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${ noResults }
259 Commits (0f62296eb6ccffbedea9ca243d4985b5ad6839dd)
Author | SHA1 | Message | Date |
---|---|---|---|
|
596f8b5186 |
Merged revisions 123113 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123113 | tilghman | 2008-06-16 14:50:12 -0500 (Mon, 16 Jun 2008) | 2 lines Port "hasvoicemail" change from SIP to other channel drivers ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@123114 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
a68a160b75 |
Implement call parking in chan_skinny.
(closes issue #11342) Reported by: DEA Patches: chan_skinny-park.txt uploaded by DEA (license 3) chan_skinny-park-v2.diff.txt uploaded by mvanbaak (license 7) Tested by: DEA, mvanbaak git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122493 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
4741113bac |
formatting changes.
A lot of whitespace issues have been resolved in this commit Also some doc updates, but that's only 6 lines git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@118824 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
1bd776b0ea |
remove unused astobj.h header file from chan_skinny.c
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@118750 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
f1e9371da8 |
- revert change to ast_queue_hangup and create ast_queue_hangup_with_cause
- make data member of the ast_frame struct a named union instead of a void Recently the ast_queue_hangup function got a new parameter, the hangupcause Feedback came in that this is no good and that instead a new function should be created. This I did. The hangupcause was stored in the seqno member of the ast_frame struct. This is not very elegant, and since there's already a data member that one should be used. Problem is, this member was a void *. Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone wants to store another type in there in the future. This commit is so massive, because all ast_frame.data uses have to be altered to ast_frame.data.data Thanks russellb and kpfleming for the feedback. (closes issue #12674) Reported by: mvanbaak git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117802 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
df6cd7a879 |
Merged revisions 116799 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116799 | file | 2008-05-16 17:28:11 -0300 (Fri, 16 May 2008) | 4 lines Check to make sure an RTP structure exists before calling ast_rtp_new_source on it. (closes issue #12669) Reported by: sbisker ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116800 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
0ebec7fa4f |
Undo inadvertent changes to chan_skinny caused by the merging of urgent messaging
support. Thanks to Damien Wedhorn for pointing out the problem. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116138 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
7daebcd610 |
Adding support for "urgent" voicemail messages. Messages which are
marked "urgent" are considered to be higher priority than other messages and so they will be played before any other messages in a user's mailbox. There are two ways to leave an urgent message. 1. send the 'U' option to VoiceMail(). 2. Set review=yes in voicemail.conf. This will give instructions for a caller to mark a message as urgent after the message has been recorded. I have tested that this works correctly with file and ODBC storage, and James Rothenberger (who wrote initial support for this feature) has tested its use with IMAP storage. (closes issue #11817) Reported by: jaroth Based on branch http://svn.digium.com/svn/asterisk/team/jrothenberger/asterisk-urgent Tested by: putnopvut, jaroth git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115588 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
dff07f833c |
s/chan_zap/chan_skinny/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114703 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
4cd37243ff |
Make MWI in chan_skinny event based modeled after chan_zap and chan_mgcp.
(closes issue #12214) Reported by: DEA Patches: chan_skinny-vm-events-v3.txt uploaded by DEA (license 3) Tested by: DEA and me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114700 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
08e674bce0 |
Pass the hangup cause all the way to the calling app/channel.
(closes issue #11328) Reported by: rain Patches: 20071207__pass_cause_in_hangup_control_frame.diff.txt uploaded by Corydon76 (license 14) brought up-to-date to trunk by me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114637 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
41fd7a6a21 |
(closes issue #6113)
Reported by: oej Tested by: jpeeler This patch implements multiple parking lots for parked calls. The default parkinglot is used by default, however setting the channel variable PARKINGLOT in the dialplan will allow use of any other configured parkinglot. See configs/features.conf.sample for more details on setting up another non-default parkinglot. Also, one can (currently) set the default parkinglot to use in the driver configuration file via the parkinglot option. Patch initially written by oej, brought up to date and finalized by mvanbaak, and then stabilized and converted to astobj2 by me. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114487 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
4a21c5dd22 |
Fix spelling of existent in a few places.
(closes issue #12409) Reported by: candlerb git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114024 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
d314fd5336 |
Move all messages wrapped in skinnydebug from debug to verbose.
(closes issue #12224) Reported by: DEA Patches: chan_skinny-debug-log.txt uploaded by DEA (license 3) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113834 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
b52ec53da7 |
Merged revisions 113504 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113504 | qwell | 2008-04-08 13:48:55 -0500 (Tue, 08 Apr 2008) | 1 line Add a little more that is required for previously added devices. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113505 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
f469ee8cf2 |
Merged revisions 113454 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113454 | qwell | 2008-04-08 13:07:49 -0500 (Tue, 08 Apr 2008) | 4 lines Add support for several new(ish) devices - most notably, 7942/7945, 7962/7965, 7975. Thanks to Greg Oliver for providing me the required information. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113455 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
63f574ceb4 |
Merged revisions 113118 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113118 | qwell | 2008-04-07 13:00:09 -0500 (Mon, 07 Apr 2008) | 8 lines Allow playback with noanswer (and add earlyrtp option). (closes issue #9077) Reported by: pj Patches: earlyrtp.diff uploaded by wedhorn (license 30) Tested by: pj, qwell, DEA, wedhorn ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113119 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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5591c696a5 |
Merged revisions 111720 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111720 | qwell | 2008-03-28 12:55:05 -0500 (Fri, 28 Mar 2008) | 1 line Remove unimplemented softkeys. Prompted by issue #12325. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111721 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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5729b0ec40 |
Update the directory of placed calls on skinny phones
when dialing a channel that does not provide progress (analog ZAP lines) The phone does handle the double update on calls to channels that do provide progress and wont insert duplicate items (closes issue #12239) Reported by: DEA Patches: chan_skinny-call-log.txt uploaded by DEA (license 3) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109168 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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377e51c4d4 |
(closes issue #6002)
Reported by: rizzo Tested by: murf Proposal of the changes to be made, and then an announcement of how they were accomplished: http://lists.digium.com/pipermail/asterisk-dev/2008-February/032065.html and: http://lists.digium.com/pipermail/asterisk-dev/2008-March/032124.html Here is a recap, file by file, of what I have done: pbx/pbx_config.c pbx/pbx_ael.c All funcs that were passed a ptr to the context list, now will ALSO be passed a hashtab ptr to the same set. Why? because (for the time being), the dialplan is stored in both, to facilitate a quick, low-cost move to hash-tables to speed up dialplan processing. If it was deemed necessary to pass the context LIST, well, it is just as necessary to have the TABLE available. This is because the list/table in question might not be the global one, but temporary ones we would use to stage the dialplan on, and then swap into the global position when things are ready. We now have one external function for apps to use, "ast_context_find_or_create()" instead of the pre-existing "find" and "create", as all existing usages used both in tandem anyway. pbx_config, and pbx_ael, will stage the reloaded dialplan into local lists and tables, and then call merge_contexts_and_delete, which will merge (now) existing contexts and priorities from other registrars into this local set by copying them. Then, merge_contexts_and_delete will lock down the contexts, swap the lists and tables, and unlock (real quick), and then destroy the old dialplan. chan_sip.c chan_iax.c chan_skinny.c All the channel drivers that would add regcontexts now use the ast_context_find_or_create now. chan_sip also includes a small fix to get rid of warnings about removing priorities that never got entered. apps/app_meetme.c apps/app_dial.c apps/app_queue.c All the apps that added a context/exten/priority were also modified to use ast_context_find_or_create instead. include/asterisk/pbx.h ast_context_create() is removed. Find_or_create_ is the new method. ast_context_find_or_create() interface gets the hashtab added. ast_merge_contexts_and_delete() gets the local hashtab arg added. ast_wrlock_contexts_version() is added so you can detect if someone else got a writelock between your readlocking and writelocking. ast_hashtab_compare_contexts was made public for use in pbx_config/pbx_ael ast_hashtab_hash_contexts was in like fashion make public. include/asterisk/pval.h ast_compile_ael2() interface changed to include the local hashtab table ptr. main/features.c For the sake of the parking context, we use ast_context_find_or_create(). main/pbx.c I changed all the "tree" names to "table" instead. That's because the original implementation was based on binary trees. (had a free library). Then I moved to hashtabs. Now, the names move forward too. refcount field added to contexts, so you can keep track of how many modules wanted this context to exist. Some log messages that are warnings were inflated from LOG_NOTICE to LOG_WARNING. Added some calls to ast_verb(3,...) for debug messages Lots of little mods to ast_context_remove_extension2, which is now excersized in ways it was not previously; one definite bug fixed. find_or_create was upgraded to handle both local lists/tables as well as the globals. context_merge() was added to do the per-context merging of the old/present contexts/extens/prios into the new/proposed local list/tables ast_merge_contexts_and_delete() was heavily modified. ast_add_extension2() was also upgraded to handle changes. the context_destroy() code was re-engineered to handle the new way of doing things, by exten/prio instead of by context. res/ael/pval.c res/ael/ael.tab.c res/ael/ael.tab.h res/ael/ael.y res/ael/ael_lex.c res/ael/ael.flex utils/ael_main.c utils/extconf.c utils/conf2ael.c utils/Makefile Had to change the interface to ast_compile_ael2(), to include the hashtab ptr. This ended up involving several external apps. The main gotcha was I had to include lock.h and hashtab.h in several places. As a side note, I tested this stuff pretty thoroughly, I replicated the problems originally reported by Luigi, and made triply sure that reloads worked, and everything worked thru "stop gracefully". I found a and fixed a few bugs as I was merging into trunk, that did not appear in my tests of bug6002. How's this for verbose commit messages? git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106757 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
496adc6fc0 |
Merged revisions 106235 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106235 | file | 2008-03-05 18:32:10 -0400 (Wed, 05 Mar 2008) | 4 lines Add a control frame to indicate the source of media has changed. Depending on the underlying technology it may need to change some things. (closes issue #12148) Reported by: jcomellas ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106239 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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decec84c56 |
Merged revisions 104920 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104920 | qwell | 2008-02-27 22:31:21 -0600 (Wed, 27 Feb 2008) | 2 lines According to a video at www.cisco.com, the 7921G supports 6 line appearances. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104921 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
62ac882f8c |
move two ast_log calls to ast_debug.
Now monitoring chan_skinny port with nagios or zabbix wont generate noise on the console. @ok tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103750 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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a684207428 |
Change chan_skinny to use debug messages as appropriate.
(closes issue #11967) Reported by: mvanbaak Patches: 2008021000-skinnydebug.diff.txt uploaded by mvanbaak (license 7) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103313 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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1ec8cb41a8 |
Merge changes from team/mvanbaak/cli-command-audit
(closes issue #8925) About a year ago, as Leif Madsen and Jim van Meggelen were going over the CLI commands in Asterisk 1.4 for the next version of their book, they documented a lot of inconsistencies. This set of changes addresses all of these issues and has been reviewed by Leif. While this does introduce even more changes to the CLI command structure, it makes everything consistent, which is the most important thing. Thanks to all that helped with this one! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103171 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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524aed3768 |
Add proper call forwarding (all and busy) support for chan_skinny.
Note: NoAnswer support is currently not implemented, as it would take a significant amount of work to figure out how to do correctly. Closes issue #11310, patches, testing, and support by DEA, mvanbaak, and myself. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98776 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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d357378397 |
Fix for potential crash with vmexten
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98773 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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17afebc1a6 |
HUGE improvements to QoS/CoS handling by IgorG
- Refer to the proper documentation - Implement separate signalling/media QoS/CoS in many channels using RTP - Improve warnings and verbose messages - Deprecate some old settings Minor modifications by me, a big effort from IgorG. Thanks! Reported by: IgorG Patches: qoscleanup-89394-4-trunk.patch uploaded by IgorG (license 20) Tested by: IgorG (closes issue #11145) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93163 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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d5b454bf8d |
Convert ast_verbose to ast_verb.
Reported by: snuffy Patch by: snuffy (Closes issue #11547) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92913 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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86476c607f |
closes issue #11285, where an unload of a module that creates a dialplan context, causes a crash when you do a 'dialplan show' of that context. This is because the registrar string is defined in the module, and the stale pointer is traversed. The reporter offered a patch that would always strdup the registrar string, which is practical, but I preferred to destroy the created contexts in each module where one is created. That seemed more symmetric. There were only 6 place in asterisk where this is done: chan_sip, chan_iax2, chan_skinny, res_features, app_dial, and app_queue. The two apps destroyed the context, but left the contexts. All is fixed now and unloads should be dialplan friendly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89513 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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7e8835e0d7 |
remove another set of redundant #include "asterisk/options.h"
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89512 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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0595b5e2aa |
include "logger.h" and errno.h from asterisk.h - usage shows that they
were included almost everywhere. Remove some of the instances. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89424 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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d82a631f9c |
more removal of duplicate #include lines
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89349 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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fdb7f7ba3d |
Start untangling header inclusion in a way that does not affect
build times - tested, there is no measureable difference before and after this commit. In this change: use asterisk/compat.h to include a small set of system headers: inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h, stdlib.h, alloca.h, stdio.h Where available, the inclusion is conditional on HAVE_FOO_H as determined by autoconf. Normally, source files should not include any of the above system headers, and instead use either "asterisk.h" or "asterisk/compat.h" which does it better. For the time being I have left alone second-level directories (main/db1-ast, etc.). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89333 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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7f8ecd2cd3 |
make the 'name' and 'value' fields in ast_variable const char *
This prevents modifying the strings in the stored variables, and catched a few instances where this was actually done. Given the differences between trunk and 1.4 (and the fact that this is effectively an API change) it is better to fix 1.4 independently. These are chan_sip.c::sip_register() chan_skinny.c:: near line 2847 config.c:: near line 1774 logger.c::make_components() res_adsi.c:: near line 1049 I may have missed some instances for modules that do not build here. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89268 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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7c56918262 |
Commit some cleanups to the format type code.
- Remove the AST_FORMAT_MAX_* types, as these are consuming 3 out of our available 32 bits. - Add a native slin16 type, so that 16kHz codecs can translate without losing resolution. (This doesn't affect anything immediately, until another codec has wb support.) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89071 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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ebe4050128 |
Switch from AST_CLI (formerly NEW_CLI) to AST_CLI_DEFINE, since the former didn't make much sense
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86820 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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b0f3e6097e |
Convert NEW_CLI to AST_CLI.
Closes issue #11039, as suggested by seanbright. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86536 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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e97a723cf1 |
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :)
(closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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d78463be1e |
Corydon posted this janitor project to the bug tracker and mvanbaak provided
a patch for it. It replaces a bunch of simple calls to snprintf with ast_copy_string (closes issue #10843) Reported by: Corydon76 Patches: 2007092900_10843.diff uploaded by mvanbaak (license 7) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@84173 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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9388173f85 |
Make the MALLOC_DEBUG output for free() useful again. After changing calls to
free to be ast_free, astmm said all calls to free were coming from utils.h git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82628 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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4baba7c951 |
Add support in chan_skinny for sending RTP directly to the endpoints.
Closes issue #9154, patch by DEA git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82401 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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f614bc7004 |
(closes issue #10377)
Reported by: mvanbaak Patches: chan_skinny_info.diff uploaded by mvanbaak (license 7) Add skinny show device, skinny show line, and skinny show settings CLI commands. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@81782 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
b5cd67adc3 |
This code was in team/murf/bug8684-trunk; it should fix bug 8684 in trunk. I didn't add it to 1.4 yet, because it's not entirely clear to me if this is a bug fix or an enhancement. A lot of files were affected by small changes like ast_variable_new getting an added arg, for the file name the var was defined in; ast_category_new gets added args of filename and lineno; ast_category and ast_variable structures now record file and lineno for each entry; a list of all #include and #execs in a config file (or any of its inclusions are now kept in the ast_config struct; at save time, each entry is put back into its proper file of origin, in order. #include and #exec directives are folded in properly. Headers indicating that the file was generated, are generated also for each included file. Some changes to main/manager.c to take care of file renaming, via the UpdateConfig command. Multiple inclusions of the same file are handled by exploding these into multiple include files, uniquely named. There's probably more, but I can't remember it right now.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@81361 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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56b9568164 |
Don't reload a configuration file if nothing has changed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@79747 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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d5eda8709c |
Merged revisions 79174 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79174 | file | 2007-08-13 11:18:04 -0300 (Mon, 13 Aug 2007) | 4 lines (closes issue #10437) Reported by: haklin Don't set the callerid name and number a second time on a newly created channel. ast_channel_alloc itself already sets it and setting it twice would cause a memory leak. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@79175 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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9b3c3f1c61 |
Use localized softkey labels.
Add some information about localization "codes". git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@79109 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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90066ea949 |
Minor fix for building under dev mode when byteswapping macro header files are not available.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@78685 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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22114b509d |
Add support for using epoll instead of poll. This should increase scalability and is done in such a way that we should be able to add support for other poll() replacements.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@78683 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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395243c923 |
Merged revisions 78375 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r78375 | qwell | 2007-08-07 13:25:15 -0500 (Tue, 07 Aug 2007) | 3 lines Properly check the capabilities count to avoid a segfault. (ASA-2007-019) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@78378 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |