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2010-03-04Use autotagged externalsv1.6.2.6-rc1lmadsen0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.2.6-rc1@250722 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-04Importing release summary for 1.6.2.6-rc1 release.lmadsen2-0/+1057
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.2.6-rc1@250721 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-04Importing files for 1.6.2.6-rc1 release.lmadsen3-0/+22465
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.2.6-rc1@250720 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-04Creating tag for the release of asterisk-1.6.2.6-rc1lmadsen0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.2.6-rc1@250719 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-03Merged revisions 250609 via svnmerge from lmadsen1-57/+461
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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@250610 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-03Merged revisions 250481 via svnmerge from jpeeler1-0/+12
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 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@250484 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-03Blocked revisions 250478 via svnmergedvossel0-0/+0
........ r250478 | dvossel | 2010-03-03 13:02:13 -0600 (Wed, 03 Mar 2010) | 2 lines Changes 0ms to <1ms in cli END results during 'test execute' ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@250479 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-03Merged revisions 250395 via svnmerge from dvossel1-3/+8
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 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@250396 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-03Merged revisions 250246 via svnmerge from dvossel1-1/+1
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. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@250253 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-03Blocked revisions 250237 via svnmergedvossel0-0/+0
........ r250237 | dvossel | 2010-03-02 18:04:28 -0600 (Tue, 02 Mar 2010) | 2 lines fixes assumption that test failed if it did not pass when generating results ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@250238 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02Blocked revisions 250235 via svnmergedvossel0-0/+0
........ r250235 | dvossel | 2010-03-02 17:38:29 -0600 (Tue, 02 Mar 2010) | 2 lines base64 unit test ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@250236 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02Blocked revisions 250141 via svnmergedvossel0-0/+0
........ r250141 | dvossel | 2010-03-02 15:58:03 -0600 (Tue, 02 Mar 2010) | 15 lines adds 'p' option to PickupChan The 'p' option allows the PickupChan app to pickup a ringing phone by looking for the first match to a partial channel name rather than requiring a full match. (closes issue #16613) Reported by: syspert Patches: pickipbycallid.patch uploaded by syspert (license 938) pickupbycallerid_v2.patch uploaded by dvossel (license 671) Tested by: dvossel, syspert ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@250142 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02Merged revisions 250051 via svnmerge from lmadsen1-0/+6
https://origsvn.digium.com/svn/asterisk/trunk ........ r250051 | lmadsen | 2010-03-02 16:09:27 -0500 (Tue, 02 Mar 2010) | 8 lines Update IMAP documentation. Update the IMAP documentation to make it clear that storing voicemails in the same folder as a large number of emails could potentially cause significant slow downs when writing or retrieving voicemails. (issue #16704) Reported by: TimeHider Tested by: lmadsen, TimeHider ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@250052 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02Merged revisions 250045 via svnmerge from lmadsen1-0/+6
https://origsvn.digium.com/svn/asterisk/trunk ................ r250045 | lmadsen | 2010-03-02 15:52:19 -0500 (Tue, 02 Mar 2010) | 15 lines Merged revisions 250043 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r250043 | lmadsen | 2010-03-02 15:51:35 -0500 (Tue, 02 Mar 2010) | 7 lines Update documentation to clarify purpose of unanswered option. (closes issue #16267) Reported by: elsto Patches: cdr.conf.sample.patch.txt uploaded by lmadsen (license 10) Tested by: davidw, elsto ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@250047 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02Merged revisions 250037 via svnmerge from lmadsen1-12/+20
https://origsvn.digium.com/svn/asterisk/trunk ........ r250037 | lmadsen | 2010-03-02 15:36:10 -0500 (Tue, 02 Mar 2010) | 4 lines Update documentation to not imply we support overriding options. (closes issue #16855) Reported by: davidw ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@250038 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02Merged revisions 249950 via svnmerge from lmadsen1-2/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r249950 | lmadsen | 2010-03-02 14:49:48 -0500 (Tue, 02 Mar 2010) | 4 lines Fix literal values wrapped in documentation. (closes issue #16145) Reported by: tilghman ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@249953 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02revert ability to exit echo appalecdavis1-17/+10
caused a regression, as only supported VOICE, not VIDEO etc. (issue #16880) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@249952 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02Merged revisions 249925 via svnmerge from lmadsen1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r249925 | lmadsen | 2010-03-02 14:24:43 -0500 (Tue, 02 Mar 2010) | 6 lines Add missing description of the PARKINGLOT variable in XML documentation. (closes issue #16743) Reported by: snuffy Patches: parkingdoc.diff uploaded by snuffy (license 35) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@249933 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02Merged revisions 249912 via svnmerge from lmadsen1-23/+76
https://origsvn.digium.com/svn/asterisk/trunk ........ r249912 | lmadsen | 2010-03-02 14:21:19 -0500 (Tue, 02 Mar 2010) | 6 lines Convert some DUNDI functions to XML documentation. (closes issue #16798) Reported by: snuffy Patches: xml_dundi.diff uploaded by snuffy (license 35) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@249916 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02Merged revisions 249893 via svnmerge from dvossel24-13/+91
https://origsvn.digium.com/svn/asterisk/trunk ........ r249893 | dvossel | 2010-03-02 13:08:38 -0600 (Tue, 02 Mar 2010) | 11 lines fixes adaptive jitterbuffer configuration When configuring the adaptive jitterbuffer, the target_extra value not only could not be set from the configuration, but was not even being set to its proper default. This value is required in order for the adaptive jitterbuffer to work correctly. To resolve this a config option has been added to expose this value to the conf files, and a default value is provided when no config specific value is present. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@249895 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02Merged revisions 249892 via svnmerge from lmadsen1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r249892 | lmadsen | 2010-03-02 14:02:56 -0500 (Tue, 02 Mar 2010) | 1 line Fix several XML documentation validate errors. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@249894 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02fixes ability to exit echo appalecdavis1-10/+17
when called from a ISDN channel, null frames prevent '#' exit. Now only echo back VOICE and DTMF frames (issue #16880) Reported by: alecdavis Patches: echo_exit_1-6-1.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@249844 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-01Merged revisions 249672 via svnmerge from seanbright1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r249672 | seanbright | 2010-03-01 14:36:30 -0500 (Mon, 01 Mar 2010) | 18 lines Merged revisions 249671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r249671 | seanbright | 2010-03-01 14:35:01 -0500 (Mon, 01 Mar 2010) | 11 lines Fix crash in app_voicemail related to message counting. We were passing a 'struct inprocess **' and treating it like a 'struct inprocess *' causing a segfault. (closes issue #16921) Reported by: whardier Patches: 20100301_issue16921.patch uploaded by seanbright (license 71) Tested by: whardier ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@249675 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-01Merged revisions 249623 via svnmerge from tilghman1-11/+12
https://origsvn.digium.com/svn/asterisk/trunk ........ r249623 | tilghman | 2010-03-01 12:36:06 -0600 (Mon, 01 Mar 2010) | 2 lines Constify a bit of app_voicemail, to make ODBC and IMAP compile once again. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@249625 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-01Merged revisions 249538 via svnmerge from jpeeler1-35/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r249538 | jpeeler | 2010-03-01 11:11:31 -0600 (Mon, 01 Mar 2010) | 18 lines Merged revisions 249536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r249536 | jpeeler | 2010-03-01 11:02:03 -0600 (Mon, 01 Mar 2010) | 11 lines Modify queued frames from local channels to not set the other side to up In this case, attended transfers were broken due to ast_feature_request_and_dial detecting the channel being set to up before the answer frame could be read and therefore failing to mark the channel as ready. This fix is a regression fix for 244785, which should continue to work properly as well. (closes issue #16816) Reported by: jamhed Tested by: jamhed, corruptor ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@249580 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-28Merged revisions 249491 via svnmerge from tilghman1-1/+4
https://origsvn.digium.com/svn/asterisk/trunk ........ r249491 | tilghman | 2010-02-28 14:50:01 -0600 (Sun, 28 Feb 2010) | 5 lines Fix unit test that Alec Davis broke. (closes issue #16927) Reported by: alecdavis ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@249493 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-28Merged revisions 249405 via svnmerge from tilghman2-8/+33
https://origsvn.digium.com/svn/asterisk/trunk ........ r249405 | tilghman | 2010-02-28 01:10:22 -0600 (Sun, 28 Feb 2010) | 2 lines Properly document voicemail API documents. Also fix a crash reported via the -dev list. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@249407 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-27overlap receiving: automatically send CALL PROCEEDING when dialplan startsalecdavis1-1/+14
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 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@249321 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-27Merged revisions 249235 via svnmerge from kpfleming1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r249235 | kpfleming | 2010-02-27 09:08:35 -0500 (Sat, 27 Feb 2010) | 9 lines Merged revisions 249234 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r249234 | kpfleming | 2010-02-27 09:07:59 -0500 (Sat, 27 Feb 2010) | 1 line add a reference to the now-published IAX2 RFC ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@249238 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-26Merged revisions 249187 via svnmerge from tilghman1-46/+62
https://origsvn.digium.com/svn/asterisk/trunk ........ r249187 | tilghman | 2010-02-26 12:41:57 -0600 (Fri, 26 Feb 2010) | 18 lines Cleanups to fix bugs in the VM count API functions. - Urgent voicemails were not attached, because the attachment code looked in the wrong folder. - Urgent voicemails were sometimes counted twice when displaying the count of new messages. - Backends were inconsistent as to which voicemails each API counted. (closes issue #15654) Reported by: tomo1657 Patches: 20100225__issue15654.diff.txt uploaded by tilghman (license 14) Tested by: tilghman (closes issue #16448) Reported by: hevad Review: https://reviewboard.asterisk.org/r/525/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@249190 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-26Blocked revisions 249186 via svnmergedvossel0-0/+0
........ r249186 | dvossel | 2010-02-26 12:41:36 -0600 (Fri, 26 Feb 2010) | 2 lines adds Time field to "test show results" cli command ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@249188 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-26Merged revisions 249101 via svnmerge from mmichelson1-0/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r249101 | mmichelson | 2010-02-26 11:04:58 -0600 (Fri, 26 Feb 2010) | 14 lines Merged revisions 249100 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r249100 | mmichelson | 2010-02-26 11:04:29 -0600 (Fri, 26 Feb 2010) | 8 lines For T.38 reINVITEs treat a 606 the same as a 488. (closes issue #16792) Reported by: vrban Patches: t38_606.patch uploaded by vrban (license 756) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@249104 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-25Merged revisions 248952 via svnmerge from jpeeler1-3/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r248952 | jpeeler | 2010-02-25 17:09:54 -0600 (Thu, 25 Feb 2010) | 24 lines Merged revisions 248860 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r248860 | jpeeler | 2010-02-25 15:22:06 -0600 (Thu, 25 Feb 2010) | 18 lines Ensure that monitor recordings are written to the correct location (again) This is an extension to 248757. As such the dialplan test has been extended: exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n, dial(sip/5001) exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n, dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b) exten => 5042, n, dial(sip/5001) exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test3,m) exten => 5043, n, changemonitor(monitor_test4) exten => 5043, n, dial(sip/5001) exten => 5044, 1, monitor(wav,monitor_test4,m) exten => 5044, n, changemonitor(tmp/jeff/monitor_test5) ; this looks to fail by design and emits a warning exten => 5044, n, dial(sip/5001) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@248955 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-25Merged revisions 248946 via svnmerge from mmichelson1-1/+8
https://origsvn.digium.com/svn/asterisk/trunk ........ r248946 | mmichelson | 2010-02-25 16:41:48 -0600 (Thu, 25 Feb 2010) | 5 lines Fix incorrect ACL behavior when CIDR notation of "/0" is used. AST-2010-003 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@248949 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-25Merged revisions 248861 via svnmerge from tilghman1-0/+12
https://origsvn.digium.com/svn/asterisk/trunk ................ r248861 | tilghman | 2010-02-25 15:22:39 -0600 (Thu, 25 Feb 2010) | 22 lines Merged revisions 248859 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r248859 | tilghman | 2010-02-25 15:21:05 -0600 (Thu, 25 Feb 2010) | 15 lines Some platforms clear /var/run at boot, which makes connecting a remote console... difficult. Previously, we only created the default /var/run/asterisk directory at install time. While we could create it in the init script, that would not work for those who start asterisk manually from the command line. So the safest thing to do is to create it as part of the Asterisk boot process. This also changes the ownership of the directory, because the pid and ctl files are created after we setuid/setgid. (closes issue #16802) Reported by: Brian Patches: 20100224__issue16802.diff.txt uploaded by tilghman (license 14) Tested by: tzafrir ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@248864 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-25Merged revisions 248793 via svnmerge from jpeeler1-9/+10
https://origsvn.digium.com/svn/asterisk/trunk ................ r248793 | jpeeler | 2010-02-25 12:37:56 -0600 (Thu, 25 Feb 2010) | 22 lines Merged revisions 248757 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r248757 | jpeeler | 2010-02-25 12:06:54 -0600 (Thu, 25 Feb 2010) | 15 lines Ensure that monitor recordings are written to the correct location. Recordings should be placed in the monitor directory when a non-absolute path is used. Exact dialplan used for testing: exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n, dial(sip/5001) exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n, dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b) exten => 5042, n, dial(sip/5001) ABE-2101 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@248797 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-24Merged revisions 248584 via svnmerge from tilghman1-3/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r248584 | tilghman | 2010-02-24 15:17:26 -0600 (Wed, 24 Feb 2010) | 14 lines Merged revisions 248582 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r248582 | tilghman | 2010-02-24 15:02:18 -0600 (Wed, 24 Feb 2010) | 7 lines Remove color code sequences from verbose messages that go to logfiles. (closes issue #16786) Reported by: dodo Patches: logger2.patch uploaded by dodo (license 989) Tested by: tilghman ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@248643 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-23Merged revisions 248397 via svnmerge from dvossel1-19/+53
https://origsvn.digium.com/svn/asterisk/trunk ................ r248397 | dvossel | 2010-02-23 10:34:39 -0600 (Tue, 23 Feb 2010) | 15 lines Merged revisions 248396 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r248396 | dvossel | 2010-02-23 10:26:05 -0600 (Tue, 23 Feb 2010) | 9 lines fixes invite with replaces deadlock (closes issue #16862) Reported by: pwalker Patches: replaces_deadlock_1.4 uploaded by dvossel (license 671) Tested by: pwalker, dvossel ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@248398 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-20Blocked revisions 248108 via svnmergeoej0-0/+0
........ r248108 | oej | 2010-02-20 23:37:22 +0100 (Lör, 20 Feb 2010) | 2 lines Improve support for RTCP reports without report blocks ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@248109 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-19Merged revisions 228798 via svnmerge from tilghman2-17/+31
https://origsvn.digium.com/svn/asterisk/trunk ........ r228798 | tilghman | 2009-11-09 01:37:52 -0600 (Mon, 09 Nov 2009) | 14 lines Fix various problems detected with Valgrind. * chan_console accessed pvts after deallocation. * The module loader did not check usecount on shutdown, which led to chan_iax2 reading a timer that was already unloaded. (closes issue #16062) Reported by: alexanderheinz Patches: 20091109__issue16062.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@248011 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-19Merged revisions 248003 via svnmerge from moy1-10/+10
https://origsvn.digium.com/svn/asterisk/trunk ........ r248003 | moy | 2010-02-19 13:38:34 -0500 (Fri, 19 Feb 2010) | 1 line mfcr2 issue 0016844 - Fix portability bit fields and make mfcr2_immediate_accept work again, reported and patched by korihor ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@248005 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-19Merged revisions 247914 via svnmerge from rmudgett1-31/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r247914 | rmudgett | 2010-02-19 11:33:33 -0600 (Fri, 19 Feb 2010) | 62 lines Merged revisions 247910 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r247910 | rmudgett | 2010-02-19 11:18:49 -0600 (Fri, 19 Feb 2010) | 55 lines Merged revision 247904 from https://origsvn.digium.com/svn/asterisk/be/branches/C.2-... .......... r247904 | rmudgett | 2010-02-19 10:49:44 -0600 (Fri, 19 Feb 2010) | 49 lines Make chan_misdn DTMF processing consistent with other channel technologies. The processing of DTMF tones on the receiving side of an ISDN channel is inconsistent with the way it is handled in other channels, especially DAHDI analog. This causes DTMF tones sent from an ISDN phone to be doubled at the connected party. We are using the following 2 options of misdn.conf 1) astdtmf=yes 2) senddtmf=yes Option one is necessary because the asterisk DSP DTMF detection is better than mISDN's internal DSP. Not as many false positives. Option two is necessary to transmit DTMF tones end to end when mISDN channels are connected to SIP channels with out of band DTMF for example. The symptom is that DTMF tones sent by an ISDN phone are doubled on the way through asterisk when two mISDN channels are connected with a Local channel in between or if it is bridged to an analog channel. The doubling of DTMF tones is because DTMF is passed inband to asterisk by the mISDN channel and passed out of band once again after the release of the DTMF tone. Passing it inband is wrong. Neither an analog channel nor SIP channel passes DTMF inband if configured to inband DTMF. Analog and SIP channels filter out the DTMF tones because they use the voice frames returned by ast_dsp_process. But chan_misdn passes the unfiltered input voice frames instead. To overcome one aspect of the problem, the doubling of DTMF tones when two mISDN channels are directly bridged, someone made an 'optimization', where in that case the DTMF tone passed out-of-band to the peer channel is not translated to an inband tone at the transmit side. This optimization is bad because it does not work in general. For example, analog channels or mISDN channels when bridged through an intermediary local channel will generate DTMF tones from out-of-band information. Also, of course, it must not be done when there is no inband DTMF available. This patch fixes the issue. Now chan_misdn will filter the received inband DTMF signal the same as other channel types. Another change included: No need to build an extra translation path because ast_process_dsp does it if required. Patches: misdn-dtmf.patch JIRA ABE-2080 ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@248004 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-19Merged revisions 247915 via svnmerge from dvossel1-21/+19
https://origsvn.digium.com/svn/asterisk/trunk ........ r247915 | dvossel | 2010-02-19 11:40:26 -0600 (Fri, 19 Feb 2010) | 7 lines handle_request_invite revise comment, fix coding guideline issues I'm working with this code right now trying to analyze a deadlock. This change is just to clean up a few things before I make a more complex patch. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@247916 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-18Merged revisions 247841 via svnmerge from tilghman1-1/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r247841 | tilghman | 2010-02-18 17:13:46 -0600 (Thu, 18 Feb 2010) | 7 lines Revert an errant part of a previous cleanup, to fix a memory corruption issue. (closes issue #16368) Reported by: thirionjwf Patches: res_speech.c.patch uploaded by thirionjwf (license 955) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@247845 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-18Merged revisions 247787 via svnmerge from tilghman1-1/+6
https://origsvn.digium.com/svn/asterisk/trunk ........ r247787 | tilghman | 2010-02-18 15:42:53 -0600 (Thu, 18 Feb 2010) | 17 lines If the peer record is from realtime, it could be set to 0, due to MySQL not representing NULL well in integer columns. NULL means the value is not specified for the column, which normally means the driver uses whatever is the default value. However, on MySQL, placing a NULL in either a float or integer column results in a retrieval of the 0 value. Hence, users get an errant error on load. This patch suppresses that error and makes the value as if it was not there. Note that this cannot be done in the realtime driver, because the lack of difference between NULL and 0 can only be intepreted correctly by the driver itself. If we did it in the realtime driver, then it would be effectively impossible to set any realtime field to 0, because it would act as if the field were unspecified and possibly take on a different value. (closes issue #16683) Reported by: wdoekes ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@247792 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-18Merged revisions 247770 via svnmerge from dvossel1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r247770 | dvossel | 2010-02-18 15:23:48 -0600 (Thu, 18 Feb 2010) | 9 lines fixes confbridge crash when no timing module is loaded. (closes issue #16471) Reported by: kjotte Patches: M16471.diff uploaded by junky (license 177) Tested by: kjotte, junky ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@247776 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-18Merged revisions 247736 via svnmerge from dvossel1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r247736 | dvossel | 2010-02-18 14:58:41 -0600 (Thu, 18 Feb 2010) | 7 lines fixes Queue with C option crash (closes issue #16475) Reported by: okrief Patches: queue_crash.diff uploaded by dvossel (license 671) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@247737 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-18Merged revisions 247652 via svnmerge from mnicholson1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r247652 | mnicholson | 2010-02-18 13:39:37 -0600 (Thu, 18 Feb 2010) | 13 lines Merged revisions 247651 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r247651 | mnicholson | 2010-02-18 13:38:09 -0600 (Thu, 18 Feb 2010) | 6 lines Copy the calling party's account code to the called party if they don't already have one. (closes issue #16331) Reported by: bluefox Tested by: mnicholson ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@247653 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-18Merged revisions 247509 via svnmerge from lmadsen1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r247509 | lmadsen | 2010-02-18 11:54:43 -0500 (Thu, 18 Feb 2010) | 9 lines Merged revisions 247508 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r247508 | lmadsen | 2010-02-18 11:53:44 -0500 (Thu, 18 Feb 2010) | 1 line Add additional link to best practices document per jsmith. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@247512 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-18Merged revisions 247503 via svnmerge from lmadsen1-0/+292
https://origsvn.digium.com/svn/asterisk/trunk ................ r247503 | lmadsen | 2010-02-18 11:41:04 -0500 (Thu, 18 Feb 2010) | 18 lines Merged revisions 247502 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r247502 | lmadsen | 2010-02-18 11:38:17 -0500 (Thu, 18 Feb 2010) | 10 lines Add best practices documentation. (issue #16808) Reported by: lmadsen (issue #16810) Reported by: Nick_Lewis Tested by: lmadsen Review: https://reviewboard.asterisk.org/r/507/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@247506 f38db490-d61c-443f-a65b-d21fe96a405b