6409 Commits (3b94cadaf95975ca7ff90a091ce4e3d3461b218d)

Author	SHA1	Message	Date
Alec L Davis	98154867e6	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@249320 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	7e2145b9ac	Merged revisions 249234 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r249234 | kpfleming | 2010-02-27 09:07:59 -0500 (Sat, 27 Feb 2010) | 1 line add a reference to the now-published IAX2 RFC ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@249235 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	86f0690571	Merged revisions 249100 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r249100 | mmichelson | 2010-02-26 11:04:29 -0600 (Fri, 26 Feb 2010) | 8 lines For T.38 reINVITEs treat a 606 the same as a 488. (closes issue #16792) Reported by: vrban Patches: t38_606.patch uploaded by vrban (license 756) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@249101 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	ba75980904	Also kill the .i files, or else the build process will not recreate them, when we ... change flags. Fixes a weird symbol problem mmichelson was having in a group branch, but also applies to trunk. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@248667 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	6568b06d29	Merged revisions 248396 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r248396 | dvossel | 2010-02-23 10:26:05 -0600 (Tue, 23 Feb 2010) | 9 lines fixes invite with replaces deadlock (closes issue #16862) Reported by: pwalker Patches: replaces_deadlock_1.4 uploaded by dvossel (license 671) Tested by: pwalker, dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@248397 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	3a422d7796	Move the REF_DEBUG comment higher in the include list. ... 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	16 years ago
Michiel van Baak	55d1fcdd02	Cleanup transmit_* functions, part 1 ... 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	16 years ago
Moises Silva	0d838691bc	mfcr2 issue 0016844 - Fix portability bit fields and make 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	16 years ago
David Vossel	fc0cb53aa5	handle_request_invite revise comment, fix coding guideline issues ... I'm working with this code right now trying to analyze a deadlock. This change is just to clean up a few things before I make a more complex patch. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247915 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	57ee669d9f	Merged revisions 247910 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r247910 | rmudgett | 2010-02-19 11:18:49 -0600 (Fri, 19 Feb 2010) | 55 lines Merged revision 247904 from https://origsvn.digium.com/svn/asterisk/be/branches/C.2-... .......... r247904 | rmudgett | 2010-02-19 10:49:44 -0600 (Fri, 19 Feb 2010) | 49 lines Make chan_misdn DTMF processing consistent with other channel technologies. The processing of DTMF tones on the receiving side of an ISDN channel is inconsistent with the way it is handled in other channels, especially DAHDI analog. This causes DTMF tones sent from an ISDN phone to be doubled at the connected party. We are using the following 2 options of misdn.conf 1) astdtmf=yes 2) senddtmf=yes Option one is necessary because the asterisk DSP DTMF detection is better than mISDN's internal DSP. Not as many false positives. Option two is necessary to transmit DTMF tones end to end when mISDN channels are connected to SIP channels with out of band DTMF for example. The symptom is that DTMF tones sent by an ISDN phone are doubled on the way through asterisk when two mISDN channels are connected with a Local channel in between or if it is bridged to an analog channel. The doubling of DTMF tones is because DTMF is passed inband to asterisk by the mISDN channel and passed out of band once again after the release of the DTMF tone. Passing it inband is wrong. Neither an analog channel nor SIP channel passes DTMF inband if configured to inband DTMF. Analog and SIP channels filter out the DTMF tones because they use the voice frames returned by ast_dsp_process. But chan_misdn passes the unfiltered input voice frames instead. To overcome one aspect of the problem, the doubling of DTMF tones when two mISDN channels are directly bridged, someone made an 'optimization', where in that case the DTMF tone passed out-of-band to the peer channel is not translated to an inband tone at the transmit side. This optimization is bad because it does not work in general. For example, analog channels or mISDN channels when bridged through an intermediary local channel will generate DTMF tones from out-of-band information. Also, of course, it must not be done when there is no inband DTMF available. This patch fixes the issue. Now chan_misdn will filter the received inband DTMF signal the same as other channel types. Another change included: No need to build an extra translation path because ast_process_dsp does it if required. Patches: misdn-dtmf.patch JIRA ABE-2080 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247914 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	6fb7e0ece7	If the peer record is from realtime, it could be set to 0, due to MySQL not representing NULL well in integer columns. ... NULL means the value is not specified for the column, which normally means the driver uses whatever is the default value. However, on MySQL, placing a NULL in either a float or integer column results in a retrieval of the 0 value. Hence, users get an errant error on load. This patch suppresses that error and makes the value as if it was not there. Note that this cannot be done in the realtime driver, because the lack of difference between NULL and 0 can only be intepreted correctly by the driver itself. If we did it in the realtime driver, then it would be effectively impossible to set any realtime field to 0, because it would act as if the field were unspecified and possibly take on a different value. (closes issue #16683) Reported by: wdoekes git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247787 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	e7a5fb5459	Make all of the various rtpqos parameters in this branch available from the 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	16 years ago
Tilghman Lesher	22b144cef4	Change the blanket rules to delete .lastclean on all CFLAGS menuselect 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	16 years ago
Tilghman Lesher	b26b519159	Allow Timer B to be set on the peer, and ensure SIP rules are followed (or 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	16 years ago
Richard Mudgett	b2a1ad7946	Restore triedtopribridge flag code removed in -r211197. ... 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	16 years ago
David Vossel	2003243947	chan_sip parse code refactoring plus two new unit tests ... 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	16 years ago
David Vossel	72dcd51cc8	fixes areas where port should be removed from domain during parsing ... A patch was committed recently that converted duplicate uri parsing code to use the parse_uri function. There were two instances where this conversion did not mimic previous behavior exactly because the port was not being parsed off the end of the domain. In order to do this, a dummy pointer argument needs to be passed into parse_uri so it will know it must parse out the port from the domain. If a port output paramenter is not present, the domain is returned with the port still attached. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246420 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	6d9c531237	fixes some test description formatting inconsistencies so log file looks nice ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246338 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	f57e5150e5	additional parse_uri test and documentation ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246249 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	556260ad93	Change channel state on local channels for busy,answer,ring. ... Previously local channels channel state never changed. This became problematic when the state of the other side of the local channel was lost, for example during a masquerade. Changing the state of the local channel allows for the scenario to be detected when the channel state is set to ringing, but the peer isn't ringing. The specific problem scenario is described in 164201. Although this was noted on one of the issues, here is the tested dialplan verified to work: exten => 9700,1,Dial(Local/*9700@default&Local/0009700@default) exten => *9700,1,Set(GLOBAL(TESTCHAN)=${CHANNEL:0:${MATH(${LEN(${CHANNEL})}-1):0:2}}1) exten => *9700,n,wait(3) ;3 works, 1 did not exten => *9700,n,Dial(SIP/5001) exten => 0009700,1,Wait(1) ;1 works, 3 did not exten => 0009700,n,ChannelRedirect(${TESTCHAN},parkedcalls,701,1) (closes issue #14992) Reported by: davidw git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246070 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	bbed34f764	Various updates to the unit test API. ... 1) It occurred to me that the difference in usage between the error ast_str and the ast_test_update_status() usage has turned out to be a bit ambiguous in practice. In a lot of cases, the same message was being sent to both. In other cases, it was only sent to one or the other. My opinion now is that in every case, I think it makes sense to do both; we should output it to the CLI as well as save it off for logging purposes. This change results in most of the changes in this diff, since it required changes to all existing unit tests. It also allowed for some simplifications of unit test API implementation code. 2) Update ast_test_status_update() to include the file, function, and line number for the code providing the update. 3) There are some formatting tweaks here and there. Hopefully they aren't too distracting for code review purposes. Reviewboard's diff viewer seems to do a pretty good job of pointing out when something is a whitespace change. 4) I moved the md5_test and sha1_test into the test_utils module. It seemed like a better approach since these tests are so tiny. 5) I changed the number of nodes used in heap_test_2 from 1 million to 100 thousand. The only reason for this was to reduce the time it took for this test to run. 6) Remove an unused function prototype that was at the bottom of utils.h. 7) Simplify test_insert() using the LIST_INSERT_SORTALPHA() macro. The one minor difference in behavior is that it no longer checks for a test registered with the same name. 8) Expand the code in test_alloc() to provide specific error messages for each failure case, to clearly inform developers if they forget to set the name, summary, description, etc. 9) Tweak the output of the "test show registered" CLI command. I swapped the name and category to have the category first. It seemed more natural since that is the sort key. 10) Don't output the status ast_str in the "test show results" CLI command. This is going to tend to be pretty verbose, so just leave that for the detailed test logs (test generate results). Review: https://reviewboard.asterisk.org/r/493/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245864 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	dd48c7eb40	fixes a merging error for the iaxs and iaxsl off by one fix ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245804 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	5be3d14c11	Merged revisions 245792 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r245792 | dvossel | 2010-02-09 16:55:38 -0600 (Tue, 09 Feb 2010) | 12 lines Fixes iaxs and iaxsl size off by one issue. 2^15 = 32768 which is the maximum allowed iax2 callnumber. Creating the iaxs and iaxsl array of size 32768 means the maximum callnumber is actually out of bounds. This causes a nasty crash. (closes issue #15997) Reported by: exarv Patches: iax_fix.diff uploaded by dvossel (license 671) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245793 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	f935799e51	This commit removes an extra newline in T.38 generated SDP packets. This bug was caused by the fix introduced in r243860. ... (closes issue #16766) Reported by: raivisr Patches: t38-sdp-newline-fix1.diff uploaded by mnicholson (license 96) Tested by: raivisr git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245727 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	e9184b8dbe	Remove object files from the channels/sip/ directory on make clean. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245597 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	2b01c7b185	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245578 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	96f382fb60	Make chan_usbradio compile. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245382 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	f199367b93	Tweak formatting and add minor updates to some comments. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245269 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	a1ac799b58	Remove parsing of constantssrc from reload_config. ... This config option is already handled by the function handle_common_options and it is unnecessary to parse the value again. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245230 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	38cb3e2ac9	Remove useless sip options related to hash table size. ... First off, these options weren't actually doing anything. By the time the options were parsed, the peer and dialog containers had already been allocated with their default values. Second, hash table size is something that doesn't really make sense to change in a config file. If a user is that interested in changing the hashtable size, he can modify the source itself. I have removed the parsing of the hash_peer, hash_user, and hash_dialog options. I have removed the hash_user_size variable altogether since it is not used at all. I also changed hash_peer_size and hash_dialog_size to be constant, and have changed the symbols to be in all caps as constants typically are. I have also removed the entire section in sip.conf.sample regarding configurable hashtable sizes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245192 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	a97e8f3908	adds total call numbers available to 'iax2 show callnumber usage' cli output ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245006 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	1810f1efff	fixes issue with sip registry not having correct default expiry ... default expiry was not being set correctly for a registry object. Thanks to ebroad for reporting the issue and testing the patch. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244924 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	f30de5ef0e	parse_moved_contact tries to parse contact_name twice ... parse_moved_contact attempts to remove a quoted string twice, and the first try wasn't even being done correctly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244769 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	a9932363a9	-----Changes ----- ... New files - channels/sip/sip.h – A new header for shared #define, enum, and struct definitions. - channels/sip/include/sip_utils.h – sip util functions shared among the all the sip APIs - channels/sip/include/config_parser.h – sip config-parser API - channels/sip/config_parser.c – Contains sip.conf parsing helper functions with unit tests. - channels/sip/include/reqresp_parser.h – sip request response parser API - channels/sip/reqresp_parser.c – Contains sip request and response parsing helper functions with unit tests. New Unit Tests - sip_parse_uri_test - sip_parse_host_test - sip_parse_register_line_test Code Refactoring - All reusable #define, enum, and struct definitions were moved out of chan_sip.c into sip.h. During this process formatting changes were made to comments in both sip.h and chan_sip.c in order to better adhere to the coding guidelines. - The beginnings of three new sip APIs, sip-utils.h, config-parser.h, reqresp-parser.h using existing chan_sip.c functions. - parse_uri() and get_calleridname() were moved from chan_sip.c to request-parser.c along with unit tests for both functions. - sip_parse_host() and sip_parse_register_line() were moved from chan_sip.c to config-parser.c along with unit tests for both functions. Changes to parse_uri() -removal of the options parameter. It was never used and did not behave correctly. -additional check for [?header] field. When this field was present, the transport type was not being set correctly. ----- Overview ----- This patch is introduced with the hope that unit tests for all our sip parsing functions will be written soon. chan_sip is a huge file, and with the addition of each unit test chan_sip is going to grow larger and harder to maintain. I'm proposing we begin refactoring chan_sip, starting with the parsing functions. With each parsing function we move into a separate helper file, a unit test should accompany it. I've attempted to lay down the ground work for this change by creating two new parser helper files (config-parser.c and reqresp-parser.c) and moving all shared structs, enums, and defines from chan_sip.c into a shared sip.h file. We can't verify everything in Asterisk using unit tests, but string parsing is one area where unit tests make the most sense. By beginning to restructure the code in this way, chan_sip not only becomes less bloated, but Asterisk as a whole will become more stable. Review: https://reviewboard.asterisk.org/r/477/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244597 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	d3ae695204	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244505 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	36bbf8f902	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244443 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	72c1b76038	Merged revisions 244070 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r244070 | tilghman | 2010-02-01 11:46:31 -0600 (Mon, 01 Feb 2010) | 16 lines Revert previous chan_local fix (r236981) and fix instead by destroying expired frames in the queue. (closes issue #16525) Reported by: kobaz Patches: 20100126__issue16525.diff.txt uploaded by tilghman (license 14) 20100129__issue16525__1.6.0.diff.txt uploaded by tilghman (license 14) Tested by: kobaz, atis (closes issue #16581) Reported by: ZX81 (closes issue #16681) Reported by: alexr1 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244071 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	397ec33284	Informational message, not an error. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243943 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	5766b06ad4	Add a missing line terminator for T.38 SDP. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243860 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	9ae1efe42c	Merged revisions 243779 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r243779 | russell | 2010-01-28 09:03:17 -0600 (Thu, 28 Jan 2010) | 2 lines Fix a bogus third argument to ast_copy_string(). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243780 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	34317fb0d3	Fix the ability to specify an OSP token for an outbound IAX2 call. ... When this patch was originally submitted, the code allowed for the token to be set via a channel variable. I decided that a cleaner approach would be to integrate it into the CHANNEL() function. Unfortunately, that is not a suitable approach. It's not possible to get the value set on the channel soon enough using that method. So, go back to the simple channel variable method. (closes issue #16711) Reported by: homesick Patches: iax-svn.diff uploaded by homesick (license 91) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243482 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	d16b89be17	RFC compliant uri and display-name encode/decode ... 1. URI Encoding This patch changes ast_uri_encode()'s behavior when doreserved is enabled. Previously when doreserved was enabled only a small set of reserved characters were encoded. This set was comprised primarily of the reserved characters defined in RFC3261 section 25.1, but contained other characters as well. Rather than only escaping the reserved set, doreserved now escapes all characters not within the unreserved set as defined by RFC 3261 and RFC 2396. Also, the 'doreserved' variable has been renamed to 'do_special_char' in attempts to avoid confusion. When doreserve is not enabled, the previous logic of only encoding the characters <= 0X1F and > 0X7f remains, except for the '%' character, which must always be encoded as it signifies a HEX escaped character during the decode process. 2. URI Decoding: Break up URI before decode. In chan_sip.c ast_uri_decode is called on the entire URI instead of it's individual parts after it is parsed. This is not good as ast_uri_decode can introduce special characters back into the URI which can mess up parsing. This patch resolves this by not decoding a URI until parsing is completely done. There are many instances where we check to see if pedantic checking is enabled before we decode a URI. In these cases a new macro, SIP_PEDANTIC_DECODE, is used on the individual parsed segments of the URI rather than constantly putting if (pedantic) { decode() } checks everywhere in the code. In the areas where ast_uri_decode is not dependent upon pedantic checking this macro is not used, but decoding is still moved to each individual part of the URI. The only behavior that should change from this patch is the time at which decoding occurs. Since I had to look over every place URI parsing occurs to create this patch, I found several places where we use duplicate code for parsing. To consolidate the code, those areas have updated to use the parse_uri() function where possible. 3. SIP display-name decoding according to RFC3261 section 25. To properly decode the display-name portion of a FROM header, chan_sip's get_calleridname() function required a complete re-write. More information about this change can be found in the comments at the beginning of this function. 4. Unit Tests. Unit tests for ast_uri_encode, ast_uri_decode, and get_calleridname() have been written. This involved the addition of the test_utils.c file for testing the utils api. (closes issue #16299) Reported by: wdoekes Patches: astsvn-16299-get_calleridname.diff uploaded by wdoekes (license 717) get_calleridname_rewrite.diff uploaded by dvossel (license 671) Tested by: wdoekes, dvossel, Nick_Lewis Review: https://reviewboard.asterisk.org/r/469/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243200 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Olle Johansson	64b76fa41a	Merged revisions 242226 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r242226 | oej | 2010-01-22 10:19:30 +0100 (Fre, 22 Jan 2010) | 3 lines Initialize notify_types to NULL ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242227 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	04aa8b1149	Formats are inconsistent between even 32-bit and 64-bit Linux. Use casts to ensure both compile. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241896 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	237f57cafe	Fix up compile breakage from ast_tvdiff_ms() API change. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241503 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Alec L Davis	c7be027151	Update CDR variables as pbx starts ... Allows CDR variables added in cdr.c:set_one_cid to become visable during the call, by executing ast_cdr_update() early in __ast_pbx run. Reverts sig_pri changes in trunk that are specific to isdn technology only. (closes issue #16638) Reported by: alecdavis Patches: cdr_update.diff3.txt uploaded by alecdavis (license 585) Tested by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241416 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	716d3880fc	small correction from 241314 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241315 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	4bb10fe6a9	Merged revisions 241227 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r241227 | jpeeler | 2010-01-19 11:22:18 -0600 (Tue, 19 Jan 2010) | 13 lines Fix deadlock in agent_read by removing call to agent_logoff. One must always lock the agents list lock before the agent private. agent_read locks the private immediately, so locking the agents list lock is not an option (which is what agent_logoff requires). Because agent_read already has access to the agent private all that is necessary is to do the required hanging up that agent_logoff performed. (closes issue #16321) Reported by: valon24 Patches: bug16321.patch uploaded by jpeeler (license 325) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241314 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Alec L Davis	800313cbd9	Update CDR variables before pbx starts (overlap dial) ... Allows CDR variables added in cdr.c:set_one_cid to become visable during the call. (issue #16638) Reported by: alecdavis Patches: cdr_update.diff2.txt uploaded by alecdavis (license 585) Tested by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241187 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	568c057c4c	Extend max call limit duration from 24.8 days to 292+ million years. ... If the limit was set past MAX_INT upon answering, the call was immediately hung up due to overflow from the return of ast_tvdiff_ms (in ast_check_hangup). The time calculation functions ast_tvdiff_sec and ast_tvdiff_ms have been changed to return an int64_t to prevent overflow. Also the reporter suggested adding a message indicating the reason for the call hanging up. Given that the new limit is so much higher, the message (which would only really be useful in the overflow scenario) has been made a debug message only. (closes issue #16006) Reported by: viraptor git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241143 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago