summaryrefslogtreecommitdiff
AgeCommit message (Collapse)Author
2010-02-24Merged revisions 248582 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r248582 | tilghman | 2010-02-24 15:02:18 -0600 (Wed, 24 Feb 2010) | 7 lines Remove color code sequences from verbose messages that go to logfiles. (closes issue #16786) Reported by: dodo Patches: logger2.patch uploaded by dodo (license 989) Tested by: tilghman ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@248584 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-24Remove unnecessary warning message, make a couple of formatting tweaksRussell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@248534 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-24Add ASTERISK_FILE_VERSION macro.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@248533 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-23Unit test for ast_str API.Mark Michelson
Review: https://reviewboard.asterisk.org/r/517 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@248489 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-23Merged revisions 248396 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r248396 | dvossel | 2010-02-23 10:26:05 -0600 (Tue, 23 Feb 2010) | 9 lines fixes invite with replaces deadlock (closes issue #16862) Reported by: pwalker Patches: replaces_deadlock_1.4 uploaded by dvossel (license 671) Tested by: pwalker, dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@248397 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-22Move the REF_DEBUG comment higher in the include list.Mark Michelson
Uncommenting the REF_DEBUG definition where it was in the source resulted in only a small part of the astobj2 references being logged to a file. Moving this up higher in the include list causes all references to be logged as they should be. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@248347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-22Blocked revisions 248268 via svnmergeOlle Johansson
........ r248268 | oej | 2010-02-22 14:52:34 +0100 (Mån, 22 Feb 2010) | 2 lines Don't log to debug unless debug is turned on ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@248269 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-22Minor tweaks to comment blocks and includes.Russell Bryant
Fix the copyright lines, tweak doxygen formatting, and remove some unnecessary includes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@248226 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-22Tweak copyright and author lines.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@248225 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-21Cleanup transmit_* functions, part 1Michiel van Baak
Break transmit_tone into transmit_start_tone and transmit_stop_tone as per the skinny protocol. (closes issue #16874) Reported by: wedhorn Patches: skinny-clean01.diff uploaded by wedhorn (license 30) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@248184 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-20Improve support for RTCP reports without report blocksOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@248108 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-20Blocked revisions 248106 via svnmergeOlle Johansson
........ r248106 | oej | 2010-02-20 23:25:42 +0100 (Lör, 20 Feb 2010) | 2 lines Make sure we support RTCP compound messages with zero reports ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@248107 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-19mfcr2 issue 0016844 - Fix portability bit fields and make ↵Moises Silva
mfcr2_immediate_accept work again, reported and patched by korihor git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@248003 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-19handle_request_invite revise comment, fix coding guideline issuesDavid Vossel
I'm working with this code right now trying to analyze a deadlock. This change is just to clean up a few things before I make a more complex patch. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247915 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-19Merged revisions 247910 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r247910 | rmudgett | 2010-02-19 11:18:49 -0600 (Fri, 19 Feb 2010) | 55 lines Merged revision 247904 from https://origsvn.digium.com/svn/asterisk/be/branches/C.2-... .......... r247904 | rmudgett | 2010-02-19 10:49:44 -0600 (Fri, 19 Feb 2010) | 49 lines Make chan_misdn DTMF processing consistent with other channel technologies. The processing of DTMF tones on the receiving side of an ISDN channel is inconsistent with the way it is handled in other channels, especially DAHDI analog. This causes DTMF tones sent from an ISDN phone to be doubled at the connected party. We are using the following 2 options of misdn.conf 1) astdtmf=yes 2) senddtmf=yes Option one is necessary because the asterisk DSP DTMF detection is better than mISDN's internal DSP. Not as many false positives. Option two is necessary to transmit DTMF tones end to end when mISDN channels are connected to SIP channels with out of band DTMF for example. The symptom is that DTMF tones sent by an ISDN phone are doubled on the way through asterisk when two mISDN channels are connected with a Local channel in between or if it is bridged to an analog channel. The doubling of DTMF tones is because DTMF is passed inband to asterisk by the mISDN channel and passed out of band once again after the release of the DTMF tone. Passing it inband is wrong. Neither an analog channel nor SIP channel passes DTMF inband if configured to inband DTMF. Analog and SIP channels filter out the DTMF tones because they use the voice frames returned by ast_dsp_process. But chan_misdn passes the unfiltered input voice frames instead. To overcome one aspect of the problem, the doubling of DTMF tones when two mISDN channels are directly bridged, someone made an 'optimization', where in that case the DTMF tone passed out-of-band to the peer channel is not translated to an inband tone at the transmit side. This optimization is bad because it does not work in general. For example, analog channels or mISDN channels when bridged through an intermediary local channel will generate DTMF tones from out-of-band information. Also, of course, it must not be done when there is no inband DTMF available. This patch fixes the issue. Now chan_misdn will filter the received inband DTMF signal the same as other channel types. Another change included: No need to build an extra translation path because ast_process_dsp does it if required. Patches: misdn-dtmf.patch JIRA ABE-2080 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247914 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-18Revert an errant part of a previous cleanup, to fix a memory corruption issue.Tilghman Lesher
(closes issue #16368) Reported by: thirionjwf Patches: res_speech.c.patch uploaded by thirionjwf (license 955) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247841 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-18If the peer record is from realtime, it could be set to 0, due to MySQL not ↵Tilghman Lesher
representing NULL well in integer columns. NULL means the value is not specified for the column, which normally means the driver uses whatever is the default value. However, on MySQL, placing a NULL in either a float or integer column results in a retrieval of the 0 value. Hence, users get an errant error on load. This patch suppresses that error and makes the value as if it was not there. Note that this cannot be done in the realtime driver, because the lack of difference between NULL and 0 can only be intepreted correctly by the driver itself. If we did it in the realtime driver, then it would be effectively impossible to set any realtime field to 0, because it would act as if the field were unspecified and possibly take on a different value. (closes issue #16683) Reported by: wdoekes git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247787 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-18fixes confbridge crash when no timing module is loaded.David Vossel
(closes issue #16471) Reported by: kjotte Patches: M16471.diff uploaded by junky (license 177) Tested by: kjotte, junky git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247770 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-18fixes Queue with C option crashDavid Vossel
(closes issue #16475) Reported by: okrief Patches: queue_crash.diff uploaded by dvossel (license 671) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247736 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-18Merged revisions 247651 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r247651 | mnicholson | 2010-02-18 13:38:09 -0600 (Thu, 18 Feb 2010) | 6 lines Copy the calling party's account code to the called party if they don't already have one. (closes issue #16331) Reported by: bluefox Tested by: mnicholson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247652 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-18Fix placing ISDN calls on hold preventing native bridging from being ↵Richard Mudgett
reexamined after a transfer. Consider the following scenario: /-- B A == * == Network \-- C Party B calls party A (EuroISDN BRI phone) Party A puts B on hold using the HOLD/RETRIEVE messages. Party A calls party C. Party A puts C on hold to talk with party B again. Party A transfers B to C by hanging up. The call does not get the opportunity to get re-transferred into the ISDN network by the native bridge because native bridging is not being reexamined after the initial transfer. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247609 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-18Merged revisions 247508 via svnmerge from Leif Madsen
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r247508 | lmadsen | 2010-02-18 11:53:44 -0500 (Thu, 18 Feb 2010) | 1 line Add additional link to best practices document per jsmith. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247509 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-18Merged revisions 247502 via svnmerge from Leif Madsen
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r247502 | lmadsen | 2010-02-18 11:38:17 -0500 (Thu, 18 Feb 2010) | 10 lines Add best practices documentation. (issue #16808) Reported by: lmadsen (issue #16810) Reported by: Nick_Lewis Tested by: lmadsen Review: https://reviewboard.asterisk.org/r/507/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247503 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-18Add a new manager event for our buddies status.Philippe Sultan
The new JabberStatus event gives a concise view of the status change to the AMI clients. Thanks fiddur! (closes issue #16760) Reported by: fiddur Patches: 244498.2.diff uploaded by fiddur (license 678) Tested by: fiddur, phsultan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247500 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-18Merged revisions 247422 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r247422 | russell | 2010-02-17 22:19:01 -0600 (Wed, 17 Feb 2010) | 10 lines Tweak argument handling for wget in the sounds Makefile. 1) Fix the check to see if we are using wget to not be full of fail. The configure script populates this variable with the absolute path to wget if it is found, so it didn't work. 2) Allow some extra arguments to be passed in for wget. This is just a simple change to allow our Bamboo build script to tell wget to be quiet and not fill up our logs with download status output. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-17Fix a couple of bugs in test tab completion.Mark Michelson
1. Add missing unlock of lists. 2. Swap order of arguments to test_cat_cmp in complete_test_name. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247381 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-17Tab completion for test categories and names for "test show registered" and ↵Mark Michelson
"test execute" CLI commands. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247380 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-17Fix two problems in ast_str functions found while writing a unit test.Mark Michelson
1. The documentation for ast_str_set and ast_str_append state that the max_len parameter may be -1 in order to limit the size of the ast_str to its current allocated size. The problem was that the max_len parameter in all cases was a size_t, which is unsigned. Thus a -1 was interpreted as UINT_MAX instead of -1. Changing the max_len parameter to be ssize_t fixed this issue. 2. Once issue 1 was fixed, there was an off-by-one error in the case where we attempted to write a string larger than the current allotted size to a string when -1 was passed as the max_len parameter. When trying to write more than the allotted size, the ast_str's __AST_STR_USED was set to 1 higher than it should have been. Thanks to Tilghman for quickly spotting the offending line of code. Oh, and the unit test that I referenced in the top line of this commit will be added to reviewboard shortly. Sit tight... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247335 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-17Add support for GROUP_MATCH_COUNT regex matching on categoryJeff Peeler
Current support for regex matching was previously only available on the group. Also, error reporting for regex failures has been added. In addition to this feature enhancement a unit test has been written to check the regular expression logic to ensure the count operation is working as expected. (closes issue #16642) Reported by: kobaz Patches: groupmatch2.patch uploaded by kobaz (license 834) Review: https://reviewboard.asterisk.org/r/503/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247295 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-17modified device2extension_test's categoryDavid Vossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247282 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-17unit test for combined device state mapping and device to exten state mappingDavid Vossel
Review: https://reviewboard.asterisk.org/r/516/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247262 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-17addition of dynamic parkinglots featureDavid Vossel
This feature allows for parkinglots to be created dynamically within the dialplan. Thanks to all who were involved with getting this patch written and tested! (closes issue #15135) Reported by: IgorG Patches: features.dynamic_park.v3.diff uploaded by IgorG (license 20) 2009090400_dynamicpark.diff.txt uploaded by mvanbaak (license 7) dynamic_parkinglot.diff uploaded by dvossel (license 671) Tested by: eliel, IgorG, acunningham, mvanbaak, zktech Review: https://reviewboard.asterisk.org/r/352/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247248 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-17Merged revisions 247168 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r247168 | mmichelson | 2010-02-17 10:24:17 -0600 (Wed, 17 Feb 2010) | 3 lines Make sure that when autofill is disabled that callers not in the front of the queue cannot place calls. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247169 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-17RTP documentation states that you can pass NULL as the module, so make sure ↵Tilghman Lesher
that's really the case. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247125 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-17Make all of the various rtpqos parameters in this branch available from the ↵Tilghman Lesher
CHANNEL function. Also includes a test for retrieving rtpqos parameters, including a NULL RTP driver. Additionally, some further separation of the SIP internal API into headers was necessary. (closes issue #16652) Reported by: kkm Patches: 20100204__issue16652.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/501/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-16Add va_end calls to __ast_str_helper.Mark Michelson
According to the man page for stdarg(3), "Each invocation of va_copy() must be matched by a corresponding invocation of va_end() in the same function." There were several cases in __ast_str_helper where va_copy was not matched with a corresponding call to va_end. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247076 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-16generate connected line info update from info in h.323 packetsAlexandr Anikin
Tested by: benngard git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247035 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-16Add some clarifying documentation to the ast_str_set and ast_str_append ↵Mark Michelson
functions. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246985 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-16swap openssl with OpenSSL in warning message.David Vossel
(issue #16673) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246981 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-16warning message if openssl support is missing while attempting tls connectionDavid Vossel
(closes issue #16673) Reported by: michaesc Patches: tls_error_msg.diff uploaded by dvossel (license 671) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246980 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-16Add unit test for dialplan pattern matching.Mark Michelson
This test works by reading input from arrays to build a sample dialplan. From there, patterns are attempted to be matched against said dialplan, with the expected match given. We then search in our example dialplan to see if we find a match and if what we find matches what we expected it to match. (closes issue #16809) Reported by: lmadsen Tested by: mmichelson Review: https://reviewboard.asterisk.org/r/504/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246942 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-16fixes sample rate conversion issue with Monitor applicationDavid Vossel
When using ast_seekstream with the read/write streams of a monitor, the number of samples we are seeking must be of the same rate as the stream or the jump calculation will be incorrect. This patch adds logic to correctly convert the number of samples to jump to the sample rate the read/write stream is using. For example, if the call is G722 (16khz) and the read/write stream is recording a 8khz wav, seeking 320 samples of 16khz audio is not the same as seeking 320 samples of 8khz audio when performing the ast_seekstream on the stream. ABE-2044 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246899 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-16Revert changes for now, pending discussionTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246863 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-16Add a few more targets for DEBUG_THREADLOCALSTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246827 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-16Change the blanket rules to delete .lastclean on all CFLAGS menuselect ↵Tilghman Lesher
targets to be more particular. This change builds upon the recent change to menuselect to add 'touch_on_change' as an attribute of both categories and members. This should allow only the most invasive defines to cause a complete rebuild, while defines which only affect a subset of modules will only cause a rebuild of that smaller set. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246789 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-16Allow Timer B to be set on the peer, and ensure SIP rules are followed (or ↵Tilghman Lesher
warn) in comparison to Timer T1. (closes issue #16643) Reported by: nahuelgreco Patches: 20100204__issue16643.diff.txt uploaded by tilghman (license 14) Tested by: oej git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246724 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-15Merged revisions 246709 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r246709 | tilghman | 2010-02-15 17:42:33 -0600 (Mon, 15 Feb 2010) | 5 lines Make the menuselect instructions correct by allowing 'make menuselect' to actually solve dependency problems. (Previously, it would fail out again with the same message about running 'make menuselect', which was NOT at all helpful.) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246710 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-15Restore triedtopribridge flag code removed in -r211197.Richard Mudgett
Ooops. Failed to note that we were inside a for loop and pri_channel_bridge() needs to be executed only once. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246669 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-15Instead of just automatically filtering out in the Makefile, give an ↵Tilghman Lesher
indication of dependencies in menuselect. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246667 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-15chan_sip parse code refactoring plus two new unit testsDavid Vossel
Code Refactoring Changes - read_to_parts() moved to reqresp_parser.c and has been renamed as get_name_and_number() - get_in_brackets() moved to reqresp_parser.c - find_closing_quotes() added to sip_utils.h Logic Changes - get_name_and_number() now uses parse_uri() and get_calleridname() for parsing. Before this change only names within quotes were found, when names not within quotes are possible. New Unit Tests -sip_get_name_and_number_test -sip_get_in_brackets_test (closes issue #16707) Reported by: Nick_Lewis Patches: issue16706.diff uploaded by dvossel (license 671) Review: https://reviewboard.asterisk.org/r/499/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-03 00:11:57 +0000