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${ noResults }
19449 Commits (3a422d77963f65aa179458d47623b0e2b80c5fdc)
Author | SHA1 | Message | Date |
---|---|---|---|
|
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 |
|
f861e34cf3 |
Blocked revisions 248268 via svnmerge
........ 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 |
16 years ago |
|
5665aa361f |
Minor tweaks to comment blocks and includes.
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 |
16 years ago |
|
29c57c6dac |
Tweak copyright and author lines.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@248225 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
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 |
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e8df30b584 |
Improve support for RTCP reports without report blocks
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@248108 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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8d80349bbf |
Blocked revisions 248106 via svnmerge
........ 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 |
16 years ago |
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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 |
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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 |
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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 |
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de1d19f511 |
Revert an errant part of a previous cleanup, to fix a memory corruption issue.
(closes issue #16368) Reported by: thirionjwf Patches: res_speech.c.patch uploaded by thirionjwf (license 955) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247841 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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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 |
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eb31f8b082 |
fixes confbridge crash when no timing module is loaded.
(closes issue #16471) Reported by: kjotte Patches: M16471.diff uploaded by junky (license 177) Tested by: kjotte, junky git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247770 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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48134df655 |
fixes Queue with C option crash
(closes issue #16475) Reported by: okrief Patches: queue_crash.diff uploaded by dvossel (license 671) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247736 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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469fed1cd3 |
Merged revisions 247651 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r247651 | mnicholson | 2010-02-18 13:38:09 -0600 (Thu, 18 Feb 2010) | 6 lines Copy the calling party's account code to the called party if they don't already have one. (closes issue #16331) Reported by: bluefox Tested by: mnicholson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247652 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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efea9ad922 |
Fix placing ISDN calls on hold preventing native bridging from being 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 |
16 years ago |
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4913cb8e75 |
Merged revisions 247508 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r247508 | lmadsen | 2010-02-18 11:53:44 -0500 (Thu, 18 Feb 2010) | 1 line Add additional link to best practices document per jsmith. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247509 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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e3f2fae59a |
Merged revisions 247502 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r247502 | lmadsen | 2010-02-18 11:38:17 -0500 (Thu, 18 Feb 2010) | 10 lines Add best practices documentation. (issue #16808) Reported by: lmadsen (issue #16810) Reported by: Nick_Lewis Tested by: lmadsen Review: https://reviewboard.asterisk.org/r/507/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247503 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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945529cae8 |
Add a new manager event for our buddies status.
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 |
16 years ago |
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230ded2388 |
Merged revisions 247422 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r247422 | russell | 2010-02-17 22:19:01 -0600 (Wed, 17 Feb 2010) | 10 lines Tweak argument handling for wget in the sounds Makefile. 1) Fix the check to see if we are using wget to not be full of fail. The configure script populates this variable with the absolute path to wget if it is found, so it didn't work. 2) Allow some extra arguments to be passed in for wget. This is just a simple change to allow our Bamboo build script to tell wget to be quiet and not fill up our logs with download status output. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247423 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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23d1c9186d |
Fix a couple of bugs in test tab completion.
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 |
16 years ago |
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dd975726fc |
Tab completion for test categories and names for "test show registered" and "test execute" CLI commands.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247380 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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2ce7eabb24 |
Fix two problems in ast_str functions found while writing a unit test.
1. The documentation for ast_str_set and ast_str_append state that the max_len parameter may be -1 in order to limit the size of the ast_str to its current allocated size. The problem was that the max_len parameter in all cases was a size_t, which is unsigned. Thus a -1 was interpreted as UINT_MAX instead of -1. Changing the max_len parameter to be ssize_t fixed this issue. 2. Once issue 1 was fixed, there was an off-by-one error in the case where we attempted to write a string larger than the current allotted size to a string when -1 was passed as the max_len parameter. When trying to write more than the allotted size, the ast_str's __AST_STR_USED was set to 1 higher than it should have been. Thanks to Tilghman for quickly spotting the offending line of code. Oh, and the unit test that I referenced in the top line of this commit will be added to reviewboard shortly. Sit tight... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247335 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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27a4cda821 |
Add support for GROUP_MATCH_COUNT regex matching on category
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 |
16 years ago |
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e8d201e870 |
modified device2extension_test's category
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247282 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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5b0b04f2d8 |
unit test for combined device state mapping and device to exten state mapping
Review: https://reviewboard.asterisk.org/r/516/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247262 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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57c819fd5e |
addition of dynamic parkinglots feature
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 |
16 years ago |
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c54f8ced1b |
Merged revisions 247168 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r247168 | mmichelson | 2010-02-17 10:24:17 -0600 (Wed, 17 Feb 2010) | 3 lines Make sure that when autofill is disabled that callers not in the front of the queue cannot place calls. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247169 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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bbb5acc65e |
RTP documentation states that you can pass NULL as the module, so make sure that's really the case.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247125 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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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 |
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68c6175f87 |
Add va_end calls to __ast_str_helper.
According to the man page for stdarg(3), "Each invocation of va_copy() must be matched by a corresponding invocation of va_end() in the same function." There were several cases in __ast_str_helper where va_copy was not matched with a corresponding call to va_end. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247076 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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13d7c1e6bd |
generate connected line info update from info in h.323 packets
Tested by: benngard git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247035 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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164549691e |
Add some clarifying documentation to the ast_str_set and ast_str_append functions.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246985 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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0f9e5ab557 |
swap openssl with OpenSSL in warning message.
(issue #16673) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246981 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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d693767226 |
warning message if openssl support is missing while attempting tls connection
(closes issue #16673) Reported by: michaesc Patches: tls_error_msg.diff uploaded by dvossel (license 671) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246980 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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0d4a8f642d |
Add unit test for dialplan pattern matching.
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 |
16 years ago |
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091f850a58 |
fixes sample rate conversion issue with Monitor application
When using ast_seekstream with the read/write streams of a monitor, the number of samples we are seeking must be of the same rate as the stream or the jump calculation will be incorrect. This patch adds logic to correctly convert the number of samples to jump to the sample rate the read/write stream is using. For example, if the call is G722 (16khz) and the read/write stream is recording a 8khz wav, seeking 320 samples of 16khz audio is not the same as seeking 320 samples of 8khz audio when performing the ast_seekstream on the stream. ABE-2044 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246899 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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919b0b5f54 |
Revert changes for now, pending discussion
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246863 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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ca8faab594 |
Add a few more targets for DEBUG_THREADLOCALS
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246827 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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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 |
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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 |
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1ac0713811 |
Merged revisions 246709 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r246709 | tilghman | 2010-02-15 17:42:33 -0600 (Mon, 15 Feb 2010) | 5 lines Make the menuselect instructions correct by allowing 'make menuselect' to actually solve dependency problems. (Previously, it would fail out again with the same message about running 'make menuselect', which was NOT at all helpful.) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246710 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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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 |
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94fd355eff |
Instead of just automatically filtering out in the Makefile, give an indication of dependencies in menuselect.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246667 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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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 |
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7d5d0311c1 |
Merged revisions 246545 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r246545 | dvossel | 2010-02-12 17:30:17 -0600 (Fri, 12 Feb 2010) | 16 lines lock channel during datastore removal On channel destruction the channel's datastores are removed and destroyed. Since there are public API calls to find and remove datastores on a channel, a lock should be held whenever datastores are removed and destroyed. This resolves a crash caused by a race condition in app_chanspy.c. (closes issue #16678) Reported by: tim_ringenbach Patches: datastore_destroy_race.diff uploaded by tim ringenbach (license 540) Tested by: dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246546 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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db861e181b |
Blocked revisions 246460 via svnmerge
........ r246460 | qwell | 2010-02-12 12:52:28 -0600 (Fri, 12 Feb 2010) | 1 line Fix some silly formatting, and remove unnecessary option_debug checks ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246461 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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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 |
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034a79c303 |
Updated doc for OSP lookup application.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246382 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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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 |