark M (putnopvut) (closes issue #12363) Reported by: jvandal Tested by: putnopvut, oej ........ ................ 2008-04-30 16:41 +0000 [r114893] Russell Bryant * /, channels/chan_console.c, channels/chan_iax2.c: Merged revisions 114892 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114892 | russell | 2008-04-30 11:34:24 -0500 (Wed, 30 Apr 2008) | 36 lines Merged revisions 114891 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114891 | russell | 2008-04-30 11:30:01 -0500 (Wed, 30 Apr 2008) | 28 lines Merge changes from team/russell/iax2_find_callno and iax2_find_callno_1.4 These changes address a critical performance issue introduced in the latest release. The fix for the latest security issue included a change that made Asterisk randomly choose call numbers to make them more difficult to guess by attackers. However, due to some inefficient (this is by far, an understatement) code, when Asterisk chose high call numbers, chan_iax2 became unusable after just a small number of calls. On a small embedded platform, it would not be able to handle a single call. On my Intel Core 2 Duo @ 2.33 GHz, I couldn't run more than about 16 IAX2 channels. Ouch. These changes address some performance issues of the find_callno() function that have bothered me for a very long time. On every incoming media frame, it iterated through every possible call number trying to find a matching active call. This involved a mutex lock and unlock for each call number checked. So, if the random call number chosen was 20000, then every media frame would cause 20000 locks and unlocks. Previously, this problem was not as obvious since Asterisk always chose the lowest call number it could. A second container for IAX2 pvt structs has been added. It is an astobj2 hash table. When we know the remote side's call number, the pvt goes into the hash table with a hash value of the remote side's call number. Then, lookups for incoming media frames are a very fast hash lookup instead of an absolutely insane array traversal. In a quick test, I was able to get more than 3600% more IAX2 channels on my machine with these changes. ........ ................ 2008-04-30 16:15 +0000 [r114889] Jeff Peeler * /, channels/chan_console.c: Merged revisions 114888 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114888 | jpeeler | 2008-04-30 11:14:43 -0500 (Wed, 30 Apr 2008) | 3 lines Fixes a bug where if a stream monitor thread was not created (caused from failure of opening or starting the stream) pthread_cancel was called with an invalid thread ID. ........ 2008-04-30 14:55 +0000 [r114877-114886] Kevin P. Fleming * /, channels/iax2.h, channels/chan_iax2.c: Merged revisions 114884 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114884 | kpfleming | 2008-04-30 09:49:51 -0500 (Wed, 30 Apr 2008) | 10 lines Merged revisions 114880 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114880 | kpfleming | 2008-04-30 09:46:57 -0500 (Wed, 30 Apr 2008) | 2 lines use the ARRAY_LEN macro for indexing through the iaxs/iaxsl arrays so that the size of the arrays can be adjusted in one place, and change the size of the arrays from 32768 calls to 2048 calls when LOW_MEMORY is defined ........ ................ * /, Makefile.rules: Merged revisions 114876 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114876 | kpfleming | 2008-04-30 07:15:43 -0500 (Wed, 30 Apr 2008) | 10 lines Merged revisions 114875 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114875 | kpfleming | 2008-04-30 07:14:07 -0500 (Wed, 30 Apr 2008) | 2 lines pay attention to *all* header files for dependency tracking, not just the local ones (inspired by r578 of asterisk-addons by tilghman) ........ ................ 2008-04-29 22:55 +0000 [r114867] Jeff Peeler * /, channels/iax2-provision.c: Merged revisions 114866 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114866 | jpeeler | 2008-04-29 17:54:14 -0500 (Tue, 29 Apr 2008) | 2 lines Fixes a problem where all the templates were marked as dead no matter what. The templates should only be marked as dead if a configuration file has been successfully loaded and has changes. Bug found while making API documentation for 1.6.0. ........ 2008-04-29 21:09 +0000 [r114850-114858] Mark Michelson * /, apps/app_queue.c: Merged revisions 114849 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114849 | mmichelson | 2008-04-29 14:42:04 -0500 (Tue, 29 Apr 2008) | 22 lines Merged revisions 114848 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114848 | mmichelson | 2008-04-29 14:40:06 -0500 (Tue, 29 Apr 2008) | 14 lines Use the MACRO_CONTEXT and MACRO_EXTEN channel variables instead of the channel's macrocontext and macroexten fields. This is needed because if macros are daisy-chained, the incorrect context and extension are placed on the new channel. I also added locking to the channel prior to accessing these variables as noted in trunk's janitor project file. (closes issue #12549) Reported by: darren1713 Patches: app_queue.c.macroextenpatch uploaded by darren1713 (license 116) (with modifications from me) Tested by: putnopvut ........ ................ 2008-04-29 19:04 +0000 [r114846] Kevin P. Fleming * /: bug is not present in this branch 2008-04-29 17:11 +0000 [r114831] Jason Parker * res/res_config_pgsql.c, /: Merged revisions 114830 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114830 | qwell | 2008-04-29 12:10:55 -0500 (Tue, 29 Apr 2008) | 9 lines Merged revisions 114829 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114829 | qwell | 2008-04-29 12:08:55 -0500 (Tue, 29 Apr 2008) | 1 line Change warning message to debug, since there are cases where 0 results is perfectly fine. ........ ................ 2008-04-29 12:55 +0000 [r114825] Kevin P. Fleming * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 114824 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114824 | kpfleming | 2008-04-29 07:54:31 -0500 (Tue, 29 Apr 2008) | 18 lines Merged revisions 114823 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r114823 | kpfleming | 2008-04-29 07:53:12 -0500 (Tue, 29 Apr 2008) | 10 lines Merged revisions 114822 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r114822 | kpfleming | 2008-04-29 07:52:32 -0500 (Tue, 29 Apr 2008) | 2 lines stop script from appending source code if run multiple times ........ ................ ................ 2008-04-28 17:04 +0000 [r114777] Matthew Fredrickson * channels/chan_zap.c, /: Merged revisions 114776 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114776 | mattf | 2008-04-28 12:00:38 -0500 (Mon, 28 Apr 2008) | 1 line Fix deadlock issue in chan_zap with libss7 due to channel variables being set with the channel pvt lock being held. #12512 ........ 2008-04-28 13:44 +0000 [r114714] Joshua Colp * /, configure, configure.ac: Merged revisions 114713 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114713 | file | 2008-04-28 10:42:13 -0300 (Mon, 28 Apr 2008) | 2 lines Update autoconf logic with latest API change for libss7. ........ 2008-04-28 04:54 +0000 [r114707-114710] Tilghman Lesher * channels/chan_jingle.c, /, channels/chan_gtalk.c: Merged revisions 114709 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114709 | tilghman | 2008-04-27 23:53:20 -0500 (Sun, 27 Apr 2008) | 13 lines Merged revisions 114708 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114708 | tilghman | 2008-04-27 23:47:39 -0500 (Sun, 27 Apr 2008) | 5 lines When modules are embedded, they take on a different name, without the ".so" extension. Specifically check for this name, when we're checking if a module is loaded. (Closes issue #12534) ........ ................ 2008-04-27 15:20 +0000 [r114701] Michiel van Baak * /, channels/chan_skinny.c: Merged revisions 114700 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk Merged to 1.6 because it fixes a crash. ........ r114700 | mvanbaak | 2008-04-27 17:17:18 +0200 (Sun, 27 Apr 2008) | 8 lines Make MWI in chan_skinny event based modeled after chan_zap and chan_mgcp. (closes issue #12214) Reported by: DEA Patches: chan_skinny-vm-events-v3.txt uploaded by DEA (license 3) Tested by: DEA and me ........ 2008-04-27 01:30 +0000 [r114697] Sean Bright * /, configure, configure.ac: Merged revisions 114696 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114696 | seanbright | 2008-04-26 21:28:32 -0400 (Sat, 26 Apr 2008) | 13 lines Merged revisions 114695 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114695 | seanbright | 2008-04-26 21:26:15 -0400 (Sat, 26 Apr 2008) | 5 lines When we don't explicitly pass a path to the --with-tds configure option, we may end up finding tds.h in /usr/local/include instead of /usr/include. If this happens, the grep that looks for the version (from tdsver.h) will fail and we'll have some problems during the build. ........ ................ 2008-04-26 15:09 +0000 [r114684-114693] Tilghman Lesher * /, contrib/scripts/vmail.cgi: Merged revisions 114690 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114690 | tilghman | 2008-04-26 08:17:19 -0500 (Sat, 26 Apr 2008) | 14 lines Merged revisions 114689 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114689 | tilghman | 2008-04-26 08:15:21 -0500 (Sat, 26 Apr 2008) | 6 lines Clicking forward without selecting a message leaves an errant .lock file. (closes issue #12528) Reported by: pukepail Patches: patch.diff uploaded by pukepail (license 431) ........ ................ 2008-04-25 22:05 +0000 [r114671-114677] Russell Bryant * /, pbx/pbx_lua.c: Merged revisions 114676 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114676 | russell | 2008-04-25 17:04:46 -0500 (Fri, 25 Apr 2008) | 7 lines Lock the channel around datastore access (closes issue #12527) Reported by: mnicholson Patches: pbx_lua4.diff uploaded by mnicholson (license 96) ........ * /, channels/chan_iax2.c: Merged revisions 114674 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114674 | russell | 2008-04-25 17:00:35 -0500 (Fri, 25 Apr 2008) | 11 lines Merged revisions 114673 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114673 | russell | 2008-04-25 16:54:40 -0500 (Fri, 25 Apr 2008) | 3 lines Use consistent logic for checking to see if a call number has been chosen yet. Also, remove some redundant logic I recently added in a fix. ........ ................ 2008-04-25 19:34 +0000 [r114664] Mark Michelson * apps/app_chanspy.c, /: Merged revisions 114663 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114663 | mmichelson | 2008-04-25 14:33:27 -0500 (Fri, 25 Apr 2008) | 12 lines Merged revisions 114662 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114662 | mmichelson | 2008-04-25 14:32:02 -0500 (Fri, 25 Apr 2008) | 4 lines Move the unlock of the spyee channel to outside the start_spying() function so that the channel is not unlocked twice when using whisper mode. ........ ................ 2008-04-25 16:26 +0000 [r114652] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 114651 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114651 | mmichelson | 2008-04-25 11:25:17 -0500 (Fri, 25 Apr 2008) | 4 lines Fix a memory leak and protect against potential dereferences of a NULL pointer. ........ 2008-04-24 22:14 +0000 [r114636] Joshua Colp * /, channels/chan_sip.c: Merged revisions 114635 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114635 | file | 2008-04-24 19:11:46 -0300 (Thu, 24 Apr 2008) | 4 lines Hey look, it builds. (closes issue #12519) Reported by: falves11 ........ 2008-04-24 21:36 +0000 [r114626-114634] Mark Michelson * /, channels/chan_sip.c: Merged revisions 114633 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114633 | mmichelson | 2008-04-24 16:35:39 -0500 (Thu, 24 Apr 2008) | 19 lines Merged revisions 114632 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114632 | mmichelson | 2008-04-24 16:35:08 -0500 (Thu, 24 Apr 2008) | 11 lines Re-invite RTP during a masquerade so that, for instance, an AMI redirect of two channels which are natively bridged will preserve audio on both channels. This prevents a problem with Asterisk not re-inviting due to one of the channels having being a zombie. (closes issue #12513) Reported by: mneuhauser Patches: asterisk-1.4-114602_restore-RTP-on-fixup.patch uploaded by mneuhauser (license 425) ........ ................ * /, apps/app_queue.c: Merged revisions 114629 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114629 | mmichelson | 2008-04-24 15:43:52 -0500 (Thu, 24 Apr 2008) | 16 lines Merged revisions 114628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114628 | mmichelson | 2008-04-24 15:43:03 -0500 (Thu, 24 Apr 2008) | 8 lines Output of channel variables when eventwhencalled=vars was set was being truncated two characters. This patch corrects the problem. (closes issue #12493) Reported by: davidw ........ ................ * channels/chan_local.c, /: Merged revisions 114625 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114625 | mmichelson | 2008-04-24 15:06:06 -0500 (Thu, 24 Apr 2008) | 18 lines Merged revisions 114624 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114624 | mmichelson | 2008-04-24 15:04:24 -0500 (Thu, 24 Apr 2008) | 10 lines Resolve a deadlock in chan_local by releasing the channel lock temporarily. (closes issue #11712) Reported by: callguy Patches: 11712.patch uploaded by putnopvut (license 60) Tested by: acunningham ........ ................ 2008-04-24 19:55 +0000 [r114619-114623] Tilghman Lesher * channels/chan_local.c, /: Merged revisions 114622 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114622 | tilghman | 2008-04-24 14:54:57 -0500 (Thu, 24 Apr 2008) | 12 lines Merged revisions 114621 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114621 | tilghman | 2008-04-24 14:53:36 -0500 (Thu, 24 Apr 2008) | 4 lines Ensure that when we set the accountcode, it actually shows up in the CDR. (Fix for AMI Originate) (Closes issue #12007) ........ ................ * /, apps/app_meetme.c: Merged revisions 114617 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114617 | tilghman | 2008-04-24 14:24:31 -0500 (Thu, 24 Apr 2008) | 6 lines Fix DST calculation, and fix bug in calculation of whether conf has started yet or not (Closes issue #12292) Reported by: DEA Patches: app_meetme-rt-dst-sched-fix.txt uploaded by DEA (license 3) ........ 2008-04-24 16:48 +0000 [r114613] Jason Parker * channels/chan_misdn.c, /: Merged revisions 114612 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114612 | qwell | 2008-04-24 11:47:01 -0500 (Thu, 24 Apr 2008) | 17 lines Merged revisions 51989 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #12496) Reported by: daniele Patches: misdn-moh-1.6.0-beta7.1.patch uploaded by daniele (license 471) Tested by: daniele Technically, I didn't use the patch above except to find out what revision to merge - but it's the same thing as this revision. ........ r51989 | crichter | 2007-01-24 06:57:22 -0600 (Wed, 24 Jan 2007) | 1 line added fix from #8899 ........ ................ 2008-04-24 15:57 +0000 [r114610] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 114609 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114609 | russell | 2008-04-24 10:56:55 -0500 (Thu, 24 Apr 2008) | 12 lines Merged revisions 114608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114608 | russell | 2008-04-24 10:55:21 -0500 (Thu, 24 Apr 2008) | 4 lines Fix a silly mistake in a change I made yesterday that caused chan_iax2 to blow up very quickly. (issue #12515) ........ ................ 2008-04-24 15:00 +0000 [r114607] Olle Johansson * channels/chan_sip.c: Merged revisions 114606 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114606 | oej | 2008-04-24 16:59:05 +0200 (Tor, 24 Apr 2008) | 11 lines Merged revisions 114603 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114603 | oej | 2008-04-24 16:55:18 +0200 (Tor, 24 Apr 2008) | 3 lines Only have one max-forwards header in outbound REFERs. Discovered in the Asterisk SIP Masterclass in Orlando. Thanks Joe! ........ ................ 2008-04-24 14:56 +0000 [r114599-114605] Russell Bryant * /, channels/chan_sip.c: Merged revisions 114604 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114604 | russell | 2008-04-24 09:55:21 -0500 (Thu, 24 Apr 2008) | 3 lines Change a verbose message to debug. (closes issue #12514) ........ * /, main/http.c: Merged revisions 114601 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114601 | russell | 2008-04-23 17:53:20 -0500 (Wed, 23 Apr 2008) | 14 lines Merged revisions 114600 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114600 | russell | 2008-04-23 17:18:12 -0500 (Wed, 23 Apr 2008) | 6 lines Improve some broken cookie parsing code. Previously, manager login over HTTP would only work if the mansession_id cookie was first. Now, the code builds a list of all of the cookies in the Cookie header. This fixes a problem observed by users of the Asterisk GUI. (closes AST-20) ........ ................ * apps/app_chanspy.c, /: Merged revisions 114598 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114598 | russell | 2008-04-23 15:53:05 -0500 (Wed, 23 Apr 2008) | 18 lines Merged revisions 114597 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114597 | russell | 2008-04-23 15:49:18 -0500 (Wed, 23 Apr 2008) | 10 lines Fix an issue that caused getting the correct next channel to not always work. Also, remove setting the amount of time to wait for a digit from 5 seconds back down to 1/10 of a second. I believe this was so the beep didn't get played over and over really fast, but a while back I put in another fix for that issue. (closes issue #12498) Reported by: jsmith Patches: app_chanspy_channel_walk.trunk.patch uploaded by jsmith (license 15) ........ ................ 2008-04-23 18:34 +0000 [r114596] Jason Parker * /, res/res_musiconhold.c: Merged revisions 114595 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114595 | qwell | 2008-04-23 13:33:28 -0500 (Wed, 23 Apr 2008) | 16 lines Merged revisions 114594 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114594 | qwell | 2008-04-23 13:28:44 -0500 (Wed, 23 Apr 2008) | 8 lines Fix reload/unload for res_musiconhold module. (closes issue #11575) Reported by: sunder Patches: M11575_14_rev3.diff uploaded by junky (license 177) bug11575_trunk.diff.txt uploaded by jamesgolovich (license 176) ........ ................ 2008-04-23 18:01 +0000 [r114589-114593] Russell Bryant * main/manager.c, /, include/asterisk/manager.h: Merged revisions 114592 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114592 | russell | 2008-04-23 13:01:00 -0500 (Wed, 23 Apr 2008) | 13 lines Merged revisions 114591 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114591 | russell | 2008-04-23 12:55:31 -0500 (Wed, 23 Apr 2008) | 5 lines Store the manager session ID explicitly as 4 byte ID instead of a ulong. The mansession_id cookie is coded to be limited to 8 characters of hex, and this could break logins from 64-bit machines in some cases. (inspired by AST-20) ........ ................ * /, channels/chan_iax2.c: Merged revisions 114588 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114588 | russell | 2008-04-23 12:18:29 -0500 (Wed, 23 Apr 2008) | 10 lines Merged revisions 114587 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114587 | russell | 2008-04-23 12:16:32 -0500 (Wed, 23 Apr 2008) | 2 lines Fix find_callno_locked() to actually return the callno locked in some more cases. ........ ................ 2008-04-23 16:57 +0000 [r114586] Olle Johansson * channels/chan_sip.c: Merged revisions 114585 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114585 | oej | 2008-04-23 18:53:34 +0200 (Ons, 23 Apr 2008) | 10 lines Merged revisions 114584 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114584 | oej | 2008-04-23 18:51:41 +0200 (Ons, 23 Apr 2008) | 2 lines Add 502 support for both directions, not only one... (see r114571) ........ ................ 2008-04-23 14:56 +0000 [r114581] Joshua Colp * main/pbx.c, /: Merged revisions 114580 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114580 | file | 2008-04-23 11:55:03 -0300 (Wed, 23 Apr 2008) | 12 lines Merged revisions 114579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114579 | file | 2008-04-23 11:54:11 -0300 (Wed, 23 Apr 2008) | 4 lines Instead of stopping dialplan execution when SayNumber attempts to say a large number that it can not print out a message informing the user and continue on. (closes issue #12502) Reported by: bcnit ........ ................ 2008-04-23 01:00 +0000 [r114576-114578] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 114575 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114575 | mmichelson | 2008-04-22 19:40:30 -0500 (Tue, 22 Apr 2008) | 10 lines Round 1 of IMAP_STORAGE-related app_voicemail changes This makes IMAP_STORAGE include the proper headers if you have specified the "system" option for --with-imap when running the configure script and your IMAP-related headers exist in /usr/include/c-client. This change is due to a hasty merge of a 1.4 change I made. ........ 2008-04-22 23:59 +0000 [r114573] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 114572 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114572 | tilghman | 2008-04-22 18:58:19 -0500 (Tue, 22 Apr 2008) | 10 lines Merged revisions 114571 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114571 | tilghman | 2008-04-22 18:51:44 -0500 (Tue, 22 Apr 2008) | 2 lines Treat a 502 just like a 503, when it comes to processing a response code ........ ................ 2008-04-22 Russell Bryant * Asterisk 1.6.0-beta8 released. 2008-04-22 22:18 +0000 [r114560] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 114559 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114559 | russell | 2008-04-22 17:17:31 -0500 (Tue, 22 Apr 2008) | 13 lines Merged revisions 114558 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114558 | russell | 2008-04-22 17:15:36 -0500 (Tue, 22 Apr 2008) | 5 lines When we receive a full frame that is supposed to contain our call number, ensure that it has the correct one. (closes issue #10078) (AST-2008-006) ........ ................ 2008-04-22 22:04 +0000 [r114556] Steve Murphy * main/pbx.c, /: Merged revisions 114553 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114553 | murf | 2008-04-22 15:57:57 -0600 (Tue, 22 Apr 2008) | 14 lines (closes issue #12469) Reported by: triccyx I had a bit a problem reproducing this in my setup (trying not to disturb my other stuff) but finally, I got it. The problem appears to be that the extension is being added in replace mode, which kinda assumes that the pattern trie has been formed, when in fact, in this case, it was not. The checks being done are not nec. when the tree is not yet formed, as changes like this will be summarized when the trie is formed in the future. I tested the fix, and the crash no longer happens. Feel free to open the bug again if this fix doesn't cure the problem. ........ 2008-04-22 21:16 +0000 [r114544-114552] Russell Bryant * main/channel.c, /: Merged revisions 114548 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114548 | russell | 2008-04-22 15:25:56 -0500 (Tue, 22 Apr 2008) | 2 lines re-add a fix that got lost with a recent change ........ 2008-04-22 18:14 +0000 [r114541] Jason Parker * main/pbx.c, /, include/asterisk/pbx.h, apps/app_queue.c: Merged revisions 114540 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114540 | qwell | 2008-04-22 13:14:09 -0500 (Tue, 22 Apr 2008) | 8 lines Allow setqueuevar=yes (et al) to work, after changes to pbx_builtin_setvar() (closes issue #12490) Reported by: bcnit Patches: 12490-queuevars-3.diff uploaded by qwell (license 4) Tested by: qwell ........ 2008-04-22 18:06 +0000 [r114534-114539] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 114538 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114538 | russell | 2008-04-22 13:04:39 -0500 (Tue, 22 Apr 2008) | 17 lines Merged revisions 114537 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114537 | russell | 2008-04-22 13:03:33 -0500 (Tue, 22 Apr 2008) | 9 lines If the dial string passed to the call channel callback does not indicate an extension, then consider the extension on the channel before falling back to the default. (closes issue #12479) Reported by: darren1713 Patches: exten_dial_fix_chan_iax2.c.patch uploaded by darren1713 (license 116) ........ ................ 2008-04-22 15:46 +0000 [r114524-114528] Russell Bryant * main/manager.c, /: Merged revisions 114527 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114527 | russell | 2008-04-22 10:46:01 -0500 (Tue, 22 Apr 2008) | 8 lines Correct action_ping() and action_events() with regards to Manager 1.1 documentation. Also, fix a bug in xml_translate(). (closes issue #11649) Reported by: ys Patches: trunk_manager.c.diff uploaded by ys (license 281) ........ 2008-04-21 20:23 +0000 [r114422] Matthew Fredrickson * channels/chan_zap.c, /: Merged revisions 114389 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114389 | mattf | 2008-04-21 13:44:35 -0500 (Mon, 21 Apr 2008) | 1 line Add support for generic name transmission (#12484) on SS7 in chan_zap ........ 2008-04-21 15:38 +0000 [r114328] Jeff Peeler * /, apps/app_authenticate.c: Merged revisions 114327 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114327 | jpeeler | 2008-04-21 10:34:37 -0500 (Mon, 21 Apr 2008) | 2 lines This removes an invalid warning message for an incorrectly entered pin, but more importantly removes an inapplicable check. If the first argument passed to app_authenticate does not contain a '/', the argument should be treated as the sole fixed "password" to match against and that is all. (Previous behavior was attempting to open a file based on the pin.) ........ 2008-04-21 14:42 +0000 [r114321-114324] Joshua Colp * /, channels/chan_sip.c: Merged revisions 114323 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114323 | file | 2008-04-21 11:40:33 -0300 (Mon, 21 Apr 2008) | 12 lines Merged revisions 114322 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114322 | file | 2008-04-21 11:39:32 -0300 (Mon, 21 Apr 2008) | 4 lines Only drop audio if we receive it without a progress indication. We allow other frames through such as DTMF because they may be needed to complete the call. (closes issue #12440) Reported by: aragon ........ ................ * /, res/res_config_ldap.c: Merged revisions 114320 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114320 | file | 2008-04-21 11:34:06 -0300 (Mon, 21 Apr 2008) | 6 lines Only print out the error message if ldap_modify_ext_s actually returns an error, and not success. (closes issue #12438) Reported by: gservat Patches: res_config_ldap.c-patch-code uploaded by gservat (license 466) ........ 2008-04-19 17:00 +0000 [r114304] Matthew Fredrickson * channels/chan_zap.c: SS7:Added - Generic Name / Access Transport / Redirecting Number handling. #12425 2008-04-18 21:51 +0000 [r114277-114286] Russell Bryant * main/manager.c, /: Merged revisions 114285 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114285 | russell | 2008-04-18 16:51:05 -0500 (Fri, 18 Apr 2008) | 10 lines Merged revisions 114284 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114284 | russell | 2008-04-18 16:48:06 -0500 (Fri, 18 Apr 2008) | 2 lines Don't destroy a manager session if poll() returns an error of EAGAIN. ........ ................ * Makefile, /: Merged revisions 114279 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114279 | russell | 2008-04-18 15:01:47 -0500 (Fri, 18 Apr 2008) | 10 lines Merged revisions 114278 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114278 | russell | 2008-04-18 15:01:09 -0500 (Fri, 18 Apr 2008) | 2 lines ensure directories are created before we try to install stuff into them ........ ................ * Makefile, /: Merged revisions 114276 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114276 | russell | 2008-04-18 14:59:17 -0500 (Fri, 18 Apr 2008) | 10 lines Merged revisions 114275 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114275 | russell | 2008-04-18 14:58:55 -0500 (Fri, 18 Apr 2008) | 2 lines SUBDIRS_INSTALL is already listed as a subtarget for bininstall ........ ................ 2008-04-18 19:36 +0000 [r114262-114272] Joshua Colp * channels/chan_unistim.c, /: Merged revisions 114271 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114271 | file | 2008-04-18 16:35:33 -0300 (Fri, 18 Apr 2008) | 4 lines Make sure ADSI is marked as unavailable on Unistim channels so voicemail does not try to do some ADSI jazz. (closes issue #12460) Reported by: PerryB ........ 2008-04-18 18:04 +0000 [r114260] Mark Michelson * channels/chan_zap.c, /, main/callerid.c: Merged revisions 114259 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114259 | mmichelson | 2008-04-18 13:03:06 -0500 (Fri, 18 Apr 2008) | 14 lines Merged revisions 114257 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114257 | mmichelson | 2008-04-18 12:44:29 -0500 (Fri, 18 Apr 2008) | 6 lines Clearing up error messages so they make a bit more sense. Also removing a redundant error message. Issue AST-15 ........ ................ 2008-04-18 16:12 +0000 [r114255] Joshua Colp * /, res/res_config_ldap.c: Merged revisions 114254 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114254 | file | 2008-04-18 13:11:27 -0300 (Fri, 18 Apr 2008) | 4 lines If the parsing of the config file fails make sure we unlock ldap_lock. (closes issue #12477) Reported by: IgorG ........ 2008-04-18 13:40 +0000 [r114247] Sean Bright * channels/chan_sip.c: Merged revisions 114246 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114246 | seanbright | 2008-04-18 09:38:07 -0400 (Fri, 18 Apr 2008) | 9 lines Merged revisions 114245 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114245 | seanbright | 2008-04-18 09:33:32 -0400 (Fri, 18 Apr 2008) | 1 line Only complete the SIP channel name once for 'sip show channel ' ........ ................ 2008-04-18 06:54 +0000 [r114244] Tilghman Lesher * apps/app_setcallerid.c, /: Merged revisions 114243 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114243 | tilghman | 2008-04-18 01:53:47 -0500 (Fri, 18 Apr 2008) | 11 lines Merged revisions 114242 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114242 | tilghman | 2008-04-18 01:49:16 -0500 (Fri, 18 Apr 2008) | 3 lines For consistency sake, ensure that the values that ${CALLINGPRES} returns are valid as an input to SetCallingPres. (Closes issue #12472) ........ ................ 2008-04-17 23:09 +0000 [r114232-114241] Russell Bryant * /, channels/chan_sip.c: Merged revisions 114151 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114151 | oej | 2008-04-15 15:39:29 -0500 (Tue, 15 Apr 2008) | 10 lines Merged revisions 114148 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114148 | oej | 2008-04-15 22:26:05 +0200 (Tis, 15 Apr 2008) | 2 lines Handle subscribe queues in all situations... Thanks to festr_ on irc for telling me about this bug. ........ ................ * /, channels/chan_sip.c: Merged revisions 114150 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114150 | oej | 2008-04-15 15:31:08 -0500 (Tue, 15 Apr 2008) | 2 lines Adding chanvar to SIPPEER from 1.4 branch ........ * main/autoservice.c, /: Merged revisions 114233 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114233 | russell | 2008-04-17 17:24:00 -0500 (Thu, 17 Apr 2008) | 14 lines Merged revisions 114230 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114230 | russell | 2008-04-17 17:15:43 -0500 (Thu, 17 Apr 2008) | 6 lines Remove redundant safety net. The check for the autoservice channel list state accomplishes the same goal in a better way. (issue #12470) Reported By: atis ........ ................ 2008-04-17 21:05 +0000 [r114228] Mark Michelson * apps/app_chanspy.c, /: Merged revisions 114227 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114227 | mmichelson | 2008-04-17 16:04:40 -0500 (Thu, 17 Apr 2008) | 17 lines Merged revisions 114226 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114226 | mmichelson | 2008-04-17 16:03:29 -0500 (Thu, 17 Apr 2008) | 9 lines Declaration of the peer channel in this scope was making it so the peer variable defined in the outer scope was never set properly, therefore making iterating through the channel list always restart from the beginning. This bug would have affected anyone who called chanspy without specifying a first argument. (closes issue #12461) Reported by: stever28 ........ ................ 2008-04-17 16:51 +0000 [r114210-114213] Mark Michelson * main/dsp.c, main/frame.c, /, include/asterisk/dsp.h, include/asterisk/frame.h: Merged revisions 114208 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114208 | mmichelson | 2008-04-17 11:40:12 -0500 (Thu, 17 Apr 2008) | 20 lines Merged revisions 114207 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114207 | mmichelson | 2008-04-17 11:28:03 -0500 (Thu, 17 Apr 2008) | 12 lines It was possible for a reference to a frame which was part of a freed DSP to still be referenced, leading to memory corruption and eventual crashes. This code change ensures that the dsp is freed when we are finished with the frame. This change is very similar to a change Russell made with translators back a month or so ago. (closes issue #11999) Reported by: destiny6628 Patches: 11999.patch uploaded by putnopvut (license 60) Tested by: destiny6628, victoryure ........ ................ 2008-04-17 16:26 +0000 [r114206] Russell Bryant * Makefile, /: Merged revisions 114205 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114205 | russell | 2008-04-17 11:25:29 -0500 (Thu, 17 Apr 2008) | 11 lines Merged revisions 114204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114204 | russell | 2008-04-17 11:23:45 -0500 (Thu, 17 Apr 2008) | 3 lines Fix the bininstall target to install from subdirs, as well. (closes issue AST-8, patch from bmd at switchvox) ........ ................ 2008-04-17 15:17 +0000 [r114203] Tilghman Lesher * doc/CODING-GUIDELINES, /: Merged revisions 114202 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114202 | tilghman | 2008-04-17 10:12:52 -0500 (Thu, 17 Apr 2008) | 2 lines fileio.h does not exist; io.h does, though. ........ 2008-04-17 13:55 +0000 [r114200] Philippe Sultan * /, res/res_jabber.c: Merged revisions 114199 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114199 | phsultan | 2008-04-17 15:46:17 +0200 (Thu, 17 Apr 2008) | 10 lines Merged revisions 114198 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114198 | phsultan | 2008-04-17 15:42:23 +0200 (Thu, 17 Apr 2008) | 2 lines Use keepalives effectively in order diagnose bug #12432. ........ ................ 2008-04-17 12:59 +0000 [r114197] Tilghman Lesher * /, res/res_agi.c: Merged revisions 114196 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114196 | tilghman | 2008-04-17 07:59:04 -0500 (Thu, 17 Apr 2008) | 16 lines Merged revisions 114195 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114195 | tilghman | 2008-04-17 07:56:38 -0500 (Thu, 17 Apr 2008) | 8 lines Add special case for when the agi cannot be executed, to comply with the documentation that we return failure in that case. (closes issue #12462) Reported by: fmueller Patches: 20080416__bug12462.diff.txt uploaded by Corydon76 (license 14) Tested by: fmueller ........ ................ 2008-04-17 10:56 +0000 [r114193] Sean Bright * apps/app_chanspy.c, /: Merged revisions 114192 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114192 | seanbright | 2008-04-17 06:55:05 -0400 (Thu, 17 Apr 2008) | 9 lines Merged revisions 114191 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114191 | seanbright | 2008-04-17 06:51:20 -0400 (Thu, 17 Apr 2008) | 1 line Make sure we have enough room for the recording's filename. ........ ................ 2008-04-16 20:48 +0000 [r114186] Kevin P. Fleming * channels/chan_zap.c, /: Merged revisions 114185 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114185 | kpfleming | 2008-04-16 15:47:30 -0500 (Wed, 16 Apr 2008) | 14 lines Merged revisions 114184 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114184 | kpfleming | 2008-04-16 15:46:38 -0500 (Wed, 16 Apr 2008) | 6 lines use the ZT_SET_DIALPARAMS ioctl properly by initializing the structure to all zeroes in case it contains fields that we don't write values into (which it does as of Zaptel 1.4.10) (closes issue #12456) Reported by: fnordian ........ ................ 2008-04-15 20:53 +0000 [r114153] Tilghman Lesher * /, cdr/cdr_pgsql.c: Merged revisions 114152 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114152 | tilghman | 2008-04-15 15:51:08 -0500 (Tue, 15 Apr 2008) | 2 lines Oops, buffer wasn't long enough for query ........ 2008-04-15 20:09 +0000 [r114147] Steve Murphy * main/pbx.c, /: Merged revisions 114146 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114146 | murf | 2008-04-15 13:59:50 -0600 (Tue, 15 Apr 2008) | 8 lines These changes: a. fix a self-found problem with SPAWN-ing an extension, where matches were not being found b. correct some wording in a comment c. Add some debug for future debugging. ........ 2008-04-15 17:22 +0000 [r114132-114142] Jason Parker * channels/chan_unistim.c, /: Merged revisions 114141 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114141 | qwell | 2008-04-15 12:21:58 -0500 (Tue, 15 Apr 2008) | 8 lines Shorten the mac address pattern, since some phones use different identifiers (such as the i2050 softphone). (closes issue #12398) Reported by: c_hans Patches: chan_unistim_svn.diff uploaded by c (license 460) Tested by: c_hans ........ * contrib/scripts/autosupport, /: Merged revisions 114139 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114139 | qwell | 2008-04-15 12:17:37 -0500 (Tue, 15 Apr 2008) | 15 lines Merged revisions 114138 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114138 | qwell | 2008-04-15 12:17:18 -0500 (Tue, 15 Apr 2008) | 7 lines Update Digium autosupport script, for more useful information. (closes issue #12452) Reported by: angler Patches: autosupport.diff uploaded by angler (license 106) ........ ................ * /, apps/app_queue.c: Merged revisions 114134 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114134 | qwell | 2008-04-15 11:18:38 -0500 (Tue, 15 Apr 2008) | 16 lines Merged revisions 114133 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114133 | qwell | 2008-04-15 11:18:08 -0500 (Tue, 15 Apr 2008) | 8 lines Allow autofill to work in the general section of queues.conf. Additionally, don't try to (re)set options when they have empty values in realtime (all unset columns would have an empty value). (closes issue #12445) Reported by: atis Patches: 12445-autofill.diff uploaded by qwell (license 4) ........ ................ 2008-04-14 18:34 +0000 [r114122] Jason Parker * /, channels/chan_h323.c: Merged revisions 114121 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114121 | qwell | 2008-04-14 13:34:17 -0500 (Mon, 14 Apr 2008) | 15 lines Merged revisions 114120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114120 | qwell | 2008-04-14 13:31:57 -0500 (Mon, 14 Apr 2008) | 7 lines The call_token on the pvt can occasionally be NULL, causing a crash. If it is NULL, we can skip this channel, since it can't the one we're looking for. (closes issue #9299) Reported by: vazir ........ ................ 2008-04-14 17:42 +0000 [r114119] Mark Michelson * main/channel.c, /: Merged revisions 114118 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114118 | mmichelson | 2008-04-14 12:42:20 -0500 (Mon, 14 Apr 2008) | 19 lines Merged revisions 114117 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114117 | mmichelson | 2008-04-14 12:41:03 -0500 (Mon, 14 Apr 2008) | 11 lines Increase the retry count when attempting to show channels. This apparently cleared an issue someone was seeing when attempting to show channels when the load was high. (closes issue #11667) Reported by: falves11 Patches: 11677.txt uploaded by russell (license 2) Tested by: falves11 ........ ................ 2008-04-14 16:33 +0000 [r114116] Tilghman Lesher * /, contrib/scripts/astcli: Merged revisions 114115 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114115 | tilghman | 2008-04-14 11:32:59 -0500 (Mon, 14 Apr 2008) | 2 lines Make tab-completion work for all cases ........ 2008-04-14 16:25 +0000 [r114114] Mark Michelson * apps/app_dial.c, /, apps/app_queue.c: Merged revisions 114113 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114113 | mmichelson | 2008-04-14 11:25:09 -0500 (Mon, 14 Apr 2008) | 17 lines Merged revisions 114112 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114112 | mmichelson | 2008-04-14 11:24:22 -0500 (Mon, 14 Apr 2008) | 9 lines If the datastore has been moved to another channel due to a masquerade, then freeing the datastore here causes an eventual double free when the new channel hangs up. We should only free the datastore if we were able to successfully remove it from the channel we are referencing (i.e. the datastore was not moved). (closes issue #12359) Reported by: pguido ........ ................ 2008-04-14 15:02 +0000 [r114108] Mark Michelson * main/channel.c, /: Merged revisions 114107 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114107 | mmichelson | 2008-04-14 10:01:36 -0500 (Mon, 14 Apr 2008) | 13 lines Merged revisions 114106 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114106 | mmichelson | 2008-04-14 09:58:02 -0500 (Mon, 14 Apr 2008) | 5 lines Save a local copy of the generate callback prior to unlocking the channel in case the generate callback goes NULL on us after the channel is unlocked. Thanks to Russell for pointing this need out to me. ........ ................ 2008-04-14 14:54 +0000 [r114102-114105] Joshua Colp * /, channels/chan_sip.c: Merged revisions 114104 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114104 | file | 2008-04-14 11:53:33 -0300 (Mon, 14 Apr 2008) | 12 lines Merged revisions 114103 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114103 | file | 2008-04-14 11:52:46 -0300 (Mon, 14 Apr 2008) | 4 lines It is possible for the remote side to say they want T38 but not give any capabilities. (closes issue #12414) Reported by: MVF ........ ................ * main/rtp.c, /: Merged revisions 114101 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114101 | file | 2008-04-14 10:53:33 -0300 (Mon, 14 Apr 2008) | 12 lines Merged revisions 114100 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114100 | file | 2008-04-14 10:52:49 -0300 (Mon, 14 Apr 2008) | 4 lines Don't change the SSRC when a new source comes into play, this might happen quite often and depending on the remote side... they might not like this. (closes issue #12353) Reported by: dimas ........ ................ 2008-04-14 02:59 +0000 [r114097-114099] Tilghman Lesher * /, contrib/scripts/astcli: Merged revisions 114098 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114098 | tilghman | 2008-04-13 21:55:41 -0500 (Sun, 13 Apr 2008) | 3 lines Add tab command-line completion (Closes issue #12428) ........ * /, apps/app_meetme.c: Merged revisions 114096 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114096 | tilghman | 2008-04-13 09:35:43 -0500 (Sun, 13 Apr 2008) | 3 lines Use ast_mkdir instead of mkdir (Closes issue #12430) ........ 2008-04-12 16:22 +0000 [r114094-114095] Matthew Fredrickson * channels/chan_zap.c: Make sure linkset is locked exiting ss7_start_call * channels/chan_zap.c: Make sure we start incoming calls on SS7 with echo cancellation enabled. Also make sure when completing a COT we call ss7_start_call with the proper locks held. Lastly, make sure if we fail to get a channel from zt_new that we don't assume it's there. 2008-04-11 23:27 +0000 [r114089-114091] Tilghman Lesher * /, cdr/cdr_pgsql.c: Merged revisions 114090 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114090 | tilghman | 2008-04-11 18:26:56 -0500 (Fri, 11 Apr 2008) | 3 lines If any field is not null, but has no default, then it must be set or the insert will fail. (Closes issue #12285) ........ * /, configs/res_ldap.conf.sample: Merged revisions 114088 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114088 | tilghman | 2008-04-11 18:21:54 -0500 (Fri, 11 Apr 2008) | 3 lines Make the sample config match the contributed LDAP schema (Closes issue #12421) ........ 2008-04-11 23:21 +0000 [r114087] Terry Wilson * /, channels/chan_iax2.c: Merged revisions 114084 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114084 | twilson | 2008-04-11 17:48:52 -0500 (Fri, 11 Apr 2008) | 15 lines Merged revisions 114083 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114083 | twilson | 2008-04-11 17:32:51 -0500 (Fri, 11 Apr 2008) | 7 lines Several places in the code called find_callno() (which releases the lock on the pvt structure) and then immediately locked the call and did things with it. Unfortunately, the call can disappear between the find_callno and the lock, causing Bad Stuff(tm) to happen. Added find_callno_locked() function to return the callno withtout unlocking for instances that it is needed. (issue #12400) Reported by: ztel ........ ................ 2008-04-11 23:13 +0000 [r114086] Tilghman Lesher * /, res/res_config_ldap.c: Merged revisions 114085 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114085 | tilghman | 2008-04-11 18:12:16 -0500 (Fri, 11 Apr 2008) | 7 lines Use the correct function for free'ing objects, and maybe we won't crash. (closes issue #12163) Reported by: gservat Patches: 20080411__bug12163.diff.txt uploaded by Corydon76 (license 14) Tested by: gservat ........ 2008-04-11 15:51 +0000 [r114065] Mark Michelson * /, main/features.c: Merged revisions 114064 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114064 | mmichelson | 2008-04-11 10:49:35 -0500 (Fri, 11 Apr 2008) | 19 lines Merged revisions 114063 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114063 | mmichelson | 2008-04-11 10:44:28 -0500 (Fri, 11 Apr 2008) | 11 lines Fix a race condition that may happen between a sip hangup and a "core show channel" command. This patch adds locking to prevent the resulting crash. (closes issue #12155) Reported by: tsearle Patches: show_channels_crash2.patch uploaded by tsearle (license 373) Tested by: tsearle ........ ................ 2008-04-11 14:56 +0000 [r114062] Tilghman Lesher * /, res/res_config_ldap.c: Merged revisions 114061 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114061 | tilghman | 2008-04-11 09:54:22 -0500 (Fri, 11 Apr 2008) | 6 lines Errors are all greater than 0 (closes issue #12422) Reported by: nito Patches: res_config_ldap_result_check_patch.diff uploaded by nito (license 340) ........ 2008-04-10 22:23 +0000 [r114056] Mark Michelson * utils/conf2ael.c, utils/check_expr.c, utils/Makefile, main/manager.c, /, utils/astman.c, utils/hashtest.c, main/utils.c, include/asterisk/lock.h, utils/ael_main.c, utils/hashtest2.c: Merged revisions 114052 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114052 | mmichelson | 2008-04-10 17:02:32 -0500 (Thu, 10 Apr 2008) | 11 lines Merged revisions 114051 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114051 | mmichelson | 2008-04-10 15:59:49 -0500 (Thu, 10 Apr 2008) | 3 lines Fix 1.4 build when LOW_MEMORY is enabled. ........ ................ 2008-04-10 19:59 +0000 [r114047] Mark Michelson * /, channels/chan_sip.c: Merged revisions 114046 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114046 | mmichelson | 2008-04-10 14:58:36 -0500 (Thu, 10 Apr 2008) | 14 lines Merged revisions 114045 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114045 | mmichelson | 2008-04-10 14:55:33 -0500 (Thu, 10 Apr 2008) | 6 lines Be sure that we're not about to set bridgepvt NULL prior to dereferencing it. (closes issue #11775) Reported by: fujin ........ ................ 2008-04-10 19:09 +0000 [r114043] Tilghman Lesher * /, contrib/scripts/astcli: Merged revisions 114042 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114042 | tilghman | 2008-04-10 14:04:29 -0500 (Thu, 10 Apr 2008) | 7 lines The hydra grows yet another head... (closes issue #12401) Reported by: davevg Patches: astcli.diff2 uploaded by davevg (license 209) Tested by: davevg, Corydon76 ........ 2008-04-10 17:27 +0000 [r114037] Jason Parker * /, main/file.c: Merged revisions 114036 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114036 | qwell | 2008-04-10 12:27:16 -0500 (Thu, 10 Apr 2008) | 18 lines Merged revisions 114035 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114035 | qwell | 2008-04-10 12:26:10 -0500 (Thu, 10 Apr 2008) | 10 lines Only try to prefix language if we are not using an absolute path (suffix it otherwise). en/var/lib/asterisk/sounds/blah.gsm is a very silly path. (closes issue #12379) Reported by: kuj Patches: 12379-absolutepath.diff uploaded by qwell (license 4) Tested by: kuj, qwell ........ ................ 2008-04-10 16:00 +0000 [r114023-114034] Joshua Colp * /, apps/app_meetme.c: Merged revisions 114030 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114030 | file | 2008-04-10 12:10:47 -0300 (Thu, 10 Apr 2008) | 14 lines Merged revisions 114029 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114029 | file | 2008-04-10 12:09:04 -0300 (Thu, 10 Apr 2008) | 6 lines Create the directory where name recordings will go if it does not exist. (closes issue #12311) Reported by: rkeene Patches: 12311-mkdir.diff uploaded by qwell (license 4) ........ ................ * apps/app_voicemail.c, /: Merged revisions 114027 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114027 | file | 2008-04-10 11:53:19 -0300 (Thu, 10 Apr 2008) | 6 lines Don't hardcode ru into the digits filename so that languageprefix can work. (closes issue #12404) Reported by: IgorG Patches: voicemail_ru_hardcoded-v1.patch uploaded by IgorG (license 20) ........ * main/rtp.c, channels/chan_unistim.c, /, channels/chan_skinny.c: Merged revisions 114024 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114024 | file | 2008-04-10 10:45:45 -0300 (Thu, 10 Apr 2008) | 4 lines Fix spelling of existent in a few places. (closes issue #12409) Reported by: candlerb ........ * /, channels/chan_sip.c: Merged revisions 114022 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114022 | file | 2008-04-10 10:28:30 -0300 (Thu, 10 Apr 2008) | 14 lines Merged revisions 114021 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114021 | file | 2008-04-10 10:27:11 -0300 (Thu, 10 Apr 2008) | 6 lines Don't add custom URI options if they don't exist OR they are empty. (closes issue #12407) Reported by: homesick Patches: uri_options-1.4.diff uploaded by homesick (license 91) ........ ................ 2008-04-09 22:34 +0000 [r113929-113982] Mark Michelson * /, apps/app_queue.c: Merged revisions 113980 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r113980 | mmichelson | 2008-04-09 17:32:32 -0500 (Wed, 09 Apr 2008) | 8 lines Fix a crash that happened due to accessing free'd memory (closes issue #12396) Reported by: tcalosi Patches: 12396.patch uploaded by putnopvut (license 60) Tested by: tcalosi ........ * /, channels/chan_sip.c: Merged revisions 113928 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113928 | mmichelson | 2008-04-09 15:56:14 -0500 (Wed, 09 Apr 2008) | 16 lines Merged revisions 113927 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113927 | mmichelson | 2008-04-09 15:54:31 -0500 (Wed, 09 Apr 2008) | 8 lines We need to set the persistant_route [sic] parameter for the sip_pvt during the initial INVITE, no matter if we're building the route set from an INVITE request or response. (closes issue #12391) Reported by: benjaminbohlmann Tested by: benjaminbohlmann ........ ................ 2008-04-09 19:02 +0000 [r113876] Tilghman Lesher * cdr/cdr_csv.c, /, configs/cdr.conf.sample: Merged revisions 113875 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113875 | tilghman | 2008-04-09 14:00:40 -0500 (Wed, 09 Apr 2008) | 12 lines Merged revisions 113874 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113874 | tilghman | 2008-04-09 13:57:33 -0500 (Wed, 09 Apr 2008) | 4 lines If the [csv] section does not exist in cdr.conf, then an unload/load sequence is needed to correct the problem. Track whether the load succeeded with a variable, so we can fix this with a simple reload event, instead. ........ ................ 2008-04-09 17:56 +0000 [r113839] Jason Parker * /, contrib/scripts/astcli: Merged revisions 113838 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r113838 | qwell | 2008-04-09 12:56:07 -0500 (Wed, 09 Apr 2008) | 2 lines Fix a small file handle "leak" pointed out by jjshoe on #asterisk. ........ 2008-04-09 17:50 +0000 [r113837] Mark Michelson * main/pbx.c, /: Merged revisions 113836 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r113836 | mmichelson | 2008-04-09 12:48:33 -0500 (Wed, 09 Apr 2008) | 14 lines There was a subtle logical difference between 1.4 and trunk with regards to how timeouts were handled. In 1.4, if the absolute timeout were reached on a call, no matter what the return value of ast_spawn_extension was, the pbx would attempt to go to the 'T' extension or hangup otherwise. The rearrangement of this function in trunk made this check only happen in the case that ast_spawn_extension returned 0. If ast_spawn_extension returned 1, then the fact that the timeout expired resulted in a no-op, and would cause an infinite loop to occur in __ast_pbx_run. This change fixes this problem. Now timeouts will behave as they did in 1.4 (closes issue #11550) Reported by: pj Tested by: putnopvut ........ 2008-04-09 16:53 +0000 [r113786] Joshua Colp * /, channels/chan_iax2.c: Merged revisions 113785 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113785 | file | 2008-04-09 13:52:04 -0300 (Wed, 09 Apr 2008) | 12 lines Merged revisions 113784 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113784 | file | 2008-04-09 13:50:45 -0300 (Wed, 09 Apr 2008) | 4 lines If we receive an AUTHREQ from the remote server and we are unable to reply (for example they have a secret configured, but we do not) then queue a hangup frame on the Asterisk channel. This will cause the channel to hangup and a HANGUP to be sent via IAX2 to the remote side which is the proper thing to do in this scenario. (closes issue #12385) Reported by: viraptor ........ ................ 2008-04-09 14:42 +0000 [r113683] Mark Michelson * /, channels/chan_sip.c: Merged revisions 113682 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113682 | mmichelson | 2008-04-09 09:41:58 -0500 (Wed, 09 Apr 2008) | 17 lines Merged revisions 113681 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113681 | mmichelson | 2008-04-09 09:40:05 -0500 (Wed, 09 Apr 2008) | 9 lines If Asterisk receives a 488 on an INVITE (not a reinvite), then we should not send a BYE. (closes issue #12392) Reported by: fnordian Patches: chan_sip.patch uploaded by fnordian (license 110) with small modification from me ........ ................ 2008-04-09 13:56 +0000 [r113648-113650] Tilghman Lesher * /, contrib/scripts/astcli: Merged revisions 113647 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r113647 | tilghman | 2008-04-09 08:23:44 -0500 (Wed, 09 Apr 2008) | 6 lines Additional enhancements (closes issue #12390) Reported by: tzafrir Patches: astcli_fixes.diff uploaded by tzafrir (license 46) ........ 2008-04-09 01:40 +0000 [r113598] Terry Wilson * /, channels/chan_iax2.c: Merged revisions 113597 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113597 | twilson | 2008-04-08 20:36:58 -0500 (Tue, 08 Apr 2008) | 10 lines Merged revisions 113596 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113596 | twilson | 2008-04-08 20:34:25 -0500 (Tue, 08 Apr 2008) | 2 lines Initialize fr->cacheable to make valgrind happy ........ ................ 2008-04-08 21:34 +0000 [r113560] Tilghman Lesher * /, contrib/scripts/astcli (added): Merged revisions 113559 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r113559 | tilghman | 2008-04-08 16:33:11 -0500 (Tue, 08 Apr 2008) | 6 lines Add commandline tool for doing CLI commands through AMI (instead of using asterisk -rx) (closes issue #12389) Reported by: davevg Patches: astcli uploaded by davevg (license 209) ........ 2008-04-08 18:49 +0000 [r113404-113506] Jason Parker * /, channels/chan_skinny.c: Merged revisions 113505 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113505 | qwell | 2008-04-08 13:49:21 -0500 (Tue, 08 Apr 2008) | 9 lines Merged revisions 113504 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113504 | qwell | 2008-04-08 13:48:55 -0500 (Tue, 08 Apr 2008) | 1 line Add a little more that is required for previously added devices. ........ ................ * /, channels/chan_skinny.c: Merged revisions 113455 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113455 | qwell | 2008-04-08 13:08:35 -0500 (Tue, 08 Apr 2008) | 12 lines Merged revisions 113454 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113454 | qwell | 2008-04-08 13:07:49 -0500 (Tue, 08 Apr 2008) | 4 lines Add support for several new(ish) devices - most notably, 7942/7945, 7962/7965, 7975. Thanks to Greg Oliver for providing me the required information. ........ ................ * main/asterisk.c, /: Merged revisions 113403 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113403 | qwell | 2008-04-08 12:00:55 -0500 (Tue, 08 Apr 2008) | 9 lines Merged revisions 113402 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113402 | qwell | 2008-04-08 11:56:52 -0500 (Tue, 08 Apr 2008) | 1 line Work around some silliness caused by sys/capability.h - this should fix compile errors a number of users have been experiencing. ........ ................ 2008-04-08 16:56 +0000 [r113350-113401] Tilghman Lesher * /, contrib/scripts/astgenkey.8: Merged revisions 113400 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113400 | tilghman | 2008-04-08 11:54:21 -0500 (Tue, 08 Apr 2008) | 14 lines Merged revisions 113399 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113399 | tilghman | 2008-04-08 11:51:28 -0500 (Tue, 08 Apr 2008) | 6 lines Add security note on astgenkey's manpage. (closes issue #12373) Reported by: lmamane Patches: 20080406__bug12373.diff.txt uploaded by Corydon76 (license 14) ........ ................ * /, channels/chan_sip.c: Merged revisions 113349 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113349 | tilghman | 2008-04-08 10:48:58 -0500 (Tue, 08 Apr 2008) | 15 lines Merged revisions 113348 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113348 | tilghman | 2008-04-08 10:39:16 -0500 (Tue, 08 Apr 2008) | 7 lines Move check for still-bridged channels out a little further, to avoid possible deadlocks. (Closes issue #12252) Reported by: callguy Patches: 20080319__bug12252.diff.txt uploaded by Corydon76 (license 14) Tested by: callguy ........ ................ 2008-04-08 15:10 +0000 [r113298-113299] Joshua Colp * /, main/audiohook.c: Merged revisions 113297 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113297 | file | 2008-04-08 12:05:35 -0300 (Tue, 08 Apr 2008) | 12 lines Merged revisions 113296 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113296 | file | 2008-04-08 12:03:43 -0300 (Tue, 08 Apr 2008) | 4 lines If audio suddenly gets fed into one side of a channel after a lapse of frames flush the other factory so that old audio does not remain in the factory causing the sync code to not execute. (closes issue #12296) Reported by: jvandal ........ ................ 2008-04-07 22:17 +0000 [r113246] Tilghman Lesher * /, configs/manager.conf.sample: Merged revisions 113245 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r113245 | tilghman | 2008-04-07 17:16:46 -0500 (Mon, 07 Apr 2008) | 2 lines Additional note ........ 2008-04-07 21:49 +0000 [r113244] Jason Parker * /, configs/manager.conf.sample: Merged revisions 113243 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r113243 | qwell | 2008-04-07 16:49:27 -0500 (Mon, 07 Apr 2008) | 1 line Document 'originate' permission in manager sample config. ........ 2008-04-07 21:36 +0000 [r113242] Jeff Peeler * /, channels/chan_sip.c: Merged revisions 113241 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113241 | jpeeler | 2008-04-07 16:35:48 -0500 (Mon, 07 Apr 2008) | 23 lines Merged revisions 113013 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113013 | jpeeler | 2008-04-07 10:18:10 -0500 (Mon, 07 Apr 2008) | 15 lines Merged revisions 113012 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113012 | jpeeler | 2008-04-07 10:16:44 -0500 (Mon, 07 Apr 2008) | 7 lines (closes issue #12362) (closes issue #12372) Reported by: vinsik Tested by: tecnoxarxa This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one. ........ ................ ................ 2008-04-07 19:10 +0000 [r113174] Jason Parker * /, channels/chan_skinny.c, configs/skinny.conf.sample: Merged revisions 113119 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113119 | qwell | 2008-04-07 13:02:51 -0500 (Mon, 07 Apr 2008) | 16 lines Merged revisions 113118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113118 | qwell | 2008-04-07 13:00:09 -0500 (Mon, 07 Apr 2008) | 8 lines Allow playback with noanswer (and add earlyrtp option). (closes issue #9077) Reported by: pj Patches: earlyrtp.diff uploaded by wedhorn (license 30) Tested by: pj, qwell, DEA, wedhorn ........ ................ 2008-04-07 19:08 +0000 [r113173] Tilghman Lesher * /, funcs/func_strings.c: Merged revisions 113172 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113172 | tilghman | 2008-04-07 14:06:46 -0500 (Mon, 07 Apr 2008) | 11 lines Merged revisions 113117 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113117 | tilghman | 2008-04-07 12:51:49 -0500 (Mon, 07 Apr 2008) | 3 lines Force ast_mktime() to check for DST, since strptime(3) does not. (Closes issue #12374) ........ ................ 2008-04-07 16:13 +0000 [r113067] Mark Michelson * main/channel.c, /: Merged revisions 113066 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113066 | mmichelson | 2008-04-07 11:12:30 -0500 (Mon, 07 Apr 2008) | 21 lines Merged revisions 113065 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113065 | mmichelson | 2008-04-07 11:08:45 -0500 (Mon, 07 Apr 2008) | 13 lines This fix prevents a deadlock that was experienced in chan_local. There was deadlock prevention in place in chan_local, but it would not work in a specific case because the channel was recursively locked. By unlocking the channel prior to calling the generator's generate callback in ast_read_generator_actions(), we prevent the recursive locking, and therefore the deadlock. (closes issue #12307) Reported by: callguy Patches: 12307.patch uploaded by putnopvut (license 60) Tested by: callguy ........ ................ 2008-04-07 15:28 +0000 [r113042] Jeff Peeler * /, channels/chan_sip.c: Merged revisions 113013 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113013 | jpeeler | 2008-04-07 10:18:10 -0500 (Mon, 07 Apr 2008) | 15 lines Merged revisions 113012 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113012 | jpeeler | 2008-04-07 10:16:44 -0500 (Mon, 07 Apr 2008) | 7 lines (closes issue #12362) (closes issue #12372) Reported by: vinsik Tested by: tecnoxarxa This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one. ........ ................ 2008-04-05 13:30 +0000 [r112973-112975] Tilghman Lesher * /, res/res_agi.c: Merged revisions 112972 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r112972 | tilghman | 2008-04-05 08:24:12 -0500 (Sat, 05 Apr 2008) | 6 lines AsyncAGI should not close the manager session on error. (closes issue #12370) Reported by: srt Patches: asterisk-12370.diff uploaded by srt (license 378) ........ 2008-04-04 19:30 +0000 [r112786-112822] Philippe Sultan * /, channels/chan_gtalk.c: Merged revisions 112821 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r112821 | phsultan | 2008-04-04 21:28:49 +0200 (Fri, 04 Apr 2008) | 9 lines Merged revisions 112820 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112820 | phsultan | 2008-04-04 21:26:15 +0200 (Fri, 04 Apr 2008) | 1 line Free newly allocated channel before returning ........ ................ * /, channels/chan_gtalk.c: Merged revisions 112785 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r112785 | phsultan | 2008-04-04 19:32:46 +0200 (Fri, 04 Apr 2008) | 15 lines Merged revisions 112766 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112766 | phsultan | 2008-04-04 19:16:59 +0200 (Fri, 04 Apr 2008) | 7 lines Prevent call connections when codecs don't match. (closes issue #10604) Reported by: keepitcool Patches: branch-1.4-10604-2.diff uploaded by phsultan (license 73) Tested by: phsultan ........ ................ 2008-04-04 01:08 +0000 [r112715] Dwayne M. Hubbard * main/asterisk.c, /: Merged revisions 112653,112656,112714 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r112653 | dhubbard | 2008-04-03 17:13:11 -0500 (Thu, 03 Apr 2008) | 1 line add a Zaptel timer check to verify the timer is responding when Zaptel support is compiled into Asterisk and Zaptel drivers are loaded. This will help people not waste their valuable time debugging side effects. ........ r112656 | dhubbard | 2008-04-03 17:19:43 -0500 (Thu, 03 Apr 2008) | 1 line satisfy buildbot ........ r112714 | dhubbard | 2008-04-03 19:57:33 -0500 (Thu, 03 Apr 2008) | 1 line sleep long enough for the zaptel timer error message to display before exit ........ 2008-04-04 00:54 +0000 [r112713] Joshua Colp * /, main/Makefile: Merged revisions 112712 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r112712 | file | 2008-04-03 21:53:19 -0300 (Thu, 03 Apr 2008) | 10 lines Merged revisions 112711 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112711 | file | 2008-04-03 21:52:36 -0300 (Thu, 03 Apr 2008) | 2 lines Pass in the path to Zaptel for systems that install Zaptel headers in a separate location. ........ ................ 2008-04-03 14:42 +0000 [r112601] Mark Michelson * channels/chan_zap.c, /: Merged revisions 112600 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r112600 | mmichelson | 2008-04-03 09:35:47 -0500 (Thu, 03 Apr 2008) | 17 lines Merged revisions 112599 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112599 | mmichelson | 2008-04-03 09:32:20 -0500 (Thu, 03 Apr 2008) | 9 lines Fix the testing of the "res" variable so that it is more logically correct and makes the correct warning and debug messages print. (closes issue #12361) Reported by: one47 Patches: chan_zap_deferred_digit.patch uploaded by one47 (license 23) ........ ................ 2008-04-02 17:37 +0000 [r112470] Mark Michelson * main/manager.c, /: Merged revisions 112469 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r112469 | mmichelson | 2008-04-02 12:36:49 -0500 (Wed, 02 Apr 2008) | 21 lines Merged revisions 112468 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112468 | mmichelson | 2008-04-02 12:36:04 -0500 (Wed, 02 Apr 2008) | 13 lines Fix a race condition in the manager. It is possible that a new manager event could be appended during a brief time when the manager is not waiting for input. If an event comes during this period, we need to set an indicator that there is an event pending so that the manager doesn't attempt to wait forever for an event that already happened. (closes issue #12354) Reported by: bamby Patches: manager_race_condition.diff uploaded by bamby (license 430) (comments added by me) ........ ................ 2008-04-02 15:27 +0000 [r112436] Joshua Colp * /, channels/chan_sip.c: Merged revisions 112431 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r112431 | file | 2008-04-02 12:26:51 -0300 (Wed, 02 Apr 2008) | 7 lines Since the SIP request structure gets reused multiple times with TCP handling we have to clear the debug state or else we will keep spitting out debug even after it has been turned off. (closes issue #12169) Reported by: pj Patches: 12169-debugoff-2.diff uploaded by qwell (license 4) Tested by: pj ........ 2008-04-02 14:33 +0000 [r112395] Mark Michelson * /, apps/app_queue.c: Merged revisions 112394 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r112394 | mmichelson | 2008-04-02 09:32:43 -0500 (Wed, 02 Apr 2008) | 14 lines Merged revisions 112393 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112393 | mmichelson | 2008-04-02 09:32:00 -0500 (Wed, 02 Apr 2008) | 6 lines Ensure that there is no timeout if none is specified. (closes issue #12349) Reported by: johnlange ........ ................ 2008-04-01 22:48 +0000 [r112359] Steve Murphy * main/pbx.c, /: Merged revisions 112357 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r112357 | murf | 2008-04-01 16:45:10 -0600 (Tue, 01 Apr 2008) | 1 line Bumped across another test set for the new exten pattern matcher, which revealed a problem with the CANMATCH/MATCHMORE modes. Direct matches were getting in the way. Fixed. ........ 2008-04-01 20:20 +0000 [r112299] Steve Murphy * main/pbx.c, /: Merged revisions 112289 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r112289 | murf | 2008-04-01 14:02:19 -0600 (Tue, 01 Apr 2008) | 21 lines (closes issue #12298) Reported by: falves11 Patches: 12298.patch1 uploaded by murf (license 17) Tested by: murf I have hopes that the changes made over the last few days will finalize and solidify this code. While there are bound to be small tweaks still needed, I feel that the job (at last) is somewhat completed. Finally, I had a chance to comprehend how the scoring of extension patterns was done in the previous version, and I've come very close to using the exact same criteria in the new pattern matching code. The left-right sorting is now replicated in the trie structure itself, such that the first match found will the 'best' match. Compared the results against 1.4 for several extensions. Replicated falves11's setup and it works. Used some devious patterns provided by jsmith, supplemented with a few of my own. Looks good. ........ 2008-04-01 18:09 +0000 [r112211] Joshua Colp * main/rtp.c, /: Merged revisions 112210 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r112210 | file | 2008-04-01 15:06:13 -0300 (Tue, 01 Apr 2008) | 12 lines Merged revisions 112209 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112209 | file | 2008-04-01 15:02:43 -0300 (Tue, 01 Apr 2008) | 4 lines Disable Packet2Packet bridging when we need to feed DTMF frames into the core. Some implementations do not like how we switch between things. (closes issue #12212) Reported by: bamby ........ ................ 2008-04-01 17:52 +0000 [r112170-112206] Joshua Colp * /, channels/chan_sip.c: Merged revisions 112205 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r112205 | file | 2008-04-01 14:48:52 -0300 (Tue, 01 Apr 2008) | 12 lines Merged revisions 112204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112204 | file | 2008-04-01 14:43:46 -0300 (Tue, 01 Apr 2008) | 4 lines Do not pass audio until the remote side has indicated they are providing early media, or if the channel has been answered. (closes issue #11823) Reported by: SDamm ........ ................ 2008-04-01 17:25 +0000 [r112157] Mark Michelson * main/dns.c, /: Merged revisions 112148 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r112148 | mmichelson | 2008-04-01 12:23:19 -0500 (Tue, 01 Apr 2008) | 18 lines Merged revisions 112138 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112138 | mmichelson | 2008-04-01 12:21:21 -0500 (Tue, 01 Apr 2008) | 10 lines Initialize the __res_state structure used for dns purposes to all 0's prior to using it. This is due to valgrind's complaints on issue #12284 as well as an excerpt found in "Description" portion of the online man page found here: http://www.iti.cs.tu-bs.de/cgi-bin/UNIXhelp/man-cgi?res_nquery+3RESOLV (pertains to issue #12284 but does not necessarily close it) ........ ................ 2008-04-01 16:57 +0000 [r112127] Joshua Colp * include/asterisk/slinfactory.h, /, main/slinfactory.c: Merged revisions 112126 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r112126 | file | 2008-04-01 13:50:37 -0300 (Tue, 01 Apr 2008) | 13 lines Merged revisions 112125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112125 | file | 2008-04-01 13:45:14 -0300 (Tue, 01 Apr 2008) | 5 lines Ensure that we do not exceed the hold's maximum size with a single frame. (closes issue #12047) Reported by: fabianoheringer Tested by: fabianoheringer ........ ................ 2008-03-31 22:17 +0000 [r112070-112072] Jason Parker * apps/app_voicemail.c, /: Merged revisions 112069 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r112069 | qwell | 2008-03-31 16:48:30 -0500 (Mon, 31 Mar 2008) | 13 lines Merged revisions 112068 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112068 | qwell | 2008-03-31 16:48:05 -0500 (Mon, 31 Mar 2008) | 5 lines Fix a silly infinite loop when choosing an invalid option. (closes issue #12315) Reported by: jmls ........ ................ 2008-03-31 21:03 +0000 [r112034-112036] Terry Wilson * /, main/http.c: Merged revisions 112033 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r112033 | twilson | 2008-03-31 15:45:05 -0500 (Mon, 31 Mar 2008) | 2 lines Handle blank prefix= in http.conf ........ 2008-03-31 17:15 +0000 [r111997-111999] Russell Bryant * Makefile, /: Merged revisions 111998 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r111998 | russell | 2008-03-31 12:14:58 -0500 (Mon, 31 Mar 2008) | 7 lines Ensure configure gets run on a clean checkout. (closes issue #12197) Reported by: juggie Patches: 12197.diff uploaded by juggie (license 24) ........ 2008-03-31 14:22 +0000 [r111962] Joshua Colp * res/res_config_sqlite.c, /: Merged revisions 111961 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r111961 | file | 2008-03-31 11:20:39 -0300 (Mon, 31 Mar 2008) | 4 lines Initialize all these here tmp pointers at declaration. They confused some compilers a wee bit. (closes issue #12333) Reported by: ovi ........ 2008-03-29 Russell Bryant * Asterisk 1.6.0-beta7.1 released. Asterisk 1.6.0-beta7 was tagged against trunk, instead of the 1.6.0 branch. 2008-03-28 21:46 +0000 [r111858] Jason Parker * codecs/gsm/inc/private.h, /: Merged revisions 111857 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111857 | qwell | 2008-03-28 16:46:02 -0500 (Fri, 28 Mar 2008) | 20 lines Merged revisions 111856 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111856 | qwell | 2008-03-28 16:45:35 -0500 (Fri, 28 Mar 2008) | 12 lines Allow gsm to compile correctly on x86 with gcc4 optimizations. (closes issue #11243) Reported by: whiskerp Patches: 11243-maybe-asm.diff uploaded by qwell (license 4) Tested by: Seggy (IRC) Note: While I did write this patch, I would not have found this if fossil had not reported and fixed issue #12253. A huge thanks to him for helping to (indirectly) find the problem here. ........ ................ 2008-03-28 19:11 +0000 [r111722-111776] Jason Parker * /, channels/chan_skinny.c: Merged revisions 111721 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111721 | qwell | 2008-03-28 12:57:12 -0500 (Fri, 28 Mar 2008) | 9 lines Merged revisions 111720 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111720 | qwell | 2008-03-28 12:55:05 -0500 (Fri, 28 Mar 2008) | 1 line Remove unimplemented softkeys. Prompted by issue #12325. ........ ................ 2008-03-28 16:21 +0000 [r111660] Jason Parker * /, formats/format_wav_gsm.c: Merged revisions 111659 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111659 | qwell | 2008-03-28 11:20:59 -0500 (Fri, 28 Mar 2008) | 16 lines Merged revisions 111658 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111658 | qwell | 2008-03-28 11:19:56 -0500 (Fri, 28 Mar 2008) | 8 lines The file size of WAV49 does not need to be an even number. (closes issue #12128) Reported by: mdu113 Patches: 12128-noevenlength.diff uploaded by qwell (license 4) Tested by: qwell, mdu113 ........ ................ 2008-03-28 14:43 +0000 [r111607-111608] Tilghman Lesher * doc/valgrind.txt, /: Merged revisions 111606 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111606 | tilghman | 2008-03-28 09:37:28 -0500 (Fri, 28 Mar 2008) | 11 lines Merged revisions 111605 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111605 | tilghman | 2008-03-28 09:35:45 -0500 (Fri, 28 Mar 2008) | 3 lines Update debugging text, since Valgrind eliminated the --log-file-exactly option. (Closes issue #12320) ........ ................ 2008-03-28 00:56 +0000 [r111566] Joshua Colp * /, apps/app_queue.c: Merged revisions 111565 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r111565 | file | 2008-03-27 21:55:47 -0300 (Thu, 27 Mar 2008) | 2 lines Forgetting to unregister a manager action is bad, mmmk? ........ 2008-03-28 00:17 +0000 [r111534] Mark Michelson * /, apps/app_queue.c: Merged revisions 111533 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r111533 | mmichelson | 2008-03-27 19:12:52 -0500 (Thu, 27 Mar 2008) | 10 lines Fix a crash that would happen when attempting to unload the app_queue module. The problem was that when the refcount on the queue hit 0, the destructor was called, and inside the destructor, another function was called which would increase the refcount back to 1 again and then decrease it again back to 0 for every member in the queue. This meant that the destructor was being recursively called, leading to a double free of the queue. This is now fixed by making sure to unlink the queue from the queues container prior to the final unref of the queue. ........ 2008-03-27 21:28 +0000 [r111498] Steve Murphy * main/pbx.c, /: Merged revisions 111497 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r111497 | murf | 2008-03-27 15:25:55 -0600 (Thu, 27 Mar 2008) | 1 line comment cleanup and iron out a really dumb mistake in handling the '.'-wildcard in the new exten pattern matcher. ........ 2008-03-27 19:30 +0000 [r111444] Tilghman Lesher * /, main/acl.c: Merged revisions 111443 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111443 | tilghman | 2008-03-27 14:26:45 -0500 (Thu, 27 Mar 2008) | 14 lines Merged revisions 111442 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111442 | tilghman | 2008-03-27 14:23:12 -0500 (Thu, 27 Mar 2008) | 6 lines For FreeBSD, at least, the ifa_addr element could be NULL. (closes issue #12300) Reported by: festr Patches: acl.c.patch uploaded by festr (license 443) ........ ................ 2008-03-27 13:42 +0000 [r111361-111411] Steve Murphy * apps/app_playback.c, main/pbx.c, /: Merged revisions 111410 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111410 | murf | 2008-03-27 07:29:41 -0600 (Thu, 27 Mar 2008) | 17 lines Merged revisions 111391 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111391 | murf | 2008-03-27 07:03:28 -0600 (Thu, 27 Mar 2008) | 9 lines These small documentation updates made in response to a query in asterisk-users, where a user was using Playback, but needed the features of Background, and had no idea that Background existed, or that it might provide the features he needed. I thought the best way to avert these kinds of queries was to provide "See Also" references in all three of "Background", "Playback", "WaitExten". Perhaps a project to do this with all related apps is in order. ........ ................ * res/ael/pval.c, /, include/asterisk/pval.h, pbx/pbx_ael.c, include/asterisk/ael_structs.h: Merged revisions 111360 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111360 | murf | 2008-03-26 22:47:12 -0600 (Wed, 26 Mar 2008) | 23 lines Merged revisions 111341 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111341 | murf | 2008-03-26 21:21:05 -0600 (Wed, 26 Mar 2008) | 15 lines (closes issue #12302) Reported by: pj Tested by: murf These changes will set a channel variable ~~EXTEN~~ just before generating code for a switch, with the value of ${EXTEN}. The exten is marked as having a switch, and ever after that, till the end of the exten, we substitute any ${EXTEN} with ${~~EXTEN~~} instead in application arguments; (and the ${EXTEN: also). The reason for this, is that because switches are coded using separate extensions to provide pattern matching, and jumping to/from these switch extensions messes up the ${EXTEN} value, which blows the minds of users. ........ ................ 2008-03-27 00:36 +0000 [r111247-111339] Jason Parker * main/frame.c, /: Merged revisions 111285 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111285 | qwell | 2008-03-26 19:25:56 -0500 (Wed, 26 Mar 2008) | 9 lines Merged revisions 111280 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111280 | qwell | 2008-03-26 19:25:13 -0500 (Wed, 26 Mar 2008) | 1 line Put this flag back so we don't change the API. ........ ................ * main/frame.c, /: Merged revisions 111246 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111246 | qwell | 2008-03-26 18:27:33 -0500 (Wed, 26 Mar 2008) | 17 lines Merged revisions 111245 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111245 | qwell | 2008-03-26 18:26:33 -0500 (Wed, 26 Mar 2008) | 9 lines Remove excessive smoother optimization that was causing audio glitches (small "pops") after (about 200ms later) an "incorrectly" sized frame was received. While it would be very nice to keep this as optimized as possible, it makes no sense for the smoother to be dropping random bits of audio like this. Isn't that the whole point of a smoother? Closes issue #12093. ........ ................ 2008-03-26 19:57 +0000 [r111131] Joshua Colp * contrib/scripts/autosupport, /: Merged revisions 111130 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111130 | file | 2008-03-26 16:56:40 -0300 (Wed, 26 Mar 2008) | 14 lines Merged revisions 111129 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111129 | file | 2008-03-26 16:55:08 -0300 (Wed, 26 Mar 2008) | 6 lines Update autosupport script. (closes issue #12310) Reported by: angler Patches: autosupport.diff uploaded by angler (license 106) ........ ................ 2008-03-26 19:53 +0000 [r111128] Kevin P. Fleming * /, UPGRADE.txt: Merged revisions 111127 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111127 | kpfleming | 2008-03-26 14:52:27 -0500 (Wed, 26 Mar 2008) | 18 lines Merged revisions 111126 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r111126 | kpfleming | 2008-03-26 14:51:24 -0500 (Wed, 26 Mar 2008) | 10 lines Merged revisions 111125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r111125 | kpfleming | 2008-03-26 14:49:30 -0500 (Wed, 26 Mar 2008) | 2 lines update UPGRADE notes to document usage of the script ........ ................ ................ 2008-03-26 19:41 +0000 [r111124] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 111123 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111123 | mmichelson | 2008-03-26 14:39:23 -0500 (Wed, 26 Mar 2008) | 12 lines Merged revisions 111121 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111121 | mmichelson | 2008-03-26 14:37:36 -0500 (Wed, 26 Mar 2008) | 4 lines This code change is made just for clarification. It does exactly the same thing as before. It just doesn't look as wrong. ........ ................ 2008-03-26 19:27 +0000 [r111072] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 111067 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111067 | mmichelson | 2008-03-26 14:26:23 -0500 (Wed, 26 Mar 2008) | 17 lines Merged revisions 111049 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111049 | mmichelson | 2008-03-26 14:22:16 -0500 (Wed, 26 Mar 2008) | 9 lines Add a lock to the vm_state structure and use the lock around mail_open calls to prevent concurrent access of the same mailstream. This, along with trunk's ability to configure TCP timeouts for IMAP storage will help to prevent crashes and hangs when using voicemail with IMAP storage. (closes issue #10487) Reported by: ewilhelmsen ........ ................ 2008-03-26 19:08 +0000 [r111026] Kevin P. Fleming * codecs/ilbc, /, contrib/scripts/get_ilbc_source.sh (added): Merged revisions 111025 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111025 | kpfleming | 2008-03-26 14:08:00 -0500 (Wed, 26 Mar 2008) | 18 lines Merged revisions 111024 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r111024 | kpfleming | 2008-03-26 14:06:56 -0500 (Wed, 26 Mar 2008) | 10 lines Merged revisions 111019 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r111019 | kpfleming | 2008-03-26 13:58:37 -0500 (Wed, 26 Mar 2008) | 2 lines add a script to make getting the iLBC source code simple for end users ........ ................ ................ 2008-03-26 19:06 +0000 [r111018-111023] Joshua Colp * /, channels/chan_sip.c: Merged revisions 111021 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111021 | file | 2008-03-26 16:05:42 -0300 (Wed, 26 Mar 2008) | 12 lines Merged revisions 111020 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111020 | file | 2008-03-26 16:04:35 -0300 (Wed, 26 Mar 2008) | 4 lines If we are requested to authenticate a reinvite make sure that it contains T38 SDP if need be. (closes issue #11995) Reported by: fall ........ ................ * /, channels/chan_iax2.c: Merged revisions 111017 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111017 | file | 2008-03-26 15:42:52 -0300 (Wed, 26 Mar 2008) | 12 lines Merged revisions 110628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4 lines Add an option (transmit_silence) which transmits silence during both Record() and DTMF generation. The reason this is an option is that in order to transmit silence we have to setup a translation path. This may not be needed/wanted in all cases. (closes issue #10058) Reported by: tracinet ........ ................ 2008-03-26 17:44 +0000 [r110964] Kevin P. Fleming * /, UPGRADE.txt: Merged revisions 110963 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110963 | kpfleming | 2008-03-26 12:44:09 -0500 (Wed, 26 Mar 2008) | 10 lines Merged revisions 110962 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110962 | kpfleming | 2008-03-26 12:43:02 -0500 (Wed, 26 Mar 2008) | 2 lines add note that the user will need to enable codec_ilbc to get it to build ........ ................ 2008-03-26 17:35 +0000 [r110959] Donny Kavanagh * /, doc/snmp.txt: Merged revisions 110911 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110911 | juggie | 2008-03-26 13:24:54 -0400 (Wed, 26 Mar 2008) | 8 lines update documentation to reflect the changes in the way configure detects net-snmp. (closes issue #12067) Reported by: juggie Patches: 12067_snmp_doc.patch uploaded by juggie (license 24) Tested by: juggie ........ 2008-03-26 17:15 +0000 [r110882] Kevin P. Fleming * codecs/ilbc/constants.h (removed), codecs/ilbc/iLBC_decode.h (removed), codecs/ilbc/iCBSearch.c (removed), codecs/Makefile, codecs/ilbc/filter.c (removed), codecs/ilbc/hpInput.c (removed), codecs/ilbc/gainquant.c (removed), codecs/ilbc/hpOutput.c (removed), codecs/ilbc/iCBSearch.h (removed), codecs/ilbc/filter.h (removed), codecs/ilbc/hpInput.h (removed), codecs/ilbc/gainquant.h (removed), codecs/ilbc/LPCencode.c (removed), codecs/ilbc/hpOutput.h (removed), codecs/ilbc/StateSearchW.c (removed), codecs/codec_ilbc.c, codecs/ilbc/LPCencode.h (removed), codecs/ilbc/iCBConstruct.c (removed), codecs/ilbc/StateSearchW.h (removed), codecs/ilbc/syntFilter.c (removed), /, codecs/ilbc/iCBConstruct.h (removed), codecs/ilbc/syntFilter.h (removed), codecs/ilbc/packing.c (removed), codecs/ilbc/StateConstructW.c (removed), codecs/ilbc/packing.h (removed), codecs/ilbc/libilbc.vcproj (removed), codecs/ilbc/StateConstructW.h (removed), codecs/ilbc/LPCdecode.c (removed), codecs/ilbc/getCBvec.c (removed), codecs/ilbc/enhancer.c (removed), codecs/ilbc/lsf.c (removed), codecs/ilbc/iLBC_encode.c (removed), codecs/ilbc/getCBvec.h (removed), codecs/ilbc/LPCdecode.h (removed), codecs/ilbc/enhancer.h (removed), codecs/ilbc/FrameClassify.c (removed), codecs/ilbc/iLBC_define.h (removed), codecs/ilbc/lsf.h (removed), codecs/ilbc/iLBC_encode.h (removed), codecs/ilbc/FrameClassify.h (removed), codecs/ilbc/helpfun.c (removed), codecs/ilbc/doCPLC.c (removed), codecs/ilbc/anaFilter.c (removed), codecs/ilbc/helpfun.h (removed), codecs/ilbc/createCB.c (removed), codecs/ilbc/doCPLC.h (removed), codecs/ilbc/anaFilter.h (removed), UPGRADE.txt, codecs/ilbc/constants.c (removed), codecs/ilbc/iLBC_decode.c (removed), codecs/ilbc/createCB.h (removed), CHANGES: Merged revisions 110881 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110881 | kpfleming | 2008-03-26 10:10:28 -0700 (Wed, 26 Mar 2008) | 18 lines Merged revisions 110880 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r110880 | kpfleming | 2008-03-26 09:42:35 -0700 (Wed, 26 Mar 2008) | 10 lines Merged revisions 110869 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r110869 | kpfleming | 2008-03-26 08:53:46 -0700 (Wed, 26 Mar 2008) | 2 lines due to licensing restrictions, we cannot distribute the source code for iLBC encoding and decoding... so remove it, and add instructions on how the user can obtain it themselves ........ ................ ................ 2008-03-26 15:33 +0000 [r110866-110868] Joshua Colp * /: Merged revisions 110726 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110726 | jpeeler | 2008-03-25 17:02:57 -0300 (Tue, 25 Mar 2008) | 2 lines This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one. ........ 2008-03-26 00:03 +0000 [r110832] Mark Michelson * main/manager.c, /: Merged revisions 110831 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110831 | mmichelson | 2008-03-25 19:02:31 -0500 (Tue, 25 Mar 2008) | 6 lines This ensures that the manager interface is not enabled by default. Prior to this change, it was possible to start Asterisk with the manager interface enabled, then either comment out the enabled option or make manager.conf unopenable and the manager interface would still be enabled. ........ 2008-03-25 22:52 +0000 [r110781] Jason Parker * cdr/cdr_custom.c, /: Merged revisions 110780 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110780 | qwell | 2008-03-25 17:51:55 -0500 (Tue, 25 Mar 2008) | 14 lines Merged revisions 110779 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110779 | qwell | 2008-03-25 17:51:17 -0500 (Tue, 25 Mar 2008) | 6 lines Make file access in cdr_custom similar to cdr_csv. Fixes issue #12268. Patch borrowed from r82344 ........ ................ 2008-03-25 22:11 +0000 [r110778] Jeff Peeler * channels/chan_sip.c: This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one. 2008-03-25 17:47 +0000 [r110690-110692] Tilghman Lesher * configs/extensions.conf.sample, /, configs/voicemail.conf.sample: Merged revisions 110691 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110691 | tilghman | 2008-03-25 12:46:34 -0500 (Tue, 25 Mar 2008) | 6 lines Update sample configurations to make virtual hosting more obvious. (closes issue #11969) Reported by: pprindeville Patches: acme-virtualpbx.1.6.patch uploaded by pprindeville (license 347) ........ * configs/extensions.conf.sample, /: Merged revisions 110689 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110689 | tilghman | 2008-03-25 12:40:28 -0500 (Tue, 25 Mar 2008) | 6 lines Update the sample configuration, to use Macro less (since it's now deprecated). (closes issue #12293) Reported by: pprindeville Patches: bugid-0012293.1.6.patch uploaded by pprindeville (license 347) ........ 2008-03-25 15:43 +0000 [r110637-110638] Mark Michelson * channels/chan_sip.c: Oops. * /, channels/chan_sip.c: Merged revisions 110636 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110636 | mmichelson | 2008-03-25 10:41:33 -0500 (Tue, 25 Mar 2008) | 15 lines Merged revisions 110635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110635 | mmichelson | 2008-03-25 10:40:33 -0500 (Tue, 25 Mar 2008) | 7 lines When reverting a commit, I accidentally left in this bit which was an experiment to see what would happen. It passed the compile test, and I didn't notice I had left this change in too. So this is a revert of a revert...sort of. ........ ................ 2008-03-25 15:39 +0000 [r110630-110634] Joshua Colp * include/asterisk/options.h, main/asterisk.c, Makefile, /, main/app.c: Merged revisions 110629 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110629 | file | 2008-03-25 11:39:45 -0300 (Tue, 25 Mar 2008) | 12 lines Merged revisions 110628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4 lines Add an option (transmit_silence) which transmits silence during both Record() and DTMF generation. The reason this is an option is that in order to transmit silence we have to setup a translation path. This may not be needed/wanted in all cases. (closes issue #10058) Reported by: tracinet ........ ................ 2008-03-24 20:14 +0000 [r110620-110622] Mark Michelson * /, channels/chan_sip.c: Merged revisions 110619 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110619 | mmichelson | 2008-03-24 14:19:37 -0500 (Mon, 24 Mar 2008) | 23 lines Merged revisions 110618 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110618 | mmichelson | 2008-03-24 14:17:41 -0500 (Mon, 24 Mar 2008) | 15 lines This is a revert for revision 108288. The reason is that that revision was not for an actual bug fix per se, and so it really should not have been in 1.4 in the first place. Plus, people who compile with DO_CRASH are more likely to encounter a crash due to this change. While I think the usage of DO_CRASH in ast_sched_del is a bit absurd, this sort of change is beyond the scope of 1.4 and should be done instead in a developer branch based on trunk so that all scheduler functions are fixed at once. I also am reverting the change to trunk and 1.6 since they also suffer from the DO_CRASH potential. (closes issue #12272) Reported by: qq12345 ........ ................ 2008-03-24 17:36 +0000 [r110616] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 110615 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110615 | russell | 2008-03-24 12:36:04 -0500 (Mon, 24 Mar 2008) | 10 lines Merged revisions 110614 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110614 | russell | 2008-03-24 12:34:56 -0500 (Mon, 24 Mar 2008) | 2 lines Turn a NOTICE into a DEBUG message. ........ ................ 2008-03-24 15:29 +0000 [r110611] Joshua Colp * /, channels/chan_sip.c: Merged revisions 110610 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110610 | file | 2008-03-24 12:28:25 -0300 (Mon, 24 Mar 2008) | 6 lines Only print out the set_address_from_contact host verbose message if debugging is enabled on the dialog. (closes issue #12280) Reported by: rjain Patches: chan_sip.c.diff uploaded by rjain (license 226) ........ 2008-03-21 21:52 +0000 [r110579] Jason Parker * /, sounds/Makefile: Merged revisions 110578 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110578 | qwell | 2008-03-21 16:52:06 -0500 (Fri, 21 Mar 2008) | 1 line Update to 1.4.11 core sounds. ........ 2008-03-21 15:25 +0000 [r110501] Russell Bryant * /, configs/sip.conf.sample, CHANGES: Merged revisions 110499 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110499 | russell | 2008-03-21 10:24:43 -0500 (Fri, 21 Mar 2008) | 3 lines Note that the TCP and TLS support is currently considered experimental and is subject to change while we work out the remaining issues. ........ 2008-03-21 14:36 +0000 [r110476] Jason Parker * /, codecs/gsm/Makefile: Merged revisions 110475 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110475 | qwell | 2008-03-21 09:36:17 -0500 (Fri, 21 Mar 2008) | 15 lines Merged revisions 110474 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110474 | qwell | 2008-03-21 09:32:52 -0500 (Fri, 21 Mar 2008) | 7 lines Don't attempt to do optimizations of gsm on mips platforms either. (closes issue #12270) Reported by: zandbelt Patches: 026-gsm-mips.patch uploaded by zandbelt (license 33) ........ ................ 2008-03-20 23:14 +0000 [r110304-110397] Russell Bryant * main/autoservice.c, /: Merged revisions 110396 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110396 | russell | 2008-03-20 18:14:13 -0500 (Thu, 20 Mar 2008) | 17 lines Merged revisions 110395 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110395 | russell | 2008-03-20 18:13:56 -0500 (Thu, 20 Mar 2008) | 9 lines Shorten the ast_waitfor() timeout from 500 ms to 50 ms in the autoservice thread. This really should not make a difference except in very rare cases. That case would be that all of the channels in autoservice are not generating any frames. In that case, this change reduces the potential amount of time that a thread waits in ast_autoservice_stop() for the autoservice thread to wrap back around to the beginning of its loop. (closes issue #12266, reported by dimas) ........ ................ * codecs/codec_g722.c, /: Merged revisions 110339 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110339 | russell | 2008-03-20 17:02:20 -0500 (Thu, 20 Mar 2008) | 3 lines Use the correct buffer for g722tolin16_sample. This shouldn't have caused any problems, but Qwell noticed the typo here. ........ * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions 110337 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110337 | russell | 2008-03-20 16:55:50 -0500 (Thu, 20 Mar 2008) | 22 lines Merged revisions 110336 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r110336 | russell | 2008-03-20 16:54:58 -0500 (Thu, 20 Mar 2008) | 14 lines Merged revisions 110335 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r110335 | russell | 2008-03-20 16:53:27 -0500 (Thu, 20 Mar 2008) | 6 lines Fix some very broken code that was introduced in 1.2.26 as a part of the security fix. The dnsmgr is not appropriate here. The dnsmgr takes a pointer to an address structure that a background thread continuously updates. However, in these cases, a stack variable was passed. That means that the dnsmgr thread would be continuously writing to bogus memory. ........ ................ ................ * /, main/file.c: Merged revisions 110303 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110303 | russell | 2008-03-20 15:08:26 -0500 (Thu, 20 Mar 2008) | 8 lines Fix a bug when using zaptel timing for playing back files that have a sample rate other than 8 kHz. The issue here is that format modules give a "whennext" sample value, which is used to calculate when to set a timer for to retrieve the next frame. However, the zaptel timer operates on 8 kHz samples, so this must be taken into account. (another part of issue #12164, reported by milazzo and jsmith, patch by me) ........ 2008-03-20 18:02 +0000 [r110273] Mark Michelson * main/dial.c, /: Merged revisions 110272 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110272 | mmichelson | 2008-03-20 13:01:36 -0500 (Thu, 20 Mar 2008) | 3 lines Add missing unlock ........ 2008-03-20 17:45 +0000 [r110269-110271] Russell Bryant * main/channel.c, /, res/res_musiconhold.c: Merged revisions 110268 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110268 | russell | 2008-03-20 12:41:22 -0500 (Thu, 20 Mar 2008) | 27 lines Add some fixes that I made in regards to wideband codec handling to get G.722 music on hold working for me. (issue #12164, reported by milazzo and jsmith, patches by me) res/res_musiconhold.c: - I moved a single line so that the sample queue update happened before ast_write(). The reason that this was a bug is that the G.722 frame originally says it has 320 samples in it (which is correct). However, when the frame is written to a channel that uses RTP, main/rtp.c modifies the frame to cut the number of samples in half before it sends it on the wire. This is to account for the stupid incorrect G.722 spec that makes it so we have to lie about the number of samples with RTP. I should probably go and re-work the RTP code so it doesn't modify the frame so that a bug like this won't happen in the future. However, this change to MOH is harmless. main/channel.c: - I made two fixes in regards to generator timing. Generators use samples for timing. However, this code assumed 8 kHz samples. In one case, it was a hard coded 160 samples, that is now written as the sample rate / 50. The other place was dealing with timing a generator based on frames coming from the other direction. However, that would have only worked if the sample rates for the formats in both directions were the same. The code now takes into account that the sample rates may differ, and scales the generator samples accordingly. ........ 2008-03-19 23:00 +0000 [r110165] Russell Bryant * /, apps/app_meetme.c: Merged revisions 110164 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110164 | russell | 2008-03-19 17:58:33 -0500 (Wed, 19 Mar 2008) | 13 lines Merged revisions 110163 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110163 | russell | 2008-03-19 17:57:59 -0500 (Wed, 19 Mar 2008) | 5 lines Fix a bug where when calls on the trunk side hang up while on hold, the state is not properly reflected. (closes issue #11990, reported by anakaoka, patched by me) ........ ................ 2008-03-19 21:06 +0000 [r110088] Jeff Peeler * /: marking rev 110087 from trunk as not applying 2008-03-19 20:37 +0000 [r110085] Mark Michelson * apps/app_chanspy.c, /: Merged revisions 110084 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110084 | mmichelson | 2008-03-19 15:34:13 -0500 (Wed, 19 Mar 2008) | 12 lines Merged revisions 110083 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110083 | mmichelson | 2008-03-19 15:33:03 -0500 (Wed, 19 Mar 2008) | 4 lines Add a missing unlock in the case that memory allocation fails in app_chanspy. Thanks to Russell for confirming that this was an issue. ........ ................ 2008-03-19 19:14 +0000 [r110037] Joshua Colp * /, res/res_musiconhold.c: Merged revisions 110036 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110036 | file | 2008-03-19 16:13:39 -0300 (Wed, 19 Mar 2008) | 12 lines Merged revisions 110035 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110035 | file | 2008-03-19 16:11:33 -0300 (Wed, 19 Mar 2008) | 4 lines Add sanity checking for position resuming. We *have* to make sure that the position does not exceed the total number of files present, and we have to make sure that the position's filename is the same as previous. These values can change if a music class is reloaded and give unpredictable behavior. (closes issue #11663) Reported by: junky ........ ................ 2008-03-19 19:00 +0000 [r110024-110032] Russell Bryant * Makefile, build_tools/cflags.xml, build_tools/cflags-devmode.xml (added), /: Merged revisions 109974 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109974 | qwell | 2008-03-19 12:15:14 -0500 (Wed, 19 Mar 2008) | 13 lines Merged revisions 109973 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109973 | qwell | 2008-03-19 12:12:52 -0500 (Wed, 19 Mar 2008) | 5 lines People report bugs about Asterisk crashing with DO_CRASH enabled was getting a little silly... Now we only show certain cflags when you run configure with --enable-dev-mode (corresponding menuselect change to follow) ........ ................ 2008-03-19 18:26 +0000 [r109971-110021] Joshua Colp * main/rtp.c, /: Merged revisions 110020 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110020 | file | 2008-03-19 15:25:33 -0300 (Wed, 19 Mar 2008) | 14 lines Merged revisions 110019 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110019 | file | 2008-03-19 15:20:28 -0300 (Wed, 19 Mar 2008) | 6 lines Make sure that the mark bit does not incorrectly cause video frame timestamps to be calculated as if they are audio frames. (closes issue #11429) Reported by: sperreault Patches: 11429-frametype.diff uploaded by qwell (license 4) ........ ................ 2008-03-19 16:46 +0000 [r109969] Steve Murphy * main/config.c, /: Merged revisions 109942 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109942 | murf | 2008-03-19 10:24:51 -0600 (Wed, 19 Mar 2008) | 80 lines Merged revisions 109908 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109908 | murf | 2008-03-19 09:41:13 -0600 (Wed, 19 Mar 2008) | 72 lines (closes issue #11442) Reported by: tzafrir Patches: 11442.patch uploaded by murf (license 17) Tested by: murf I didn't give tzafrir very much time to test this, but if he does still have remaining issues, he is welcome to re-open this bug, and we'll do what is called for. I reproduced the problem, and tested the fix, so I hope I am not jumping by just going ahead and committing the fix. The problem was with what file_save does with templates; firstly, it tended to print out multiple options: [my_category](!)(templateref) instead of [my_category](!,templateref) which is fixed by this patch. Nextly, the code to suppress output of duplicate declarations that would occur because the reader copies inherited declarations down the hierarchy, was not working. Thus: [master-template](!) mastervar = bar [template](!,master-template) tvar = value [cat](template) catvar = val would be rewritten as: ;! ;! Automatically generated configuration file ;! Filename: experiment.conf (/etc/asterisk/experiment.conf) ;! Generator: Manager ;! Creation Date: Tue Mar 18 23:17:46 2008 ;! [master-template](!) mastervar = bar [template](!,master-template) mastervar = bar tvar = value [cat](template) mastervar = bar tvar = value catvar = val This has been fixed. Since the config reader 'explodes' inherited vars into the category, users may, in certain circumstances, see output different from what they originally entered, but it should be both correct and equivalent. ........ ................ 2008-03-19 04:06 +0000 [r109834-109840] Russell Bryant * /, main/utils.c: Merged revisions 109839 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109839 | russell | 2008-03-18 23:06:31 -0500 (Tue, 18 Mar 2008) | 10 lines Merged revisions 109838 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109838 | russell | 2008-03-18 23:06:05 -0500 (Tue, 18 Mar 2008) | 2 lines Tweak spacing in a recent change because I'm very picky. ........ ................ * apps/app_chanspy.c, /: Merged revisions 109764 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109764 | russell | 2008-03-18 17:36:02 -0500 (Tue, 18 Mar 2008) | 11 lines Merged revisions 109763 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109763 | russell | 2008-03-18 17:34:42 -0500 (Tue, 18 Mar 2008) | 3 lines Fix one place where the chanspy datastore isn't removed from a channel. (issue #12243, reported by atis, patch by me) ........ ................ 2008-03-18 23:23 +0000 [r109779] Tilghman Lesher * /, configs/res_ldap.conf.sample, res/res_config_ldap.c: Merged revisions 109775 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109775 | tilghman | 2008-03-18 18:22:25 -0500 (Tue, 18 Mar 2008) | 3 lines Change back to using ldap_initialize() and let the user specify a URL directly, instead of trying to piece it together, badly. ........ 2008-03-18 21:03 +0000 [r109716] Mark Michelson * /, apps/app_queue.c: Merged revisions 109714 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109714 | mmichelson | 2008-03-18 15:59:02 -0500 (Tue, 18 Mar 2008) | 20 lines Merged revisions 109713 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109713 | mmichelson | 2008-03-18 15:52:15 -0500 (Tue, 18 Mar 2008) | 12 lines This patch makes it so that all queue member status changes are handled through device state code. This removes several problems people were seeing where their queue members would get into an "unknown" state. Huge props go to atis on this one since he was the one who found the code section that was causing the problem and proposed the solution. I just wrote what he suggested :) (closes issue #12127) Reported by: atis Patches: 12127v3.patch uploaded by putnopvut (license 60) Tested by: atis, jvandal ........ ................ 2008-03-18 20:14 +0000 [r109684] Tilghman Lesher * /, res/res_config_ldap.c: Merged revisions 109683 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109683 | tilghman | 2008-03-18 15:13:40 -0500 (Tue, 18 Mar 2008) | 4 lines Set protocol version, port number correctly. (closes issue #12211, closes issue #12209) Reported by: sylvain ........ 2008-03-18 19:24 +0000 [r109654] Jason Parker * /, codecs/log2comp.h: Merged revisions 109651 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109651 | qwell | 2008-03-18 14:24:15 -0500 (Tue, 18 Mar 2008) | 15 lines Merged revisions 109648 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109648 | qwell | 2008-03-18 14:23:44 -0500 (Tue, 18 Mar 2008) | 7 lines Allow codecs that use log2comp (g726) to compile correctly on x86 with gcc4 optimizations. (closes issue #12253) Reported by: fossil Patches: log2comp.patch uploaded by fossil (license 140) ........ ................ 2008-03-18 19:00 +0000 [r109546-109622] Mark Michelson * /, channels/chan_agent.c: Merged revisions 109576 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109576 | mmichelson | 2008-03-18 12:59:18 -0500 (Tue, 18 Mar 2008) | 14 lines Merged revisions 109575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109575 | mmichelson | 2008-03-18 12:58:11 -0500 (Tue, 18 Mar 2008) | 6 lines Make sure an agent doesn't try to send dtmf to a NULL channel closes issue #12242 Reported by Yourname ........ ................ * include/asterisk/astmm.h, /: Merged revisions 109545 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109545 | mmichelson | 2008-03-18 12:00:53 -0500 (Tue, 18 Mar 2008) | 3 lines Add format attribute to printf-style functions in astmm.h ........ 2008-03-18 Russell Bryant * Asterisk 1.6.0-beta6 released. 2008-03-18 17:01 +0000 [r109546] Mark Michelson * include/asterisk/astmm.h, /: Merged revisions 109545 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109545 | mmichelson | 2008-03-18 12:00:53 -0500 (Tue, 18 Mar 2008) | 3 lines Add format attribute to printf-style functions in astmm.h ........ 2008-03-18 16:26 +0000 [r109487] Kevin P. Fleming * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, channels/chan_misdn.c, /: Merged revisions 109475 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109475 | kpfleming | 2008-03-18 11:23:05 -0500 (Tue, 18 Mar 2008) | 2 lines fix up various warnings found via the addition of format string checking... some of these were really, really bad code ........ 2008-03-18 15:58 +0000 [r109454-109459] Russell Bryant * Makefile, channels/chan_misdn.c, include/asterisk/strings.h, res/res_indications.c, utils/extconf.c, main/asterisk.c, apps/app_voicemail.c, utils/check_expr.c, cdr/cdr_sqlite3_custom.c, apps/app_meetme.c, /, res/res_phoneprov.c, main/utils.c, channels/chan_iax2.c, utils/frame.c, main/cli.c, funcs/func_enum.c, main/manager.c, include/asterisk/astobj.h, res/res_agi.c, main/features.c, apps/app_minivm.c, res/res_realtime.c, res/res_config_ldap.c, include/asterisk/utils.h, channels/chan_sip.c, apps/app_festival.c, main/translate.c, main/jitterbuf.c, utils/astman.c, include/jitterbuf.h, apps/app_queue.c: Merged revisions 109447 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109447 | twilson | 2008-03-18 10:43:34 -0500 (Tue, 18 Mar 2008) | 3 lines Go through and fix a bunch of places where character strings were being interpreted as format strings. Most of these changes are solely to make compiling with -Wsecurity and -Wformat=2 happy, and were not actual problems, per se. I also added format attributes to any printf wrapper functions I found that didn't have them. -Wsecurity and -Wmissing-format-attribute added to --enable-dev-mode. ........ * configs/sip_notify.conf.sample, /: Merged revisions 109111 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109111 | qwell | 2008-03-17 11:37:31 -0500 (Mon, 17 Mar 2008) | 10 lines Add sample events for aastra phones. aastra-check-cfg is the same as the other check-cfg entries, and aastra-xml is to load a pre-configured xml script. (closes issue #12229) Reported by: gowen72 Patches: aastra.patch uploaded by gowen72 (license 432) ........ 2008-03-18 15:50 +0000 [r109453] Kevin P. Fleming * /, configure, include/asterisk/autoconfig.h.in, acinclude.m4: Merged revisions 109451 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109451 | kpfleming | 2008-03-18 10:50:29 -0500 (Tue, 18 Mar 2008) | 2 lines ensure that dependencies on AST_C_DEFINE_CHECK symbols work properly ........ 2008-03-18 15:50 +0000 [r109448-109452] Russell Bryant * main/dial.c, /: Merged revisions 108962 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108962 | mvanbaak | 2008-03-16 16:50:58 -0500 (Sun, 16 Mar 2008) | 15 lines Merged revisions 108961 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108961 | mvanbaak | 2008-03-16 22:47:10 +0100 (Sun, 16 Mar 2008) | 7 lines add missing break to case AST_CONTROL_SRCUPDATE (closes issue #12228) Reported by: andrew Patches: SRC.patch uploaded by andrew (license 240) ........ ................ 2008-03-18 15:16 +0000 [r109398] Joshua Colp * main/manager.c, /, main/logger.c: Merged revisions 109396 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109396 | file | 2008-03-18 12:13:07 -0300 (Tue, 18 Mar 2008) | 3 lines Make sure values are interpreted as character strings and not format strings. (AST-2008-004) ........ 2008-03-18 15:14 +0000 [r109397] Steve Murphy * pbx/ael/ael-test/ael-ntest23 (added), pbx/ael/ael-test/ael-ntest23/t1/a.ael, pbx/ael/ael-test/ael-ntest23/t1/b.ael, pbx/ael/ael-test/ael-ntest23/t1/c.ael, pbx/ael/ael-test/ael-ntest23/t2/d.ael, pbx/ael/ael-test/ael-ntest23/t2/e.ael, pbx/ael/ael-test/ael-ntest23/t2/f.ael, res/ael/ael_lex.c, pbx/ael/ael-test/ref.ael-ntest23 (added), pbx/ael/ael-test/ael-ntest23/t3/g.ael, pbx/ael/ael-test/ael-ntest23/t3/h.ael, pbx/ael/ael-test/ael-ntest23/t3/i.ael, res/ael/ael.flex, pbx/ael/ael-test/ael-ntest23/t3/j.ael, pbx/ael/ael-test/ael-ntest23/qq.ael, pbx/ael/ael-test/ael-ntest23/t1, pbx/ael/ael-test/ael-ntest23/t2, pbx/ael/ael-test/ael-ntest23/t3, /, pbx/ael/ael-test/ael-ntest23/extensions.ael: Merged revisions 109357 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109357 | murf | 2008-03-18 08:09:50 -0600 (Tue, 18 Mar 2008) | 25 lines Merged revisions 109309 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109309 | murf | 2008-03-18 00:37:15 -0600 (Tue, 18 Mar 2008) | 17 lines (closes issue #11903) Reported by: atis Many thanks to atis for spotting this problem and reporting it. The fix was to straighten out how items are placed on and removed from the file stack. Regressions as well as the provided test case helped to straighten out all code paths. valgrind was used to make sure all memory allocated was freed. Sorry for not solving this earlier. I got distracted. Added the ntest23 regression test, which is mainly a copy of ntest22, but with a few juicy errors thrown in, to replicate the kind of error that atis spotted. ........ ................ 2008-03-18 15:11 +0000 [r109395] Jason Parker * /, channels/chan_sip.c: Merged revisions 109389 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109389 | qwell | 2008-03-18 10:07:04 -0500 (Tue, 18 Mar 2008) | 3 lines Do not return with a successful authentication if the From header ends up empty. (AST-2008-003) ........ 2008-03-18 15:09 +0000 [r109392] Joshua Colp * main/rtp.c, /, channels/chan_sip.c: Merged revisions 109390 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109390 | file | 2008-03-18 12:08:09 -0300 (Tue, 18 Mar 2008) | 11 lines Merged revisions 109386 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109386 | file | 2008-03-18 11:58:39 -0300 (Tue, 18 Mar 2008) | 3 lines Put a maximum limit on the number of payloads accepted, and also make sure a given payload does not exceed our maximum value. (AST-2008-002) ........ ................ 2008-03-18 00:40 +0000 [r109283] Sean Bright * /, configure, configure.ac: Merged revisions 109282 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109282 | seanbright | 2008-03-17 20:28:39 -0400 (Mon, 17 Mar 2008) | 1 line Fix a typo ........ 2008-03-17 22:24 +0000 [r109254] Terry Wilson * build_tools/cflags.xml, /, build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, main/Makefile, configure.ac, main/http.c, main/minimime (removed), build_tools/make_buildopts_h, makeopts.in: Merged revisions 109229 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109229 | twilson | 2008-03-17 17:10:06 -0500 (Mon, 17 Mar 2008) | 5 lines Replace minimime with superior GMime library so that the entire contents of an http post are not read into memory. This does introduce a dependency on the GMime library for handling HTTP POSTs, but it is available in most distros. If the library is present, then the compile flag for ENABLE_UPLOADS is enabled by default in menuselect. ........ 2008-03-17 22:07 +0000 [r109228] Mark Michelson * /, main/utils.c: Merged revisions 109227 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109227 | mmichelson | 2008-03-17 17:06:44 -0500 (Mon, 17 Mar 2008) | 20 lines Merged revisions 109226 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109226 | mmichelson | 2008-03-17 17:05:49 -0500 (Mon, 17 Mar 2008) | 12 lines Fix a logic flaw in the code that stores lock info which is displayed via the "core show locks" command. The idea behind this section of code was to remove the previous lock from the list if it was a trylock that had failed. Unfortunately, instead of checking the status of the previous lock, we were referencing the index immediately following the previous lock in the lock_info->locks array. The result of this problem, under the right circumstances, was that the lock which we currently in the process of attempting to acquire could "overwrite" the previous lock which was acquired. While this does not in any way affect typical operation, it *could* lead to misleading "core show locks" output. ........ ................ 2008-03-17 18:11 +0000 [r109175] Michiel van Baak * /, channels/chan_skinny.c: Merged revisions 109168 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109168 | mvanbaak | 2008-03-17 18:43:46 +0100 (Mon, 17 Mar 2008) | 11 lines Update the directory of placed calls on skinny phones when dialing a channel that does not provide progress (analog ZAP lines) The phone does handle the double update on calls to channels that do provide progress and wont insert duplicate items (closes issue #12239) Reported by: DEA Patches: chan_skinny-call-log.txt uploaded by DEA (license 3) ........ 2008-03-17 17:42 +0000 [r109167] Kevin P. Fleming * Makefile, /, configure, configure.ac, acinclude.m4: Merged revisions 109166 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109166 | kpfleming | 2008-03-17 12:31:46 -0500 (Mon, 17 Mar 2008) | 3 lines don't define Zaptel features as libraries, they aren't, and we don't want '--with-zaptel-' configure options for them also some minor cleanups ........ 2008-03-17 16:47 +0000 [r109109-109114] Joshua Colp * /, channels/chan_sip.c: Merged revisions 109108 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109108 | file | 2008-03-17 13:26:36 -0300 (Mon, 17 Mar 2008) | 12 lines Merged revisions 109107 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109107 | file | 2008-03-17 13:24:29 -0300 (Mon, 17 Mar 2008) | 4 lines 200 OKs in response to a reinvite need to be sent reliably. If the remote side does not receive one the dialog will be torn down. (closes issue #12208) Reported by: atrash ........ ................ 2008-03-17 14:21 +0000 [r109027] Mark Michelson * apps/app_chanspy.c, /: Merged revisions 109024 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109024 | mmichelson | 2008-03-17 09:21:14 -0500 (Mon, 17 Mar 2008) | 14 lines Merged revisions 109012 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109012 | mmichelson | 2008-03-17 09:18:26 -0500 (Mon, 17 Mar 2008) | 6 lines Make sure that we release the lock on the spyee channel if the spyee or spy has hung up (closes issue #12232) Reported by: atis ........ ................ 2008-03-16 17:56 +0000 [r108928-108930] Russell Bryant * apps/app_voicemail.c, /: Merged revisions 108927 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r108927 | russell | 2008-03-16 12:53:46 -0500 (Sun, 16 Mar 2008) | 7 lines Fix polling for mailbox changes in mailboxes that are not in the default vm context. (closes issue #12223) Reported by: DEA Patches: vm-polled-imap.txt uploaded by DEA (license 3) ........ 2008-03-15 16:21 +0000 [r108741-108895] Russell Bryant * Makefile, /: Merged revisions 108799 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r108799 | russell | 2008-03-14 15:14:06 -0500 (Fri, 14 Mar 2008) | 8 lines Make sure configure is run before menuselect on a clean checkout (closes issue #12197) Reported by: juggie Patches: 12197.diff uploaded by juggie (license 24) ........ * channels/chan_oss.c, /: Merged revisions 108797 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108797 | russell | 2008-03-14 15:09:37 -0500 (Fri, 14 Mar 2008) | 13 lines Merged revisions 108796 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108796 | russell | 2008-03-14 15:09:22 -0500 (Fri, 14 Mar 2008) | 5 lines Fix a channel name issue. chan_oss registers the "Console" channel type, but it created channels with an "OSS" prefix. (closes issue #12194, reported by davidw, patched by me) ........ ................ * contrib/init.d/rc.suse.asterisk, /: Merged revisions 108793 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108793 | russell | 2008-03-14 15:04:56 -0500 (Fri, 14 Mar 2008) | 12 lines Merged revisions 108792 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108792 | russell | 2008-03-14 15:04:35 -0500 (Fri, 14 Mar 2008) | 4 lines Update the SuSE init script to start networking before asterisk, as well. (closes issue #12200, reported by and change suggested by reinerotto) ........ ................ * /, configure, acinclude.m4: Merged revisions 108740 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r108740 | russell | 2008-03-14 12:05:11 -0500 (Fri, 14 Mar 2008) | 5 lines Do a link test in AST_EXT_TOOL_CHECK() to ensure we have all the required libs reported by the tool. (closes issue #12067, reported by Juggie, patched by me) ........ 2008-03-14 16:54 +0000 [r108739] Mark Michelson * /, channels/chan_sip.c: Merged revisions 108738 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108738 | mmichelson | 2008-03-14 11:52:51 -0500 (Fri, 14 Mar 2008) | 41 lines Merged revisions 108737 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108737 | mmichelson | 2008-03-14 11:44:08 -0500 (Fri, 14 Mar 2008) | 33 lines Fix a race condition in the SIP packet scheduler which could cause a crash. chan_sip uses the scheduler API in order to schedule retransmission of reliable packets (such as INVITES). If a retransmission of a packet is occurring, then the packet is removed from the scheduler and retrans_pkt is called. Meanwhile, if a response is received from the packet as previously transmitted, then when we ACK the response, we will remove the packet from the scheduler and free the packet. The problem is that both the ACK function and retrans_pkt attempt to acquire the same lock at the beginning of the function call. This means that if the ACK function acquires the lock first, then it will free the packet which retrans_pkt is about to read from and write to. The result is a crash. The solution: 1. If the ACK function fails to remove the packet from the scheduler and the retransmit id of the packet is not -1 (meaning that we have not reached the maximum number of retransmissions) then release the lock and yield so that retrans_pkt may acquire the lock and operate. 2. Make absolutely certain that the ACK function does not recursively lock the lock in question. If it does, then releasing the lock will do no good, since retrans_pkt will still be unable to acquire the lock. (closes issue #12098) Reported by: wegbert (closes issue #12089) Reported by: PTorres Patches: 12098-putnopvutv3.patch uploaded by putnopvut (license 60) Tested by: jvandal ........ ................ 2008-03-14 14:33 +0000 [r108684] Jason Parker * /, res/res_musiconhold.c: Merged revisions 108683 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108683 | qwell | 2008-03-14 09:32:55 -0500 (Fri, 14 Mar 2008) | 12 lines Merged revisions 108682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108682 | qwell | 2008-03-14 09:29:05 -0500 (Fri, 14 Mar 2008) | 4 lines Fix a potential segfault if chan (or chan->music_state) is NULL. Closes issue #12210, credit to edantie for pointing this out. ........ ................ 2008-03-13 21:48 +0000 [r108587] Mark Michelson * main/manager.c, /: Merged revisions 108586 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r108586 | mmichelson | 2008-03-13 16:47:55 -0500 (Thu, 13 Mar 2008) | 3 lines Make this compile ........ 2008-03-13 21:41 +0000 [r108585] Russell Bryant * apps/app_chanspy.c, main/channel.c, /, include/asterisk/channel.h: Merged revisions 108584 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108584 | russell | 2008-03-13 16:40:43 -0500 (Thu, 13 Mar 2008) | 19 lines Merged revisions 108583 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108583 | russell | 2008-03-13 16:38:16 -0500 (Thu, 13 Mar 2008) | 11 lines Fix another issue that was causing crashes in chanspy. This introduces a new datastore callback, called chan_fixup(). The concept is exactly like the fixup callback that is used in the channel technology interface. This callback gets called when the owning channel changes due to a masquerade. Before this was introduced, if a masquerade happened on a channel being spyed on, the channel pointer in the datastore became invalid. (closes issue #12187) (reported by, and lots of testing from atis) (props to file for the help with ideas) ........ ................ 2008-03-13 21:31 +0000 [r108582] Mark Michelson * main/manager.c, /: Merged revisions 108529 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r108529 | mmichelson | 2008-03-13 15:59:00 -0500 (Thu, 13 Mar 2008) | 11 lines Fixing a potential buffer overflow in the manager command ModuleCheck. Though this overflow is exploitable remotely, we are NOT issuing a security advisory for this since in order to exploit the overflow, the attacker would have to establish an authenticated manager session AND have the system privilege. By gaining this privilege, the attacker already has more powerful weapons at his disposal than overflowing a buffer with a malformed manager header, so the vulnerability in this case really lies with the authentication method that allowed the attacker to gain the system privilege in the first place. ........ 2008-03-13 21:07 +0000 [r108347-108532] Russell Bryant * /, channels/chan_sip.c: Merged revisions 108531 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108531 | russell | 2008-03-13 16:06:52 -0500 (Thu, 13 Mar 2008) | 18 lines Merged revisions 108530 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108530 | russell | 2008-03-13 16:06:33 -0500 (Thu, 13 Mar 2008) | 10 lines Make a tweak that gets the LEDs on polycom phones to blink when an extension that has been subscribed to goes on hold. Otherwise, they just stay on like it does when an extension is in use. (closes issue #11263) Reported by: russell Patches: notify_hold.rev1.txt uploaded by russell (license 2) Tested by: russell ........ ................ * apps/app_voicemail.c, /: Merged revisions 108508 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r108508 | russell | 2008-03-13 15:35:28 -0500 (Thu, 13 Mar 2008) | 2 lines Fix a place where configuration values could cause an overflow of a buffer. ........ * /, apps/app_followme.c: Merged revisions 108472 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108472 | russell | 2008-03-13 15:26:59 -0500 (Thu, 13 Mar 2008) | 12 lines Merged revisions 108469 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108469 | russell | 2008-03-13 15:26:28 -0500 (Thu, 13 Mar 2008) | 4 lines Fix a couple uses of sprintf. The second one could actually cause an overflow of a stack buffer. It's not a security issue though, it only depends on your configuration. ........ ................ * /, main/features.c: Merged revisions 107465 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r107465 | file | 2008-03-11 10:05:17 -0500 (Tue, 11 Mar 2008) | 4 lines Clarify comment about masquerading and playback of the parking slot. (closes issue #12180) Reported by: davidw ........ * /, channels/chan_sip.c: Merged revisions 107157 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r107157 | file | 2008-03-10 15:00:21 -0500 (Mon, 10 Mar 2008) | 4 lines If we receive a 488 on a T38 request reinvite back to audio. As well reinvite across a bridge back to audio if one side doesn't negotiate to T38. (closes issue #8677) Reported by: alex-911 ........ * /: Merged revisions 106892 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r106892 | mattf | 2008-03-07 16:36:49 -0600 (Fri, 07 Mar 2008) | 1 line Make sure we don't start a call when we have already done so in response to a COT message ........ * /, main/editline/Makefile.in: Merged revisions 106843 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r106843 | qwell | 2008-03-07 16:15:20 -0600 (Fri, 07 Mar 2008) | 13 lines Merged revisions 106842 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106842 | qwell | 2008-03-07 16:14:45 -0600 (Fri, 07 Mar 2008) | 5 lines Fix hardcoded grep in editline, were GNU grep is required. (closes issue #12124) Reported by: dmartin ........ ................ * include/asterisk/http.h, main/tcptls.c, main/manager.c, /, channels/chan_sip.c, res/res_phoneprov.c, main/http.c, include/asterisk/tcptls.h: Merged revisions 108295 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r108295 | russell | 2008-03-12 17:13:18 -0500 (Wed, 12 Mar 2008) | 3 lines Rename ast_tcptls_server_instance to session_instance, since this pertains to server and client usage. ........ * /, main/http.c: Merged revisions 108346 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r108346 | russell | 2008-03-12 17:49:26 -0500 (Wed, 12 Mar 2008) | 4 lines Make the default prefix empty, like it was in Asterisk 1.4. (closes issue #12198, reported by bkruse, patched by me) ........ 2008-03-12 22:10 +0000 [r108246-108294] Mark Michelson * /, channels/chan_sip.c: Merged revisions 108293 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r108293 | mmichelson | 2008-03-12 17:09:52 -0500 (Wed, 12 Mar 2008) | 3 lines Let's get this to compile ........ * /, channels/chan_sip.c: Merged revisions 108289 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108289 | mmichelson | 2008-03-12 16:57:41 -0500 (Wed, 12 Mar 2008) | 22 lines Merged revisions 108288 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108288 | mmichelson | 2008-03-12 16:53:46 -0500 (Wed, 12 Mar 2008) | 14 lines Change AST_SCHED_DEL use to ast_sched_del for autocongestion in chan_sip. The scheduler callback will always return 0. This means that this id is never rescheduled, so it makes no sense to loop trying to delete the id from the scheduler queue. If we fail to remove the item from the queue once, it will fail every single time. (Yes I realize that in this case, the macro would exit early because the id is set to -1 in the callback, but it still makes no sense to use that macro in favor of calling ast_sched_del once and being done with it) This is the first of potentially several such fixes. ........ ................ * /, include/asterisk/sched.h: Merged revisions 108238 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108238 | mmichelson | 2008-03-12 16:19:30 -0500 (Wed, 12 Mar 2008) | 20 lines Merged revisions 108227 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108227 | mmichelson | 2008-03-12 16:16:28 -0500 (Wed, 12 Mar 2008) | 12 lines Added a large comment before the AST_SCHED_DEL macro to explain its purpose as well as when it is appropriate and when it is not appropriate to use it. I also removed the part of the debug message that mentions that this is probably a bug because there are some perfectly legitimate places where ast_sched_del may fail to delete an entry (e.g. when the scheduler callback manually reschedules with a new id instead of returning non-zero to tell the scheduler to reschedule with the same idea). I also raised the debug level of the debug message in AST_SCHED_DEL since it seems like it could come up quite frequently since the macro is probably being used in several places where it shouldn't be. Also removed the redundant line, file, and function information since that is provided by ast_log. ........ ................ 2008-03-12 20:29 +0000 [r108205] Kevin P. Fleming * /, channels/chan_sip.c: Merged revisions 108191 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108191 | kpfleming | 2008-03-12 15:27:01 -0500 (Wed, 12 Mar 2008) | 14 lines Merged revisions 108086 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108086 | kpfleming | 2008-03-12 14:16:07 -0500 (Wed, 12 Mar 2008) | 6 lines if we receive an INVITE with a Content-Length that is not a valid number, or is zero, then don't process the rest of the message body looking for an SDP closes issue #11475 Reported by: andrebarbosa ........ ................ 2008-03-12 19:59 +0000 [r108138] Russell Bryant * apps/app_chanspy.c, main/channel.c, /: Merged revisions 108137 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108137 | russell | 2008-03-12 14:59:05 -0500 (Wed, 12 Mar 2008) | 48 lines Merged revisions 108135 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108135 | russell | 2008-03-12 14:57:42 -0500 (Wed, 12 Mar 2008) | 40 lines (closes issue #12187, reported by atis, fixed by me after some brainstorming on the issue with mmichelson) - Update copyright info on app_chanspy. - Fix a race condition that caused app_chanspy to crash. The issue was that the chanspy datastore magic that was used to ensure that spyee channels did not disappear out from under the code did not completely solve the problem. It was actually possible for chanspy to acquire a channel reference out of its datastore to a channel that was in the middle of being destroyed. That was because datastore destruction in ast_channel_free() was done near the end. So, this left the code in app_chanspy accessing a channel that was partially, or completely invalid because it was in the process of being free'd by another thread. The following sort of shows the code path where the race occurred: ============================================================================= Thread 1 (PBX thread for spyee chan) || Thread 2 (chanspy) --------------------------------------||------------------------------------- ast_channel_free() || - remove channel from channel list || - lock/unlock the channel to ensure || that no references retrieved from || the channel list exist. || --------------------------------------||------------------------------------- || channel_spy() - destroy some channel data || - Lock chanspy datastore || - Retrieve reference to channel || - lock channel || - Unlock chanspy datastore --------------------------------------||------------------------------------- - destroy channel datastores || - call chanspy datastore d'tor || which NULL's out the ds' || - Operate on the channel ... reference to the channel || || - free the channel || || || - unlock the channel --------------------------------------||------------------------------------- ============================================================================= ........ ................ 2008-03-12 18:31 +0000 [r108085] Joshua Colp * apps/app_mixmonitor.c, /, include/asterisk/audiohook.h, main/audiohook.c: Merged revisions 108084 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108084 | file | 2008-03-12 15:29:33 -0300 (Wed, 12 Mar 2008) | 12 lines Merged revisions 108083 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108083 | file | 2008-03-12 15:26:37 -0300 (Wed, 12 Mar 2008) | 4 lines Add a trigger mode that triggers on both read and write. The actual function that returns the combined audio frame though will wait until both sides have fed in audio, or until one side stops (such as the case when you call Wait). (closes issue #11945) Reported by: xheliox ........ ................ 2008-03-12 17:03 +0000 [r108033] Russell Bryant * main/channel.c, /: Merged revisions 108032 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108032 | russell | 2008-03-12 12:02:57 -0500 (Wed, 12 Mar 2008) | 12 lines Merged revisions 108031 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108031 | russell | 2008-03-12 11:59:07 -0500 (Wed, 12 Mar 2008) | 4 lines Destroy the channel lock after the channel datastores. (inspired by issue #12187) ........ .......