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19842 Commits (d38c6459f55adb90d8e7eb29d9f11fdf0777f3e5)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Matthew Nicholson
|
d38c6459f5 |
Merged revisions 264334 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r264334 | mnicholson | 2010-05-19 15:01:38 -0500 (Wed, 19 May 2010) | 5 lines Set quieted flag when receiving a dtmf tone during playback in speechbackground. (closes issue #16966) Reported by: asackheim ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264335 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
David Vossel
|
0407208b6d |
fixes crash in check_rtp_timeout
During deadlock avoidance the sip dialog pvt is locked and unlocked. When this occurs we have no guarantee the pvt's owner is still valid. We were trying to access the pvt's owner after this without checking to see if it still existed first. (closes issue #17271) Reported by: under Patches: check_rtp_timeout.diff uploaded by under (license 914) Tested by: dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264331 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Tilghman Lesher
|
b5a629624a |
Merged revisions 264248 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r264248 | tilghman | 2010-05-19 12:41:29 -0500 (Wed, 19 May 2010) | 17 lines Internal timing is now on by default, if you're using DAHDI 2.3 or above. The reason for ensuring DAHDI 2.3 or above is that this version ensures that a timer is always available, whereas in previous versions, it was possible for DAHDI to be loaded, but have no drivers to actually generate timing. If internal_timing was turned on in this circumstance, a complete lack of audio would result. This is the reason why internal_timing was not on by default. However, now that DAHDI ensures the availability of a timer, there is no reason for this setting to be off (and in fact, it solves a great many initial user problems). (closes issue #15932) Reported by: dimas Patches: 20100519__issue15932.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264249 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
|
07df131a7f |
Keep track of digit duration, when we're decoding inband to pass DTMF frames.
(closes issue #17235) Reported by: frawd Patches: new_dtmf_dsp_len.patch uploaded by frawd (license 610) 20100518__issue17235.diff.txt uploaded by tilghman (license 14) Tested by: frawd git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264204 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Leif Madsen
|
e3c9e6ae86 |
Fix compilation problem with previous commit.
(issue #16009) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264161 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Kevin P. Fleming
|
e77efbc12e |
Add ability for logger channels to include *all* levels.
Now that Asterisk modules can dynamically create and destroy logger levels on demand, it's useful to be able to configure a logger channel (console, file, whatever) to be able to accept log messages from *all* levels, even levels created dynamically. This patch adds support for this, by allowing the '*' level name to be used in logger.conf. Review: https://reviewboard.asterisk.org/r/663/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264160 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Leif Madsen
|
a8a1961be7 |
Add ability to hangup all channels from the CLI.
Added the keyword 'all' to the 'channel hangup request' CLI command so that you can request all channels to be hungup without having to restart Asterisk. (closes issue #16009) Reported by: moy Patches: hangup-all-rev-221688.patch uploaded by moy (license 222) Tested by: moy, russell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264117 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
David Vossel
|
51e7ee235b |
fixes crash during dtmf
During the processing of Cisco dtmf the dtmf samples were not being calculated correctly. In an attempt to determine what sample rate was being used, a NULL frame was processed which caused a crash. This patch resolves this. (closes issue #17248) Reported by: falves11 Patches: issue_17248.diff uploaded by dvossel (license 671) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264114 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Alec L Davis
|
30e9a9794c |
fix incorrectly typed indications for [nz] stutter and dialrecall
(closes issue #17359) Reported by: alecdavis Patches: bug17359.diff.txt uploaded by alecdavis (license 585) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264031 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
|
f55aff74ed |
Merged revisions 263949 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r263949 | tilghman | 2010-05-19 01:32:27 -0500 (Wed, 19 May 2010) | 8 lines Because progress is called multiple times, across several frames, we must persist states when detecting multitone sequences. (closes issue #16749) Reported by: dant Patches: dsp.c-bug16749-1.patch uploaded by dant (license 670) Tested by: dant ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263950 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
|
19f4ca6176 |
Add an sha1sum-workalike for platforms which don't have it (like Mac OS X)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263905 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
David Vossel
|
10789ef88a |
fixes segfault on logging
(closes issue #17331) Reported by: under Patches: utils.diff uploaded by under (license 914) segfault_on_logging.diff uploaded by dvossel (license 671) Tested by: under, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263904 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Mark Michelson
|
7814913d86 |
Be sure to heap-allocate the redirecting to tag so as not to cause crashiness.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263860 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
|
a21192f4a7 |
Make happy green color come back
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263858 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Mark Michelson
|
2b2439dede |
Fix memory leaks in redirecting structures in chan_sip.c
Thanks to Richard for pointing this out. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263810 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Jeff Peeler
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115f5076f5 |
put changes with the correct version
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263808 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Jeff Peeler
|
94df424e1d |
Merged revisions 263769 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r263769 | jpeeler | 2010-05-18 13:54:58 -0500 (Tue, 18 May 2010) | 10 lines Modify directory name reading to be interrupted with operator or pound escape. In the case of accidentally entering the wrong first three letters for the reading, users could be very frustrated if the name listing is very long. This allows interrupting the reading by pressing 0 or #. 0 will attempt to execute a configured operator (o) extension and # will exit and proceed in the dialplan. ABE-2200 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263807 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
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4c034c1f72 |
Cache sound tarfiles in a common directory, such that a clean reinstall does not force a re-download of the tarballs.
(closes issue #15370) Reported by: pprindeville Patches: asterisk-trunk-bugid15370.patch uploaded by pprindeville (license 347) Tested by: pprindeville, tilghman, seanbright git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263724 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Mark Michelson
|
e3ac20a7f6 |
Merged revisions 263639 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r263639 | mmichelson | 2010-05-17 17:00:28 -0500 (Mon, 17 May 2010) | 10 lines Fix logic error when checking for a devstate provider. When using strsep, if one of the list of specified separators is not found, it is the first parameter to strsep which is now NULL, not the pointer returned by strsep. This issue isn't especially severe in that the worst it is likely to do is waste some cycles when a device with no '/' and no ':' is passed to ast_device_state. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263640 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Mark Michelson
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7160f0af45 |
Blocked revisions 263637 via svnmerge
........ r263637 | mmichelson | 2010-05-17 16:48:46 -0500 (Mon, 17 May 2010) | 8 lines Remove arbitrary size limitation for hints. (closes issue #17257) Reported by: tim_ringenbach Patches: hints_crash_fix.diff uploaded by tim ringenbach (license 540) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263638 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
|
fa8e44f232 |
With IMAP backend, messages in INBOX were counted twice for MWI.
(closes issue #17135) Reported by: edhorton Patches: 20100513__issue17135.diff.txt uploaded by tilghman (license 14) 17135_2.diff uploaded by ebroad (license 878) Tested by: edhorton, ebroad git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263589 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Mark Michelson
|
b5d5cc565f |
Enhancements to connected line and redirecting work.
From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Leif Madsen
|
fa5350f7d7 |
Missing newlines added to Set-Cookie line in manager.c
Sean Bright pointed out that we lost a set of newline characters in commit 190349 on a line I had recently changed. Yay for code review on commits. (issue #17231, #10961) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263460 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Leif Madsen
|
193d495a8a |
Recorded merge of revisions 263456 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r263456 | lmadsen | 2010-05-17 09:35:18 -0500 (Mon, 17 May 2010) | 11 lines Manager cookies are not compatible with RFC2109. The Version field in the cookies we're setting contain quotes around the version number which is not compatible with RFC2109 and breaks some implementations. (closes issue #17231) Reported by: ecarruda Patches: manager_rfc2109-trunk-v1.patch uploaded by ecarruda (license 559) manager_rfc2109-1.6.2-v1.patch uploaded by ecarruda (license 559) Tested by: ecarruda, russell ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263457 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Leif Madsen
|
3f1fc9e354 |
Merged revisions 263374 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r263374 | lmadsen | 2010-05-17 09:04:57 -0500 (Mon, 17 May 2010) | 8 lines Update link to new version of core sounds. The latest version of the core sounds files 1.4.19 now includes the missing queue-minute sound file which is called by app_queue but which has been missing. (closes issue #17123) Reported by: n8ideas ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263375 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
David Vossel
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96d3e573c9 |
Update CHANGES to reflect DAHDI buffer dialstring option backport to 1.6.2
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263294 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Tzafrir Cohen
|
b8ea6e7500 |
live_ast: add commands 'rsync' and 'gen-live-asterisk'
This adds the following two commands to live_ast: * rsync [user]@host directory Copy over all generated files to <directory> at remote host. Would allow running live_ast there. Hence allows separating a build machine from a test machine. * gen-live-asteris: regenerate live/asterisk . Useful if copying over files to a different directory. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263250 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Kevin P. Fleming
|
c44da92360 |
Improve some very confusing structure names in astobj2.c
As pointed out by 'akshayb' on #asterisk-dev, the code here called a list of bucket entries a 'bucket', and the entries within the bucket were called 'bucket_list'. This made the code very hard to understand without reading all of it... so I've renamed 'bucket_list' to 'bucket_entry' to clarify the purpose of the structure. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263208 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Vossel
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cddc244c97 |
fix iax_frame double free
Very unfortunate things happen if we add an iax_frame to the frame queue and let go of the lock before scheduling the frame's transmit... There is a race condition that exists where the frame can be removed from the frame_queue and freed before the transmit is scheduled if we do not hold on to that lock. This results in a freed frame being scheduled for transmit later. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263151 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Richard Mudgett
|
274eb8960c |
Fix inverted logic in cli command: ss7 set debug on/off
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263069 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tzafrir Cohen
|
85299754b1 |
Remove "untested" feature PRI_VERSION
Nobody seems to actually test PRI_VERSION. It is only useful for failing PRI support in chan_dahdi. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263028 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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113c677257 |
For FreeBSD
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262987 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
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88a8703c37 |
Hmmm, probably should have read the manpage more thoroughly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262940 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Russell Bryant
|
c26cd3aaac |
Fix an off by one error that causes a crash.
Thanks to Raymond Burke for pointing it out. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262897 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Russell Bryant
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420acb8f0a |
Fix build on linux.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262896 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Russell Bryant
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7c4a95f2ea |
Fix build on linux.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262895 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
|
8d6ee962c7 |
Add kqueue(2) implementation to Asterisk in various places.
This will save a considerable amount of CPU on the BSDs, including Mac OS X, as it eliminates several places in the code that we previously used a busy loop. Additionally, this adds a res_timing interface, using kqueue timers. Review: https://reviewboard.asterisk.org/r/543/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262852 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Paul Belanger
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7d53dc86d6 |
Notify CLI when modules is loaded / unloaded
(closes issue #17308) Reported by: pabelanger Patches: cli.modules.patch uploaded by pabelanger (license 224) Tested by: pabelanger, russell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262800 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Leif Madsen
|
c17cda109a |
Revert previous WARNING message removal.
Marquis42 suggested a better method of doing what I wanted because I ended up removing the WARNING message for all instances when really I just wanted to remove it for the 'return' keyword, not everything. (issue #17145) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262798 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Leif Madsen
|
881450ec82 |
Remove unnecessary WARNING message in ael/pval.c
(closes issue #17145) Reported by: okrief git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262796 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Vossel
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a0b12a5666 |
Merged revisions 262662 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r262662 | dvossel | 2010-05-12 12:00:04 -0500 (Wed, 12 May 2010) | 11 lines fixes app_meetme dsp error We attempted to detect silence after translating a frame from signed linear. This caused a flooding of errors. To resolve this the code to detect silence was moved before the translation. (closes issue #17133) Reported by: jsdyer ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262744 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Richard Mudgett
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093dbfdd52 |
Don't crash when destroying chan_dahdi pseudo channels.
Must do a deep copy of the cc_params in duplicate_pseudo(). Otherwise, when the duplicate pseudo channel is destroyed, it frees the original pseudo channel cc_params. The original pseudo channel is then left with a dangling pointer for when the next duplicated pseudo channel is created. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262743 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Richard Mudgett
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e2336b73ef |
Merged revisions 262657,262660 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r262660 | rmudgett | 2010-05-12 11:46:47 -0500 (Wed, 12 May 2010) | 4 lines Forgot some conditionals around the callrerouting facility help text. JIRA ABE-2223 .......... r262657 | rmudgett | 2010-05-12 11:26:49 -0500 (Wed, 12 May 2010) | 22 lines Add mISDN Call rerouting facility for point-to-point ISDN lines (exchange line) In the case of ISDN point-to-multipoint (multidevice) you can use the mISDN "facility calldeflect" application for call diversions from external (PSTN) to external (PSTN). In that case this is the only way to get rid of the two call legs to the PBX and let the calling number at the C party become the number of the A party. In the case of ISDN point-to-point (exchange line) the call deflection facility may not be used. Instead a call rerouting facility has to be used. This patch for chan_misdn.c is an extension to realize this service (facility rerouting application). It can accept either spelling: "callrerouting" or "callrerouteing". The patch is tested towards Deutsche Telekom and requires a modified version of mISDN from Digium, Inc. Patches: misdn_rerouteing_corrected.patch (Slightly modified.) JIRA ABE-2223 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262661 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
|
1d7a548ae6 |
Ensure the arguments are initialized. Also miscellaneous CG cleanup.
(closes issue #16576) Reported by: uxbod Patches: 20100505__issue16576.diff.txt uploaded by tilghman (license 14) Tested by: uxbod git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262656 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Paul Belanger
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4b1d9f85a7 |
Convert to AST_CLI_YESNO and AST_CLI_ONOFF
Clean up chan_sip.c to use new AST_CLI functions (closes issue #17287) Reported by: pabelanger Patches: issue17287.patch uploaded by pabelanger (license 224) Tested by: russell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262613 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Richard Mudgett
|
9534f72cb0 |
Dialing an invalid extension causes incomplete hangup sequence.
Revision -r1489 of the libpri 1.4 branch corrected a deviation from Q.931 Section 5.3.2. However, this resulted in an unexpected behaviour change to the upper layer (Asterisk). This change uses pri_hangup_fix_enable() to follow Q.931 Section 5.3.2 call hangup better if the version of libpri supports it. (issue #17104) Reported by: shawkris Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262569 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
|
2c10997e99 |
Move cause 200 to cause 26, as specified in Q.850.
Also cleanup the formatting and add a few more that seem like good candidates. (closes issue #16157) Reported by: wimpy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262513 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Jason Parker
|
d8dea9e76a |
Merged revisions 262421 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r262421 | qwell | 2010-05-11 14:55:42 -0500 (Tue, 11 May 2010) | 11 lines Use a less silly method for modifying a flex-generated file. The sed syntax that was used wasn't actually valid, causing some versions to choke. This is the method that is used in 1.6.x+ for similar changes. (closes issue #16696) Reported by: bklang Patches: 16696-sedfix.diff uploaded by qwell (license 4) Tested by: qwell ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262422 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Paul Belanger
|
663a368a87 |
Improve logging by displaying line number
(closes issue #16303) Reported by: dant Patches: issue16303.patch.v2 uploaded by pabelanger (license 224) Tested by: dant, lmadsen, pabelanger git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262419 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Paul Belanger
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9c012b460f |
Improve logging information for misconfigured contexts
(closes issue #17238) Reported by: pprindeville Patches: chan_sip-bug17238.patch uploaded by pprindeville (license 347) Tested by: pprindeville git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262414 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |