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authorrussell <russell@f38db490-d61c-443f-a65b-d21fe96a405b>2008-09-03 14:56:56 +0000
committerrussell <russell@f38db490-d61c-443f-a65b-d21fe96a405b>2008-09-03 14:56:56 +0000
commit7ed78f04b53d344afa90ce0c0a51c749e2a3807f (patch)
tree4681a61e84c40398770f7f2d1bd83be1eb5aeafb
parente4ad898ea5bf4d5c7739eb10543a795160077dcb (diff)
parentea098db1f6eee69e908ec28c078cee1e5f0815a3 (diff)
Creating tag for the release of asterisk-1.6.0-rc4
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.0-rc4@140897 f38db490-d61c-443f-a65b-d21fe96a405b
-rw-r--r--.lastclean1
-rw-r--r--.version1
-rw-r--r--ChangeLog46549
-rw-r--r--main/cdr.c11
-rw-r--r--main/channel.c6
-rw-r--r--main/pbx.c1
-rw-r--r--main/poll.c10
7 files changed, 12 insertions, 46567 deletions
diff --git a/.lastclean b/.lastclean
deleted file mode 100644
index 8f92bfdd4..000000000
--- a/.lastclean
+++ /dev/null
@@ -1 +0,0 @@
-35
diff --git a/.version b/.version
deleted file mode 100644
index 1a93a4424..000000000
--- a/.version
+++ /dev/null
@@ -1 +0,0 @@
-1.6.0-rc4
diff --git a/ChangeLog b/ChangeLog
deleted file mode 100644
index 5a5cda1c8..000000000
--- a/ChangeLog
+++ /dev/null
@@ -1,46549 +0,0 @@
-2008-09-03 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.6.0-rc4 released.
-
-2008-09-02 18:17 +0000 [r140607] Sean Bright <sean.bright@gmail.com>
-
- * /, channels/chan_iax2.c: Merged revisions 140606 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r140606 | seanbright | 2008-09-02 14:15:54 -0400
- (Tue, 02 Sep 2008) | 16 lines Merged revisions 140605 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r140605 | seanbright | 2008-09-02 14:14:57 -0400 (Tue, 02 Sep
- 2008) | 8 lines Make sure to use the correct length of the
- mohinterpret and mohsuggest buffers when copying configuration
- values. (closes issue #13336) Reported by:
- decryptus_proformatique Patches:
- chan_iax2_mohinterpret_mohsuggest_general_settings.patch uploaded
- by decryptus (license 555) ........ ................
-
-2008-09-02 15:12 +0000 [r140564-140567] Russell Bryant <russell@digium.com>
-
- * apps/app_jack.c, /, codecs/codec_resample.c: Merged revisions
- 140566 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r140566 |
- russell | 2008-09-02 10:11:53 -0500 (Tue, 02 Sep 2008) | 2 lines
- Update instructions for getting libresample ........
-
-2008-08-27 20:15 +0000 [r140302-140304] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: Revert commit 140302. Should not be merging
- changes like that into a release-candidate branch
-
- * channels/chan_sip.c: Merged revisions 140301 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r140301 | mmichelson | 2008-08-27 15:11:22 -0500 (Wed, 27 Aug
- 2008) | 19 lines Merged revisions 140299 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r140299 | mmichelson | 2008-08-27 14:49:20 -0500 (Wed, 27 Aug
- 2008) | 11 lines Fix tag checking in get_sip_pvt_byid_locked when
- in pedantic mode. The problem was that the wrong tags would be
- compared depending on the direction of the call. (closes issue
- #13353) Reported by: flefoll Patches:
- chan_sip.c.br14.139015.patch-refer-pedantic uploaded by flefoll
- (license 244) ........ ................
-
-2008-08-26 18:12 +0000 [r140170] Russell Bryant <russell@digium.com>
-
- * Makefile, /: Merged revisions 140169 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r140169 |
- russell | 2008-08-26 13:11:49 -0500 (Tue, 26 Aug 2008) | 4 lines
- Fix building menuselect-tree with PRINT_DIR set. We _must_ use
- the --quiet flag here, or else some arbitrary text will end up in
- the resulting menuselect-tree file and things will explode.
- ........
-
-2008-08-25 21:33 +0000 [r139918] Sean Bright <sean.bright@gmail.com>
-
- * build_tools/get_moduleinfo, /, build_tools/get_makeopts: Merged
- revisions 139915 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r139915 | seanbright | 2008-08-25 17:32:10 -0400 (Mon, 25 Aug
- 2008) | 17 lines Merged revisions 139909 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r139909 | seanbright | 2008-08-25 17:31:03 -0400 (Mon, 25 Aug
- 2008) | 9 lines Some versions of awk (nawk, for example) don't
- like empty regular expressions so be slightly more verbose.
- (closes issue #13374) Reported by: dougm Patches: 13374.diff
- uploaded by seanbright (license 71) Tested by: dougm ........
- ................
-
-2008-08-25 21:05 +0000 [r139872] Terry Wilson <twilson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 139870 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r139870 | twilson | 2008-08-25 15:59:58 -0500 (Mon, 25 Aug 2008)
- | 10 lines Merged revisions 139869 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r139869 | twilson | 2008-08-25 15:46:10 -0500 (Mon, 25 Aug 2008)
- | 2 lines Make SIPADDHEADER() propagate indefinitely ........
- ................
-
-2008-08-25 16:00 +0000 [r139774] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, /, main/features.c: Merged revisions 139770 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r139770 | murf | 2008-08-25 09:54:18 -0600 (Mon,
- 25 Aug 2008) | 17 lines Merged revisions 139764 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r139764 | murf | 2008-08-25 09:33:14 -0600 (Mon, 25 Aug 2008) | 9
- lines This patch reverts the changes made via 139347, and 139635,
- as users are seeing adverse difference. I will un-close 13251.
- Back to the drawing board/ concept/ beginning/ whatever! ........
- ................
-
-2008-08-24 16:30 +0000 [r139705-139708] Tilghman Lesher <tlesher@digium.com>
-
- * /, cdr/cdr_pgsql.c: Merged revisions 139707 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r139707 |
- tilghman | 2008-08-24 11:26:48 -0500 (Sun, 24 Aug 2008) | 2 lines
- Memory leak ........
-
-2008-08-22 22:35 +0000 [r139628-139671] Steve Murphy <murf@digium.com>
-
- * /, main/features.c: Merged revisions 139662 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r139662 | murf | 2008-08-22 16:32:35 -0600 (Fri, 22 Aug 2008) |
- 14 lines Merged revisions 139635 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r139635 | murf | 2008-08-22 16:24:02 -0600 (Fri, 22 Aug 2008) | 6
- lines I found some problems with the code I committed earlier,
- when I merged them into trunk, so I'm coming back to clean up.
- And, in the process, I found an error in the code I added to
- trunk and 1.6.x, that I'll fix using this patch also. ........
- ................
-
- * apps/app_dial.c, main/pbx.c, /, main/features.c: Merged revisions
- 139627 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r139627 | murf | 2008-08-22 16:03:13 -0600 (Fri, 22 Aug 2008) |
- 59 lines Merged revisions 139347 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r139347 | murf | 2008-08-21 17:03:50 -0600 (Thu, 21 Aug 2008) |
- 47 lines (closes issue #13251) Reported by: sergee Tested by:
- murf THis is a bold move for a static release fix, but I wouldn't
- have made it if I didn't feel confident (at least a *bit*
- confident) that it wouldn't mess everyone up. The reasoning goes
- something like this: 1. We simply cannot do anything with CDR's
- at the current point (in pbx.c, after the __ast_pbx_run loop).
- It's way too late to have any affect on the CDRs. The CDR is
- already posted and gone, and the remnants have been cleared. 2. I
- was very much afraid that moving the running of the 'h' extension
- down into the bridge code (where it would be now practical to do
- it), would result in a lot more calls to the 'h' exten, so I
- implemented it as another exten under another name, but found, to
- my pleasant surprise, that there was a 1:1 correspondence to the
- running of the 'h' exten in the pbx_run loop, and the new spot at
- the end of the bridge. So, I ifdef'd out the current 'h' loop,
- and moved it into the bridge code. The only difference I can see
- is the stuff about the AST_PBX_KEEPALIVE, and hopefully, if this
- is still an important decision point, I can replicate it if there
- are complaints. To be perfectly honest, the KEEPALIVE situation
- is not totally clear to me, and how it relates to a post-bridge
- situation is less clear. I suspect the users will point out
- everything in total clarity if this steps on anyone's toes! 3. I
- temporarily swap the bridge_cdr into the channel before running
- the 'h' exten, which makes it possible for users to edit the cdr
- before it goes out the door. And, of course, with the
- endbeforehexten config var set, the users can also get at the
- billsec/duration vals. After the h exten finishes, the cdr is
- swapped back and processing continues as normal. Please, all who
- deal with CDR's, please test this version of Asterisk, and file
- bug reports as appropriate! ........ I also made a little fix to
- the app_dial's 'e' option, that is related to my updates.
- ................
-
-2008-08-22 20:21 +0000 [r139458-139564] Mark Michelson <mmichelson@digium.com>
-
- * include/asterisk/threadstorage.h, /: Merged revisions 139554 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r139554 | mmichelson | 2008-08-22 14:45:41 -0500
- (Fri, 22 Aug 2008) | 16 lines Merged revisions 139553 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r139553 | mmichelson | 2008-08-22 14:45:19 -0500 (Fri, 22 Aug
- 2008) | 8 lines Fix compilation when DEBUG_THREAD_LOCALS is
- selected (closes issue #13298) Reported by: snuffy Patches:
- bug13298_20080822.diff uploaded by snuffy (license 35) ........
- ................
-
- * /, channels/chan_iax2.c: Merged revisions 139469 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r139469 | mmichelson | 2008-08-22 12:25:12 -0500
- (Fri, 22 Aug 2008) | 11 lines Merged revisions 139466 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r139466 | mmichelson | 2008-08-22 12:24:47 -0500 (Fri, 22 Aug
- 2008) | 3 lines Fix the build. Thanks, mvanbaak! ........
- ................
-
- * /, channels/chan_iax2.c: Merged revisions 139457 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r139457 | mmichelson | 2008-08-22 11:58:21 -0500
- (Fri, 22 Aug 2008) | 15 lines Merged revisions 139456 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r139456 | mmichelson | 2008-08-22 11:57:38 -0500 (Fri, 22 Aug
- 2008) | 7 lines Prevent a deadlock in chan_iax2 resulting from
- incorrect locking order between iax2_pvt and ast_channel
- structures. AST-13 ........ ................
-
-2008-08-21 23:46 +0000 [r139400] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 139391 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r139391 | jpeeler | 2008-08-21 18:41:50 -0500
- (Thu, 21 Aug 2008) | 11 lines Merged revisions 139387 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r139387 | jpeeler | 2008-08-21 18:39:31 -0500 (Thu, 21 Aug 2008)
- | 3 lines Fixes loop that could possibly never exit in the event
- of a channel never being able to be opened or specify after a
- restart. (closes issue #11017) ........ ................
-
-2008-08-21 10:02 +0000 [r139282] Philippe Sultan <philippe.sultan@gmail.com>
-
- * /, channels/chan_gtalk.c: Merged revisions 139281 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r139281 | phsultan | 2008-08-21 11:55:31 +0200 (Thu, 21 Aug 2008)
- | 5 lines Fix two memory leaks in chan_gtalk, thanks Eliel!
- (closes issue #13310) Reported by: eliel Patches:
- chan_gtalk.c.patch uploaded by eliel (license 64) ........
-
-2008-08-20 Kevin P. Fleming <kpfleming@digium.com>
-
- * Asterisk 1.6.0-rc3 released.
-
-2008-08-20 22:17 +0000 [r139216] Russell Bryant <russell@digium.com>
-
- * apps/app_chanspy.c, /: Merged revisions 139215 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r139215 | russell | 2008-08-20 17:16:36 -0500 (Wed, 20 Aug 2008)
- | 19 lines Merged revisions 139213 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r139213 | russell | 2008-08-20 17:14:35 -0500 (Wed, 20 Aug 2008)
- | 11 lines Fix a crash in the ChanSpy application. The issue here
- is that if you call ChanSpy and specify a spy group, and sit in
- the application long enough looping through the channel list, you
- will eventually run out of stack space and the application with
- exit with a seg fault. The backtrace was always inside of a
- harmless snprintf() call, so it was tricky to track down.
- However, it turned out that the call to snprintf() was just the
- biggest stack consumer in this code path, so it would always be
- the first one to hit the boundary. (closes issue #13338) Reported
- by: ruddy ........ ................
-
-2008-08-20 20:12 +0000 [r139155] Shaun Ruffell <sruffell@digium.com>
-
- * codecs/codec_dahdi.c: Fix bug where the samples were not accurate
- when in G723 mode, which would cause the timestamp field of the
- RTP header to be invalid.
-
-2008-08-20 17:30 +0000 [r139104] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, /: Merged revisions 139083 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r139083 | murf | 2008-08-20 11:25:07 -0600 (Wed, 20 Aug 2008) |
- 20 lines Merged revisions 139074 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r139074 | murf | 2008-08-20 11:14:55 -0600 (Wed, 20 Aug 2008) |
- 12 lines (closes issue #13263) Reported by: brainy Tested by:
- murf The specialized reset routine is tromping on the flags field
- of the CDR. I made a change to not reset the DISABLED bit. This
- should get rid of this problem. ........ ................
-
-2008-08-20 15:39 +0000 [r138889-139017] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 139016 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r139016 | mmichelson | 2008-08-20 10:38:47 -0500 (Wed, 20 Aug
- 2008) | 14 lines Merged revisions 139015 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r139015 | mmichelson | 2008-08-20 10:37:56 -0500 (Wed, 20 Aug
- 2008) | 6 lines sip_read should properly handle a NULL return
- from sip_rtp_read. (closes issue #13257) Reported by: travishein
- ........ ................
-
- * apps/app_chanspy.c: Manually add revision 138887 from trunk to
- the 1.6.0 branch. I had misunderstood the policy for when to
- merge to 1.6.0 since it moved to rc status.
-
-2008-08-19 16:38 +0000 [r138846-138847] Steve Murphy <murf@digium.com>
-
- * utils/conf2ael.c, /, res/ael/ael.tab.c, res/ael/ael.y,
- res/ael/ael.tab.h, utils/ael_main.c: Merged revisions 138845 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r138845 | murf | 2008-08-19 10:31:24 -0600 (Tue, 19 Aug
- 2008) | 1 line Oops. put a decl in a generated file. My bad, but
- fixed now. ........
-
- * main/pbx.c, /, res/ael/ael.tab.c, res/ael/ael.y,
- res/ael/ael.tab.h: Merged revisions 138815 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r138815 |
- murf | 2008-08-19 09:59:12 -0600 (Tue, 19 Aug 2008) | 19 lines
- These changes are in regards to bug 13249, where users are being
- surprised by the changes made to the Set app in trunk/1.6.x, as
- they come from the 1.4 world. They are only bitten if they write
- their AEL dialplan in the 1.4 world, and then carry it over to a
- trunk/1.6.x installation where a "make samples" was executed, or
- where they hand-edited the asterisk.conf file and added the
- [compat] category with app_set = 1.6 (or higher). (this commit
- does not totally solve 13249, at least not yet) The change
- involves issueing a single warning while the AEL file is loading,
- if: 1. app_set is present in the config file, and set to 1.6 or
- higher. 2. there are double quotes in an assignment statement (eg
- x = "hi there";) 3. the warning was not already issued. The
- standalone app, aelparse, does not (yet) issue this warning. I'd
- have to have it read in the asterisk.conf file, and that's a bit
- of hassle. I'll add it if users request it, tho. ........
-
-2008-08-19 00:15 +0000 [r138776-138781] Sean Bright <sean.bright@gmail.com>
-
- * /, channels/chan_sip.c: Merged revisions 138778-138780 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r138778 | seanbright | 2008-08-18 20:08:27 -0400 (Mon,
- 18 Aug 2008) | 1 line While we're at it, make this machine
- parseable too. ........ r138779 | seanbright | 2008-08-18
- 20:09:38 -0400 (Mon, 18 Aug 2008) | 1 line And remove code we
- don't need anymore. ........ r138780 | seanbright | 2008-08-18
- 20:10:56 -0400 (Mon, 18 Aug 2008) | 1 line Let it compile now,
- too (woops) ........
-
- * /, channels/chan_sip.c: Merged revisions 138775 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r138775 |
- seanbright | 2008-08-18 19:42:36 -0400 (Mon, 18 Aug 2008) | 3
- lines Change event header to RegistrationTime to be more
- consistent (and avoid breaking existing frameworks). Pointed out
- by Laureano on #asterisk-dev. ........
-
-2008-08-18 20:23 +0000 [r138688-138695] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 138687 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r138687 | mmichelson | 2008-08-18 15:04:10 -0500 (Mon, 18 Aug
- 2008) | 18 lines Merged revisions 138685 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r138685 | mmichelson | 2008-08-18 15:01:14 -0500 (Mon, 18 Aug
- 2008) | 10 lines Change the inequalities used in app_queue with
- regards to timeouts from being strict to non-strict for more
- accuracy. (closes issue #13239) Reported by: atis Patches:
- app_queue_timeouts_v2.patch uploaded by atis (license 242)
- ........ ................
-
-2008-08-18 15:54 +0000 [r138632] Jason Parker <jparker@digium.com>
-
- * Makefile, /: Merged revisions 138631 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r138631 |
- qwell | 2008-08-18 10:54:07 -0500 (Mon, 18 Aug 2008) | 1 line
- Remove option that isn't valid here. ........
-
-2008-08-18 02:14 +0000 [r138519] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 138518 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r138518 | jpeeler | 2008-08-17 21:13:04 -0500 (Sun, 17 Aug 2008)
- | 1 line add missing define for SS7 in dahdi_restart ........
-
-2008-08-17 14:14 +0000 [r138443-138483] Sean Bright <sean.bright@gmail.com>
-
- * /, main/features.c: Merged revisions 138482 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r138482 |
- seanbright | 2008-08-17 10:12:11 -0400 (Sun, 17 Aug 2008) | 6
- lines Move Uniqueid to the end of the event for those that rely
- on the position of the name/value pairs, pointed out by
- snuffy-home on #asterisk-commits. For those of you who rely on
- the position of name/value pairs in manager events... stop...
- that is why associative arrays were invented. ........
-
- * /, main/features.c: Merged revisions 138479 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r138479 |
- seanbright | 2008-08-17 09:51:08 -0400 (Sun, 17 Aug 2008) | 7
- lines Add Uniqueid header to ParkedCall manager event. (closes
- issue #13323) Reported by: srt Patches:
- 13323_unique_id_for_parkedcalls_event.diff uploaded by srt
- (license 378) ........
-
- * main/rtp.c, /: Merged revisions 138476 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r138476 |
- seanbright | 2008-08-17 09:40:36 -0400 (Sun, 17 Aug 2008) | 7
- lines Add missing colons to RTCPReceived and RTCPSent manager
- events. (closes issue #13319) Reported by: srt Patches:
- 13319_rtcp_manager_event_headers.diff uploaded by srt (license
- 378) ........
-
- * /, channels/chan_iax2.c: Merged revisions 138473 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r138473 | seanbright | 2008-08-17 09:31:54 -0400 (Sun, 17 Aug
- 2008) | 7 lines Fix the output of the JitterBufStats manager
- event. (closes issue #13324) Reported by: srt Patches:
- 13324_missing_nl_in_jitterbufstats_event_2.diff uploaded by srt
- (license 378) ........
-
- * configs/cdr_tds.conf.sample, /: Merged revisions 138442 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r138442 | seanbright | 2008-08-16 12:40:43 -0400 (Sat,
- 16 Aug 2008) | 4 lines Since it's introduction in revision 3497,
- cdr_tds has *never* read the port configuration option from
- cdr_tds.conf. So go ahead and remove it from the sample config.
- ........
-
-2008-08-16 13:07 +0000 [r138410-138413] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 138412 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r138412 | tilghman | 2008-08-16 08:07:08 -0500 (Sat, 16 Aug 2008)
- | 2 lines Fix compilation warnings (found with dev-mode) ........
-
-2008-08-16 01:14 +0000 [r138333-138362] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 138361 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r138361 | jpeeler | 2008-08-15 20:13:26 -0500
- (Fri, 15 Aug 2008) | 9 lines Merged revisions 138360 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r138360 | jpeeler | 2008-08-15 20:12:18 -0500 (Fri, 15
- Aug 2008) | 1 line fixes use count to properly decrement if an
- active dahdi channel is destroyed allowing module to be unloaded
- ........ ................
-
- * channels/chan_dahdi.c, /: Merged revisions 138311 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r138311 | jpeeler | 2008-08-15 18:46:09 -0500
- (Fri, 15 Aug 2008) | 20 lines Merged revisions
- 138119,138151,138238 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r138119 | jpeeler | 2008-08-15 14:21:51 -0500 (Fri, 15 Aug 2008)
- | 4 lines Fixes the dahdi restart functionality. Dahdi restart
- allows one to restart all DAHDI channels, even if they are
- currently in use. This is different from unloading and then
- loading the module since unloading requires the use count to be
- zero. Reloading the module is different in that the signalling is
- not changed from what it was originally configured. Also, this
- fixes not closing all the file descriptors for D-channels upon
- module unload (which would prevent loading the module
- afterwards). (closes issue #11017) ........ r138151 | jpeeler |
- 2008-08-15 14:41:29 -0500 (Fri, 15 Aug 2008) | 1 line declared
- static mutexes using AST_MUTEX_DEFINE_STATIC macro ........
- r138238 | jpeeler | 2008-08-15 16:28:26 -0500 (Fri, 15 Aug 2008)
- | 1 line initialize condition variable ss_thread_complete using
- ast_cond_init ........ ................
-
-2008-08-15 23:03 +0000 [r138207-138262] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
- 138260 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r138260 | tilghman | 2008-08-15 17:54:57 -0500 (Fri, 15 Aug 2008)
- | 16 lines Merged revisions 138258 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r138258 | tilghman | 2008-08-15 17:33:42 -0500 (Fri, 15 Aug 2008)
- | 8 lines More fixes for realtime peers. (closes issue #12921)
- Reported by: Nuitari Patches: 20080804__bug12921.diff.txt
- uploaded by Corydon76 (license 14) 20080815__bug12921.diff.txt
- uploaded by Corydon76 (license 14) Tested by: Corydon76 ........
- ................
-
- * configs/extensions.conf.sample, main/pbx.c, /: Merged revisions
- 138206 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r138206 |
- tilghman | 2008-08-15 15:35:24 -0500 (Fri, 15 Aug 2008) | 4 lines
- Remove deprecated syntax from sample config file (closes issue
- #13314) Reported by: kue ........
-
-2008-08-15 20:20 +0000 [r138156-138157] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c: rename all zfd instances in chan_dahdi to
- dfd to match 1.4 (left over from DAHDI transition)
-
-2008-08-15 15:12 +0000 [r138029] Russell Bryant <russell@digium.com>
-
- * main/autoservice.c, /: Merged revisions 138028 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r138028 | russell | 2008-08-15 10:09:46 -0500 (Fri, 15 Aug 2008)
- | 17 lines Merged revisions 138027 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r138027 | russell | 2008-08-15 10:07:16 -0500 (Fri, 15 Aug 2008)
- | 9 lines Ensure that when a hangup occurs in autoservice, that a
- hangup frame gets properly deferred to be read from the channel
- owner when it gets taken out of autoservice. (closes issue
- #12874) Reported by: dimas Patches: v1-12874.patch uploaded by
- dimas (license 88) ........ ................
-
-2008-08-15 15:04 +0000 [r138025] Tilghman Lesher <tlesher@digium.com>
-
- * /, funcs/func_strings.c: Merged revisions 138024 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r138024 | tilghman | 2008-08-15 10:03:32 -0500
- (Fri, 15 Aug 2008) | 16 lines Merged revisions 138023 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r138023 | tilghman | 2008-08-15 09:51:12 -0500 (Fri, 15 Aug 2008)
- | 8 lines Additional check for more string specifiers than
- arguments. (closes issue #13299) Reported by: adomjan Patches:
- 20080813__bug13299.diff.txt uploaded by Corydon76 (license 14)
- func_strings.c-sprintf.patch uploaded by adomjan (license 487)
- Tested by: adomjan ........ ................
-
-2008-08-14 22:43 +0000 [r137988] Russell Bryant <russell@digium.com>
-
- * /, doc/tex/Makefile: Merged revisions 137987 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r137987 |
- russell | 2008-08-14 17:43:15 -0500 (Thu, 14 Aug 2008) | 2 lines
- Fix a bashism that causes an error when trying to build the pdf
- on ubuntu ........
-
-2008-08-14 18:48 +0000 [r137934] Sean Bright <sean.bright@gmail.com>
-
- * cdr/cdr_sqlite3_custom.c, /: Merged revisions 137933 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r137933 | seanbright | 2008-08-14 14:47:28 -0400 (Thu, 14 Aug
- 2008) | 8 lines Fix memory leak in cdr_sqlite3_custom. (closes
- issue #13304) Reported by: eliel Patches: sqlite.patch uploaded
- by eliel (license 64) (Slightly modified by me) ........
-
-2008-08-14 17:01 +0000 [r137849-137852] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 137848 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r137848 | tilghman | 2008-08-14 11:52:43 -0500
- (Thu, 14 Aug 2008) | 17 lines Merged revisions 137847 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r137847 | tilghman | 2008-08-14 11:47:30 -0500 (Thu, 14 Aug 2008)
- | 9 lines When creating the secondary subchannel name, it is
- necessary to compare to the existing channel name without the
- "Zap/" or "DAHDI/" prefix, since our test string is also without
- that prefix. (closes issue #13027) Reported by: dferrer Patches:
- chan_zap-1.4.21.1_fix2.patch uploaded by dferrer (license 525)
- (Slightly modified by me, to compensate for both names) ........
- ................
-
-2008-08-14 Jason Parker <jparker@digium.com>
-
- * Asterisk 1.6.0-rc2 released.
-
-2008-08-14 15:37 +0000 [r137814] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 137812 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r137812 |
- qwell | 2008-08-14 10:32:16 -0500 (Thu, 14 Aug 2008) | 8 lines
- Make sure we set the socket port, so we don't try to use <ip
- address>:0. (closes issue #13255) Reported by: falves11 Patches:
- 13255-socketport.diff uploaded by qwell (license 4) Tested by:
- falves11 ........
-
-2008-08-14 15:20 +0000 [r137783] Russell Bryant <russell@digium.com>
-
- * /, configs/sip.conf.sample: Merged revisions 137732 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r137732 | russell | 2008-08-14 09:15:50 -0500
- (Thu, 14 Aug 2008) | 12 lines Merged revisions 137731 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r137731 | russell | 2008-08-14 09:05:23 -0500 (Thu, 14 Aug 2008)
- | 4 lines Comments in this config file were aligned only if your
- tab size was set to 8. So, convert tabs to spaces so that things
- should be aligned regardless of what tab size you use in your
- editor. ........ ................
-
-2008-08-14 15:05 +0000 [r137781] Sean Bright <sean.bright@gmail.com>
-
- * cdr/cdr_tds.c, /: Merged revisions 137780 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r137780 |
- seanbright | 2008-08-14 11:03:03 -0400 (Thu, 14 Aug 2008) | 8
- lines If we detect that we are no longer connected, try to
- reconnect a few times before giving up. This relies on the
- timeout settings in the freetds.conf file and, unfortunately, on
- a recent version of FreeTDS (0.82 or newer). I either need to
- change the current execs to be non-blocking (which I do not want
- to do) or we have to force people to run with the latest and
- greatest of FreeTDS. I'm on the fence... ........
-
-2008-08-14 02:04 +0000 [r137681] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, Zaptel-to-DAHDI.txt: Merged revisions 137680 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r137680 | kpfleming | 2008-08-13 21:03:47 -0500 (Wed, 13 Aug
- 2008) | 9 lines Merged revisions 137679 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r137679 | kpfleming | 2008-08-13 21:03:04 -0500 (Wed, 13 Aug
- 2008) | 1 line forgot one module name that changed ........
- ................
-
-2008-08-13 Kevin P. Fleming <kpfleming@digium.com>
-
- * Asterisk 1.6.0-rc1 released.
-
-2008-08-13 23:00 +0000 [r137631-137641] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, build_tools/prep_tarball: Merged revisions 137640 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r137640 | kpfleming | 2008-08-13 18:00:37 -0500 (Wed, 13 Aug
- 2008) | 1 line make this script actually work ........
-
- * /, Zaptel-to-DAHDI.txt (added), UPGRADE.txt: Merged revisions
- 137627 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r137627 | kpfleming | 2008-08-13 17:33:32 -0500 (Wed, 13 Aug
- 2008) | 9 lines Merged revisions 137530 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r137530 | kpfleming | 2008-08-13 15:46:19 -0500 (Wed, 13 Aug
- 2008) | 1 line add document describing what users will need to be
- aware of when upgrading to this version and using DAHDI ........
- ................
-
-2008-08-13 21:09 +0000 [r137497-137533] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 137532 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r137532 |
- qwell | 2008-08-13 16:08:58 -0500 (Wed, 13 Aug 2008) | 8 lines
- Correctly end locally ended calls. (closes issue #12170) Reported
- by: pj Patches: 20080702__issue12170_clear_pendinginvite.diff
- uploaded by bbryant (license 36) Tested by: bbryant, pabelanger
- ........
-
- * /, apps/app_fax.c: Merged revisions 137496 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r137496 |
- qwell | 2008-08-13 15:05:50 -0500 (Wed, 13 Aug 2008) | 6 lines
- Add FAXMODE variable with what fax transport was used. (closes
- issue #13252) Patches: v1-13252.patch uploaded by dimas (license
- 88) ........
-
-2008-08-13 14:47 +0000 [r137350-137407] Sean Bright <sean.bright@gmail.com>
-
- * /, doc/tex/cdrdriver.tex: Merged revisions 137406 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r137406 | seanbright | 2008-08-13 10:41:49 -0400
- (Wed, 13 Aug 2008) | 9 lines Merged revisions 137405 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r137405 | seanbright | 2008-08-13 10:33:49 -0400 (Wed,
- 13 Aug 2008) | 1 line Update docs to reflect the change to
- cdr_tds ........ ................
-
- * cdr/cdr_tds.c, /: Merged revisions 137403 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r137403 |
- seanbright | 2008-08-13 10:22:47 -0400 (Wed, 13 Aug 2008) | 1
- line Use the ast_vasprintf macro instead of vasprintf directly.
- ........
-
-2008-08-12 19:48 +0000 [r137300-137302] Russell Bryant <russell@digium.com>
-
- * doc/tex/asterisk.tex, /: Merged revisions 137301 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r137301 | russell | 2008-08-12 14:48:38 -0500 (Tue, 12 Aug 2008)
- | 2 lines Grammar hax from Qwell ........
-
- * doc/tex/asterisk.tex, /: Merged revisions 137299 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r137299 | russell | 2008-08-12 14:40:35 -0500 (Tue, 12 Aug 2008)
- | 3 lines Note that developer documentation belongs in doxygen,
- and not integrated with the user manual stuff in doc/tex/.
- ........
-
-2008-08-11 16:15 +0000 [r137240] Russell Bryant <russell@digium.com>
-
- * Makefile, /: Merged revisions 137239 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r137239 |
- russell | 2008-08-11 11:14:29 -0500 (Mon, 11 Aug 2008) | 2 lines
- Make PRINT_DIR work as advertised. ........
-
-2008-08-11 14:31 +0000 [r137217] Sean Bright <sean.bright@gmail.com>
-
- * cdr/cdr_tds.c, /, UPGRADE.txt: Merged revisions 137203 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r137203 | seanbright | 2008-08-11 10:25:15 -0400 (Mon,
- 11 Aug 2008) | 7 lines Log the userfield CDR variable like the
- other CDR backends, assuming the column is actually there. If
- it's not, we still log everything else as before. (closes issue
- #13281) Reported by: falves11 ........
-
-2008-08-11 00:27 +0000 [r137160] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_odbc.c, /: Merged revisions 137150 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r137150 | tilghman | 2008-08-10 19:25:28 -0500 (Sun, 10 Aug 2008)
- | 13 lines Merged revisions 137138 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r137138 | tilghman | 2008-08-10 19:20:38 -0500 (Sun, 10 Aug 2008)
- | 5 lines Deallocate database connection handle on disconnect, as
- we allocate another one on connect. (closes issue #13271)
- Reported by: dveiga ........ ................
-
-2008-08-09 15:27 +0000 [r136948] Tilghman Lesher <tlesher@digium.com>
-
- * /, include/asterisk/compat.h, include/asterisk/astobj2.h: Merged
- revisions 136947 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r136947 | tilghman | 2008-08-09 10:26:27 -0500 (Sat, 09 Aug 2008)
- | 18 lines Merged revisions 136946 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r136946 | tilghman | 2008-08-09 10:25:36 -0500
- (Sat, 09 Aug 2008) | 10 lines Merged revisions 136945 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r136945 | tilghman | 2008-08-09 10:24:36 -0500 (Sat, 09 Aug 2008)
- | 2 lines Regression fixes for Solaris ........ ................
- ................
-
-2008-08-09 01:16 +0000 [r136860] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_agi.c: Merged revisions 136859 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r136859 |
- tilghman | 2008-08-08 20:15:38 -0500 (Fri, 08 Aug 2008) | 4 lines
- Update documentation as to the behavior of AGI in 1.6.0 and
- higher. Also, add an OOB message that answers the question of, if
- AGI no longer shuts down the connection on hangup, how will
- FastAGI know when to stop processing the call? ........
-
-2008-08-08 15:33 +0000 [r136785] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 136784 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r136784 | mmichelson | 2008-08-08 10:31:31 -0500 (Fri, 08 Aug
- 2008) | 3 lines Fix compilation for ODBC voicemail ........
-
-2008-08-08 06:45 +0000 [r136778] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18,
- pbx/ael/ael-test/ref.ael-test19,
- pbx/ael/ael-test/ref.ael-vtest13, res/ael/pval.c, /,
- pbx/ael/ael-test/ref.ael-ntest10, include/asterisk/ael_structs.h,
- utils/ael_main.c: Merged revisions 136746 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r136746 | murf | 2008-08-07 18:48:35 -0600 (Thu, 07 Aug 2008) |
- 40 lines Merged revisions 136726 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r136726 | murf | 2008-08-07 18:15:34 -0600 (Thu, 07 Aug 2008) |
- 32 lines (closes issue #13236) Reported by: korihor Wow, this one
- was a challenge! I regrouped and ran a new strategy for setting
- the ~~MACRO~~ value; I set it once per extension, up near the
- top. It is only set if there is a switch in the extension. So, I
- had to put in a chunk of code to detect a switch in the pval
- tree. I moved the code to insert the set of ~~exten~~ up to the
- beginning of the gen_prios routine, instead of down in the switch
- code. I learned that I have to push the detection of the switches
- down into the code, so everywhere I create a new exten in
- gen_prios, I make sure to pass onto it the values of the
- mother_exten first, and the exten next. I had to add a couple
- fields to the exten struct to accomplish this, in the
- ael_structs.h file. The checked field makes it so we don't repeat
- the switch search if it's been done. I also updated the
- regressions. ........ ................
-
-2008-08-08 02:36 +0000 [r136753] Tilghman Lesher <tlesher@digium.com>
-
- * /: Merged revisions 136751 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r136751 |
- tilghman | 2008-08-07 21:34:17 -0500 (Thu, 07 Aug 2008) | 2 lines
- Removing bad properties ........
-
-2008-08-07 23:42 +0000 [r136719-136724] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: This is weird. Either SVN or vim tabbed a
- bunch of functions over one level during a merge.
-
- * apps/app_voicemail.c, /: Merged revisions 136722 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r136722 | mmichelson | 2008-08-07 18:39:50 -0500 (Thu, 07 Aug
- 2008) | 3 lines Remove one last batch of debug messages ........
-
- * apps/app_voicemail.c, /: Merged revisions 136715 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r136715 | mmichelson | 2008-08-07 17:25:50 -0500 (Thu, 07 Aug
- 2008) | 18 lines Merging the imap_consistency_trunk branch to
- trunk. For an explanation of what "imap_consistency" is, please
- see svn revision 134223 to the 1.4 branch. Coincidentally, this
- also fixes a recent bug report regarding the inability to save
- messages to the new folder when using IMAP storage since they
- will would be flagged as "seen" and not be recognized as new
- messages. (closes issue #13234) Reported by: jaroth ........
-
-2008-08-07 20:41 +0000 [r136672-136674] Shaun Ruffell <sruffell@digium.com>
-
- * codecs/codec_dahdi.c: Removing code that was commented out.
-
- * codecs/codec_dahdi.c: Updated codec_dahdi to use the transcoder
- interface in the DAHDI. (Issue: DAHDI-42)
-
-2008-08-07 20:26 +0000 [r136632-136663] Mark Michelson <mmichelson@digium.com>
-
- * /, main/features.c: Merged revisions 136660 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r136660 |
- mmichelson | 2008-08-07 15:25:43 -0500 (Thu, 07 Aug 2008) | 4
- lines Bump a LOG_NOTICE message to LOG_DEBUG since it appears
- once for every bridged call ........
-
- * main/pbx.c, /: Merged revisions 136635 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r136635 |
- mmichelson | 2008-08-07 14:58:32 -0500 (Thu, 07 Aug 2008) | 5
- lines Don't allow Answer() to accept a negative argument.
- Negative argument means an infinite delay and we don't want that.
- ........
-
- * main/channel.c, /: Merged revisions 136633 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r136633 |
- mmichelson | 2008-08-07 14:54:27 -0500 (Thu, 07 Aug 2008) | 7
- lines Fix a calculation error I had made in the poll. The poll
- would reset to 500 ms every time a non-voice frame was received.
- The total time we poll should be 500 ms, so now we save the
- amount of time left after the poll returned and use that as our
- argument for the next call to poll ........
-
- * main/channel.c, /: Merged revisions 136631 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r136631 |
- mmichelson | 2008-08-07 14:36:46 -0500 (Thu, 07 Aug 2008) | 13
- lines Scrap the 500 ms delay when Asterisk auto-answers a
- channel. Instead, poll the channel until receiving a voice frame.
- The cap on this poll is 500 ms. The optional delay is still
- allowable in the Answer() application, but the delay has been
- moved back to its original position, after the call to the
- channel's answer callback. The poll for the voice frame will not
- happen if a delay is specified when calling Answer(). (closes
- issue #12708) Reported by: kactus ........
-
-2008-08-07 19:19 +0000 [r136598] Richard Mudgett <rmudgett@digium.com>
-
- * channels/misdn_config.c, channels/chan_misdn.c, /,
- configs/misdn.conf.sample: Merged revisions 136594 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r136594 | rmudgett | 2008-08-07 14:01:03 -0500
- (Thu, 07 Aug 2008) | 13 lines Merged revisions 136241 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r136241 | rmudgett | 2008-08-06 16:18:53 -0500 (Wed, 06 Aug 2008)
- | 5 lines * The allowed_bearers setting in misdn.conf misspelled
- one of its options: digital_restricted. * Fixed some other
- spelling errors and typos. ........ ................
-
-2008-08-07 17:44 +0000 [r136506-136543] Kevin P. Fleming <kpfleming@digium.com>
-
- * include/asterisk/doxyref.h, /: Merged revisions 136542 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r136542 | kpfleming | 2008-08-07 12:44:20 -0500
- (Thu, 07 Aug 2008) | 6 lines Merged revisions 136541 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- ........ ................
-
-2008-08-07 16:57 +0000 [r136490] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 136489 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r136489 | tilghman | 2008-08-07 11:55:57 -0500 (Thu, 07 Aug 2008)
- | 15 lines Merged revisions 136488 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r136488 | tilghman | 2008-08-07 11:50:47 -0500 (Thu, 07 Aug 2008)
- | 7 lines Update persistent state on all exit conditions. (closes
- issue #12916) Reported by: sgenyuk Patches: app_queue.patch.txt
- uploaded by neutrino88 (license 297) Tested by: sgenyuk, aragon
- ........ ................
-
-2008-08-06 20:16 +0000 [r136113-136192] Tilghman Lesher <tlesher@digium.com>
-
- * /, contrib/init.d/rc.redhat.asterisk: Merged revisions 136191 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r136191 | tilghman | 2008-08-06 15:15:34 -0500
- (Wed, 06 Aug 2008) | 12 lines Merged revisions 136190 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r136190 | tilghman | 2008-08-06 15:14:54 -0500 (Wed, 06 Aug 2008)
- | 4 lines -C option takes a filename, not a directory path.
- (closes issue #13007) Reported by: klaus3000 ........
- ................
-
- * /, funcs/func_dialgroup.c: Merged revisions 136112 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r136112 | tilghman | 2008-08-06 11:58:42 -0500 (Wed, 06 Aug 2008)
- | 7 lines Persist DIALGROUP() values in astdb (closes issue
- #13138) Reported by: Corydon76 Patches:
- 20080725__bug13138.diff.txt uploaded by Corydon76 (license 14)
- Tested by: pj ........
-
-2008-08-06 16:00 +0000 [r136064] Mark Michelson <mmichelson@digium.com>
-
- * main/rtp.c, /, channels/chan_skinny.c: Merged revisions 136063
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r136063 | mmichelson | 2008-08-06 10:59:29 -0500
- (Wed, 06 Aug 2008) | 24 lines Merged revisions 136062 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r136062 | mmichelson | 2008-08-06 10:58:40 -0500 (Wed, 06 Aug
- 2008) | 16 lines Since adding the AST_CONTROL_SRCUPDATE frame
- type, there are places where ast_rtp_new_source may be called
- where the tech_pvt of a channel may not yet have an rtp structure
- allocated. This caused a crash in chan_skinny, which was fixed
- earlier, but now the same crash has been reported against
- chan_h323 as well. It seems that the best solution is to modify
- ast_rtp_new_source to not attempt to set the marker bit if the
- rtp structure passed in is NULL. This change to
- ast_rtp_new_source also allows the removal of what is now a
- redundant pointer check from chan_skinny. (closes issue #13247)
- Reported by: pj ........ ................
-
-2008-08-06 13:59 +0000 [r136006] Olle Johansson <oej@edvina.net>
-
- * /, res/res_jabber.c: Merged revisions 136005 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r136005 |
- oej | 2008-08-06 15:34:08 +0200 (Ons, 06 Aug 2008) | 6 lines -
- Formatting - Changing debug messages from VERBOSE to DEBUG
- channel - Adding a few todo's - Adding a few more "XMPP"'s to
- compliment Jabber... ........
-
-2008-08-06 03:56 +0000 [r135901-135951] Tilghman Lesher <tlesher@digium.com>
-
- * main/channel.c, /: Merged revisions 135950 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r135950 | tilghman | 2008-08-05 22:55:49 -0500 (Tue, 05 Aug 2008)
- | 12 lines Merged revisions 135949 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r135949 | tilghman | 2008-08-05 22:53:36 -0500 (Tue, 05 Aug 2008)
- | 4 lines Fix a longstanding bug in channel walking logic, and
- fix the explanation to make sense. (Closes issue #13124) ........
- ................
-
- * /, main/translate.c: Merged revisions 135938 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r135938 | tilghman | 2008-08-05 22:29:42 -0500 (Tue, 05 Aug 2008)
- | 12 lines Merged revisions 135915 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r135915 | tilghman | 2008-08-05 22:24:56 -0500 (Tue, 05 Aug 2008)
- | 4 lines Since powerof() can return an error condition, it's
- foolhardy not to detect and deal with that condition. (Related to
- issue #13240) ........ ................
-
- * include/asterisk/threadstorage.h, include/asterisk/utils.h, /:
- Merged revisions 135900 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r135900 | tilghman | 2008-08-05 22:04:01 -0500 (Tue, 05 Aug 2008)
- | 12 lines Merged revisions 135899 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r135899 | tilghman | 2008-08-05 22:02:59 -0500 (Tue, 05 Aug 2008)
- | 4 lines 1) Bugfix for debugging code 2) Reduce compiler
- warnings for another section of debugging code (Closes issue
- #13237) ........ ................
-
-2008-08-06 00:31 +0000 [r135852] Mark Michelson <mmichelson@digium.com>
-
- * include/asterisk/abstract_jb.h, main/channel.c, /,
- main/abstract_jb.c, main/fixedjitterbuf.h: Merged revisions
- 135851 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r135851 | mmichelson | 2008-08-05 19:30:53 -0500 (Tue, 05 Aug
- 2008) | 48 lines Merged revisions 135841,135847,135850 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r135841 | mmichelson | 2008-08-05 19:25:10 -0500 (Tue, 05 Aug
- 2008) | 27 lines Merging the issue11259 branch. The purpose of
- this branch was to take into account "burps" which could cause
- jitterbuffers to misbehave. One such example is if the L option
- to Dial() were used to inject audio into a bridged conversation
- at regular intervals. Since the audio here was not passed through
- the jitterbuffer, it would cause a gap in the jitterbuffer's
- timestamps which would cause a frames to be dropped for a brief
- period. Now ast_generic_bridge will empty and reset the
- jitterbuffer each time it is called. This causes injected audio
- to be handled properly. ast_generic_bridge also will empty and
- reset the jitterbuffer if it receives an AST_CONTROL_SRCUPDATE
- frame since the change in audio source could negatively affect
- the jitterbuffer. All of this was made possible by adding a new
- public API call to the abstract_jb called ast_jb_empty_and_reset.
- (closes issue #11259) Reported by: plack Tested by: putnopvut
- ........ r135847 | mmichelson | 2008-08-05 19:27:54 -0500 (Tue,
- 05 Aug 2008) | 4 lines Revert inadvertent changes to app_skel
- that occurred when I was testing for a memory leak ........
- r135850 | mmichelson | 2008-08-05 19:29:54 -0500 (Tue, 05 Aug
- 2008) | 3 lines Remove properties that should not be here
- ........ ................
-
-2008-08-05 23:52 +0000 [r135822] Steve Murphy <murf@digium.com>
-
- * apps/app_dial.c, main/cdr.c, main/channel.c, /, main/features.c,
- include/asterisk/cdr.h: Merged revisions 135821 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r135821 | murf | 2008-08-05 17:45:32 -0600 (Tue, 05 Aug 2008) |
- 42 lines Merged revisions 135799 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r135799 | murf | 2008-08-05 17:13:20 -0600 (Tue, 05 Aug 2008) |
- 34 lines (closes issue #12982) Reported by: bcnit Tested by: murf
- I discovered that also, in the previous bug fixes and changes,
- the cdr.conf 'unanswered' option is not being obeyed, so I fixed
- this. And, yes, there are two 'answer' times involved in this
- scenario, and I would agree with you, that the first answer time
- is the time that should appear in the CDR. (the second 'answer'
- time is the time that the bridge was begun). I made the necessary
- adjustments, recording the first answer time into the peer cdr,
- and then using that to override the bridge cdr's value. To get
- the 'unanswered' CDRs to appear, I purposely output them, using
- the dial cmd to mark them as DIALED (with a new flag), and
- outputting them if they bear that flag, and you are in the right
- mode. I also corrected one small mention of the Zap device to
- equally consider the dahdi device. I heavily tested 10-sec-wait
- macros in dial, and without the macro call; I tested hangups
- while the macro was running vs. letting the macro complete and
- the bridge form. Looks OK. Removed all the instrumentation and
- debug. ........ ................
-
-2008-08-05 21:38 +0000 [r135749] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 135748 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r135748 | tilghman | 2008-08-05 16:37:35 -0500
- (Tue, 05 Aug 2008) | 17 lines Merged revisions 135747 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r135747 | tilghman | 2008-08-05 16:34:46 -0500 (Tue, 05 Aug 2008)
- | 9 lines In a conversion to use ast_strlen_zero, the meaning of
- the flag IAX_HASCALLERID was perverted. This change reverts IAX2
- to the original meaning, which was, that the callerid set on the
- client should be overridden on the server, even if that means the
- resulting callerid is blank. In other words, if you set
- "callerid=" in the IAX config, then the callerid should be
- overridden to blank, even if set on the client. Note that there's
- a distinction, even on realtime, between the field not existing
- (NULL in databases) and the field existing, but set to blank
- (override callerid to blank). ........ ................
-
-2008-08-05 13:27 +0000 [r135599] Sean Bright <sean.bright@gmail.com>
-
- * main/cli.c, /: Merged revisions 135598 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r135598 | seanbright | 2008-08-05 09:26:34 -0400 (Tue, 05 Aug
- 2008) | 9 lines Merged revisions 135597 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r135597 | seanbright | 2008-08-05 09:25:00 -0400 (Tue, 05 Aug
- 2008) | 1 line Use PATH_MAX for filenames ........
- ................
-
-2008-08-04 20:15 +0000 [r135538] Russell Bryant <russell@digium.com>
-
- * configs/chan_dahdi.conf.sample, /: Merged revisions 135537 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r135537 | russell | 2008-08-04 15:15:27 -0500
- (Mon, 04 Aug 2008) | 10 lines Merged revisions 135536 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r135536 | russell | 2008-08-04 15:15:03 -0500 (Mon, 04 Aug 2008)
- | 2 lines fix a config sample typo ........ ................
-
-2008-08-04 17:12 +0000 [r135478-135486] Tilghman Lesher <tlesher@digium.com>
-
- * contrib/init.d/rc.mandriva.asterisk (added), Makefile,
- contrib/init.d/rc.mandrake.asterisk (removed), /,
- contrib/init.d/rc.mandriva.zaptel (added),
- contrib/init.d/rc.mandrake.zaptel (removed): Merged revisions
- 135485 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r135485 |
- tilghman | 2008-08-04 12:12:15 -0500 (Mon, 04 Aug 2008) | 3 lines
- Rename Mandrake scripts to Mandriva (Closes issue #13221)
- ........
-
- * contrib/init.d/rc.mandrake.asterisk, /: Merged revisions 135483
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r135483 | tilghman | 2008-08-04 12:08:42 -0500
- (Mon, 04 Aug 2008) | 11 lines Merged revisions 135482 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r135482 | tilghman | 2008-08-04 12:07:52 -0500 (Mon, 04 Aug 2008)
- | 2 lines Define ASTSBINDIR for script (Closes issue #13221)
- ........ ................
-
- * apps/app_voicemail.c, /: Merged revisions 135480 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r135480 | tilghman | 2008-08-04 11:58:29 -0500
- (Mon, 04 Aug 2008) | 14 lines Merged revisions 135479 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r135479 | tilghman | 2008-08-04 11:56:19 -0500 (Mon, 04 Aug 2008)
- | 6 lines Memory leak on unload (closes issue #13231) Reported
- by: eliel Patches: app_voicemail.leak.patch uploaded by eliel
- (license 64) ........ ................
-
-2008-08-04 16:28 +0000 [r135440-135475] Russell Bryant <russell@digium.com>
-
- * configs/chan_dahdi.conf.sample, /: Merged revisions 135474 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r135474 | russell | 2008-08-04 11:28:07 -0500
- (Mon, 04 Aug 2008) | 10 lines Merged revisions 135473 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r135473 | russell | 2008-08-04 11:26:17 -0500 (Mon, 04 Aug 2008)
- | 2 lines Add a minor clarification to the documentation of
- mohinterpret and mohsuggest ........ ................
-
- * /, channels/chan_console.c: Merged revisions 135439 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r135439 | russell | 2008-08-04 10:02:12 -0500 (Mon, 04 Aug 2008)
- | 4 lines Be explicit that we don't want a result from this
- callback. The callback would never indicate a match, so nothing
- would have been returned anyway, but it was still a poor example
- of proper usage. ........
-
-2008-08-02 05:15 +0000 [r135266] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, /: Merged revisions 135265 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r135265 |
- murf | 2008-08-01 22:51:29 -0600 (Fri, 01 Aug 2008) | 31 lines
- (closes issue #13202) Reported by: falves11 Tested by: murf
- falves11 == The changes I introduce here seem to clear up the
- problem for me. However, if they do not for you, please reopen
- this bug, and we'll keep digging. The root of this problem seems
- to be a subtle memory corruption introduced when creating an
- extension with an empty extension name. While valgrind cannot
- detect it outside of DEBUG_MALLOC mode, when compiled with
- DEBUG_MALLOC, this is certain death. The code in main/features.c
- is a puzzle to me. On the initial module load, the code is
- attempting to add the parking extension before the features.conf
- file has even been opened! I just wrapped the offending call with
- an if() that will not try to add the extension if the extension
- name is empty. THis seems to solve the corruption, and let the
- "memory show allocations" work as one would expect. But, really,
- adding an extension with an empty name is a seriously bad thing
- to allow, as it will mess up all the pattern matching algorithms,
- etc. So, I added a statement to the add_extension2 code to return
- a -1 if this is attempted. in 1.6.0, the changes to only
- main/pbx.c were applicable, as apparently the code added to
- main/features by jpeeler were not included in 1.6.0. ........
-
-2008-08-01 19:30 +0000 [r135198] Sean Bright <sean.bright@gmail.com>
-
- * channels/chan_mgcp.c, /: Merged revisions 135197 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r135197 | seanbright | 2008-08-01 15:29:26 -0400 (Fri, 01 Aug
- 2008) | 6 lines Remove some code that used to do something but
- does not anymore, mainly to get rid of a shadow warning (but this
- seemed legitimate enough to fix here instead of in my branch).
- Thanks to putnopvut for taking a look as well. ........
-
-2008-08-01 17:10 +0000 [r135127-135129] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 135128 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r135128 |
- tilghman | 2008-08-01 12:09:50 -0500 (Fri, 01 Aug 2008) | 2 lines
- Picky, picky, buildbot ........
-
- * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
- 135126 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r135126 |
- tilghman | 2008-08-01 11:39:51 -0500 (Fri, 01 Aug 2008) | 9 lines
- SIP should use the transport type set in the Moved Temporarily
- for the next invite. (closes issue #11843) Reported by:
- pestermann Patches:
- 20080723__issue11843_302_ignores_transport_16branch.diff uploaded
- by bbryant (license 36)
- 20080723__issue11843_302_ignores_transport_trunk.diff uploaded by
- bbryant (license 36) Tested by: pabelanger ........
-
-2008-08-01 14:43 +0000 [r135070] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged
- revisions 135067-135068 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r135067 |
- mmichelson | 2008-08-01 09:29:48 -0500 (Fri, 01 Aug 2008) | 13
- lines IMAP storage functioned under the assumption that folders
- such as "Work" and "Family" would be subfolders of the INBOX.
- This is an invalid assumption to make, but it could be desirable
- to set up folders in this manner, so a new option for
- voicemail.conf, "imapparentfolder" has been added to allow for
- this. (closes issue #13142) Reported by: jaroth Patches:
- parentfolder.patch uploaded by jaroth (license 50) ........
- r135068 | mmichelson | 2008-08-01 09:42:24 -0500 (Fri, 01 Aug
- 2008) | 3 lines IMAP-specific items must go in IMAP_STORAGE
- defines... ........
-
-2008-08-01 12:18 +0000 [r135057-135062] Michiel van Baak <michiel@vanbaak.info>
-
- * /, apps/app_ices.c: Merged revisions 135059 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r135059 | mvanbaak | 2008-08-01 13:47:34 +0200 (Fri, 01 Aug 2008)
- | 10 lines Merged revisions 135058 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r135058 | mvanbaak | 2008-08-01 13:43:46 +0200 (Fri, 01 Aug 2008)
- | 2 lines make app_ices compile on OpenBSD. ........
- ................
-
- * /, channels/chan_skinny.c: Merged revisions 135056 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r135056 | mvanbaak | 2008-08-01 13:00:13 +0200
- (Fri, 01 Aug 2008) | 16 lines Merged revisions 135055 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r135055 | mvanbaak | 2008-08-01 12:55:27 +0200 (Fri, 01 Aug 2008)
- | 8 lines fix some potential deadlocks in chan_skinny (closes
- issue #13215) Reported by: qwell Patches:
- 2008080100_bug13215.diff.txt uploaded by mvanbaak (license 7)
- Tested by: mvanbaak ........ ................
-
-2008-07-31 22:34 +0000 [r135034] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, main/http.c: Merged revisions 135016 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r135016 | kpfleming | 2008-07-31 17:28:42 -0500 (Thu, 31 Jul
- 2008) | 11 lines Merged revisions 134983 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r134983 | kpfleming | 2008-07-31 17:18:11 -0500 (Thu, 31 Jul
- 2008) | 3 lines accomodate users who seem to lack a sense of
- humor :-) ........ ................
-
-2008-07-31 21:58 +0000 [r134926-134981] Tilghman Lesher <tlesher@digium.com>
-
- * sample.call, main/manager.c, pbx/pbx_spool.c, /: Merged revisions
- 134980 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r134980 | tilghman | 2008-07-31 16:55:42 -0500 (Thu, 31 Jul 2008)
- | 16 lines Blocked revisions 134976 via svnmerge ........ r134976
- | tilghman | 2008-07-31 16:53:19 -0500 (Thu, 31 Jul 2008) | 9
- lines Specify codecs in callfiles and manager, to allow video
- calls to be set up from callfiles and AMI. (closes issue #9531)
- Reported by: Geisj Patches: 20080715__bug9531__1.4.diff.txt
- uploaded by Corydon76 (license 14)
- 20080715__bug9531__1.6.0.diff.txt uploaded by Corydon76 (license
- 14) Tested by: Corydon76 ........ ................
-
- * res/res_config_sqlite.c, /: Merged revisions 134977 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r134977 | tilghman | 2008-07-31 16:53:59 -0500 (Thu, 31 Jul 2008)
- | 2 lines Switch command order, to meet with current specs
- ........
-
-2008-07-31 19:54 +0000 [r134923] Steve Murphy <murf@digium.com>
-
- * /, main/features.c: Merged revisions 134922 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r134922 | murf | 2008-07-31 13:48:08 -0600 (Thu, 31 Jul 2008) |
- 63 lines Merged revisions 134883 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r134883 | murf | 2008-07-31 13:23:42 -0600 (Thu, 31 Jul 2008) |
- 51 lines (closes issue #11849) Reported by: greyvoip Tested by:
- murf OK, a few days of debugging, a bunch of instrumentation in
- chan_sip, main/channel.c, main/pbx.c, etc. and 5 solid notebook
- pages of notes later, I have made the small tweek necc. to get
- the start time right on the second CDR when: A Calls B B answ. A
- hits Xfer button on sip phone, A dials C and hits the OK button,
- A hangs up C answers ringing phone B and C converse B and/or C
- hangs up But does not harm the scenario where: A Calls B B answ.
- B hits xfer button on sip phone, B dials C and hits the OK
- button, B hangs up C answers ringing phone A and C converse A
- and/or C hangs up The difference in start times on the second CDR
- is because of a Masquerade on the B channel when the xfer number
- is sent. It ends up replacing the CDR on the B channel with a
- duplicate, which ends up getting tossed out. We keep a pointer to
- the first CDR, and update *that* after the bridge closes. But,
- only if the CDR has changed. I hope this change is specific
- enough not to muck up any current CDR-based apps. In my defence,
- I assert that the previous information was wrong, and this change
- fixes it, and possibly other similar scenarios. I wonder if I
- should be doing the same thing for the channel, as I did for the
- peer, but I can't think of a scenario this might affect. I leave
- it, then, as an exersize for the users, to find the scenario
- where the chan's CDR changes and loses the proper start time.
- ........ ................
-
-2008-07-31 19:41 +0000 [r134918] Russell Bryant <russell@digium.com>
-
- * /, apps/app_ices.c: Merged revisions 134917 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r134917 | russell | 2008-07-31 14:39:50 -0500 (Thu, 31 Jul 2008)
- | 17 lines Merged revisions 134915 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r134915 | russell | 2008-07-31 14:37:26 -0500 (Thu, 31 Jul 2008)
- | 9 lines Get app_ices working again (closes issue #12981)
- Reported by: dlogan Patches:
- 20080709__app_ices_v2_update_trunk.diff uploaded by bbryant
- (license 36) 20080709__app_ices_v2_update_14.diff uploaded by
- bbryant (license 36) Tested by: bbryant ........ ................
-
-2008-07-31 16:53 +0000 [r134816] Russell Bryant <russell@digium.com>
-
- * channels/iax2-parser.c: Merged revisions 134815 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r134815 | russell | 2008-07-31 11:50:10 -0500 (Thu, 31 Jul 2008)
- | 15 lines Merged revisions 134814 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r134814 | russell | 2008-07-31 11:45:31 -0500 (Thu, 31 Jul 2008)
- | 7 lines In case we have some processing threads that free more
- frames than they allocate, do not let the frame cache grow
- forever. (closes issue #13160) Reported by: tavius Tested by:
- tavius, russell ........ ................
-
-2008-07-31 16:07 +0000 [r134760] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 134759 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r134759 | mmichelson | 2008-07-31 11:05:12 -0500 (Thu, 31 Jul
- 2008) | 24 lines Merged revisions 134758 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r134758 | mmichelson | 2008-07-31 10:56:18 -0500 (Thu, 31 Jul
- 2008) | 16 lines Add more timeout checks into app_queue,
- specifically targeting areas where an unknown and potentially
- long time has just elapsed. Also added a check to try_calling()
- to return early if the timeout has elapsed instead of potentially
- setting a negative timeout for the call (thus making it have *no*
- timeout at all). (closes issue #13186) Reported by:
- miquel_cabrespina Patches: 13186.diff uploaded by putnopvut
- (license 60) Tested by: miquel_cabrespina ........
- ................
-
-2008-07-30 22:41 +0000 [r134651-134707] Tilghman Lesher <tlesher@digium.com>
-
- * main/sched.c, /, include/asterisk/sched.h: Merged revisions
- 134703 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r134703 |
- tilghman | 2008-07-30 17:38:58 -0500 (Wed, 30 Jul 2008) | 2 lines
- Oops, wrong define ........
-
- * /, configure, configure.ac: Merged revisions 134650 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r134650 | tilghman | 2008-07-30 16:40:08 -0500
- (Wed, 30 Jul 2008) | 12 lines Merged revisions 134649 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r134649 | tilghman | 2008-07-30 16:38:50 -0500 (Wed, 30 Jul 2008)
- | 4 lines Qwell pointed out, via IRC, that the previous fix only
- worked when explicitly set. When nothing is set, and the option
- is implied, it breaks, because configure sets the prefix to
- 'NONE'. Fixing. ........ ................
-
-2008-07-30 21:06 +0000 [r134599] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 134598 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r134598 | mmichelson | 2008-07-30 16:05:37 -0500 (Wed, 30 Jul
- 2008) | 15 lines Merged revisions 134556 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r134556 |
- mmichelson | 2008-07-30 15:24:40 -0500 (Wed, 30 Jul 2008) | 7
- lines Fix the parsing of the "reason" parameter in the Diversion:
- header. (closes issue #13195) Reported by: woodsfsg ........
- ................
-
-2008-07-30 20:39 +0000 [r134597] Russell Bryant <russell@digium.com>
-
- * /, pbx/pbx_dundi.c: Merged revisions 134596 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r134596 | russell | 2008-07-30 15:38:35 -0500 (Wed, 30 Jul 2008)
- | 14 lines Merged revisions 134595 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r134595 | russell | 2008-07-30 15:37:17 -0500 (Wed, 30 Jul 2008)
- | 6 lines Reduce stack consumption by 12.5% of the max stack size
- to fix a crash when compiled with LOW_MEMORY. (closes issue
- #13154) Reported by: edantie ........ ................
-
-2008-07-30 20:25 +0000 [r134561] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 134556 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r134556 |
- mmichelson | 2008-07-30 15:24:40 -0500 (Wed, 30 Jul 2008) | 7
- lines Fix the parsing of the "reason" parameter in the Diversion:
- header. (closes issue #13195) Reported by: woodsfsg ........
-
-2008-07-30 19:56 +0000 [r134542] Russell Bryant <russell@digium.com>
-
- * funcs/func_curl.c, /: Merged revisions 134541 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r134541 | russell | 2008-07-30 14:55:31 -0500 (Wed, 30 Jul 2008)
- | 12 lines Merged revisions 134540 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r134540 | russell | 2008-07-30 14:52:53 -0500 (Wed, 30 Jul 2008)
- | 4 lines Fix a memory leak in func_curl. Every thread that used
- this function leaked an allocation the size of a pointer.
- (reported by jmls in #asterisk-dev) ........ ................
-
-2008-07-30 19:49 +0000 [r134482-134539] Tilghman Lesher <tlesher@digium.com>
-
- * /, configure, configure.ac: Merged revisions 134538 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r134538 | tilghman | 2008-07-30 14:48:37 -0500
- (Wed, 30 Jul 2008) | 12 lines Merged revisions 134536 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r134536 | tilghman | 2008-07-30 14:47:16 -0500 (Wed, 30 Jul 2008)
- | 4 lines Only override sysconfdir and mandir when prefix=/usr
- (closes issue #13093) Reported by: pabelanger ........
- ................
-
- * /, apps/app_queue.c: Merged revisions 134483 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r134483 |
- tilghman | 2008-07-30 14:17:38 -0500 (Wed, 30 Jul 2008) | 4 lines
- Let "roundrobin" also reference rrmemory, for the 1.6 release (as
- described in UPGRADE-1.4.txt) (Closes issue #13181) ........
-
- * /, res/res_agi.c: Merged revisions 134481 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r134481 | tilghman | 2008-07-30 14:05:35 -0500 (Wed, 30 Jul 2008)
- | 13 lines Merged revisions 134480 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r134480 | tilghman | 2008-07-30 14:03:44 -0500 (Wed, 30 Jul 2008)
- | 5 lines launch_netscript sometimes returns -1, which fails to
- set AGISTATUS. Map failure to -1, so that AGISTATUS is always
- set. (closes issue #13199) Reported by: smw1218 ........
- ................
-
-2008-07-30 18:33 +0000 [r134477] Mark Michelson <mmichelson@digium.com>
-
- * /, main/app.c: Merged revisions 134476 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r134476 | mmichelson | 2008-07-30 13:33:12 -0500 (Wed, 30 Jul
- 2008) | 12 lines Merged revisions 134475 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r134475 | mmichelson | 2008-07-30 13:31:47 -0500 (Wed, 30 Jul
- 2008) | 4 lines Fix a spot where a function could return without
- bringing a channel out of autoservice. ........ ................
-
-2008-07-30 15:34 +0000 [r134356] Kevin P. Fleming <kpfleming@digium.com>
-
- * Makefile, /: Merged revisions 134355 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r134355 | kpfleming | 2008-07-30 10:32:14 -0500 (Wed, 30 Jul
- 2008) | 10 lines Merged revisions 134352 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r134352 | kpfleming | 2008-07-30 10:29:17 -0500 (Wed, 30 Jul
- 2008) | 2 lines use the proper method for building version.h
- ........ ................
-
-2008-07-29 22:29 +0000 [r134283] Kevin P. Fleming <kpfleming@digium.com>
-
- * apps/app_rpt.c, apps/app_dahdibarge.c, channels/chan_dahdi.c, /,
- apps/app_meetme.c, apps/app_dahdiscan.c, apps/app_dahdiras.c:
- Merged revisions 134260 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r134260 |
- kpfleming | 2008-07-29 17:22:13 -0500 (Tue, 29 Jul 2008) | 2
- lines build against the now-typedef-free dahdi/user.h, and remove
- some #ifdefs for features that will always be present in DAHDI
- ........
-
-2008-07-28 22:16 +0000 [r134164] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 134163 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r134163 | tilghman | 2008-07-28 17:07:12 -0500
- (Mon, 28 Jul 2008) | 15 lines Merged revisions 134161 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r134161 | tilghman | 2008-07-28 16:50:50 -0500 (Mon, 28 Jul 2008)
- | 7 lines Detect when sox fails to raise the volume, because sox
- can't read the file. (closes issue #12939) Reported by:
- rickbradley Patches: 20080728__bug12939.diff.txt uploaded by
- Corydon76 (license 14) Tested by: rickbradley ........
- ................
-
-2008-07-28 19:55 +0000 [r134126] Mark Michelson <mmichelson@digium.com>
-
- * /, configure, main/Makefile, configure.ac, CHANGES: Merged
- revisions 134125 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r134125 |
- mmichelson | 2008-07-28 14:53:56 -0500 (Mon, 28 Jul 2008) | 27
- lines This commit compensates for buggy poll(2) implementations.
- Asterisk has, for a long time, had its own implementation of
- poll(2) which just used the input arguments to call select(2). In
- 1.4, this internal implementation was used for Darwin systems.
- This was removed in Asterisk trunk at some point, but it seems as
- though this was not the right move to make. On Mac OS X, it
- appears as though the poll used to gather CLI input does not
- respond properly when connecting via a remote Asterisk console.
- Reverting to the use of Asterisk's poll fixed the issue. Also,
- there is now an option for the configure script,
- --enable-internal-poll, which will allow for anyone to use
- Asterisk's internal poll implementation in case they suspect that
- their system's poll implementation is buggy. closes issue #11928)
- Reported by: adriavidal Patches: 1.6.0-configurev2.patch uploaded
- by putnopvut (license 60) ........
-
-2008-07-28 16:49 +0000 [r134087] Kevin P. Fleming <kpfleming@digium.com>
-
- * apps/app_parkandannounce.c, main/loader.c, sample.call,
- contrib/scripts/autosupport, build_tools/cflags.xml,
- main/channel.c, apps/app_dahdibarge.c, channels/chan_dahdi.c,
- configs/chan_dahdi.conf.sample, doc/ss7.txt, /, main/features.c,
- doc/osp.txt, main/file.c, pbx/pbx_config.c: Merged revisions
- 134086 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r134086 |
- kpfleming | 2008-07-28 11:42:00 -0500 (Mon, 28 Jul 2008) | 3
- lines remove remaining Zaptel references in various places
- ........
-
-2008-07-28 16:13 +0000 [r134052] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_chanspy.c, include/asterisk/options.h, main/asterisk.c,
- /, apps/app_meetme.c, apps/app_dahdiscan.c: Merged revisions
- 134050 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r134050 |
- mmichelson | 2008-07-28 11:00:19 -0500 (Mon, 28 Jul 2008) | 3
- lines merging the zap_and_dahdi_trunk branch up to trunk ........
-
-2008-07-26 15:34 +0000 [r133942-133982] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c, include/asterisk/doxyref.h, /: Include the
- licensing page in 1.6.0 as well. Now, this page exists in 1.4,
- trunk, and 1.6.0.
-
- * /: unblock 133575
-
- * /, main/devicestate.c: Merged revisions 133945-133946 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r133945 | russell | 2008-07-26 10:15:14 -0500 (Sat, 26
- Jul 2008) | 6 lines ast_device_state() gets called in two
- different ways. The first way is when called from elsewhere in
- Asterisk to find the current state of a device. In that case, we
- want to use the cached value if it exists. The other way is when
- processing a device state change. In that case, we do not want to
- check the cache because returning the last known state is counter
- productive. ........ r133946 | russell | 2008-07-26 10:16:20
- -0500 (Sat, 26 Jul 2008) | 1 line actually use the cache_cache
- argument ........
-
-2008-07-25 22:09 +0000 [r133863-133905] Tilghman Lesher <tlesher@digium.com>
-
- * contrib/scripts/asterisk.ldif, /: Merged revisions 133902 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r133902 | tilghman | 2008-07-25 16:59:39 -0500 (Fri, 25
- Jul 2008) | 6 lines Update version (closes issue #13163) Reported
- by: suretec Patches: asterisk.ldif uploaded by suretec (license
- 70) ........
-
-2008-07-25 19:37 +0000 [r133804-133806] Brandon Kruse <bkruse@digium.com>
-
- * /: Blocking revert of code changes in r133770
-
- * main/http.c: Include the http_decode function from trunk to
- replace the + with a space.
-
-2008-07-25 17:33 +0000 [r133694] Brandon Kruse <bkruse@digium.com>
-
- * /: Blocking a fix from trunk for the function http_decode. 1.6.0
- does not have this function.
-
-2008-07-25 17:26 +0000 [r133671] Tilghman Lesher <tlesher@digium.com>
-
- * main/channel.c, /, channels/chan_agent.c, main/devicestate.c:
- Merged revisions 133665 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r133665 | tilghman | 2008-07-25 12:24:43 -0500 (Fri, 25 Jul 2008)
- | 16 lines Merged revisions 133649 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r133649 | tilghman | 2008-07-25 12:19:39 -0500 (Fri, 25 Jul 2008)
- | 8 lines Fix some errant device states by making the devicestate
- API more strict in terms of the device argument (only without the
- unique identifier appended). (closes issue #12771) Reported by:
- davidw Patches: 20080717__bug12771.diff.txt uploaded by Corydon76
- (license 14) Tested by: davidw, jvandal, murf ........
- ................
-
-2008-07-25 15:01 +0000 [r133576-133580] Russell Bryant <russell@digium.com>
-
- * /, LICENSE: Merged revisions 133579 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r133579 | russell | 2008-07-25 10:00:49 -0500 (Fri, 25 Jul 2008)
- | 18 lines Merged revisions 133578 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r133578 | russell | 2008-07-25 10:00:31 -0500
- (Fri, 25 Jul 2008) | 10 lines Merged revisions 133577 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r133577 | russell | 2008-07-25 10:00:13 -0500 (Fri, 25 Jul 2008)
- | 2 lines Fix the IAX2 URI for calling Digium ........
- ................ ................
-
-2008-07-25 14:41 +0000 [r133571-133574] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 133573 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r133573 | mmichelson | 2008-07-25 09:40:52 -0500 (Fri, 25 Jul
- 2008) | 15 lines Merged revisions 133572 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r133572 | mmichelson | 2008-07-25 09:40:10 -0500 (Fri, 25 Jul
- 2008) | 7 lines We need to make sure to null-terminate the "name"
- portion of SIP URI parameters so that there are no bogus
- comparisons. Thanks to bbryant for pointing this out. ........
- ................
-
-2008-07-25 13:25 +0000 [r133567-133569] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 133568 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r133568 |
- russell | 2008-07-25 08:01:59 -0500 (Fri, 25 Jul 2008) | 4 lines
- Minor coding guidelines tweaks ... - Use ast_strlen_zero in one
- place - check for successful string comparison the way most of
- Asterisk code does it ........
-
-2008-07-24 21:31 +0000 [r133524] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 133509 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r133509 | tilghman | 2008-07-24 16:27:06 -0500 (Thu, 24 Jul 2008)
- | 11 lines Merged revisions 133488 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r133488 | tilghman | 2008-07-24 16:17:55 -0500 (Thu, 24 Jul 2008)
- | 3 lines Fix rtautoclear and rtcachefriends (Closes issue
- #12707) ........ ................
-
-2008-07-24 20:41 +0000 [r133487] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 133486 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r133486 | russell | 2008-07-24 15:40:15 -0500 (Thu, 24 Jul 2008)
- | 3 lines I made this change from DEVICE_STATE to
- DEVICE_STATE_CHANGE, but I had it backwards, this is the right
- event to subscribe to ... ........
-
-2008-07-24 19:55 +0000 [r133449] Mark Michelson <mmichelson@digium.com>
-
- * /, main/logger.c: Merged revisions 133448 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r133448 |
- mmichelson | 2008-07-24 14:53:37 -0500 (Thu, 24 Jul 2008) | 12
- lines Print the correct PID in log messages. Prior to this
- commit, only the logger thread's PID would be printed. (closes
- issue #13150) Reported by: atis Patches: log_pid.diff uploaded by
- putnopvut (license 60) Tested by: eliel ........
-
-2008-07-24 05:21 +0000 [r133392-133405] Tilghman Lesher <tlesher@digium.com>
-
- * contrib/scripts/asterisk.logrotate, Makefile, /: Merged revisions
- 133400 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r133400 |
- tilghman | 2008-07-24 00:21:00 -0500 (Thu, 24 Jul 2008) | 3 lines
- Build the logrotate script according to paths (Closes issue
- #13147) ........
-
- * Makefile, /: Merged revisions 133391 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r133391 |
- tilghman | 2008-07-23 23:51:42 -0500 (Wed, 23 Jul 2008) | 3 lines
- Optionally install logrotate file (Closes issue #13148) ........
-
-2008-07-23 22:07 +0000 [r133300] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, /: Merged revisions 133299 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r133299 |
- murf | 2008-07-23 16:03:48 -0600 (Wed, 23 Jul 2008) | 27 lines
- (closes issue #13144) Reported by: murf Tested by: murf For: J.
- Geis The 'data' field in the ast_exten struct was being 'moved'
- from the current dialplan to the replacement dialplan. This was
- not good, as the current dialplan could have problems in the time
- between the change and when the new dialplan is swapped in. So, I
- modified the merge_and_delete code to strdup the 'data' field
- (the args to the app call), and then it's freed as normal. I
- improved a few messages; I added code to limit the number of
- calls to the context_merge_incls_swits_igps_other_registrars() to
- one per context. I don't think having it called multiple times
- per context was doing anything bad, but it was inefficient. I
- hope this fixes the problems Mr. Geiss was noting in
- asterisk-users, see
- http://lists.digium.com/pipermail/asterisk-users/2008-July/215634.html
- ........
-
-2008-07-23 21:50 +0000 [r133297] Jason Parker <jparker@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 133296 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r133296 | qwell | 2008-07-23 16:50:20 -0500
- (Wed, 23 Jul 2008) | 9 lines Merged revisions 133295 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r133295 | qwell | 2008-07-23 16:49:03 -0500 (Wed, 23 Jul
- 2008) | 1 line inbandrelease is gone - it's now inbanddisconnect
- ........ ................
-
-2008-07-23 20:39 +0000 [r133218] Brett Bryant <bbryant@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 133197 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r133197 |
- bbryant | 2008-07-23 15:33:22 -0500 (Wed, 23 Jul 2008) | 2 lines
- Fix issue where tcp in sip is enabled by default, despite what it
- says in the config sample file. Also fix "sip show settings" for
- tcp connections. ........
-
-2008-07-23 19:50 +0000 [r133042-133172] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_chanspy.c, include/asterisk/options.h, main/asterisk.c,
- /: Merged revisions 133171 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r133171 | mmichelson | 2008-07-23 14:48:03 -0500 (Wed, 23 Jul
- 2008) | 20 lines Merged revisions 133169 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r133169 | mmichelson | 2008-07-23 14:39:47 -0500 (Wed, 23 Jul
- 2008) | 12 lines As suggested by seanbright, the PSEUDO_CHAN_LEN
- in app_chanspy should be set at load time, not at compile time,
- since dahdi_chan_name is determined at load time. Also changed
- the next_unique_id_to_use to have the static qualifier. Also
- added the dahdi_chan_name_len variable so that
- strlen(dahdi_chan_name) isn't necessary. Thanks to seanbright for
- the suggestion. ........ ................
-
- * apps/app_chanspy.c, /: Merged revisions 133106 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r133106 | mmichelson | 2008-07-23 14:07:56 -0500 (Wed, 23 Jul
- 2008) | 13 lines Merged revisions 133104 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r133104 | mmichelson | 2008-07-23 14:06:16 -0500 (Wed, 23 Jul
- 2008) | 5 lines Zap/pseudo is ten characters, but DAHDI/pseudo is
- twelve. The strncmp call in next_channel should account for this.
- ........ ................
-
- * apps/app_chanspy.c, /: Merged revisions 133102 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r133102 | mmichelson | 2008-07-23 13:58:37 -0500 (Wed, 23 Jul
- 2008) | 14 lines Merged revisions 133101 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r133101 | mmichelson | 2008-07-23 13:57:17 -0500 (Wed, 23 Jul
- 2008) | 6 lines Update the "last" channel in next_channel in
- app_chanspy so that the same pseudo channel isn't constantly
- returned. related to issue #13124 ........ ................
-
- * channels/chan_dahdi.c, /: Merged revisions 133041 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r133041 | mmichelson | 2008-07-23 12:54:03 -0500
- (Wed, 23 Jul 2008) | 15 lines Merged revisions 133038 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r133038 | mmichelson | 2008-07-23 12:50:01 -0500 (Wed, 23 Jul
- 2008) | 7 lines Small cleanup. Move the declaration of the
- DAHDI_SPANINFO variable to the block where it is used. This
- allows one less #ifdef HAVE_PRI to clutter things up. Thanks to
- Tzafrir for pointing this out on #asterisk-dev ........
- ................
-
-2008-07-23 17:21 +0000 [r132978-132983] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 132981 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r132981 | tilghman | 2008-07-23 12:20:43 -0500 (Wed, 23 Jul 2008)
- | 6 lines Yet another conversion of '|' to ',' (closes issue
- #13137) Reported by: eliel Patches: chan_iax2trunk-IAXPEER.patch
- uploaded by eliel (license 64) ........
-
- * contrib/scripts/asterisk.logrotate (added), /: Merged revisions
- 132977 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r132977 |
- tilghman | 2008-07-23 12:14:56 -0500 (Wed, 23 Jul 2008) | 6 lines
- Add logrotate script for Asterisk (closes issue #13085) Reported
- by: pabelanger Patches: logrotate uploaded by pabelanger (license
- 224) ........
-
-2008-07-23 16:42 +0000 [r132965-132967] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/misdn/isdn_lib.c, /: Merged revisions 132883,132966 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r132883 | crichter | 2008-07-23 07:07:15 -0500
- (Wed, 23 Jul 2008) | 9 lines Merged revisions 132826 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r132826 | crichter | 2008-07-23 13:37:50 +0200 (Mi, 23
- Jul 2008) | 1 line another Fix because of r119585, this commit
- has broken high frequented BRI Ports, there was a possibility
- that a channel, that was marked as in_use would be reused later,
- the corresponding port could got stuck then. So it is recommended
- to upgrade for chan_misdn users. ........ ................
- r132966 | kpfleming | 2008-07-23 11:38:28 -0500 (Wed, 23 Jul
- 2008) | 2 lines use correct function name... please compile with
- --enable-dev-mode ................
-
- * include/asterisk/stringfields.h, /, main/utils.c: Merged
- revisions 132964 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r132964 | kpfleming | 2008-07-23 11:30:18 -0500 (Wed, 23 Jul
- 2008) | 10 lines Merged revisions 132872 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r132872 | kpfleming | 2008-07-23 06:52:18 -0500 (Wed, 23 Jul
- 2008) | 2 lines minor optimization for stringfields: when a field
- is being set to a larger value than it currently contains and it
- happens to be the most recent field allocated from the currentl
- pool, it is possible to 'grow' it without having to waste the
- space it is currently using (or potentially even allocate a new
- pool) ........ ................
-
-2008-07-23 08:18 +0000 [r132824] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 132823 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r132823 |
- oej | 2008-07-23 10:13:07 +0200 (Ons, 23 Jul 2008) | 8 lines
- Well, the content of a channel variable may be longer than the
- size of a pointer... Thanks, eliel! Reported by: eliel Patches:
- chan_siptrunk.SIPPEER.patch uploaded by eliel (license 64)
- (closes issue #13135) ........
-
-2008-07-22 22:20 +0000 [r132797] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 132795 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r132795 | mmichelson | 2008-07-22 17:17:09 -0500 (Tue, 22 Jul
- 2008) | 11 lines Merged revisions 132777 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ Allow
- Spiraled INVITEs to work correctly within Asterisk. Prior to this
- change, a spiraled INVITE would cause a 482 Loop Detected to be
- sent to the caller. With this change, if a potential loop is
- detected, the Request-URI is inspected to see if it has changed
- from what was originally received. If pedantic mode is on, then
- this inspection is fully RFC 3261 compliant. If pedantic mode is
- not on, then a string comparison is used to test the equality of
- the two R-URIs. This has been tested by using OpenSER to rewrite
- the R-URI and send the INVITE back to Asterisk. (closes issue
- #7403) Reported by: stephen_dredge Modified:
- branches/1.4/channels/chan_sip.c ........ ................
-
-2008-07-22 22:15 +0000 [r132793] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 132791 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r132791 | kpfleming | 2008-07-22 17:14:37 -0500 (Tue, 22 Jul
- 2008) | 2 lines correct fix made in r132777... the code *did*
- compile in dev-mode, as long as libpri was installed and enabled
- ........
-
-2008-07-22 21:59 +0000 [r132782] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c, doc/sip-retransmit.txt (added): Merged
- revisions 132703 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r132703 | oej | 2008-07-22 22:46:11 +0200 (Tis, 22 Jul 2008) | 17
- lines Merged revisions 132645 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r132645 | oej | 2008-07-22 22:10:26 +0200 (Tis, 22 Jul 2008) | 9
- lines The most common question on the #asterisk iRC channel and
- on mailing lists seems to be in regards to an error message when
- retransmit fails. This is frequently misunderstood as a failure
- of Asterisk, not a failure of the network to reach the other
- party. This document tries to assist the Asterisk user in sorting
- out these issues by explaining the logic and pointing at some
- possible causes. Hopefully, we will get other questions now :-)
- ........ ................
-
-2008-07-22 21:55 +0000 [r132780] Tilghman Lesher <tlesher@digium.com>
-
- * configs/iax.conf.sample, /, channels/chan_iax2.c: Merged
- revisions 132778 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r132778 | tilghman | 2008-07-22 16:53:40 -0500 (Tue, 22 Jul 2008)
- | 18 lines Merged revisions 132713 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r132713 | tilghman | 2008-07-22 16:19:39 -0500
- (Tue, 22 Jul 2008) | 10 lines Merged revisions 132711 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r132711 | tilghman | 2008-07-22 16:14:10 -0500 (Tue, 22 Jul 2008)
- | 2 lines Fixes for AST-2008-010 and AST-2008-011 ........
- ................ ................
-
-2008-07-22 21:54 +0000 [r132779] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 132777 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r132777 | mmichelson | 2008-07-22 16:52:24 -0500 (Tue, 22 Jul
- 2008) | 3 lines Get chan_dahdi to compile in devmode ........
-
-2008-07-22 21:23 +0000 [r132574-132729] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 132721 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r132721 | kpfleming | 2008-07-22 16:21:56 -0500
- (Tue, 22 Jul 2008) | 14 lines Merged revisions 132712 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r132712 | kpfleming | 2008-07-22 16:17:23 -0500 (Tue, 22 Jul
- 2008) | 6 lines ensure that if any alarms exist at channel
- creation time, they are handled identically to if they occurred
- later, so that later alarm clearing will work properly and 'make
- sense' (closes issue #12160) Reported by: tzafrir ........
- ................
-
- * /, configure, configure.ac, acinclude.m4: Merged revisions 132705
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r132705 | kpfleming | 2008-07-22 15:54:07 -0500
- (Tue, 22 Jul 2008) | 10 lines Merged revisions 132704 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r132704 | kpfleming | 2008-07-22 15:49:41 -0500 (Tue, 22 Jul
- 2008) | 2 lines make AST_C_COMPILE_CHECK able to print a 'pretty'
- description of what it is doing ........ ................
-
- * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /,
- configure, include/asterisk/autoconfig.h.in, configure.ac: Merged
- revisions 132643 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r132643 | kpfleming | 2008-07-22 14:59:10 -0500 (Tue, 22 Jul
- 2008) | 10 lines Merged revisions 132641 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r132641 | kpfleming | 2008-07-22 14:49:11 -0500 (Tue, 22 Jul
- 2008) | 2 lines use renamed libpri API call for controlling this
- feature (was improperly named before) ........ ................
-
- * channels/chan_dahdi.c, /: Merged revisions 132573 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r132573 | kpfleming | 2008-07-21 17:51:16 -0500
- (Mon, 21 Jul 2008) | 10 lines Merged revisions 132571 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r132571 | kpfleming | 2008-07-21 17:45:16 -0500 (Mon, 21 Jul
- 2008) | 2 lines teach chan_dahdi how to find the D-channel on BRI
- spans, and don't attempt to use channel 24 as a D-channel on
- spans of unexpected sizes ........ ................
-
-2008-07-21 21:13 +0000 [r132515] Brett Bryant <bbryant@digium.com>
-
- * configs/features.conf.sample, configs/gtalk.conf.sample, /,
- configs/jingle.conf.sample, configs/manager.conf.sample: Merged
- revisions 132514 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r132514 |
- bbryant | 2008-07-21 16:12:51 -0500 (Mon, 21 Jul 2008) | 8 lines
- Update configuration files to add missing options for jingle,
- gtalk, manager.conf, and features.conf. (closes issue #13128)
- Reported by: caio1982 Patches: missing_options1.diff uploaded by
- caio1982 (license 22) ........
-
-2008-07-21 21:02 +0000 [r132512-132513] Tilghman Lesher <tlesher@digium.com>
-
- * main/fskmodem.c (added), /, include/asterisk/fskmodem.h (added):
- Merged revisions 132511 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r132511 |
- tilghman | 2008-07-21 16:00:47 -0500 (Mon, 21 Jul 2008) | 2 lines
- (Step 2 of 2) ........
-
- * main/fskmodem.c (removed), include/asterisk/fskmodem_int.h
- (added), build_tools/cflags.xml, main/fskmodem_float.c (added),
- /, main/tdd.c, include/asterisk/fskmodem.h (removed),
- main/fskmodem_int.c (added), main/callerid.c,
- include/asterisk/fskmodem_float.h (added): Merged revisions
- 132510 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r132510 |
- tilghman | 2008-07-21 15:59:03 -0500 (Mon, 21 Jul 2008) | 5 lines
- Optionally build integer-based routines for FSK tone decoding
- (but default to the more accurate float-based routines). (Closes
- issue #11679) (Step 1 of 2) ........
-
-2008-07-21 20:55 +0000 [r132467-132509] Brett Bryant <bbryant@digium.com>
-
- * /, apps/app_sendtext.c: Merged revisions 132508 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r132508 |
- bbryant | 2008-07-21 15:54:09 -0500 (Mon, 21 Jul 2008) | 9 lines
- Fix a bug where SENDTEXTSTATUS isn't set properly when it isn't
- supported on a channel (yet _another_ useful patch by eliel).
- (closes issue #13081) Reported by: eliel Patches:
- app_sendtext.c.patch uploaded by eliel (license 64) Tested by:
- eliel ........
-
- * /, channels/chan_sip.c: Merged revisions 132468 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r132468 |
- bbryant | 2008-07-21 12:42:45 -0500 (Mon, 21 Jul 2008) | 5 lines
- Fix bug where ast_parse_arg would inadvertantly enable sip tcp
- when parsing a tcpbindaddr if it was disabled. (closes issue
- #13117) Reported by: pj ........
-
- * /, channels/chan_iax2.c: Merged revisions 132466 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r132466 | bbryant | 2008-07-21 12:22:02 -0500 (Mon, 21 Jul 2008)
- | 3 lines Fix an issue in iax2 where a call that's been rejected
- still kept an open channel on the side that attempted to make the
- call (not the side of the call that rejected the call). Changes
- were load tested and also approved by Russell. ........
-
-2008-07-21 15:34 +0000 [r132426] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 132425 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r132425 | jpeeler | 2008-07-21 10:33:13 -0500 (Mon, 21 Jul 2008)
- | 2 lines make buffers config option (chan_dahdi.conf) parsing
- safer and added logging in case of failure ........
-
-2008-07-21 14:48 +0000 [r132389-132391] Russell Bryant <russell@digium.com>
-
- * apps/app_jack.c, include/asterisk/libresample.h (removed), /,
- build_tools/menuselect-deps.in, configure,
- include/asterisk/autoconfig.h.in, main/Makefile, main/libresample
- (removed), configure.ac, codecs/codec_resample.c, makeopts.in:
- Merged revisions 132390 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r132390 |
- russell | 2008-07-21 09:47:41 -0500 (Mon, 21 Jul 2008) | 16 lines
- Remove libresample from the Asterisk source tree. It is now
- available in its own repository, and must be installed like any
- other library for Asterisk to use. The two modules that require
- it are codec_resample and app_jack. To install libresample: $ svn
- co http://svn.digium.com/svn/libresample/trunk libresample $ cd
- libresample $ ./configure $ make $ sudo make install This code is
- currently in our own repository because the build system did not
- include the appropriate targets for building a dynamic library or
- for installing the library. ........
-
- * apps/app_jack.c, /, codecs/codec_resample.c: Merged revisions
- 132388 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r132388 |
- russell | 2008-07-21 08:51:05 -0500 (Mon, 21 Jul 2008) | 3 lines
- Enable higher quality resampling, as it doesn't have a noticeable
- performance impact on my machine .. ........
-
-2008-07-19 16:47 +0000 [r132313] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, LICENSE: Merged revisions 132312 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r132312 | kpfleming | 2008-07-19 11:46:33 -0500 (Sat, 19 Jul
- 2008) | 10 lines Merged revisions 132311 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r132311 | kpfleming | 2008-07-19 11:45:52 -0500 (Sat, 19 Jul
- 2008) | 2 lines grant a license exception to allow distribution
- of Asterisk binaries that use the UW IMAP Toolkit (which is
- licensed under a non-GPL-compatible license) ........
- ................
-
-2008-07-19 10:47 +0000 [r132278] Michiel van Baak <michiel@vanbaak.info>
-
- * res/res_config_sqlite.c, /: Merged revisions 132277 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r132277 | mvanbaak | 2008-07-19 12:46:12 +0200 (Sat, 19 Jul 2008)
- | 7 lines fix a couple of comments in sqlite resource driver.
- (closes issue #13110) Reported by: gknispel_proformatique
- Patches: res_config_sqlite_comments.patch uploaded by gknispel
- (license 261) ........
-
-2008-07-18 22:20 +0000 [r132245] Brett Bryant <bbryant@digium.com>
-
- * main/manager.c, /: Merged revisions 132242 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r132242 |
- bbryant | 2008-07-18 17:19:56 -0500 (Fri, 18 Jul 2008) | 4 lines
- Fixes problem where manager users loaded from users.conf would be
- removed early (before the routine to load the configuration was
- finished) because a variable wasn't initialized. ........
-
-2008-07-18 20:58 +0000 [r132114-132207] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/say.c: Merged revisions 132113 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r132113 | tilghman | 2008-07-18 14:09:39 -0500 (Fri, 18 Jul 2008)
- | 14 lines Merged revisions 132112 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r132112 | tilghman | 2008-07-18 14:06:10 -0500 (Fri, 18 Jul 2008)
- | 6 lines Fix for Taiwanese number syntax (closes issue #12319)
- Reported by: CharlesWang Patches: saynumber-tw-1.4.18.1.patch
- uploaded by CharlesWang (license 444) ........ ................
-
-2008-07-18 18:53 +0000 [r132111] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 132108 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r132108 | mattf | 2008-07-18 13:50:00 -0500 (Fri, 18 Jul 2008) |
- 1 line Make sure we break the poll so that messages queued will
- be sent on the SS7 when using CLI commands for blocking and
- blocking of CICs and linksets. ........
-
-2008-07-18 18:51 +0000 [r132110] Tilghman Lesher <tlesher@digium.com>
-
- * main/config.c, /: Merged revisions 132109 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r132109 | tilghman | 2008-07-18 13:50:37 -0500 (Fri, 18 Jul 2008)
- | 14 lines Merged revisions 132107 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r132107 | tilghman | 2008-07-18 13:47:50 -0500 (Fri, 18 Jul 2008)
- | 6 lines Textual clarification (closes issue #13106) Reported
- by: flefoll Patches: config.c.br14.120173.patch-unknown-directive
- uploaded by flefoll (license 244) ........ ................
-
-2008-07-18 17:56 +0000 [r132051] Brett Bryant <bbryant@digium.com>
-
- * main/hashtab.c, /, cdr/cdr_radius.c: Merged revisions 132050 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r132050 | bbryant | 2008-07-18 12:55:41 -0500 (Fri, 18
- Jul 2008) | 8 lines Fix magic Revision keywords in hashtab.c and
- change cdr_radius.c to use the same keyword as the other files
- (patch by eliel). (closes issue #13104) Reported by: eliel
- Patches: revision.patch uploaded by eliel (license 64) ........
-
-2008-07-18 17:40 +0000 [r131984-132047] Tilghman Lesher <tlesher@digium.com>
-
- * main/sched.c, /: Merged revisions 131989 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r131989 | tilghman | 2008-07-18 12:10:34 -0500 (Fri, 18 Jul 2008)
- | 10 lines Merged revisions 131988 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r131988 | tilghman | 2008-07-18 12:10:01 -0500 (Fri, 18 Jul 2008)
- | 2 lines Oops ........ ................
-
- * main/sched.c, /, include/asterisk/sched.h: Merged revisions
- 131986 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r131986 | tilghman | 2008-07-18 11:48:18 -0500 (Fri, 18 Jul 2008)
- | 10 lines Merged revisions 131985 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r131985 | tilghman | 2008-07-18 11:46:23 -0500 (Fri, 18 Jul 2008)
- | 2 lines Preserve ABI compatibility with last change ........
- ................
-
- * main/sched.c, /, include/asterisk/sched.h, channels/chan_iax2.c:
- Merged revisions 131982 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r131982 | tilghman | 2008-07-18 11:33:56 -0500 (Fri, 18 Jul 2008)
- | 10 lines Merged revisions 131970 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r131970 | tilghman | 2008-07-18 11:30:31 -0500 (Fri, 18 Jul 2008)
- | 2 lines Make the ast_assert call within ast_sched_del report
- something useful. ........ ................
-
-2008-07-18 16:16 +0000 [r131924] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/dlfcn.c (removed), main/loader.c, /, main/Makefile,
- include/asterisk/dlfcn-compat.h (removed): Merged revisions
- 131923 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r131923 | kpfleming | 2008-07-18 11:16:12 -0500 (Fri, 18 Jul
- 2008) | 10 lines Merged revisions 131921 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r131921 | kpfleming | 2008-07-18 11:15:41 -0500 (Fri, 18 Jul
- 2008) | 2 lines remove the dlfcn compatibility stuff, because no
- platforms that Asterisk currently runs on it use it, and it
- doesn't build anyway ........ ................
-
-2008-07-18 15:39 +0000 [r131917] Brett Bryant <bbryant@digium.com>
-
- * /, main/features.c: Merged revisions 131916 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r131916 | bbryant | 2008-07-18 10:38:22 -0500 (Fri, 18 Jul 2008)
- | 12 lines Merged revisions 131915 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r131915 | bbryant | 2008-07-18 10:34:42 -0500 (Fri, 18 Jul 2008)
- | 4 lines Fix a bug in blind transfers where the BLINDTRANSFER
- variable isn't always set to the other end of the blind transfer.
- (closes issue #12586) ........ ................
-
-2008-07-17 22:45 +0000 [r131869] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 131868 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r131868 | jpeeler | 2008-07-17 17:40:00 -0500 (Thu, 17 Jul 2008)
- | 6 lines Add configuration option to chan_dahdi.conf to allow
- buffering policy and number of buffers to be configured per
- channel. Syntax: buffers=<num of buffers>,<policy> Where the
- number of buffers is some non-negative integer and the policy is
- either "full", "half", or "immediate". ........
-
-2008-07-17 21:27 +0000 [r131830] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_senddtmf.c: Merged revisions 131824 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r131824 |
- mmichelson | 2008-07-17 16:26:41 -0500 (Thu, 17 Jul 2008) | 10
- lines Document that the duration of dtmf may be passed to the
- SendDTMF application. Also correct the default pause between
- digits. (closes issue #13102) Reported by: eliel Patches:
- app_senddtmf.c.patch uploaded by eliel (license 64) ........
-
-2008-07-17 20:38 +0000 [r131754-131792] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 131791 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r131791 | tilghman | 2008-07-17 15:37:14 -0500
- (Thu, 17 Jul 2008) | 15 lines Merged revisions 131790 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r131790 | tilghman | 2008-07-17 15:35:44 -0500 (Thu, 17 Jul 2008)
- | 7 lines Revert part of issue #5620 (revision 6965) as it
- appears that it was in error. This should fix talk call progress
- on analog lines. (closes issue #12178) Reported by: michael-fig
- Patches: 20080717__bug12178.diff.txt uploaded by Corydon76
- (license 14) ........ ................
-
- * res/res_config_sqlite.c, /: Merged revisions 131753 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r131753 | tilghman | 2008-07-17 13:36:34 -0500 (Thu, 17 Jul 2008)
- | 6 lines Fix memory leaks (closes issue #13099) Reported by:
- gknispel_proformatique Patches:
- res_config_sqlite_leak_on_error.patch uploaded by gknispel
- (license 261) ........
-
-2008-07-17 18:15 +0000 [r131718] Brett Bryant <bbryant@digium.com>
-
- * /, main/features.c: Merged revisions 131717 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r131717 |
- bbryant | 2008-07-17 13:14:42 -0500 (Thu, 17 Jul 2008) | 8 lines
- Fix a memory leak in register_group_feature when attempting to
- register a feature without specifying a group or feature to
- register. (closes issue #13101) Reported by: eliel Patches:
- features.c.patch uploaded by eliel (license 64) ........
-
-2008-07-17 15:46 +0000 [r131682] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_config_sqlite.c, /: Merged revisions 131681 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r131681 | tilghman | 2008-07-17 10:45:25 -0500 (Thu, 17 Jul 2008)
- | 4 lines Fix memory leak. (Closes issue #13096) Reported by
- gknispel_proformatique ........
-
-2008-07-16 23:56 +0000 [r131571] Steve Murphy <murf@digium.com>
-
- * /: The commit from 131570 should not be applied to 1.6.0, as it
- is not as necessary, because log_show_lock in trunk is not
- available in 1.6.0, and is estimated to be the only function that
- might care about the lock_addr values.
-
-2008-07-16 22:18 +0000 [r131493] Brett Bryant <bbryant@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 131492 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r131492 | bbryant | 2008-07-16 17:17:36 -0500
- (Wed, 16 Jul 2008) | 14 lines Merged revisions 131491 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r131491 | bbryant | 2008-07-16 17:17:07 -0500 (Wed, 16 Jul 2008)
- | 6 lines Fix a bug in iax2 registration that allowed peers to
- register with case-insensitive names (user_cmp_cb and peer_cmp_cb
- are now both case-sensitive). (closes issue #13091) ........
- ................
-
-2008-07-16 21:54 +0000 [r131455-131486] Brett Bryant <bbryant@digium.com>
-
- * /, funcs/func_sysinfo.c: Merged revisions 131484 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r131484 | bbryant | 2008-07-16 16:54:08 -0500 (Wed, 16 Jul 2008)
- | 4 lines Fixes sysinfo operator issue also fixed elsewhere in
- r131445. (issue #13057) ........
-
- * main/asterisk.c, /: Merged revisions 131445 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r131445 |
- bbryant | 2008-07-16 16:24:18 -0500 (Wed, 16 Jul 2008) | 9 lines
- Fixes an issue with "core show sysinfo" that used the wrong
- operator to calculate the number of bytes from a sysinfo
- structure. unsigned long. (closes issue #13057) Reported by:
- eliel Patches: asterisk.c.patch uploaded by eliel (license 64)
- ........
-
-2008-07-16 20:48 +0000 [r131423] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 131422 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r131422 | russell | 2008-07-16 15:48:27 -0500
- (Wed, 16 Jul 2008) | 15 lines Merged revisions 131421 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r131421 | russell | 2008-07-16 15:47:53 -0500 (Wed, 16 Jul 2008)
- | 7 lines Always ensure that the channel's tech_pvt reference is
- NULL after calling the destroy callback. (closes issue #13060)
- Reported by: jpgrayson Patches: chan_iax2_tech_pvt_crash.patch
- uploaded by jpgrayson (license 492) ........ ................
-
-2008-07-16 20:24 +0000 [r131301-131378] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 131375 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r131375 | mmichelson | 2008-07-16 15:24:12 -0500 (Wed, 16 Jul
- 2008) | 22 lines Merged revisions 131369 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r131369 | mmichelson | 2008-07-16 15:23:02 -0500 (Wed, 16 Jul
- 2008) | 14 lines Move the init_queue call back to where it used
- to be (changed Sept 12 last year). It was moved then to prevent a
- memory leak. Since then, the same memory leak recurred and was
- fixed in a better way. Now it has been found that the placement
- of this init_queue call can cause problems if a realtime queue
- has values changed to an empty string. The problem is that the
- default value for that queue parameter would not be set. (closes
- issue #13084) Reported by: elbriga ........ ................
-
- * res/res_config_sqlite.c, /: Merged revisions 131361 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r131361 | mmichelson | 2008-07-16 14:57:02 -0500 (Wed, 16 Jul
- 2008) | 9 lines Don't try to dereference the dbfile pointer if we
- know that it's NULL. (closes issue #13092) Reported by:
- gknispel_proformatique Patches:
- trunk_sqlite_check_vars_null.patch uploaded by gknispel (license
- 261) ........
-
- * /, apps/app_queue.c: Merged revisions 131358 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r131358 | mmichelson | 2008-07-16 14:37:42 -0500 (Wed, 16 Jul
- 2008) | 14 lines Merged revisions 131357 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r131357 | mmichelson | 2008-07-16 14:37:08 -0500 (Wed, 16 Jul
- 2008) | 6 lines Apparently, "thread safety" is important,
- whatever that means. :P (Thanks Russell!) ........
- ................
-
- * /, apps/app_queue.c: Merged revisions 131300 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r131300 | mmichelson | 2008-07-16 13:59:27 -0500 (Wed, 16 Jul
- 2008) | 21 lines Merged revisions 131299 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r131299 | mmichelson | 2008-07-16 13:57:34 -0500 (Wed, 16 Jul
- 2008) | 13 lines Make absolutely certain that the transfer
- datastore is removed from the calling channel once the caller is
- finished in the queue. This could have weird con- sequences when
- dialing local queue members when multiple transfers occur on a
- single call. Also fixed a memory leak that would occur when an
- attended transfer occurred from a queue member. (closes issue
- #13047) Reported by: festr ........ ................
-
-2008-07-16 18:20 +0000 [r131248] Steve Murphy <murf@digium.com>
-
- * res/ael/pval.c, /: Merged revisions 131243 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r131243 | murf | 2008-07-16 11:59:33 -0600 (Wed, 16 Jul 2008) |
- 27 lines Merged revisions 131242 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r131242 | murf | 2008-07-16 11:53:43 -0600 (Wed, 16 Jul 2008) |
- 19 lines (closes issue #13090) Reported by: murf The problem was
- that, esoteric as it is, because the hangerupper context
- immediately preceded the std-priv-extent macro, that the checking
- code accidentally would fall from traversing hangerupper into the
- std-priv-exten macro, where it would hit the hangerupper in the
- 'includes', and proceed into an infinite recursion. A small fix
- to traverse into the statements of the context instead of the
- context solves this issue. I also added some commented out
- printfs for debug, which were pretty handy in the face of a dorky
- gdb. This was a problem around since the package was first
- written; but evidently pretty rare in turning up in the field.
- ........ ................
-
-2008-07-16 15:04 +0000 [r131206] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_agent.c: add missing terminator argument for
- ast_event_subscribe(). Without it the function will randomly walk
- on the stack possibly causing a panic
-
-2008-07-16 00:54 +0000 [r131168] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/logger.c: Merged revisions 131166 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r131166 |
- tilghman | 2008-07-15 19:52:48 -0500 (Tue, 15 Jul 2008) | 3 lines
- Fix rotate strategy (Closes issue #13086) ........
-
-2008-07-15 23:41 +0000 [r131131] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, /: Merged revisions 131129 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r131129 |
- murf | 2008-07-15 17:36:19 -0600 (Tue, 15 Jul 2008) | 21 lines
- (closes issue #12960) Reported by: mnicholson Spent most of the
- day on this bug, and the solution was so simple. Just had to find
- and understand the problem. The problem was, that the routine to
- copy the existing switches, includes, and ignorepats from the old
- context to the new one, wasn't getting called when the context is
- already existent. (In other words, if AEL is adding a new context
- to the mix, they get copied, but if pbx_config already defined a
- context, then the copy wasn't happening. This made no sense, so I
- moved the call to copy the includes & etc, no matter the case.
- ........
-
-2008-07-15 18:47 +0000 [r131073] Russell Bryant <russell@digium.com>
-
- * /, res/res_agi.c: Merged revisions 131072 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r131072 |
- russell | 2008-07-15 13:46:40 -0500 (Tue, 15 Jul 2008) | 5 lines
- Fix a couple of places in res_agi where the agi_commands lock
- would not be released, causing a deadlock. (Reported by mvanbaak
- in #asterisk-dev, discovered by bbryant's change to the lock
- tracking code to yell at you if a thread exits with a lock still
- held) ........
-
-2008-07-15 18:29 +0000 [r131060] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c, main/manager.c, /, channels/chan_sip.c: Merged
- revisions 131044 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r131044 | tilghman | 2008-07-15 13:25:34 -0500 (Tue, 15 Jul 2008)
- | 16 lines Merged revisions 130959 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r130959 | tilghman | 2008-07-15 12:19:13 -0500 (Tue, 15 Jul 2008)
- | 8 lines astman_send_error does not need a newline appended --
- the API takes care of that for us. (closes issue #13068) Reported
- by: gknispel_proformatique Patches:
- asterisk_1_4_astman_send.patch uploaded by gknispel (license 261)
- asterisk_trunk_astman_send.patch uploaded by gknispel (license
- 261) ........ ................
-
-2008-07-15 18:00 +0000 [r131014] Michiel van Baak <michiel@vanbaak.info>
-
- * main/cdr.c, /: Merged revisions 131013 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r131013 | mvanbaak | 2008-07-15 19:49:48 +0200 (Tue, 15 Jul 2008)
- | 15 lines Merged revisions 131012 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r131012 | mvanbaak | 2008-07-15 19:47:15 +0200 (Tue, 15 Jul 2008)
- | 7 lines remove 4 lines of redundant code. (closes issue #13080)
- Reported by: gknispel_proformatique Patches:
- trunk_ast_cdr_setapp.patch uploaded by gknispel (license 261)
- ........ ................
-
-2008-07-15 13:14 +0000 [r130946] Steve Murphy <murf@digium.com>
-
- * utils/conf2ael.c, utils/Makefile, res/ael/pval.c,
- channels/chan_skinny.c, res/ael/ael.tab.c, main/features.c,
- pbx/pbx_ael.c, res/ael/ael_lex.c, res/ael/ael.tab.h,
- utils/ael_main.c, include/asterisk/pbx.h, utils/extconf.c,
- res/ael/ael.flex, pbx/pbx_config.c, apps/app_stack.c,
- apps/app_dial.c, main/pbx.c, include/asterisk/pval.h, /,
- channels/chan_sip.c, apps/app_meetme.c, res/ael/ael.y,
- channels/chan_iax2.c, apps/app_queue.c: Merged revisions 130145
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- Merging this rev from trunk to 1.6.0 was not simple. Why? Because
- we've enhanced trunk to do a [fast] merge-and-delete operation
- which also solved problems with contexts having entries from
- different registrars. Fast as in the amount of time the contexts
- are locked down. That *is* fast, but traversing the entire
- dialplan looking for priorities to delete takes more time
- overall. This particular fix involved pulling in those
- enhancements from trunk, along with all the various fixes and
- refinements made along the way. Merging all this from trunk into
- 1.6 involved: a. mergetrunk6 in the stuff from 130145; b. revert
- all but the prop changes c. catalog all revisions to pbx.c since
- 1.6.0 was forked (at rev 105596). d. catalog all revisions to
- pbx.c in trunk since 1.6.0 was forked, making special note of all
- revs that were not merged into 1.6.0. e. study each rev in trunk
- not applied to 1.6.0, and determine if it was involved in the
- merge_and_delete enhancements in trunk. 25 commits were done in
- 1.6.0, all but one (106306) was a merge from trunk. Trunk had 22
- additional changes, of which 7 were involved in the
- merge_and_delete enhancements: 106757 108894 109169 116461 123358
- 130145 130297 f. Go to trunk and collect patches, one by one, of
- the changes made by each rev across the entire source tree, using
- svn diff -c <num> > pfile g. Apply each patch in order to 1.6.0,
- and resolve all failures and compilation problems before
- proceding to the next patch. h. test the stuff. i. profit!
- ........ r130145 | murf | 2008-07-11 12:24:31 -0600 (Fri, 11 Jul
- 2008) | 40 lines (closes issue #13041) Reported by: eliel Tested
- by: murf (closes issue #12960) Reported by: mnicholson In this
- 'omnibus' fix, I **think** I solved both the problem in 13041,
- where unloading pbx_ael.so caused crashes, or incomplete removal
- of previous registrar'ed entries. And I added code to completely
- remove all includes, switches, and ignorepats that had a matching
- registrar entry, which should appease 12960. I also added a lot
- of seemingly useless brackets around single statement if's, which
- helped debug so much that I'm leaving them there. I added a
- routine to check the correlation between the extension tree lists
- and the hashtab tables. It can be amazingly helpful when you have
- lots of dialplan stuff, and need to narrow down where a problem
- is occurring. It's ifdef'd out by default. I cleaned up the code
- around the new CIDmatch code. It was leaving hanging extens with
- bad ptrs, getting confused over which objects to remove, etc. I
- tightened up the code and changed the call to remove_exten in the
- merge_and_delete code. I added more conditions to check for empty
- context worthy of deletion. It's not empty if there are any
- includes, switches, or ignorepats present. If I've missed
- anything, please re-open this bug, and be prepared to supply
- example dialplan code. ........
-
-2008-07-15 00:00 +0000 [r130891] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 130890 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r130890 | tilghman | 2008-07-14 18:59:54 -0500
- (Mon, 14 Jul 2008) | 16 lines Merged revisions 130889 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r130889 | tilghman | 2008-07-14 18:59:13 -0500 (Mon, 14 Jul 2008)
- | 8 lines Override the callerid in all cases when the callerid is
- set in the user, not just when a remote callerid is set. Also, if
- not set in the user, allow the remote CallerID to pass through.
- (closes issue #12875) Reported by: dimas Patches:
- 20080714__bug12875.diff.txt uploaded by Corydon76 (license 14)
- ........ ................
-
-2008-07-14 22:24 +0000 [r130795-130855] Mark Michelson <mmichelson@digium.com>
-
- * main/asterisk.c, /: Merged revisions 130854 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r130854 |
- mmichelson | 2008-07-14 17:22:57 -0500 (Mon, 14 Jul 2008) | 9
- lines Fix a memory leak in the case that /dev/null cannot be
- opened when running startup commands from cli.conf (closes issue
- #13066) Reported by: eliel Patches: asterisk.c.patch uploaded by
- eliel (license 64) ........
-
- * apps/app_dial.c, /: Merged revisions 130794 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r130794 | mmichelson | 2008-07-14 12:54:11 -0500 (Mon, 14 Jul
- 2008) | 16 lines Merged revisions 130792 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r130792 | mmichelson | 2008-07-14 12:50:21 -0500 (Mon, 14 Jul
- 2008) | 8 lines Add a check to the CAN_EARLY_BRIDGE macro in
- app_dial to be sure there are no audiohooks present on the
- channels involved. This fixed a one-way audio situation I had in
- my test setup. I couldn't find any open issues that suggested
- one-way audio with regards to mixmonitor (or other audiohook)
- usage, though. ........ ................
-
-2008-07-14 17:22 +0000 [r130752] Michiel van Baak <michiel@vanbaak.info>
-
- * main/dnsmgr.c, /: Merged revisions 130744 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r130744 | mvanbaak | 2008-07-14 19:21:18 +0200 (Mon, 14 Jul 2008)
- | 18 lines Merged revisions 130735 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r130735 | mvanbaak | 2008-07-14 19:10:21 +0200 (Mon, 14 Jul 2008)
- | 10 lines notify the user that dnsmgr refresh wont work when
- dnsmgr is not enabled. Previously this command would
- automagically appear and disappear. This was confusing. (closes
- issue #12796) Reported by: chappell Patches:
- dnsmgr_refresh_3.diff uploaded by chappell (license 8) Tested by:
- russell, chappell, mvanbaak ........ ................
-
-2008-07-14 10:40 +0000 [r130636-130637] Russell Bryant <russell@digium.com>
-
- * /, include/asterisk/astobj.h: Merged revisions 129987 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r129987 | russell | 2008-07-11 09:22:44 -0500
- (Fri, 11 Jul 2008) | 10 lines Merged revisions 129970 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r129970 | russell | 2008-07-11 09:18:43 -0500 (Fri, 11 Jul 2008)
- | 2 lines add a simple ASTOBJ_TRYWRLOCK macro ... ........
- ................
-
- * /, main/audiohook.c: Merged revisions 130635 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r130635 | russell | 2008-07-14 05:39:23 -0500 (Mon, 14 Jul 2008)
- | 10 lines Merged revisions 130634 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r130634 | russell | 2008-07-14 05:38:14 -0500 (Mon, 14 Jul 2008)
- | 2 lines Bump up the debug level for a message. ........
- ................
-
-2008-07-13 23:20 +0000 [r130575-130582] Michiel van Baak <michiel@vanbaak.info>
-
- * /, doc/tex/Makefile, build_tools/prep_tarball, res/Makefile:
- Merged revisions 130578 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r130578 |
- mvanbaak | 2008-07-14 01:14:03 +0200 (Mon, 14 Jul 2008) | 15
- lines Make all sed calls Posix sed compatible. To make sure
- nobody commits script-modified files we first make a backup of
- asterisk.tex, run the script, generate the pdf and / or html, and
- put the original asterisk.tex back. This will guard us for the
- stuff that happened before that someone committed a locally
- modified asterisk.tex, with changes done by this script. (closes
- issue #13062) Reported by: mvanbaak Patches:
- sed_without-i-v3.diff uploaded by mvanbaak (license 7) Tested by:
- mvanbaak Feedback from Corydon. Thanks for taking the time to go
- through this. ........
-
- * main/manager.c, /: Merged revisions 130574 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r130574 | mvanbaak | 2008-07-14 00:50:31 +0200 (Mon, 14 Jul 2008)
- | 16 lines Merged revisions 130573 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r130573 | mvanbaak | 2008-07-14 00:48:51 +0200 (Mon, 14 Jul 2008)
- | 8 lines fix memory leak when originate from manager cannot
- create a thread (closes issue #13069) Reported by:
- gknispel_proformatique Patches:
- asterisk_trunk_action_originate.patch uploaded by gknispel
- (license 261) Tested by: gknispel_proformatique, mvanbaak
- ........ ................
-
-2008-07-13 17:59 +0000 [r130516] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 130515 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r130515 | tilghman | 2008-07-13 12:58:47 -0500
- (Sun, 13 Jul 2008) | 12 lines Merged revisions 130514 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r130514 | tilghman | 2008-07-13 12:56:10 -0500 (Sun, 13 Jul 2008)
- | 4 lines Reverting 2 changesets, as it breaks incoming IAX2
- calls (Related to issue #12963) Reported by: mvanbaak ........
- ................
-
-2008-07-13 15:00 +0000 [r130480] Michiel van Baak <michiel@vanbaak.info>
-
- * doc/tex/asterisk.tex, /: Merged revisions 130479 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r130479 | mvanbaak | 2008-07-13 16:58:40 +0200 (Sun, 13 Jul 2008)
- | 3 lines restore ASTERISKVERSION marker to asterisk.tex. This
- got lost in commit 97634 ........
-
-2008-07-13 02:35 +0000 [r130445] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 130444 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r130444 | tilghman | 2008-07-12 21:34:32 -0500 (Sat, 12 Jul 2008)
- | 2 lines Unlock list before returning ........
-
-2008-07-11 21:39 +0000 [r130294] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 130293 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r130293 | mattf | 2008-07-11 16:36:26 -0500 (Fri, 11 Jul 2008) |
- 1 line Support new TRANSPORT definitions in libss7 ........
-
-2008-07-11 20:04 +0000 [r130238] Mark Michelson <mmichelson@digium.com>
-
- * /, main/audiohook.c: Merged revisions 130237 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r130237 | mmichelson | 2008-07-11 15:03:55 -0500 (Fri, 11 Jul
- 2008) | 11 lines Merged revisions 130236 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r130236 | mmichelson | 2008-07-11 15:03:23 -0500 (Fri, 11 Jul
- 2008) | 3 lines Remove redundant logic ........ ................
-
-2008-07-11 19:57 +0000 [r130231-130235] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_dahdi.c, /, channels/chan_agent.c, utils/astman.c:
- Merged revisions 130230 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r130230 |
- tilghman | 2008-07-11 14:40:55 -0500 (Fri, 11 Jul 2008) | 2 lines
- Fix trunk breakage ........
-
-2008-07-11 19:14 +0000 [r130175] Mark Michelson <mmichelson@digium.com>
-
- * /, main/audiohook.c: Merged revisions 130174 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r130174 | mmichelson | 2008-07-11 14:14:15 -0500 (Fri, 11 Jul
- 2008) | 15 lines Merged revisions 130173 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r130173 | mmichelson | 2008-07-11 14:13:29 -0500 (Fri, 11 Jul
- 2008) | 7 lines Fix a typo in audiohook_read_frame_both. While
- this change has not been proven to fix any specific issue, it is
- incorrect and could cause unforeseen problems. ........
- ................
-
-2008-07-11 18:53 +0000 [r130171] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 130170 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r130170 | tilghman | 2008-07-11 13:52:42 -0500
- (Fri, 11 Jul 2008) | 15 lines Merged revisions 130169 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r130169 | tilghman | 2008-07-11 13:51:56 -0500 (Fri, 11 Jul 2008)
- | 7 lines Ensure that a destination callno of 0 will not match
- for frames that do not start a dialog (new, lagrq, and ping).
- (closes issue #12963) Reported by: russellb Patches:
- chan_iax2_dup_new_fix4.patch uploaded by jpgrayson (license 492)
- ........ ................
-
-2008-07-11 18:33 +0000 [r130168] Sean Bright <sean.bright@gmail.com>
-
- * /, channels/chan_sip.c: Merged revisions 130167 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r130167 |
- seanbright | 2008-07-11 14:32:26 -0400 (Fri, 11 Jul 2008) | 1
- line Missed one. Formatting only. ........
-
-2008-07-11 18:14 +0000 [r130130] Brett Bryant <bbryant@digium.com>
-
- * main/cli.c, channels/chan_jingle.c, channels/chan_dahdi.c,
- channels/chan_skinny.c, main/abstract_jb.c, apps/app_minivm.c,
- codecs/codec_resample.c, codecs/codec_dahdi.c,
- apps/app_chanspy.c, main/asterisk.c, apps/app_milliwatt.c,
- main/dsp.c, codecs/codec_g722.c, /, channels/chan_sip.c,
- main/threadstorage.c, utils/astman.c, main/utils.c,
- channels/chan_gtalk.c, pbx/dundi-parser.c: Merged revisions
- 130129 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r130129 |
- bbryant | 2008-07-11 13:09:35 -0500 (Fri, 11 Jul 2008) | 8 lines
- Janitor patch to change uses of sizeof to ARRAY_LEN (closes issue
- #13054) Reported by: pabelanger Patches: ARRAY_LEN.patch2
- uploaded by pabelanger (license 224) Tested by: seanbright
- ........
-
-2008-07-11 17:30 +0000 [r130127] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 130126 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r130126 | tilghman | 2008-07-11 12:29:24 -0500
- (Fri, 11 Jul 2008) | 17 lines Merged revisions 130102 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r130102 | tilghman | 2008-07-11 11:50:42 -0500 (Fri, 11 Jul 2008)
- | 9 lines Pass the devicestate from an underlying channel up
- through the Agent channel. This should make the Agent always
- report the correct device state, even when the underlying channel
- is used for other purposes. (closes issue #12773) Reported by:
- davidw Patches: 20080710__bug12773.diff.txt uploaded by Corydon76
- (license 14) Tested by: davidw ........ ................
-
-2008-07-11 16:18 +0000 [r129936-130045] Kevin P. Fleming <kpfleming@digium.com>
-
- * doc/ss7.txt, /, contrib/utils/zones2indications.c, CHANGES:
- Merged revisions 130044 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r130044 |
- kpfleming | 2008-07-11 11:18:01 -0500 (Fri, 11 Jul 2008) | 2
- lines clean up a bunch more Zaptel-related references ........
-
- * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /,
- configure, include/asterisk/autoconfig.h.in, configure.ac: Merged
- revisions 130040 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r130040 | kpfleming | 2008-07-11 10:57:17 -0500 (Fri, 11 Jul
- 2008) | 12 lines Merged revisions 130039 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r130039 | kpfleming | 2008-07-11 10:41:56 -0500 (Fri, 11 Jul
- 2008) | 4 lines add support for a configuration parameter for
- 'inband audio during RELEASE', which is currently mandatory in
- libpri-1.4.4 but will become configurable in libpri-1.4.5 later
- today (related to issue #13042) ........ ................
-
- * /, main/astmm.c: Merged revisions 129968 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r129968 | kpfleming | 2008-07-11 09:16:15 -0500 (Fri, 11 Jul
- 2008) | 18 lines Merged revisions 129966 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r129966 | kpfleming | 2008-07-11 09:03:52 -0500 (Fri, 11 Jul
- 2008) | 5 lines fix a flaw found while experimenting with
- structure alignment and padding; low-fence checking would not
- work properly on 64-bit platforms, because the compiler was
- putting 4 bytes of padding between the fence field and the
- allocation memory block added a very obvious runtime warning if
- this condition reoccurs, so the developer who broke it can be
- chastised into fixing it :-) ........ r129967 | kpfleming |
- 2008-07-11 09:03:52 -0500 (Fri, 11 Jul 2008) | 5 lines simplify
- calculation ........ ................
-
- * /, sounds/Makefile: Merged revisions 129916 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r129916 | kpfleming | 2008-07-11 07:21:29 -0500 (Fri, 11 Jul
- 2008) | 10 lines Merged revisions 129907 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r129907 | kpfleming | 2008-07-11 07:15:42 -0500 (Fri, 11 Jul
- 2008) | 2 lines don't attempt to set user/group ownership of
- extracted sound files (reported on asterisk-users) ........
- ................
-
-2008-07-11 01:01 +0000 [r129865] Sean Bright <sean.bright@gmail.com>
-
- * res/res_config_pgsql.c, /, res/res_config_ldap.c: Merged
- revisions 129864 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r129864 |
- seanbright | 2008-07-10 20:55:06 -0400 (Thu, 10 Jul 2008) | 1
- line Fix some usages of snprintf, and clarify a couple variable
- names. ........
-
-2008-07-10 22:07 +0000 [r129764-129805] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 129804 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r129804 | tilghman | 2008-07-10 17:06:07 -0500
- (Thu, 10 Jul 2008) | 16 lines Merged revisions 129803 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r129803 | tilghman | 2008-07-10 16:57:05 -0500 (Thu, 10 Jul 2008)
- | 8 lines Correctly deal with duplicate NEW frames (due to
- retransmission). Also, fixup the destination call number matching
- to be more strict and reliable. (closes issue #12963) Reported
- by: jpgrayson Patches: chan_iax2_dup_new_fix3.patch uploaded by
- jpgrayson (license 492) Tested by: jpgrayson, Corydon76 ........
- ................
-
- * res/res_config_odbc.c, /: Merged revisions 129758 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r129758 | tilghman | 2008-07-10 16:23:23 -0500
- (Thu, 10 Jul 2008) | 10 lines Merged revisions 129741 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r129741 | tilghman | 2008-07-10 16:19:48 -0500 (Thu, 10 Jul 2008)
- | 2 lines Oops ........ ................
-
-2008-07-10 21:05 +0000 [r129739] Terry Wilson <twilson@digium.com>
-
- * Makefile, /: Merged revisions 129738 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r129738 |
- twilson | 2008-07-10 15:56:20 -0500 (Thu, 10 Jul 2008) | 2 lines
- Move phoneprov config files to be installed with 'make samples'
- so changes aren't inadvertently lost on a 'make install' ........
-
-2008-07-10 19:14 +0000 [r129685] Brett Bryant <bbryant@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 129684 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r129684 |
- bbryant | 2008-07-10 14:13:12 -0500 (Thu, 10 Jul 2008) | 8 lines
- Fixes a bug where the interface for a queue member gets reloaded
- as the state_interface, if a state_interface was set, on reload
- because the state_interface isn't stored in the ast_db. (closes
- issue #13043) Reported by: jvandal Patches: app_queue.patch
- uploaded by jvandal (license 413) ........
-
-2008-07-10 18:20 +0000 [r129640-129647] Sean Bright <sean.bright@gmail.com>
-
- * /, channels/chan_sip.c: Merged revisions 129642 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r129642 |
- seanbright | 2008-07-10 14:19:17 -0400 (Thu, 10 Jul 2008) | 1
- line A couple more minor text changes ........
-
- * /, channels/chan_sip.c: Merged revisions 129638 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r129638 |
- seanbright | 2008-07-10 14:16:21 -0400 (Thu, 10 Jul 2008) | 1
- line Remove extraneous \n. Pointed out by eliel on #asterisk-dev.
- ........
-
-2008-07-10 16:13 +0000 [r129570] Russell Bryant <russell@digium.com>
-
- * sample.call, /: Merged revisions 129569 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r129569 | russell | 2008-07-10 11:12:51 -0500 (Thu, 10 Jul 2008)
- | 11 lines Merged revisions 129567 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r129567 | russell | 2008-07-10 11:03:59 -0500 (Thu, 10 Jul 2008)
- | 3 lines Note that pbx_spool.so is the module used for call
- files (inspired by a question in #asterisk) ........
- ................
-
-2008-07-10 14:09 +0000 [r129504-129507] Sean Bright <sean.bright@gmail.com>
-
- * /, main/editline: Merged revisions 129503 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r129503 |
- seanbright | 2008-07-10 09:54:29 -0400 (Thu, 10 Jul 2008) | 2
- lines Update svn:ignore ........
-
-2008-07-09 19:41 +0000 [r129438] Mark Michelson <mmichelson@digium.com>
-
- * main/rtp.c, /: Merged revisions 129437 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r129437 | mmichelson | 2008-07-09 14:40:30 -0500 (Wed, 09 Jul
- 2008) | 21 lines Merged revisions 129436 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r129436 | mmichelson | 2008-07-09 14:32:20 -0500 (Wed, 09 Jul
- 2008) | 13 lines Fix a problem where inbound rfc2833 audio would
- be sent to the core instead of being P2P bridged. When the core
- regenerated the rfc2833 packet for the outbound leg, the SSRC
- would be different than the RTP audio on the call leg causing
- DTMF detection issues on the far end. (closes issue #12955)
- Reported by: tonyredstone Patches: dynamic_rtp.patch uploaded by
- tsearle (license 373) Tested by: tonyredstone ........
- ................
-
-2008-07-09 16:01 +0000 [r129400] Matthew Fredrickson <creslin@digium.com>
-
- * main/pbx.c, /: Merged revisions 129399 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r129399 |
- mattf | 2008-07-09 10:57:06 -0500 (Wed, 09 Jul 2008) | 1 line Add
- Proceeding() application (#13025) ........
-
-2008-07-09 13:46 +0000 [r129345] Sean Bright <sean.bright@gmail.com>
-
- * main/editline/makelist (removed), main/editline/makelist.in
- (added), /, main/editline/configure, main/editline/Makefile.in,
- main/editline/configure.in: Merged revisions 129344 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r129344 | seanbright | 2008-07-09 09:44:43 -0400
- (Wed, 09 Jul 2008) | 12 lines Merged revisions 129343 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r129343 | seanbright | 2008-07-09 09:41:21 -0400 (Wed, 09 Jul
- 2008) | 4 lines Look for the system installed awk instead of
- assuming it's at /usr/bin/awk. Pointed out by jmls via
- #asterisk-dev. ........ ................
-
-2008-07-08 22:56 +0000 [r129160-129271] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 129270 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r129270 | mmichelson | 2008-07-08 17:56:12 -0500 (Tue, 08 Jul
- 2008) | 3 lines Fix compilation error when IMAP storage is
- enabled ........
-
-2008-07-08 21:04 +0000 [r129157] Brett Bryant <bbryant@digium.com>
-
- * main/dns.c, main/srv.c, /: Merged revisions 129156 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r129156 | bbryant | 2008-07-08 16:00:01 -0500 (Tue, 08 Jul 2008)
- | 6 lines Fix a bug in SRV lookups where dnsmgr would discard
- everything but the first SRV result from DNS before processing
- weights and priorities and dns_parse_answer wouldn't report that
- there were no records in DNS unless a failure occured. Also fixed
- a bug where dnsmgr_refresh would report that a entry was being
- changed when ast_gethostbyname had failed. ........
-
-2008-07-08 20:31 +0000 [r129049-129153] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_dial.c, /, channels/chan_sip.c,
- include/asterisk/causes.h: Merged revisions 129152 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r129152 | tilghman | 2008-07-08 15:30:29 -0500
- (Tue, 08 Jul 2008) | 16 lines Merged revisions 129149 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r129149 | tilghman | 2008-07-08 15:27:47 -0500 (Tue, 08 Jul 2008)
- | 8 lines Cause SIP to return a 480 instead of a 404 when a sip
- peer exists, but is not registered. (closes issue #12885)
- Reported by: ibc Patches: 20080701__bug12885__2.diff.txt uploaded
- by Corydon76 (license 14) Tested by: ibc ........
- ................
-
- * /, channels/chan_iax2.c: Merged revisions 129048 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r129048 | tilghman | 2008-07-08 11:49:01 -0500
- (Tue, 08 Jul 2008) | 15 lines Merged revisions 129047 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r129047 | tilghman | 2008-07-08 11:45:23 -0500 (Tue, 08 Jul 2008)
- | 7 lines Timestamp decoding for video mini-frames is bogus,
- because the timestamp only includes 15 bits, unlike voice frames,
- which contain a 16-bit timestamp. (closes issue #13013) Reported
- by: jpgrayson Patches: chan_iax2_unwrap_ts.patch uploaded by
- jpgrayson (license 492) ........ ................
-
-2008-07-08 16:41 +0000 [r129041-129046] Brett Bryant <bbryant@digium.com>
-
- * main/rtp.c, main/channel.c, channels/chan_dahdi.c,
- main/manager.c, formats/format_pcm.c, main/logger.c,
- main/callerid.c, apps/app_parkandannounce.c, apps/app_adsiprog.c,
- main/pbx.c, main/frame.c, /, channels/chan_sip.c,
- apps/app_meetme.c, channels/h323/ast_h323.cxx, res/res_limit.c,
- main/acl.c, channels/iax2-provision.c, pbx/dundi-parser.c,
- channels/chan_iax2.c: Merged revisions 129045 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r129045 |
- bbryant | 2008-07-08 11:40:28 -0500 (Tue, 08 Jul 2008) | 7 lines
- Janitor project to convert sizeof to ARRAY_LEN macro. (closes
- issue #13002) Reported by: caio1982 Patches:
- janitor_arraylen5.diff uploaded by caio1982 (license 22) ........
-
- * /, channels/chan_sip.c: Merged revisions 127621 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r127621 |
- bbryant | 2008-07-02 17:16:29 -0500 (Wed, 02 Jul 2008) | 1 line
- Update transport= in sip so that the option is not broken from a
- recent commit. ........
-
- * /, channels/chan_sip.c: Merged revisions 127434 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r127434 |
- bbryant | 2008-07-02 12:27:36 -0500 (Wed, 02 Jul 2008) | 1 line
- Fix to sip_parse_host so that it passes the correct information
- to sip_registry. ........
-
-2008-07-08 14:18 +0000 [r129007] Russell Bryant <russell@digium.com>
-
- * /, apps/app_fax.c: Merged revisions 129006 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r129006 |
- russell | 2008-07-08 09:17:37 -0500 (Tue, 08 Jul 2008) | 9 lines
- Update app_fax for better compatibility with spandsp 0.0.5. Add a
- call to t38_terminal_release, and make sure that the phase E
- handler gets called with proper status. (closes issue #13020)
- Reported by: dimas Patches: v1-appfax.patch uploaded by dimas
- (license 88) ........
-
-2008-07-08 10:06 +0000 [r128913-128952] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 128951 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r128951 | oej | 2008-07-08 12:02:12 +0200 (Tis, 08 Jul 2008) | 19
- lines Merged revisions 128950 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r128950 | oej | 2008-07-08 11:52:21 +0200 (Tis, 08 Jul 2008) | 11
- lines Don't hangup the call if we can't resolve the Contact if
- there's a proxy route set for the call. ---- This comment was
- added a while ago and today it hit me badly. /* OEJ: Possible
- issue that may need a check: If we have a proxy route between us
- and the device, should we care about resolving the contact or
- should we just send it? */ ........ ................
-
- * /, channels/chan_sip.c: Merged revisions 128927 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r128927 | oej | 2008-07-08 11:26:37 +0200 (Tis, 08 Jul 2008) | 15
- lines Merged revisions 128912 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r128912 | oej | 2008-07-08 11:06:08 +0200 (Tis, 08 Jul 2008) | 7
- lines Fix issues where repeated messages where ignored, but
- retransmitted reliably instead of unreliably. Reported by: johan
- Patches: 12746.txt uploaded by oej (license 306) Tested by: johan
- (issue #12746) ........ ................
-
-2008-07-08 00:03 +0000 [r128855-128858] Tilghman Lesher <tlesher@digium.com>
-
- * /: Merged revisions 128857 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r128857 | tilghman | 2008-07-07 19:02:11 -0500 (Mon, 07 Jul 2008)
- | 15 lines Merged revisions 128856 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r128856 | tilghman | 2008-07-07 19:01:30 -0500 (Mon, 07 Jul 2008)
- | 7 lines Check for non-NULL before stripping characters. (closes
- issue #12954) Reported by: bfsworks Patches:
- 20080701__bug12954.diff.txt uploaded by Corydon76 (license 14)
- Tested by: deti ........ ................
-
- * apps/app_voicemail.c, /: Merged revisions 128830 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r128830 | tilghman | 2008-07-07 18:25:39 -0500
- (Mon, 07 Jul 2008) | 10 lines Merged revisions 128812 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r128812 | tilghman | 2008-07-07 18:21:52 -0500 (Mon, 07 Jul 2008)
- | 2 lines Stop using deprecated method, as requested by Kevin.
- ........ ................
-
-2008-07-07 22:44 +0000 [r128797] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 128796 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r128796 | russell | 2008-07-07 17:42:30 -0500
- (Mon, 07 Jul 2008) | 16 lines Merged revisions 128795 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r128795 | russell | 2008-07-07 17:41:48 -0500 (Mon, 07 Jul 2008)
- | 8 lines Fix handling of when a pvt disappears. Properly return
- the pvt locked and don't hold the pvt lock while destroying the
- ast_channel. (closes issue #13014) Reported by: jpgrayson
- Patches: chan_iax2_ast_iax2_new2.patch uploaded by jpgrayson
- (license 492) ........ ................
-
-2008-07-07 20:51 +0000 [r128739] Sean Bright <sean.bright@gmail.com>
-
- * /, channels/chan_iax2.c: Merged revisions 128738 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r128738 | seanbright | 2008-07-07 16:50:29 -0400
- (Mon, 07 Jul 2008) | 17 lines Merged revisions 128737 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r128737 | seanbright | 2008-07-07 16:47:56 -0400 (Mon, 07 Jul
- 2008) | 9 lines Remove spurious trailing whitespace from log
- messages and fix a spelling error in a log message. (closes issue
- #13017) Reported by: jpgrayson Patches:
- chan_iax2_space_after_newline.patch uploaded by jpgrayson
- (license 492) chan_iax2_spelling.patch uploaded by jpgrayson
- (license 492) ........ ................
-
-2008-07-07 20:31 +0000 [r128601-128735] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 128733 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r128733 | mmichelson | 2008-07-07 15:30:46 -0500 (Mon, 07 Jul
- 2008) | 3 lines Crap ........
-
- * apps/app_voicemail.c, /: Merged revisions 128731 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r128731 | mmichelson | 2008-07-07 15:28:33 -0500 (Mon, 07 Jul
- 2008) | 7 lines If imapfolder=foo were set in voicemail.conf,
- then when calling VoiceMailMain, app_voicemail would attempt to
- play a file called vm-foo instead of playing vm-INBOX to play the
- "new" sound file. This commit fixes that issue. This may fix one
- of the problems reported in issue #12987 ........
-
- * /, channels/chan_iax2.c: Merged revisions 128640 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r128640 | mmichelson | 2008-07-07 12:09:11 -0500
- (Mon, 07 Jul 2008) | 18 lines Merged revisions 128639 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r128639 | mmichelson | 2008-07-07 12:02:28 -0500 (Mon, 07 Jul
- 2008) | 10 lines By using the iaxdynamicthreadcount to identify a
- thread, it was possible for thread identifiers to be duplicated.
- By using a globally-unique monotonically- increasing integer,
- this is now avoided. (closes issue #13009) Reported by: jpgrayson
- Patches: chan_iax2_dyn_threadnum.patch uploaded by jpgrayson
- (license 492) ........ ................
-
- * configs/extensions.conf.sample, /, doc/tex/extensions.tex: Merged
- revisions 128599 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r128599 |
- mmichelson | 2008-07-07 09:35:27 -0500 (Mon, 07 Jul 2008) | 6
- lines Update a few instances of "extensions reload" to "dialplan
- reload" in the documentation. Patch provided by caio1982 (license
- 22) ........
-
-2008-07-06 20:22 +0000 [r128288-128543] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
- 128524 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r128524 |
- oej | 2008-07-06 22:11:37 +0200 (Sön, 06 Jul 2008) | 5 lines -
- Fixing issues with "sip show settings" - Adding IP address for
- TCP and/or TLS too if auto-domain is enabled and binding to a
- different IP address - Fixing documentation in sip.conf.sample
- ........
-
- * /, channels/chan_sip.c: Merged revisions 128491 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r128491 |
- oej | 2008-07-06 21:14:06 +0200 (Sön, 06 Jul 2008) | 3 lines -
- Remove unused variable "expiry" - Set global_outboundproxy.force
- at reload. ........
-
- * doc/realtimetext.txt (added), /: The following patch with
- references to t140red removed, since it only exists in trunk.
- Merged revisions 128417 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r128417 |
- oej | 2008-07-06 12:13:45 +0200 (Sön, 06 Jul 2008) | 3 lines
- Adding documentation on the T.140 support in Asterisk. This is a
- function that we're the reference implementation on now. :-)
- ........
-
- * /: Merged revisions 128343 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r128343 |
- oej | 2008-07-06 10:10:27 +0200 (Sön, 06 Jul 2008) | 2 lines
- Removing the CLI dumpdb command (see asterisk-dev discussion and
- decision) ........
-
- * /, channels/chan_sip.c: Merged revisions 128290 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r128290 |
- oej | 2008-07-05 23:55:57 +0200 (Lör, 05 Jul 2008) | 5 lines
- Adding doxygen comments to missing parts, moving some #define
- ...trying to get my head around the thoughts behind the TCP/TLS
- stuff and figure out what needs to be done to make it useful...
- ........
-
- * /, channels/chan_sip.c: Merged revisions 128287 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r128287 |
- oej | 2008-07-05 23:37:57 +0200 (Lör, 05 Jul 2008) | 3 lines
- Adding TCP and TLS to "sip show settings". TLS needs to have one
- configuration per configured domain at some point. ........
-
- * /: Blocking changes in trunk.
-
-2008-07-05 21:02 +0000 [r128238-128243] Olle Johansson <oej@edvina.net>
-
- * /: Keep the "sip-user" structure in 1.6.0, while testing new
- funky stuff in trunk.
-
- * /: Blocking the AGi changes from 1.6.0. Let's test them for a
- while in trunk before a release.
-
- * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
- 128237 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r128237 |
- oej | 2008-07-05 22:39:54 +0200 (Lör, 05 Jul 2008) | 2 lines
- Make TCP disabled by default (it's considered experimental)
- ........
-
- * /, configs/sip.conf.sample: Merged revisions 128236 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r128236 | oej | 2008-07-05 22:37:53 +0200 (Lör, 05 Jul 2008) | 2
- lines Reformatting the config sample ........
-
-2008-07-05 15:19 +0000 [r128161] Tilghman Lesher <tlesher@digium.com>
-
- * contrib/scripts/asterisk.ldap-schema,
- contrib/scripts/asterisk.ldif, /: Merged revisions 128160 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r128160 | tilghman | 2008-07-05 10:17:51 -0500 (Sat, 05
- Jul 2008) | 7 lines LDAP schema updates (closes issue #12860)
- Reported by: flyn Patches: asterisk.ldif uploaded by suretec
- (license 70) asterisk.schema uploaded by suretec (license 70)
- ........
-
-2008-07-05 03:40 +0000 [r128124-128127] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 128125 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r128125 | mattf | 2008-07-04 22:39:07 -0500 (Fri, 04 Jul 2008) |
- 1 line It would help if we actually parsed the ss7_explicitacm
- option in the config file... ........
-
- * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged
- revisions 128122 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r128122 |
- mattf | 2008-07-04 22:26:42 -0500 (Fri, 04 Jul 2008) | 1 line Add
- option to wait to be able to explicitly send ACM via the
- Proceeding() application in the dialplan. Also minor
- documentation update explaining how to setup multiple signalling
- links within a linkset ........
-
-2008-07-04 16:12 +0000 [r128028-128031] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c, /, include/asterisk/pbx.h, pbx/pbx_config.c: Merged
- revisions 128027 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r128027 | tilghman | 2008-07-04 11:06:34 -0500 (Fri, 04 Jul 2008)
- | 16 lines Merged revisions 127973 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r127973 | tilghman | 2008-07-03 22:30:30 -0500 (Thu, 03 Jul 2008)
- | 8 lines Fix the 'dialplan remove extension' logic, so that it
- a) works with cidmatch, and b) completes contexts correctly when
- the extension is ambiguous. (closes issue #12980) Reported by:
- licedey Patches: 20080703__bug12980.diff.txt uploaded by
- Corydon76 (license 14) Tested by: Corydon76 ........
- ................
-
-2008-07-03 22:23 +0000 [r127905] Kevin P. Fleming <kpfleming@digium.com>
-
- * Makefile, /, apps/Makefile, main/editline/np/vis.c: Merged
- revisions 127903 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r127903 | kpfleming | 2008-07-03 17:23:04 -0500 (Thu, 03 Jul
- 2008) | 20 lines Merged revisions 127892,127895 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r127892 | kpfleming | 2008-07-03 17:18:38 -0500 (Thu, 03 Jul
- 2008) | 6 lines a couple of small Solaris-related fixes (closes
- issue #11885) Reported by: snuffy, asgaroth ........ r127895 |
- kpfleming | 2008-07-03 17:20:16 -0500 (Thu, 03 Jul 2008) | 3
- lines remove this, it has been moved to the main Makefile
- ........ ................
-
-2008-07-03 19:12 +0000 [r127830] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, main/channel.c, channels/chan_dahdi.c, main/pbx.c, /,
- channels/chan_sip.c, main/features.c, include/asterisk/cdr.h:
- Merged revisions 127793 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r127793 | murf | 2008-07-03 11:16:44 -0600 (Thu, 03 Jul 2008) |
- 38 lines Merged revisions 127663 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r127663 | murf | 2008-07-02 18:16:25 -0600 (Wed, 02 Jul 2008) |
- 30 lines The CDRfix4/5/6 omnibus cdr fixes. (closes issue #10927)
- Reported by: murf Tested by: murf, deeperror (closes issue
- #12907) Reported by: falves11 Tested by: murf, falves11 (closes
- issue #11849) Reported by: greyvoip As to 11849, I think these
- changes fix the core problems brought up in that bug, but perhaps
- not the more global problems created by the limitations of CDR's
- themselves not being oriented around transfers. Reopen if necc,
- but bug reports are not the best medium for enhancement
- discussions. We need to start a second-generation CDR
- standardization effort to cover transfers. (closes issue #11093)
- Reported by: rossbeer Tested by: greyvoip, murf ........
- ................
-
-2008-07-03 16:50 +0000 [r127790-127792] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 127791 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r127791 |
- oej | 2008-07-03 18:48:23 +0200 (Tor, 03 Jul 2008) | 5 lines Make
- sure we stop session timers as soon as we start hanging up an
- active call. May fix issue 12919. ........
-
- * /, channels/chan_sip.c: Merged revisions 127779 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r127779 |
- oej | 2008-07-03 18:25:59 +0200 (Tor, 03 Jul 2008) | 4 lines
- Revert some logic for session timers. We do send in-dialog
- requests that should not have session-timer require headers, like
- MESSAGE and REFER. So in the future, only add them on requests
- and responses that are related to INVITEs and re-INVITEs.
- ........
-
-2008-07-03 16:24 +0000 [r127778] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
- acinclude.m4: Merged revisions 127767 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r127767 |
- kpfleming | 2008-07-03 11:22:02 -0500 (Thu, 03 Jul 2008) | 2
- lines some minor fixes found while working on issue #12911 (and
- block the rev from 1.4 since the equivalent is already here)
- ........
-
-2008-07-02 21:10 +0000 [r127567] Mark Michelson <mmichelson@digium.com>
-
- * /, doc/janitor-projects.txt: Merged revisions 127566 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r127566 | mmichelson | 2008-07-02 16:09:18 -0500 (Wed, 02 Jul
- 2008) | 4 lines Add a janitor project to use ARRAY_LEN instead of
- in-line sizeof() and division. ........
-
-2008-07-02 20:49 +0000 [r127559-127563] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 127562 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r127562 | mmichelson | 2008-07-02 15:49:08 -0500
- (Wed, 02 Jul 2008) | 11 lines Merged revisions 127560 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r127560 | mmichelson | 2008-07-02 15:47:38 -0500 (Wed, 02 Jul
- 2008) | 3 lines Fix thread-safety of some of the
- pbx_builtin_getvar_helper calls ........ ................
-
-2008-07-02 19:48 +0000 [r127467-127503] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/acl.c: Merged revisions 127466 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r127466 |
- tilghman | 2008-07-02 13:31:11 -0500 (Wed, 02 Jul 2008) | 6 lines
- Solaris fix (closes issue #12949) Reported by: snuffy Patches:
- bug_12949.diff uploaded by snuffy (license 35) ........
-
-2008-07-02 14:30 +0000 [r127396-127399] Sean Bright <sean.bright@gmail.com>
-
- * cdr/cdr_tds.c, /: Merged revisions 127398 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r127398 |
- seanbright | 2008-07-02 10:30:09 -0400 (Wed, 02 Jul 2008) | 1
- line Fix a bug I noticed while doing the previous merge ........
-
- * cdr/cdr_tds.c, /, doc/tex/freetds.tex, configure,
- include/asterisk/autoconfig.h.in, configure.ac, UPGRADE.txt:
- Merged revisions 126226,126513 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r126226 |
- seanbright | 2008-06-28 17:28:16 -0400 (Sat, 28 Jun 2008) | 8
- lines Merge in changes from my cdr-tds-conversion branch. This
- changes the internal implementation from using the volatile
- libtds, to using the db-lib front end. The unintended side effect
- of this is that we support (at least) versions 0.62 through 0.82
- of the FreeTDS distribution without any #ifdef ugliness. (closes
- issue #12844) Reported by: jcollie ........ r126513 | seanbright
- | 2008-06-30 07:57:42 -0400 (Mon, 30 Jun 2008) | 4 lines Cast a
- few more strings to char *, so that we can compile cleanly
- against FreeTDS 0.60. Update the docs to reflect that we can now
- compile and run against all modern releases of FreeTDS (0.60
- through 0.82) ........
-
- * /: Unblock some revisions so I can merge the cdr_tds changes from
- trunk
-
-2008-07-02 12:09 +0000 [r127364] Russell Bryant <russell@digium.com>
-
- * doc/CODING-GUIDELINES, /: Merged revisions 127363 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r127363 | russell | 2008-07-02 07:08:33 -0500 (Wed, 02 Jul 2008)
- | 13 lines Add a locking section to the coding guidelines
- document. This section covers some locking fundamentals, as well
- as some information on locking as it is used in Asterisk. It
- describes some of the ways that are used and could be used to
- achieve deadlock avoidance. It also demonstrates the unfortunate
- conclusion that with the use of recursive locks, none of the
- constructs in use today are failsafe from deadlocks. Finally, it
- makes some recommendations for new code being written. As proper
- locking strategies is a complex subject, this section still has
- room for expansion and improvement. This is a result of
- collaboration between Luigi Rizzo and myself on the asterisk-dev
- mailing list. ........
-
-2008-07-02 02:49 +0000 [r127298] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 127297 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r127297 |
- tilghman | 2008-07-01 21:48:43 -0500 (Tue, 01 Jul 2008) | 12
- lines Change the global timer B to be dependent on the value of
- the T1 timer, as recommended in RFC 3261, instead of being
- hardcoded to 32 seconds. This is important for LANs, as it allows
- autocongestion to occur much more quickly, if desired by the
- local PBX administrator. It also corrects a bug: if the T1 timer
- was increased beyond 500ms, then timer B would have been set at a
- much lower value than recommended. (closes issue #12544) Reported
- by: kactus Patches: 20080616__bug12544.diff.txt uploaded by
- Corydon76 (license 14) Tested by: kactus ........
-
-2008-07-01 23:39 +0000 [r127246] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 127245 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r127245 | mmichelson | 2008-07-01 18:38:12 -0500
- (Tue, 01 Jul 2008) | 13 lines Merged revisions 127244 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r127244 | mmichelson | 2008-07-01 18:36:40 -0500 (Tue, 01 Jul
- 2008) | 5 lines Add error message to failed open(2) calls inside
- the copy() function of app_voicemail. This idea came as part of
- my work in helping to resolve issue #12764. ........
- ................
-
-2008-07-01 21:19 +0000 [r127163] Brett Bryant <bbryant@digium.com>
-
- * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
- 127154 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r127154 |
- bbryant | 2008-07-01 16:03:52 -0500 (Tue, 01 Jul 2008) | 2 lines
- Add a configuration option so the global outboundproxy can use
- tcptls without it being defined by each sip user. ........
-
-2008-07-01 21:16 +0000 [r127156-127158] Mark Michelson <mmichelson@digium.com>
-
- * main/channel.c, /: Merged revisions 127157 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r127157 |
- mmichelson | 2008-07-01 16:16:00 -0500 (Tue, 01 Jul 2008) | 8
- lines Place the delay in __ast_answer prior to the
- channel-specific answer callback. This change differs from commit
- 127113 in that now the channel is not set to AST_STATE_UP until
- after the answer callback. (closes issue #12924) Reported by:
- snyfer ........
-
- * main/channel.c, /: Merging Revision 127113 from trunk
-
-2008-07-01 20:52 +0000 [r127153] Jason Parker <jparker@digium.com>
-
- * Makefile, /: Merged revisions 127152 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r127152 |
- qwell | 2008-07-01 15:51:43 -0500 (Tue, 01 Jul 2008) | 7 lines
- Fix a typo that caused this asterisk.conf to not get correctly
- generated. (closes issue #12966) Reported by: ibc Patches:
- 12966.patch uploaded by bkruse (license 132) ........
-
-2008-07-01 20:29 +0000 [r127085-127149] Tilghman Lesher <tlesher@digium.com>
-
- * build_tools/cflags.xml, /, channels/chan_iax2.c: Merged revisions
- 127143 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r127143 | tilghman | 2008-07-01 15:28:54 -0500 (Tue, 01 Jul 2008)
- | 10 lines Merged revisions 127133 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r127133 | tilghman | 2008-07-01 15:25:37 -0500 (Tue, 01 Jul 2008)
- | 2 lines Disable the old, slow search for matching callno in
- chan_iax2 (but allow it to be reenabled for debugging) ........
- ................
-
- * /, channels/chan_iax2.c: Merged revisions 127074 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r127074 | tilghman | 2008-07-01 14:20:25 -0500
- (Tue, 01 Jul 2008) | 16 lines Merged revisions 127068 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r127068 | tilghman | 2008-07-01 13:52:53 -0500 (Tue, 01 Jul 2008)
- | 8 lines Change around how we schedule pings and lagrqs, and fix
- a reason why the jobs were not getting properly cancelled.
- (closes issue #12903) Reported by: stevedavies Patches:
- 20080620__bug12903__2.diff.txt uploaded by Corydon76 (license 14)
- Tested by: stevedavies ........ ................
-
-2008-07-01 16:53 +0000 [r127001] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 127000 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r127000 | tilghman | 2008-07-01 11:52:29 -0500
- (Tue, 01 Jul 2008) | 10 lines Merged revisions 126999 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r126999 | tilghman | 2008-07-01 11:50:46 -0500 (Tue, 01 Jul 2008)
- | 2 lines Suppress annoying warning by finding the remaining
- cases where the callno is not in the hash. ........
- ................
-
-2008-07-01 15:05 +0000 [r126756-126904] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 126903 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r126903 | oej | 2008-07-01 17:03:59 +0200 (Tis, 01 Jul 2008) | 15
- lines Merged revisions 126902 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r126902 | oej | 2008-07-01 16:59:31 +0200 (Tis, 01 Jul 2008) | 7
- lines Use domain part of SIP uri in register= configuration as
- fromdomain. Reported by: one47 Patches: sip-reg-fromdom2.dpatch
- uploaded by one47 (license 23) (closes issue #12474) ........
- ................
-
- * /, channels/chan_sip.c: Merged revisions 126900 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r126900 | oej | 2008-07-01 16:32:15 +0200 (Tis, 01 Jul 2008) | 16
- lines Merged revisions 126899 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r126899 | oej | 2008-07-01 16:27:33 +0200 (Tis, 01 Jul 2008) | 8
- lines Handle escaped URI's in call pickups. Patch by oej and
- IgorG. Reported by: IgorG Patches: bug12299-11062-v2.patch
- uploaded by IgorG (license 20) Tested by: IgorG, oej (closes
- issue #12299) ........ ................
-
- * /, configs/sip.conf.sample: Merged revisions 126845 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r126845 | oej | 2008-07-01 14:54:57 +0200 (Tis,
- 01 Jul 2008) | 14 lines Merged revisions 126844 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r126844 | oej | 2008-07-01 14:53:01 +0200 (Tis, 01 Jul 2008) | 5
- lines Clear up documentation on "domain=" setting in sip.conf
- Reported by: davidw (closes issue #12413) ........
- ................
-
- * /, channels/chan_sip.c: Merged revisions 126790 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r126790 | oej | 2008-07-01 13:58:17 +0200 (Tis, 01 Jul 2008) | 14
- lines Merged revisions 126789 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r126789 | oej | 2008-07-01 13:51:38 +0200 (Tis, 01 Jul 2008) | 6
- lines Report 200 OK to all in-dialog OPTIONs requests (to confirm
- that the dialog exist). Don't bother checking the request URI.
- (closes issue #11264) Reported by: ibc ........ ................
-
- * /, channels/chan_sip.c: Merged revisions 126755 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r126755 | oej | 2008-07-01 11:51:22 +0200 (Tis, 01 Jul 2008) | 15
- lines Merged revisions 126735 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r126735 | oej | 2008-07-01 09:49:15 +0200 (Tis, 01 Jul 2008) | 7
- lines Fix bad XML for hold notification. Reported by: gowen72
- Patches: hold.patch uploaded by gowen72 (license 432) (closes
- issue #12942) ........ ................
-
-2008-06-30 22:34 +0000 [r126676] Jeff Peeler <jpeeler@digium.com>
-
- * configs/zapata.conf.sample (removed),
- configs/chan_dahdi.conf.sample (added), /: Merged revisions
- 126675 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r126675 |
- jpeeler | 2008-06-30 17:34:08 -0500 (Mon, 30 Jun 2008) | 1 line
- rename zapata.conf.sample to chan_dahdi.conf.sample ........
-
-2008-06-30 20:32 +0000 [r126638] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 126637 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r126637 | mattf | 2008-06-30 15:25:46 -0500 (Mon, 30 Jun 2008) |
- 1 line Add support to see MTP2 down events when the link layer
- drops in SS7 ........
-
-2008-06-30 16:09 +0000 [r126575] Russell Bryant <russell@digium.com>
-
- * /, include/asterisk/lock.h: Merged revisions 126574 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r126574 | russell | 2008-06-30 11:07:25 -0500
- (Mon, 30 Jun 2008) | 18 lines Merged revisions 126573 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r126573 | russell | 2008-06-30 11:05:08 -0500 (Mon, 30 Jun 2008)
- | 10 lines Fix a typo in the non-DEBUG_THREADS version of the
- recently added DEADLOCK_AVOIDANCE() macro. This caused the lock
- to not actually be released, and as a result, not avoid deadlocks
- at all. This resolves the issues reported in the last while about
- Asterisk locking up all over the place (and most commonly, in
- chan_iax2). (closes issue #12927) (closes issue #12940) (closes
- issue #12925) (potentially closes others ...) ........
- ................
-
-2008-06-30 13:07 +0000 [r126518] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 126517 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r126517 | oej | 2008-06-30 15:03:53 +0200 (MÃ¥n, 30 Jun 2008) |
- 20 lines The following patch with some changes for trunk...
- Merged revisions 126516 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r126516 | oej | 2008-06-30 14:50:55 +0200 (MÃ¥n, 30 Jun 2008) |
- 10 lines Send all responses to an INVITE reliably, so that we
- retransmit if we don't get an ACK and also fail if we don't get
- the very same precious ACK. Based on patch by tsearle, with my
- own additions. (closes issue #12951) Reported by: tsearle
- Patches: busy_retransmit.patch uploaded by tsearle (license 373)
- ........ ................
-
-2008-06-29 17:02 +0000 [r126362-126364] Kevin P. Fleming <kpfleming@digium.com>
-
- * apps/app_zapbarge.c (removed): finish converting this module
-
- * pbx/pbx_gtkconsole.c, /, configure, configure.ac, pbx/pbx_lua.c,
- pbx/Makefile: Merged revisions 126356 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r126356 |
- kpfleming | 2008-06-29 09:19:29 -0700 (Sun, 29 Jun 2008) | 9
- lines various minor fixes created while i worked on getting
- *every* Asterisk module to build on laptop in dev mode: remove
- weird pre-setting of LUA paths; they are not necessary; also use
- the proper path for including the files in pbx_lua.c make the
- compiler shut up about some ignored function results in
- pbx_gtkconsole; this module is badly coded anyway ........
-
- * apps/app_dahdibarge.c (added): don't know how this got missed in
- the DAHDI conversion of this branch
-
-2008-06-29 13:20 +0000 [r126227-126322] Sean Bright <sean.bright@gmail.com>
-
- * /, cdr/cdr_pgsql.c: Merged revisions 126274 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r126274 |
- seanbright | 2008-06-29 08:06:46 -0400 (Sun, 29 Jun 2008) | 6
- lines Quote column names when inserting CDRs into postgres to
- avoid conflicts with reserved words. (closes issue #12947)
- Reported by: panolex ........
-
-2008-06-28 15:58 +0000 [r126155-126188] Kevin P. Fleming <kpfleming@digium.com>
-
- * Makefile, /: update this branch to use the trunk goodness version
- of menuselect
-
-2008-06-27 22:43 +0000 [r126058-126112] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 126057 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r126057 | tilghman | 2008-06-27 17:10:34 -0500 (Fri, 27 Jun 2008)
- | 12 lines Merged revisions 126056 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r126056 | tilghman | 2008-06-27 17:01:09 -0500 (Fri, 27 Jun 2008)
- | 4 lines When we get a 408 Timeout, don't stop trying to
- re-register. (closes issue #12863) Reported by: ricvil ........
- ................
-
-2008-06-27 21:00 +0000 [r126023] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Port revisions 124661 and 123650 from trunk to
- 1.6.0 Thanks to Atis Lezdins for pointing this out on the
- asterisk-dev mailing list
-
-2008-06-27 19:20 +0000 [r125994] Russell Bryant <russell@digium.com>
-
- * /, doc/siptls.txt: Merged revisions 125988 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r125988 |
- russell | 2008-06-27 14:19:08 -0500 (Fri, 27 Jun 2008) | 3 lines
- Fix a typo. Someone on IRC copied this literally and then
- wondered why it wasn't working. :) ........
-
-2008-06-27 19:06 +0000 [r125981-125985] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 125984 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r125984 | mattf | 2008-06-27 14:05:40 -0500 (Fri, 27 Jun 2008) |
- 1 line Revert this part of the fix. We'll fix it in libss7
- ........
-
- * channels/chan_dahdi.c, /: Merged revisions 125982 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r125982 | mattf | 2008-06-27 14:00:44 -0500 (Fri, 27 Jun 2008) |
- 1 line Obviously somebody didn't compile with libss7 support when
- doing the DAHDI conversion. ........
-
- * channels/chan_dahdi.c, /: Merged revisions 125980 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r125980 | mattf | 2008-06-27 13:32:17 -0500 (Fri, 27 Jun 2008) |
- 1 line Add support for new commands to block/unblock all CICs on
- a linkset ........
-
-2008-06-27 17:36 +0000 [r125948] Brett Bryant <bbryant@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 125947 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r125947 |
- bbryant | 2008-06-27 12:35:41 -0500 (Fri, 27 Jun 2008) | 1 line
- Small error in the function that converts peer transports to a
- string. ........
-
-2008-06-27 16:29 +0000 [r125892] Brett Bryant <bbryant@digium.com>
-
- * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
- 125891 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r125891 |
- bbryant | 2008-06-27 11:28:06 -0500 (Fri, 27 Jun 2008) | 6 lines
- Change the way that the transport option works for sip users.
- transport will now take multiple arguments, the first one listed
- will be the one used for new dialogs, and the rest listed will be
- acceptable ways for that peer to contact us. This fixes a minor
- bug where, because SIP TCP/UDP run on the same port, could cause
- a TCP peer to be saved in the ast_db. There will also be warnings
- when a transport is changed for an unexpected reason. (issue
- #12799) ........
-
-2008-06-27 16:19 +0000 [r125859-125863] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 125855 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r125855 |
- mmichelson | 2008-06-27 11:16:13 -0500 (Fri, 27 Jun 2008) | 5
- lines Ensure the thread-safety of the monexec variable in
- app_queue. Thanks to Russell for pointing out the problem
- ........
-
-2008-06-27 16:01 +0000 [r125854] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_local.c, /: Merged revisions 125853 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r125853 | tilghman | 2008-06-27 11:00:05 -0500 (Fri, 27 Jun 2008)
- | 3 lines Revert half of the fix, as this part may have been
- unnecessary (related to issue #12914) Requested here:
- http://lists.digium.com/pipermail/asterisk-dev/2008-June/033658.html
- ........
-
-2008-06-27 14:57 +0000 [r125800-125852] Mark Michelson <mmichelson@digium.com>
-
- * main/asterisk.c, main/channel.c, channels/chan_iax2.c: Make sure
- to only include dahdi/user.h if we have installed DAHDI.
-
- * channels/chan_iax2.c: I accidentally committed a change to
- chan_iax2.c in addition to a change to app_queue.c. Reverting the
- change to chan_iax2.c, even though it may turn out that this
- change is necessary.
-
- * utils/Makefile, /: Merged revisions 125799 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r125799 |
- mmichelson | 2008-06-27 09:14:09 -0500 (Fri, 27 Jun 2008) | 3
- lines Remove an unneeded target from the Makefile ........
-
-2008-06-27 14:09 +0000 [r125742-125797] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/utils.c, include/asterisk/lock.h: Merged revisions 125794
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r125794 | tilghman | 2008-06-27 08:54:13 -0500
- (Fri, 27 Jun 2008) | 10 lines Merged revisions 125793 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r125793 | tilghman | 2008-06-27 08:45:03 -0500 (Fri, 27 Jun 2008)
- | 2 lines In this debugging function, copy to a buffer instead of
- using potentially unsafe pointers. ........ ................
-
- * channels/chan_local.c, /: Merged revisions 125741 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r125741 | tilghman | 2008-06-27 07:28:38 -0500
- (Fri, 27 Jun 2008) | 15 lines Merged revisions 125740 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r125740 | tilghman | 2008-06-27 07:19:39 -0500 (Fri, 27 Jun 2008)
- | 7 lines Add proper deadlock avoidance. (closes issue #12914)
- Reported by: ozan Patches: 20080625__bug12914.diff.txt uploaded
- by Corydon76 (license 14) Tested by: ozan ........
- ................
-
-2008-06-27 07:41 +0000 [r125704] Philippe Sultan <philippe.sultan@gmail.com>
-
- * /, include/asterisk/jabber.h, res/res_jabber.c: Merged revisions
- 125703 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r125703 |
- phsultan | 2008-06-27 09:28:17 +0200 (Fri, 27 Jun 2008) | 1 line
- Fix a compile time error that occurs if OpenSSL is not installed.
- Reported by Noel Morais on the users mailing list ........
-
-2008-06-27 01:09 +0000 [r125648-125684] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c, channels/chan_iax2.c: SENTINEL is not defined
- in 1.6.0
-
- * /, apps/app_queue.c: Merged revisions 125666 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r125666 |
- mmichelson | 2008-06-26 19:22:03 -0500 (Thu, 26 Jun 2008) | 3
- lines Make this compile with dev-mode on ........
-
- * /, apps/app_queue.c: Merged revisions 125649 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r125649 |
- mmichelson | 2008-06-26 19:15:54 -0500 (Thu, 26 Jun 2008) | 15
- lines The monitor-join option for queues was deprecated in favor
- of using MixMonitor to mix audio. However, it was pointed out to
- me that because of this, the command set for the MONITOR_EXEC
- variable is ignored as well. This means that people can't do
- their own custom mixing commands at the end of recordings in
- order to make, for instance, stereo recordings of calls. With
- this patch, app_queue will set the "joinfiles" variable for the
- channel's monitor if MONITOR_EXEC is not zero-length. This means
- that for normal audio mixing, MixMonitor is still the preferred
- choice, but we allow custom mixing to be done with the two
- Monitor streams if desired. (closes issue #12923) Reported by:
- snyfer ........
-
-2008-06-26 23:06 +0000 [r125592] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 125591 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r125591 |
- mmichelson | 2008-06-26 18:06:18 -0500 (Thu, 26 Jun 2008) | 3
- lines Fix a really stupid mistake ........
-
-2008-06-26 23:05 +0000 [r125590] Jason Parker <jparker@digium.com>
-
- * /, main/utils.c: Merged revisions 125589 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r125589 | qwell | 2008-06-26 18:04:18 -0500 (Thu, 26 Jun 2008) |
- 9 lines Merged revisions 125587 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r125587 | qwell | 2008-06-26 18:03:15 -0500 (Thu, 26 Jun 2008) |
- 1 line Make sure to unlock the lock_info lock (huh?). Possible
- deadlock? ........ ................
-
-2008-06-26 23:04 +0000 [r125588] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 125586 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r125586 | mmichelson | 2008-06-26 18:01:02 -0500 (Thu, 26 Jun
- 2008) | 19 lines Merged revisions 125585 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r125585 | mmichelson | 2008-06-26 17:52:39 -0500 (Thu, 26 Jun
- 2008) | 11 lines Add the interface of a queue member to the
- output of the "queue show" command so that it can easily be
- associated with a queue member's name. This helps so that the
- appropriate queue member can be removed or paused since the
- interface is required, not the member's name. (closes issue
- #12783) Reported by: davevg Patches: app_queue.diff uploaded by
- davevg (license 209) with small mod from me ........
- ................
-
-2008-06-26 22:50 +0000 [r125584] Tilghman Lesher <tlesher@digium.com>
-
- * /, contrib/scripts/astcli: Merged revisions 125583 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r125583 | tilghman | 2008-06-26 17:49:16 -0500 (Thu, 26 Jun 2008)
- | 2 lines Don't hang if the command is blank ........
-
-2008-06-26 22:06 +0000 [r125478-125532] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 125477 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r125477 | mmichelson | 2008-06-26 15:57:41 -0500 (Thu, 26 Jun
- 2008) | 19 lines Merged revisions 125476 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r125476 | mmichelson | 2008-06-26 15:56:01 -0500 (Thu, 26 Jun
- 2008) | 11 lines Prior to this patch, the "queue show" command
- used cached information for realtime queues instead of giving
- up-to-date info. Now realtime is queried for the latest and
- greatest in queue info. (closes issue #12858) Reported by: bcnit
- Patches: queue_show.patch uploaded by putnopvut (license 60)
- ........ ................
-
-2008-06-26 17:07 +0000 [r125388] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 125385 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r125385 | oej | 2008-06-26 18:54:22 +0200 (Tor, 26 Jun 2008) | 12
- lines Merged revisions 125384 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r125384 | oej | 2008-06-26 18:32:08 +0200 (Tor, 26 Jun 2008) | 3
- lines Add support for peer realm based auth (a few missing lines,
- the rest is well documented but never worked) ........
- ................
-
-2008-06-26 15:52 +0000 [r125280-125334] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 125333 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r125333 | kpfleming | 2008-06-26 10:50:07 -0500
- (Thu, 26 Jun 2008) | 13 lines Merged revisions 125327 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r125327 | kpfleming | 2008-06-26 10:30:33 -0500 (Thu, 26 Jun
- 2008) | 5 lines ensure that (whenever possible) if we generate a
- log message because an ioctl() call to DAHDI/Zaptel failed, that
- we include the reason it failed by including the stringified
- error number (issue AST-80) ........ ................
-
- * /, res/res_musiconhold.c: Merged revisions 125279 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r125279 | kpfleming | 2008-06-26 07:09:24 -0500 (Thu, 26 Jun
- 2008) | 2 lines fix compile failure found by buildbot (go,
- buildbot!) ........
-
-2008-06-26 11:08 +0000 [r125192-125278] Tilghman Lesher <tlesher@digium.com>
-
- * main/rtp.c, /: Merged revisions 125277 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r125277 | tilghman | 2008-06-26 06:02:11 -0500 (Thu, 26 Jun 2008)
- | 15 lines Merged revisions 125276 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r125276 | tilghman | 2008-06-26 06:01:21 -0500 (Thu, 26 Jun 2008)
- | 7 lines Check for rtcp structure before trying to delete
- schedule. (closes issue #12872) Reported by: destiny6628 Patches:
- 20080621__bug12872.diff.txt uploaded by Corydon76 (license 14)
- Tested by: destiny6628 ........ ................
-
- * configs/agents.conf.sample, /: Merged revisions 125223 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r125223 | tilghman | 2008-06-25 20:25:16 -0500
- (Wed, 25 Jun 2008) | 12 lines Merged revisions 125218 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r125218 | tilghman | 2008-06-25 20:24:26 -0500 (Wed, 25 Jun 2008)
- | 4 lines Document ackcall=always. (closes issue #12852) Reported
- by: davidw ........ ................
-
- * configs/http.conf.sample, /: Merged revisions 125191 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r125191 | tilghman | 2008-06-25 20:11:43 -0500 (Wed, 25 Jun 2008)
- | 6 lines Update sample configuration to match what are now the
- defaults for the prefix. (closes issue #12838, related to issue
- #12198) Reported by: pabelanger Patches: http.conf.diff2 uploaded
- by pabelanger (license 224) ........
-
-2008-06-25 23:20 +0000 [r125146] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/channel.c, channels/chan_dahdi.c, apps/app_flash.c,
- configure, codecs/codec_dahdi.c, apps/app_rpt.c, main/asterisk.c,
- /, apps/app_meetme.c, main/Makefile, apps/app_dahdiscan.c,
- apps/app_dahdiras.c, configure.ac, include/asterisk/dahdi.h
- (removed), res/res_musiconhold.c, channels/chan_iax2.c: Merged
- revisions 125138 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r125138 | kpfleming | 2008-06-25 18:05:28 -0500 (Wed, 25 Jun
- 2008) | 18 lines Merged revisions 125132 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r125132 | kpfleming | 2008-06-25 17:21:30 -0500 (Wed, 25 Jun
- 2008) | 10 lines allow tonezone to live in a different place than
- DAHDI/Zaptel, since dahdi-tools and dahdi-linux are now separate
- packages and can be installed in different places don't include
- tonezone.h in dahdi_compat.h, because only a couple of modules
- need it get app_rpt building again after the DAHDI changes
- (closes issue #12911) Reported by: tzafrir ........
- ................
-
-2008-06-25 01:13 +0000 [r124964-124967] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_dahdi.c, /, include/asterisk/lock.h: Merged
- revisions 124966 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r124966 | tilghman | 2008-06-24 20:08:37 -0500 (Tue, 24 Jun 2008)
- | 15 lines Merged revisions 124965 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r124965 | tilghman | 2008-06-24 19:46:24 -0500 (Tue, 24 Jun 2008)
- | 7 lines Pvt deadlock causes some channels to get stuck in
- Reserved status. (closes issue #12621) Reported by:
- fabianoheringer Patches: 20080612__bug12621.diff.txt uploaded by
- Corydon76 (license 14) Tested by: fabianoheringer ........
- ................
-
- * apps/app_voicemail.c, /: Merged revisions 124912 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r124912 | tilghman | 2008-06-24 16:18:52 -0500
- (Tue, 24 Jun 2008) | 16 lines Merged revisions 124910 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r124910 | tilghman | 2008-06-24 16:08:52 -0500 (Tue, 24 Jun 2008)
- | 8 lines Occasionally control characters find their way into
- CallerID. These need to be stripped prior to placing CallerID in
- the headers of an email. (closes issue #12759) Reported by: RobH
- Patches: 20080602__bug12759__2.diff.txt uploaded by Corydon76
- (license 14) Tested by: RobH ........ ................
-
-2008-06-24 17:52 +0000 [r124871-124873] Philippe Sultan <philippe.sultan@gmail.com>
-
- * /, res/res_jabber.c: Merged revisions 124872 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r124872 |
- phsultan | 2008-06-24 19:50:22 +0200 (Tue, 24 Jun 2008) | 6 lines
- Subscribe to buddy's presence only if we really need to. That is,
- if the corresponding roster item has a subscription value set to
- "none" or "from". Make the code more readable. ........
-
- * /, res/res_jabber.c: Merged revisions 124870 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r124870 |
- phsultan | 2008-06-24 19:28:39 +0200 (Tue, 24 Jun 2008) | 1 line
- Code simplification ........
-
-2008-06-23 15:44 +0000 [r124708] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * /: blocked revision 124707, taskprocessors are not in 1.6.0
-
-2008-06-22 03:18 +0000 [r124542] Steve Murphy <murf@digium.com>
-
- * apps/app_forkcdr.c, /: Merged revisions 124541 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r124541 | murf | 2008-06-21 20:58:06 -0600 (Sat, 21 Jun 2008) |
- 17 lines Merged revisions 124540 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r124540 | murf | 2008-06-21 20:54:52 -0600 (Sat, 21 Jun 2008) | 9
- lines (closes issue #12910) Reported by: chris-mac Sorry, my
- testing did not contain the simple case of forkCDR(v), I am much
- embarrassed to admit. If I had, I would have more solidly
- initialized the opts element for varset. ........
- ................
-
-2008-06-21 12:54 +0000 [r124397-124506] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_config_ldap.c: Merged revisions 124505 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r124505 | tilghman | 2008-06-21 07:53:48 -0500 (Sat, 21 Jun 2008)
- | 4 lines Reduce warning to debug, otherwise we flood the log
- when we (legitimately) can't find a record. (Closes issue #12908)
- ........
-
- * apps/app_rpt.c, /: Merged revisions 124451 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r124451 | tilghman | 2008-06-20 18:13:21 -0500 (Fri, 20 Jun 2008)
- | 14 lines Merged revisions 124450 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r124450 | tilghman | 2008-06-20 18:12:33 -0500 (Fri, 20 Jun 2008)
- | 6 lines usleep with a value over 1,000,000 is nonportable.
- Changing to use sleep() instead. (closes issue #12814) Reported
- by: pputman Patches: app_rtp_sleep.patch uploaded by pputman
- (license 81) ........ ................
-
- * /, main/app.c: Merged revisions 124396 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r124396 | tilghman | 2008-06-20 17:04:37 -0500 (Fri, 20 Jun 2008)
- | 11 lines Merged revisions 124395 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r124395 | tilghman | 2008-06-20 17:02:55 -0500 (Fri, 20 Jun 2008)
- | 3 lines If the last character in a string to be parsed is the
- delimiter, then we should count that final empty string as an
- additional argument. ........ ................
-
-2008-06-20 21:48 +0000 [r124394] Jeff Gehlbach <jeffg@opennms.org>
-
- * doc/asterisk-mib.txt, /, doc/digium-mib.txt: Merged revisions
- 124392-124393 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r124392 | jeffg | 2008-06-20 17:36:01 -0400 (Fri, 20 Jun 2008) |
- 9 lines Merged revisions 124372 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r124372 | jeffg | 2008-06-20 17:14:40 -0400 (Fri, 20 Jun 2008) |
- 1 line Fix issues in digium-mib.txt and asterisk-mib.txt to
- placate smilint - bug 12905 ........ ................ r124393 |
- jeffg | 2008-06-20 17:43:18 -0400 (Fri, 20 Jun 2008) | 12 lines
- (Missed committing . on previous commit.....) Merged revisions
- 124372 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r124372 | jeffg | 2008-06-20 17:14:40 -0400 (Fri, 20 Jun 2008) |
- 1 line Fix issues in digium-mib.txt and asterisk-mib.txt to
- placate smilint - bug 12905 ........ ................
- ................
-
-2008-06-20 20:18 +0000 [r124317] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_local.c, /: Merged revisions 124316 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r124316 | tilghman | 2008-06-20 15:17:04 -0500
- (Fri, 20 Jun 2008) | 16 lines Merged revisions 124315 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r124315 | tilghman | 2008-06-20 15:16:02 -0500 (Fri, 20 Jun 2008)
- | 8 lines When using a Local channel, started by a call file,
- with a destination of an AGI script, the AGI script does not
- always get notified of a hangup if the underlying channel hangs
- up early. (closes issue #11833) Reported by: IgorG Patches:
- local_hangup-v1.diff uploaded by IgorG (license 20) ........
- ................
-
-2008-06-20 16:31 +0000 [r124244-124279] Mark Michelson <mmichelson@digium.com>
-
- * main/ast_expr2.fl, include/asterisk/doxyref.h, /,
- main/ast_expr2f.c: Merged revisions 124278 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r124278 |
- mmichelson | 2008-06-20 11:30:18 -0500 (Fri, 20 Jun 2008) | 6
- lines Change references to doc/channelvariables.txt to
- doc/tex/channelvariables.tex. This issue came up on the
- asterisk-dev mailing list. ........
-
- * /, channels/chan_sip.c: Merged revisions 124243 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r124243 |
- mmichelson | 2008-06-20 10:20:11 -0500 (Fri, 20 Jun 2008) | 9
- lines Add a missing "ChannelType" header to one of the
- "PeerStatus" manager events in chan_sip (closes issue #12904)
- Reported by: eliel Patches: chan_sip.c.patch uploaded by eliel
- (license 64) ........
-
-2008-06-19 23:02 +0000 [r124184] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 124183 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r124183 | tilghman | 2008-06-19 17:59:41 -0500
- (Thu, 19 Jun 2008) | 15 lines Merged revisions 124182 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r124182 | tilghman | 2008-06-19 17:53:22 -0500 (Thu, 19 Jun 2008)
- | 7 lines It's possible for a hangup to be received, even just
- after the initial cid spill. (closes issue #12453) Reported by:
- Alex728 Patches: 20080604__bug12453.diff.txt uploaded by
- Corydon76 (license 14) ........ ................
-
-2008-06-19 20:32 +0000 [r124124] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 124121 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r124121 | mmichelson | 2008-06-19 15:30:23 -0500
- (Thu, 19 Jun 2008) | 16 lines Merged revisions 124112 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r124112 | mmichelson | 2008-06-19 15:28:41 -0500 (Thu, 19 Jun
- 2008) | 8 lines Fix IMAP forwarding so that messages are sent to
- the proper mailbox. (closes issue #12897) Reported by: jaroth
- Patches: destination_forward.patch uploaded by jaroth (license
- 50) ........ ................
-
-2008-06-19 19:49 +0000 [r124065] Brett Bryant <bbryant@digium.com>
-
- * /, main/utils.c: Merged revisions 124064 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r124064 |
- bbryant | 2008-06-19 14:48:26 -0500 (Thu, 19 Jun 2008) | 2 lines
- Add errors that report any locks held by threads when they are
- being closed. ........
-
-2008-06-19 18:57 +0000 [r124026] Brett Bryant <bbryant@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 124024 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r124024 |
- bbryant | 2008-06-19 13:57:04 -0500 (Thu, 19 Jun 2008) | 2 lines
- Fix bug in sip registration that sets the default port to 5060
- for tls. ........
-
-2008-06-19 17:58 +0000 [r123871-123989] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_config_ldap.c: Merged revisions 123952 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r123952 | tilghman | 2008-06-19 12:22:27 -0500 (Thu, 19 Jun 2008)
- | 6 lines Don't change pointers that need to be later passed back
- for deallocation. (closes issue #12572) Reported by: flyn
- Patches: 20080613__bug12572.diff.txt uploaded by Corydon76
- (license 14) ........
-
- * main/channel.c, /: Merged revisions 123931 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r123931 | tilghman | 2008-06-19 12:02:54 -0500 (Thu, 19 Jun 2008)
- | 13 lines Merged revisions 123930 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r123930 | tilghman | 2008-06-19 11:58:19 -0500 (Thu, 19 Jun 2008)
- | 5 lines Change informative messages to use the _multiple
- variant when multiple formats are possible. (Closes issue #12848)
- Reported by klaus3000 ........ ................
-
- * /, build_tools/strip_nonapi: Merged revisions 123913 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r123913 | tilghman | 2008-06-19 11:26:50 -0500
- (Thu, 19 Jun 2008) | 13 lines Merged revisions 123909 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r123909 | tilghman | 2008-06-19 11:26:03 -0500 (Thu, 19 Jun 2008)
- | 5 lines Only process 40 arguments (20 files) at once with
- xargs, because some older shells may force xargs to separate on
- an odd boundary. (Closes issue #12883) Reported by Nik Soggia
- ........ ................
-
- * /, configs/sip.conf.sample: Merged revisions 123887 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r123887 | tilghman | 2008-06-19 11:21:32 -0500
- (Thu, 19 Jun 2008) | 12 lines Merged revisions 123883 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r123883 | tilghman | 2008-06-19 11:20:41 -0500 (Thu, 19 Jun 2008)
- | 4 lines Correct description of notifyringing option. (Closes
- issue #12890) Reported by gminet ........ ................
-
- * main/asterisk.c, /: Merged revisions 123870 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r123870 | tilghman | 2008-06-19 11:08:29 -0500 (Thu, 19 Jun 2008)
- | 14 lines Merged revisions 123869 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r123869 | tilghman | 2008-06-19 11:07:23 -0500 (Thu, 19 Jun 2008)
- | 6 lines The RDTSC instruction was introduced on the Pentium
- line of microprocessors, and is not compatible with certain 586
- clones, like Cyrix. Hence, asking for i386 compatibility was
- always incorrect. See http://en.wikipedia.org/wiki/RDTSC (Closes
- issue #12886) Reported by tecnoxarxa ........ ................
-
-2008-06-18 22:18 +0000 [r123718-123772] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/say.c, doc/lang (added), doc/lang/hebrew.ods: Merged
- revisions 123770 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r123770 | tilghman | 2008-06-18 17:17:17 -0500 (Wed, 18 Jun 2008)
- | 16 lines Merged revisions 123769 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r123769 | tilghman | 2008-06-18 17:08:30 -0500 (Wed, 18 Jun 2008)
- | 8 lines Add support for saying numbers in Hebrew. (closes issue
- #11662) Reported by: greenfieldtech Patches: say.c.patch-12042008
- uploaded by greenfieldtech (license 369) Hebrew-Sounds.ods
- uploaded by greenfieldtech (with signficant changes to the
- spreadsheet by me) ........ ................
-
- * pbx/pbx_spool.c, /: Merged revisions 123715 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r123715 | tilghman | 2008-06-18 15:23:58 -0500 (Wed, 18 Jun 2008)
- | 15 lines Merged revisions 123710 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r123710 | tilghman | 2008-06-18 15:22:42 -0500 (Wed, 18 Jun 2008)
- | 7 lines Set the variables top-down, so that if a script sets a
- variable more than once, the last one will take precedence.
- (closes issue #12673) Reported by: phber Patches:
- 20080519__bug12673.diff.txt uploaded by Corydon76 (license 14)
- ........ ................
-
-2008-06-18 20:08 +0000 [r123693] Brett Bryant <bbryant@digium.com>
-
- * main/tcptls.c, /: Merged revisions 123692 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r123692 |
- bbryant | 2008-06-18 15:07:56 -0500 (Wed, 18 Jun 2008) | 2 lines
- Fix a crash in tcp and tls connections related to reference
- counts. ........
-
-2008-06-18 15:09 +0000 [r123651-123653] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 123652 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r123652 |
- mmichelson | 2008-06-18 10:08:56 -0500 (Wed, 18 Jun 2008) | 7
- lines A portion of the code which handled the 'c' queue option
- had been removed. No telling when it happened. Anyway, it's back
- in now and works properly. (Based on issue reported on mailing
- list) ........
-
-2008-06-18 12:34 +0000 [r123646-123647] Russell Bryant <russell@digium.com>
-
- * apps/app_fax.c: don't use trunk only API for frame data (closes
- issue #12881)
-
- * apps/app_fax.c (added): re-add app_fax ... it got accidentally
- removed (closes issue #12881)
-
-2008-06-17 21:57 +0000 [r123547] Brett Bryant <bbryant@digium.com>
-
- * main/tcptls.c, main/manager.c, /, channels/chan_sip.c,
- main/http.c, include/asterisk/tcptls.h: Merged revisions 123546
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r123546 | bbryant | 2008-06-17 16:46:57 -0500 (Tue, 17
- Jun 2008) | 5 lines Updates all usages of
- ast_tcptls_session_instance to be managed by reference counts so
- that they only get destroyed when all threads are done using
- them, and memory does not get free'd causing strange issues with
- SIP. This code was originally written by russellb in the
- team/group/issue_11972/ branch. ........
-
-2008-06-17 21:34 +0000 [r123487-123542] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 123486 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r123486 | mmichelson | 2008-06-17 15:28:47 -0500 (Tue, 17 Jun
- 2008) | 12 lines Merged revisions 123485 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r123485 | mmichelson | 2008-06-17 15:26:38 -0500 (Tue, 17 Jun
- 2008) | 4 lines Make chan_sip build under dev mode with compilers
- >= GCC 4.2 Thanks to jpeeler for alerting me of this ........
- ................
-
-2008-06-17 20:23 +0000 [r123473] Steve Murphy <murf@digium.com>
-
- * /: block 123448 from trunk; it doesn't apply here.
-
-2008-06-17 19:01 +0000 [r123394] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 123392 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r123392 | tilghman | 2008-06-17 13:57:45 -0500
- (Tue, 17 Jun 2008) | 11 lines Merged revisions 123391 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r123391 | tilghman | 2008-06-17 13:56:53 -0500 (Tue, 17 Jun 2008)
- | 3 lines Fix 3 more places where failure to lock the structure
- could cause the wrong lock to be unlocked. (Closes issue #12795)
- ........ ................
-
-2008-06-17 18:28 +0000 [r123382-123387] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 123238 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r123238 | jpeeler | 2008-06-16 18:05:18 -0500 (Mon, 16 Jun 2008)
- | 1 line Fix some (more) variables that were forgotten to be
- renamed, related to 117658 ........
-
-2008-06-17 18:10 +0000 [r123335] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 123334 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r123334 | mmichelson | 2008-06-17 13:09:54 -0500 (Tue, 17 Jun
- 2008) | 19 lines Merged revisions 123333 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r123333 | mmichelson | 2008-06-17 13:09:16 -0500 (Tue, 17 Jun
- 2008) | 11 lines Cisco BTS sends SIP responses with a tab between
- the Cseq number and SIP request method in the Cseq: header.
- Asterisk did not handle this properly, but with this patch, all
- is well. (closes issue #12834) Reported by: tobias_e Patches:
- 12834.patch uploaded by putnopvut (license 60) Tested by:
- tobias_e ........ ................
-
-2008-06-17 18:08 +0000 [r123332] Jeff Peeler <jpeeler@digium.com>
-
- * doc/tex/configuration.tex, configs/zapata.conf.sample, Makefile,
- doc/janitor-projects.txt, configs/vpb.conf.sample, doc/sms.txt,
- contrib/scripts/loadtest.tcl, codecs/codec_dahdi.c (added),
- configs/smdi.conf.sample, pbx/pbx_config.c, apps/app_chanspy.c,
- main/asterisk.c, configs/users.conf.sample, doc/ss7.txt,
- apps/app_meetme.c, configs/rpt.conf.sample, doc/backtrace.txt,
- doc/tex/queues-with-callback-members.tex,
- include/asterisk/dahdi.h (added), configs/extensions.ael.sample,
- res/res_musiconhold.c, configs/meetme.conf.sample,
- codecs/codec_zap.c (removed), contrib/init.d/rc.mandrake.zaptel,
- cdr/cdr_csv.c, main/channel.c, doc/tex/manager.tex,
- doc/tex/sla.tex, include/asterisk/dsp.h,
- doc/tex/localchannel.tex, apps/app_rpt.c, channels/chan_mgcp.c,
- contrib/scripts/autosupport, doc/manager_1_1.txt,
- channels/chan_zap.c (removed), doc/asterisk.8, doc/tex/ael.tex,
- doc/tex/channelvariables.tex, apps/app_getcpeid.c,
- doc/tex/enum.tex, apps/app_queue.c, configs/sla.conf.sample,
- doc/tex/security.tex, include/asterisk/zapata.h (removed),
- doc/tex/privacy.tex, build_tools/menuselect-deps.in,
- apps/app_flash.c, main/file.c, doc/osp.txt,
- contrib/utils/zones2indications.c, utils/extconf.c, makeopts.in,
- configs/extensions.conf.sample, doc/asterisk.sgml, README,
- contrib/init.d/rc.mandrake.asterisk, /,
- include/asterisk/autoconfig.h.in, apps/app_dahdiscan.c (added),
- apps/app_chanisavail.c, channels/chan_iax2.c,
- configs/muted.conf.sample, main/loader.c, channels/chan_dahdi.c
- (added), include/asterisk/doxyref.h, configure,
- doc/tex/backtrace.tex, apps/app_zapscan.c (removed),
- doc/tex/app-sms.tex, apps/app_zapras.c (removed),
- configs/extensions.lua.sample, include/asterisk/options.h,
- contrib/init.d/rc.suse.asterisk, apps/app_dial.c,
- apps/app_page.c, doc/tex/hardware.tex, apps/app_fax.c (removed),
- apps/app_dahdiras.c (added), configure.ac,
- configs/queues.conf.sample, include/asterisk/channel.h: Goodbye
- Zaptel, hello DAHDI. Removes Zaptel driver support with DAHDI.
- Configuration file and dialplan backwards compatability has been
- put in place where appropiate. Release announcement to follow.
-
-2008-06-17 15:58 +0000 [r123276] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 123275 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r123275 | mmichelson | 2008-06-17 10:57:43 -0500 (Tue, 17 Jun
- 2008) | 20 lines Merged revisions 123274 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r123274 | mmichelson | 2008-06-17 10:56:55 -0500 (Tue, 17 Jun
- 2008) | 12 lines davidw pointed out that the holdtime calculation
- used by app_queue does not use "boxcar" filtering as the comments
- say. The term "boxcar" means that the number of samples used to
- calculate stays constant, with new samples replacing the oldest
- ones. The queue holdtime calculation uses all holdtime samples
- collected since the queue was loaded, so the comment has been
- changed to be accurate. (closes issue #12781) Reported by: davidw
- ........ ................
-
-2008-06-17 15:52 +0000 [r123273] Russell Bryant <russell@digium.com>
-
- * main/astobj2.c, /: Merged revisions 123272 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r123272 | russell | 2008-06-17 10:52:13 -0500 (Tue, 17 Jun 2008)
- | 12 lines Merged revisions 123271 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r123271 | russell | 2008-06-17 10:48:31 -0500 (Tue, 17 Jun 2008)
- | 4 lines Fix a memory leak in astobj2 that was pointed out by
- seanbright. When a container got destroyed, the underlying bucket
- list entry for each object that was in the container at that time
- did not get free'd. ........ ................
-
-2008-06-16 21:20 +0000 [r123178] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_zap.c: Fix some variables that were forgotten to be
- renamed, related to 117658. Couldn't merge from trunk since the
- chan_dahdi transition has not occurred here yet
-
-2008-06-16 21:19 +0000 [r123173] Steve Murphy <murf@digium.com>
-
- * apps/app_stack.c, apps/app_dial.c, main/pbx.c, /,
- main/features.c, include/asterisk/pbx.h, apps/app_queue.c: Merged
- revisions 123165 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r123165 |
- murf | 2008-06-16 14:43:46 -0600 (Mon, 16 Jun 2008) | 19 lines
- (closes issue #12689) Reported by: ys Many thanks to ys for doing
- the research on this problem. I didn't think it would be best to
- unlock the contexts and then relock them after the
- remove_extension2() call, so I added an extra arg to
- remove_extension2() and set it appropriately in each call. There
- were not that many. I considered forcing the code to lock the
- contexts before the call to remove_extension2(), but that would
- require a slightly greater degree of changes, especially since
- the find_context_locked is local to pbx.c I did a simple sanity
- test to make sure the code doesn't mess things up in general.
- ........
-
-2008-06-16 20:03 +0000 [r123112-123116] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_mgcp.c, /, channels/chan_sip.c,
- channels/chan_skinny.c, channels/chan_h323.c,
- channels/chan_iax2.c: Merged revisions 123114 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r123114 | tilghman | 2008-06-16 14:57:05 -0500 (Mon, 16 Jun 2008)
- | 10 lines 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 ........ ................
-
- * /, channels/chan_sip.c: Merged revisions 123111 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r123111 | tilghman | 2008-06-16 14:23:51 -0500 (Mon, 16 Jun 2008)
- | 16 lines Merged revisions 123110 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r123110 | tilghman | 2008-06-16 14:21:58 -0500 (Mon, 16 Jun 2008)
- | 8 lines People expect that if "hasvoicemail" is set in
- users.conf, even if "mailbox" isn't set, that SIP will detect a
- mailbox. (closes issue #12855) Reported by: PLL Patches:
- 20080614__bug12855__2.diff.txt uploaded by Corydon76 (license 14)
- Tested by: PLL ........ ................
-
-2008-06-16 17:29 +0000 [r123075] Chris Tooley <chris@tooley.com>
-
- * apps/app_externalivr.c: Fixes and closes bug number 12804
-
-2008-06-16 12:32 +0000 [r122871-122921] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 122920 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r122920 | file | 2008-06-16 09:32:02 -0300 (Mon, 16 Jun 2008) |
- 14 lines Merged revisions 122919 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r122919 | file | 2008-06-16 09:31:09 -0300 (Mon, 16 Jun 2008) | 6
- lines Only compare the first 15 characters so that even if the
- charset is specified we still accept it as SDP. (closes issue
- #12803) Reported by: lanzaandrea Patches: chan_sip.c.diff
- uploaded by lanzaandrea (license 496) ........ ................
-
- * /, channels/chan_sip.c: Merged revisions 122870 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r122870 | file | 2008-06-16 09:09:54 -0300 (Mon, 16 Jun 2008) |
- 14 lines Merged revisions 122869 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r122869 | file | 2008-06-16 09:08:28 -0300 (Mon, 16 Jun 2008) | 6
- lines Don't send a BYE on a dialog that is already gone during a
- REFER. (closes issue #12865) Reported by: flefoll Patches:
- chan_sip.c.br14.121495.patch-ALREADYGONE uploaded by flefoll
- (license 244) ........ ................
-
-2008-06-13 21:47 +0000 [r122715] Mark Michelson <mmichelson@digium.com>
-
- * main/autoservice.c, /: Merged revisions 122714 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r122714 | mmichelson | 2008-06-13 16:45:21 -0500 (Fri, 13 Jun
- 2008) | 17 lines Merged revisions 122713 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r122713 | mmichelson | 2008-06-13 16:44:53 -0500 (Fri, 13 Jun
- 2008) | 9 lines Short circuit the loop in autoservice_run if
- there are no channels to poll. If we continued, then the result
- would be calling poll() with a NULL pollfd array. While this is
- fine with POSIX's poll(2) system call, those who use Asterisk's
- internal poll mechanism (Darwin systems) would have a failed
- assertion occur when poll is called. (related to issue #10342)
- ........ ................
-
-2008-06-13 14:15 +0000 [r122558] Tilghman Lesher <tlesher@digium.com>
-
- * main/dial.c, /: Merged revisions 122557 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r122557 |
- tilghman | 2008-06-13 09:15:07 -0500 (Fri, 13 Jun 2008) | 7 lines
- Convert one more delimiter to use comma. (closes issue #12850)
- Reported by: bcnit Patches: 20080613__bug12850.diff.txt uploaded
- by Corydon76 (license 14) Tested by: bcnit ........
-
-2008-06-13 00:18 +0000 [r122467] Jeff Peeler <jpeeler@digium.com>
-
- * apps/app_parkandannounce.c, /, main/features.c: Merged revisions
- 122433 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r122433 |
- jpeeler | 2008-06-12 18:08:37 -0500 (Thu, 12 Jun 2008) | 4 lines
- (closes issue 0012193) Reported by: davidw Patch by: Corydon76,
- modified by me to work properly with ParkAndAnnounce app ........
-
-2008-06-12 18:54 +0000 [r122313] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 122312 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r122312 | mmichelson | 2008-06-12 13:53:17 -0500 (Thu, 12 Jun
- 2008) | 17 lines Merged revisions 122311 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r122311 | mmichelson | 2008-06-12 13:50:58 -0500 (Thu, 12 Jun
- 2008) | 9 lines Properly play a holdtime message if the
- announce-holdtime option is set to "once." (closes issue #12842)
- Reported by: ramonpeek Patches: patch001.diff uploaded by
- ramonpeek (license 266) ........ ................
-
-2008-06-12 18:24 +0000 [r122242-122266] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 122262 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r122262 | russell | 2008-06-12 13:23:54 -0500
- (Thu, 12 Jun 2008) | 11 lines Merged revisions 122259 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r122259 | russell | 2008-06-12 13:22:44 -0500 (Thu, 12 Jun 2008)
- | 3 lines Fix some race conditions that cause ast_assert() to
- report that chan_iax2 tried to remove an entry that wasn't in the
- scheduler ........ ................
-
-2008-06-12 15:27 +0000 [r122132-122180] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 122174 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r122174 | tilghman | 2008-06-12 10:26:07 -0500 (Thu, 12 Jun 2008)
- | 16 lines Merged revisions 122137 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r122137 | tilghman | 2008-06-12 10:18:39 -0500 (Thu, 12 Jun 2008)
- | 8 lines Flipflop the sections for two options, since the
- section for 'X' (exit context) may otherwise absorb keypresses
- meant for 's' (admin/user menu). (closes issue #12836) Reported
- by: blitzrage Patches: 20080611__bug12836.diff.txt uploaded by
- Corydon76 (license 14) Tested by: blitzrage ........
- ................
-
- * main/channel.c, /: Merged revisions 122131 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r122131 | tilghman | 2008-06-12 10:14:37 -0500 (Thu, 12 Jun 2008)
- | 12 lines Merged revisions 122130 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r122130 | tilghman | 2008-06-12 10:11:30 -0500 (Thu, 12 Jun 2008)
- | 4 lines Occasionally, the alertpipe loses its nonblocking
- status, so detect and correct that situation before it causes a
- deadlock. (Reported and tested by ctooley via #asterisk-dev)
- ........ ................
-
-2008-06-12 15:01 +0000 [r122126-122129] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, apps/app_forkcdr.c, /, CHANGES: Merged revisions
- 122128 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r122128 | murf | 2008-06-12 08:56:26 -0600 (Thu, 12 Jun 2008) | 9
- lines Merged revisions 122127 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r122127 | murf | 2008-06-12 08:51:44 -0600 (Thu, 12 Jun 2008) | 1
- line Arkadia tried to warn me, but the code added to
- ast_cdr_busy, _failed, and _noanswer was redundant. Didn't spot
- it until I was resolving conflicts in trunk. Ugh. Redundant code
- removed. It wasn't harmful. Just dumb. ........ ................
-
- * main/cdr.c, apps/app_forkcdr.c, /, funcs/func_cdr.c,
- include/asterisk/cdr.h, CHANGES: Merged revisions 122091 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r122091 | murf | 2008-06-12 08:28:01 -0600 (Thu,
- 12 Jun 2008) | 45 lines Merged revisions 122046 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r122046 | murf | 2008-06-12 07:47:34 -0600 (Thu, 12 Jun 2008) |
- 37 lines (closes issue #10668) Reported by: arkadia Tested by:
- murf, arkadia Options added to forkCDR() app and the CDR() func
- to remove some roadblocks for CDR applications. The "show
- application ForkCDR" output was upgraded to more fully explain
- the inner workings of forkCDR. The A option was added to forkCDR
- to force the CDR system to NOT change the disposition on the
- original CDR, after the fork. This involves ast_cdr_answer,
- _busy, _failed, and so on. The T option was added to forkCDR to
- force obedience of the cdr LOCKED flag in the ast_cdr_end, all
- the disposition changing funcs (ast_cdr_answer, etc), and in the
- ast_cdr_setvar func. The CHANGES file was updated to explain ALL
- the new options added to satisfy this bug report (and some
- requests made verbally and via email, irc, etc, over the past
- months/year) The 's' option was added to the CDR() func, to force
- it to skip LOCKED cdr's in the chain. Again, the new options
- should be totally transparent to existing apps! Current behavior
- of CDR, forkCDR, and the rest of the CDR system should not change
- one little bit. Until you add the new options, at least! ........
- ................
-
-2008-06-11 18:57 +0000 [r121915] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 121914 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r121914 |
- mattf | 2008-06-11 13:53:10 -0500 (Wed, 11 Jun 2008) | 1 line Fix
- pseudo channel allocation errors on startup when using SS7
- ........
-
-2008-06-11 18:20 +0000 [r121872] Tilghman Lesher <tlesher@digium.com>
-
- * main/sched.c, main/channel.c, /, channels/chan_agent.c,
- main/abstract_jb.c: Merged revisions 121867 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r121867 | tilghman | 2008-06-11 13:19:24 -0500 (Wed, 11 Jun 2008)
- | 11 lines Merged revisions 121861 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r121861 | tilghman | 2008-06-11 13:18:16 -0500 (Wed, 11 Jun 2008)
- | 3 lines Make calls to ast_assert() actually test something, so
- that the error message printed is not nonsensical (reported by
- mvanbaak via #asterisk-bugs). ........ ................
-
-2008-06-11 17:59 +0000 [r121858] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 121857 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r121857 |
- mattf | 2008-06-11 12:50:17 -0500 (Wed, 11 Jun 2008) | 1 line
- Make sure we hangup any calls we have and NULL out the ss7call
- value when we get a reset circuit message. Fixes crash bug
- ........
-
-2008-06-11 17:45 +0000 [r121856] Tilghman Lesher <tlesher@digium.com>
-
- * contrib/scripts/realtime_pgsql.sql, /, UPGRADE.txt,
- include/asterisk/cdr.h: Merged revisions 121855 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r121855 |
- tilghman | 2008-06-11 12:44:39 -0500 (Wed, 11 Jun 2008) | 3 lines
- Expand CDR uniqueid field to 150 chars, to account for maximum
- systemname. (Closes issue #12831) ........
-
-2008-06-11 16:13 +0000 [r121806] Jeff Peeler <jpeeler@digium.com>
-
- * /, doc/backtrace.txt: Merged revisions 121805 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r121805 | jpeeler | 2008-06-11 11:11:40 -0500 (Wed, 11 Jun 2008)
- | 9 lines Merged revisions 121804 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r121804 | jpeeler | 2008-06-11 11:11:09 -0500 (Wed, 11 Jun 2008)
- | 1 line add instructions for logging gdb output via set logging
- on ........ ................
-
-2008-06-10 18:36 +0000 [r121598] Sean Bright <sean.bright@gmail.com>
-
- * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 121597
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r121597 | seanbright | 2008-06-10 14:35:37 -0400
- (Tue, 10 Jun 2008) | 14 lines Merged revisions 121596 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r121596 | seanbright | 2008-06-10 14:34:45 -0400 (Tue, 10 Jun
- 2008) | 6 lines Fixes a problem with some buggy versions of GNU
- awk (3.1.3) not liking carriage returns in scripts. (closes issue
- #12749) Reported by: alinux Tested by: Laureano (on
- #asterisk-dev), juggie ........ ................
-
-2008-06-10 12:55 +0000 [r121445] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /: Merged revisions 121444 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r121444 | file | 2008-06-10 09:54:39 -0300 (Tue, 10 Jun 2008) |
- 12 lines Merged revisions 121442 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r121442 | file | 2008-06-10 09:52:06 -0300 (Tue, 10 Jun 2008) | 4
- lines Update BRIDGEPEER variable before we do a generic bridge in
- case we just broke out of a native bridge and fell through to
- generic. (closes issue #12815) Reported by: ramonpeek ........
- ................
-
-2008-06-10 00:53 +0000 [r121404-121408] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 121407 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r121407 | russell | 2008-06-09 19:52:46 -0500 (Mon, 09 Jun 2008)
- | 2 lines Bump up the debug level of a couple of messages
- ........
-
-2008-06-09 16:37 +0000 [r121283] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /: Merged revisions 121282 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r121282 | russell | 2008-06-09 11:37:08 -0500 (Mon, 09 Jun 2008)
- | 18 lines Merged revisions 121280 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r121280 | russell | 2008-06-09 11:35:40 -0500 (Mon, 09 Jun 2008)
- | 10 lines Do not attempt to do emulation if an END digit is
- received and the length is less than the defined minimum digit
- length, and the other end only wants END digits (SIP INFO, for
- example). (closes issue #12778) Reported by: tsearle Patches:
- 12778.rev1.txt uploaded by russell (license 2) Tested by: tsearle
- ........ ................
-
-2008-06-09 16:36 +0000 [r121281] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c, /: Merged revisions 121279 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r121279 |
- tilghman | 2008-06-09 11:35:06 -0500 (Mon, 09 Jun 2008) | 6 lines
- Implement FINDLABEL matching for the new extension matching
- engine. (closes issue #12800) Reported by: chris-mac Patches:
- 20080608__bug12800.diff.txt uploaded by Corydon76 (license 14)
- ........
-
-2008-06-09 15:10 +0000 [r121231] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 121230 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r121230 | mmichelson | 2008-06-09 10:08:58 -0500
- (Mon, 09 Jun 2008) | 27 lines Merged revisions 121229 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (Note that
- this is being merged to trunk/1.6.0 because it may affect
- non-callback agents with ackcall set) ........ r121229 |
- mmichelson | 2008-06-09 10:02:37 -0500 (Mon, 09 Jun 2008) | 16
- lines A unique situation of timeouts brought forth a failure
- situation for autologoff in chan_agent. If using
- AgentCallbackLogin-style agents, then if the timeout specified by
- the Dial() to reach the agent's phone was shorter than the
- timeout specified in queues.conf, then autologoff would only work
- if the caller hung up while the agent's phone was ringing. This
- patch allows autologoff to work in this situation when the call
- in queue transfers to the next available agent (as it would have
- if the timeout in queues.conf were less than the timeout in the
- Dial()). (closes issue #12754) Reported by: Rodrigo Patches:
- 12754.patch uploaded by putnopvut (license 60) Tested by: Rodrigo
- ........ ................
-
-2008-06-08 01:43 +0000 [r121138-121164] Jeff Peeler <jpeeler@digium.com>
-
- * /, channels/chan_console.c: Merged revisions 121163 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r121163 | jpeeler | 2008-06-07 20:41:59 -0500 (Sat, 07 Jun 2008)
- | 4 lines This was accidentally reverted. Fixes a bug where if a
- stream monitor thread was not created (caused from failure of
- opening or starting the stream) pthread_cancel was called with an
- invalid thread ID. ........
-
- * apps/app_parkandannounce.c, /: Merged revisions 121131 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r121131 | jpeeler | 2008-06-07 20:16:25 -0500 (Sat, 07
- Jun 2008) | 2 lines Fixes segfault when using ParkAndAnnounce.
- Also, loop made more efficient as announce template only needs to
- be checked until the number of colon separated arguments run out,
- not the entire pointer storage array. Was done in a similiar
- fashion in 1.4, but here we're using less variables. ........
-
-2008-06-07 14:19 +0000 [r121080] Russell Bryant <russell@digium.com>
-
- * channels/chan_local.c, /, channels/chan_agent.c: Merged revisions
- 121079 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r121079 | russell | 2008-06-07 09:18:44 -0500 (Sat, 07 Jun 2008)
- | 15 lines Merged revisions 121078 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r121078 | russell | 2008-06-07 09:10:56 -0500 (Sat, 07 Jun 2008)
- | 7 lines Don't run LIST_HEAD_DESTROY on a STATIC list (closes
- issue #12807) Reported by: ys Patches: chan_agent_local.diff
- uploaded by ys (license 281) ........ ................
-
-2008-06-06 20:25 +0000 [r121011-121047] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c, /: Merged revisions 121010 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r121010 |
- tilghman | 2008-06-06 14:55:08 -0500 (Fri, 06 Jun 2008) | 6 lines
- Make extension match characters case-insensitive. (closes issue
- #12777) Reported by: jsmith Patches:
- lower_case_patterns-trunk-v1.patch uploaded by jsmith (license
- 15) ........
-
-2008-06-06 18:31 +0000 [r120907-120961] Jeff Peeler <jpeeler@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 120960 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r120960 | jpeeler | 2008-06-06 13:30:17 -0500 (Fri, 06 Jun 2008)
- | 9 lines Merged revisions 120959 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r120959 | jpeeler | 2008-06-06 13:29:14 -0500 (Fri, 06 Jun 2008)
- | 1 line add another LOW_MEMORY define I forgot ........
- ................
-
- * /, channels/chan_sip.c: Merged revisions 120909 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r120909 | jpeeler | 2008-06-06 13:06:06 -0500 (Fri, 06 Jun 2008)
- | 9 lines Merged revisions 120908 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r120908 | jpeeler | 2008-06-06 13:05:15 -0500 (Fri, 06 Jun 2008)
- | 1 line only define thread storage variable if necessary for
- LOW_MEMORY ........ ................
-
- * channels/chan_sip.c: Merged revisions 120906 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r120906 | jpeeler | 2008-06-06 12:50:05 -0500 (Fri, 06 Jun 2008)
- | 16 lines Merged revisions 120863,120885 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r120863 | jpeeler | 2008-06-06 10:33:15 -0500 (Fri, 06 Jun 2008)
- | 3 lines This fixes a crash when LOW_MEMORY is turned on. Two
- allocations of the ast_rtp struct that were previously allocated
- on the stack have been modified to use thread local storage
- instead. ........ r120885 | jpeeler | 2008-06-06 11:39:20 -0500
- (Fri, 06 Jun 2008) | 2 lines Correction to commmit 120863, make
- sure proper destructor function is called as well define two
- thread storage local variables. ........ ................
-
-2008-06-06 17:35 +0000 [r120864-120905] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_exec.c: Merged revisions 120904 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r120904 |
- tilghman | 2008-06-06 12:34:21 -0500 (Fri, 06 Jun 2008) | 3 lines
- For the purpose of making the changed syntax to ExecIf easier to
- transition, allow the deprecated syntax (fixed for jmls on -dev).
- ........
-
-2008-06-05 21:39 +0000 [r120829] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, /: Merged revisions 120828 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r120828 |
- murf | 2008-06-05 15:34:42 -0600 (Thu, 05 Jun 2008) | 1 line a
- small fix for a crash that occurs when compiling AEL with global
- vars ........
-
-2008-06-05 17:17 +0000 [r120677] Philippe Sultan <philippe.sultan@gmail.com>
-
- * /, res/res_jabber.c: Merged revisions 120676 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r120676 | phsultan | 2008-06-05 19:02:39 +0200 (Thu, 05 Jun 2008)
- | 10 lines Merged revisions 120675 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r120675 | phsultan | 2008-06-05 18:56:15 +0200 (Thu, 05 Jun 2008)
- | 2 lines Ignore appended resource when comparing JIDs. ........
- ................
-
-2008-06-05 16:42 +0000 [r120643-120674] Brett Bryant <bbryant@digium.com>
-
-2008-06-05 16:01 +0000 [r120566-120603] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_stack.c, main/loader.c, /, res/res_agi.c: Merged
- revisions 120602 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r120602 |
- tilghman | 2008-06-05 10:58:11 -0500 (Thu, 05 Jun 2008) | 4 lines
- Conditionally load the AGI command gosub, depending on whether or
- not res_agi has been loaded, fix a return value in the loader,
- and ensure that the help workhorse header does not print on load.
- ........
-
- * /, UPGRADE.txt: Merged revisions 120567 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r120567 |
- tilghman | 2008-06-05 09:35:47 -0500 (Thu, 05 Jun 2008) | 2 lines
- Add info on the [compat] section of asterisk.conf. ........
-
- * apps/app_fax.c: Fix frame API for 1.6.0 (Closes issue #12793)
-
-2008-06-04 22:08 +0000 [r120515] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 120514 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r120514 | mmichelson | 2008-06-04 17:07:37 -0500 (Wed, 04 Jun
- 2008) | 14 lines Merged revisions 120513 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r120513 | mmichelson | 2008-06-04 17:05:33 -0500 (Wed, 04 Jun
- 2008) | 6 lines Make sure that the string we set will survive the
- unref of the queue member. Thanks to Russell, who pointed this
- out. ........ ................
-
-2008-06-04 20:35 +0000 [r120478] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c, /: Merged revisions 120477 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r120477 |
- tilghman | 2008-06-04 15:34:52 -0500 (Wed, 04 Jun 2008) | 2 lines
- MSet doesn't necessarily need chan to be set ........
-
-2008-06-04 15:38 +0000 [r120338] Joshua Colp <jcolp@digium.com>
-
- * /, pbx/pbx_config.c: Merged revisions 120337 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r120337 |
- file | 2008-06-04 12:38:00 -0300 (Wed, 04 Jun 2008) | 2 lines We
- like tabs. ........
-
-2008-06-04 14:13 +0000 [r120287] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 120286 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r120286 | mmichelson | 2008-06-04 09:12:45 -0500 (Wed, 04 Jun
- 2008) | 15 lines Merged revisions 120285 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r120285 | mmichelson | 2008-06-04 09:11:12 -0500 (Wed, 04 Jun
- 2008) | 7 lines Tab completion when removing a member should give
- the member's interface, not the name, since the interface is what
- is expected for the command. (closes issue #12783) Reported by:
- davevg ........ ................
-
-2008-06-04 13:34 +0000 [r120284] Joshua Colp <jcolp@digium.com>
-
- * /, pbx/pbx_config.c: Merged revisions 120283 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r120283 | file | 2008-06-04 10:33:59 -0300 (Wed, 04 Jun 2008) |
- 14 lines Merged revisions 120282 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r120282 | file | 2008-06-04 10:31:09 -0300 (Wed, 04 Jun 2008) | 6
- lines Fix a log message and add a message for when the dialplan
- is done reloading. (closes issue #12716) Reported by: chappell
- Patches: dialplan_reload_2.diff uploaded by chappell (license 8)
- ........ ................
-
-2008-06-03 23:18 +0000 [r120228-120234] Tilghman Lesher <tlesher@digium.com>
-
- * pbx/pbx_loopback.c, /: Merged revisions 120227 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r120227 | tilghman | 2008-06-03 17:42:03 -0500 (Tue, 03 Jun 2008)
- | 16 lines Merged revisions 120226 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r120226 | tilghman | 2008-06-03 17:41:04 -0500 (Tue, 03 Jun 2008)
- | 8 lines Due to incorrect use of the AST_LIST_INSERT_HEAD()
- macro the loopback switch cannot perform any translation on the
- extension number before searching for it in the target context.
- (closes issue #12473) Reported by: chappell Patches:
- pbx_loopback.c.diff uploaded by chappell (license 8) ........
- ................
-
-2008-06-03 22:18 +0000 [r120178] Jeff Peeler <jpeeler@digium.com>
-
- * main/config.c: Merged revisions 120174 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r120174 | jpeeler | 2008-06-03 17:17:07 -0500 (Tue, 03 Jun 2008)
- | 14 lines Merged revisions 120173 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r120173 | jpeeler | 2008-06-03 17:15:33 -0500 (Tue, 03 Jun 2008)
- | 6 lines (closes issue #11594) Reported by: yem Tested by: yem
- This change decreases the buffer size allocated on the stack
- substantially in config_text_file_load when LOW_MEMORY is turned
- on. This change combined with the fix from revision 117462
- (making mkintf not copy the zt_chan_conf structure) was enough to
- prevent the crash. ........ ................
-
-2008-06-03 22:08 +0000 [r120172] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/options.h, main/asterisk.c, Makefile,
- main/pbx.c, /, res/res_agi.c, pbx/pbx_realtime.c,
- configs/pbx_realtime.conf (removed): Merged revisions 120171 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r120171 | tilghman | 2008-06-03 17:05:16 -0500 (Tue, 03
- Jun 2008) | 5 lines Move compatibility options into
- asterisk.conf, default them to on for upgrades, and off for new
- installations. This includes the translation from pipes to commas
- for pbx_realtime and the EXEC command for AGI, as well as the
- change to the Set application not to support multiple variables
- at once. ........
-
-2008-06-03 21:35 +0000 [r120170] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 120169 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r120169 | russell | 2008-06-03 16:35:11 -0500
- (Tue, 03 Jun 2008) | 12 lines Merged revisions 120168 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r120168 | russell | 2008-06-03 16:34:55 -0500 (Tue, 03 Jun 2008)
- | 4 lines Fix another place where peer->callno could change at a
- very bad time, and also fix a place where a peer was used after
- the reference was released. (inspired by rev 120001) ........
- ................
-
-2008-06-03 16:24 +0000 [r120034] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 120012 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r120012 | tilghman | 2008-06-03 11:19:35 -0500
- (Tue, 03 Jun 2008) | 17 lines Merged revisions 120001 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r120001 | tilghman | 2008-06-03 11:10:53 -0500 (Tue, 03 Jun 2008)
- | 9 lines Save the callno when we're poking, because our peer
- structure could change during destruction (and thus we unlock the
- wrong callno, causing a cascade failure). (closes issue #12717)
- Reported by: gewfie Patches: 20080525__bug12717.diff.txt uploaded
- by Corydon76 (license 14) Tested by: gewfie ........
- ................
-
-2008-06-03 15:57 +0000 [r119931-120000] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18,
- pbx/ael/ael-test/ref.ael-vtest21,
- pbx/ael/ael-test/ref.ael-test19,
- pbx/ael/ael-test/ref.ael-vtest13,
- pbx/ael/ael-test/ref.ael-vtest17,
- pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
- pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-test5,
- pbx/ael/ael-test/ref.ael-test15: Merged revisions 119998 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r119998 | murf | 2008-06-03 09:49:34 -0600 (Tue,
- 03 Jun 2008) | 16 lines Merged revisions 119966 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r119966 | murf | 2008-06-03 09:26:56 -0600 (Tue, 03 Jun 2008) | 8
- lines Updated the regressions on AEL. Hadn't updated this for the
- changes I made to preserve ${EXTEN} in switches, which affected
- several tests because it adds extra priorities, and at least one
- needed to be updated because of the removal of the empty
- extension warning message. ........ ................
-
- * res/ael/pval.c, /: Merged revisions 119930 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r119930 | murf | 2008-06-03 09:07:20 -0600 (Tue, 03 Jun 2008) |
- 24 lines Merged revisions 119929 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r119929 | murf | 2008-06-03 08:49:46 -0600 (Tue, 03 Jun 2008) |
- 16 lines as per
- http://lists.digium.com/pipermail/asterisk-users/2008-June/212934.html,
- which is a message from Philipp Kempgen, requesting that the
- WARNING that an extension is empty be reduced to a NOTICE or
- less, as empty extensions are syntactically possible, and no big
- deal. With which I agree, and have removed that WARNING message
- entirely. I think it is not necessary to see this message. It
- didn't state that a NoOp() was inserted automatically on your
- behalf, and really, as users, who cares? Why freak out dialplan
- writers with unnecessary warnings? The details of the
- machinations a compiler goes thru to produce working assembly
- code is of little interest to most programmers-- we will follow
- the unix principal of doing our work silently. ........
- ................
-
-2008-06-03 14:48 +0000 [r119928] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 119927 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r119927 | file | 2008-06-03 11:47:54 -0300 (Tue, 03 Jun 2008) |
- 10 lines Merged revisions 119926 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r119926 | file | 2008-06-03 11:46:24 -0300 (Tue, 03 Jun 2008) | 2
- lines Treat ECONNREFUSED as an error that will stop further
- retransmissions. (issue #AST-58, patch from Switchvox) ........
- ................
-
-2008-06-03 13:30 +0000 [r119745-119893] Russell Bryant <russell@digium.com>
-
- * /, main/logger.c: Merged revisions 119892 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r119892 |
- russell | 2008-06-03 08:29:16 -0500 (Tue, 03 Jun 2008) | 9 lines
- Do a deep copy of file and function strings to avoid a potential
- crash when modules are unloaded. (closes issue #12780) Reported
- by: ys Patches: logger.diff uploaded by ys (license 281) --
- modified by me for coding guidelines ........
-
- * /, channels/chan_iax2.c: Merged revisions 119839 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r119839 | russell | 2008-06-02 15:08:24 -0500
- (Mon, 02 Jun 2008) | 15 lines Merged revisions 119838 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r119838 | russell | 2008-06-02 15:08:04 -0500 (Mon, 02 Jun 2008)
- | 7 lines Revert a change made for issue #12479. This change
- caused a regression such that a dial string such as (IAX2/foo)
- did not automatically fall back to dialing the 's' extension
- anymore. (closes issue #12770) Reported by: dagmoller ........
- ................
-
- * /, apps/app_fax.c (added): Merged revisions 119801 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r119801 | russell | 2008-06-02 11:14:15 -0500 (Mon, 02 Jun 2008)
- | 4 lines Add app_fax from asterisk-addons, with some additional
- changes to resolve compiler warnings, as well as update to the
- APIs in spandsp 0.0.5. Spandsp 0.0.5 is being distributed under
- the LGPL, so we can move this module into the main tree. ........
-
- * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
- Merged revisions 119799 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r119799 |
- russell | 2008-06-02 10:57:43 -0500 (Mon, 02 Jun 2008) | 4 lines
- After determining that the version of spandsp installed is an
- acceptable version, do a build and link test to ensure that the
- library is usable, and that libtiff is also available ........
-
- * /, build_tools/menuselect-deps.in, configure,
- include/asterisk/autoconfig.h.in, configure.ac, makeopts.in:
- Merged revisions 119795 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r119795 |
- russell | 2008-06-02 10:43:40 -0500 (Mon, 02 Jun 2008) | 2 lines
- Add a configure script check for spandsp ........
-
- * main/manager.c, /: Merged revisions 119744 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r119744 | russell | 2008-06-02 09:41:55 -0500 (Mon, 02 Jun 2008)
- | 13 lines Merged revisions 119742 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r119742 | russell | 2008-06-02 09:39:45 -0500 (Mon, 02 Jun 2008)
- | 5 lines Improve CLI command blacklist checking for the command
- manager action. Previously, it did not handle case or whitespace
- properly. This made it possible for blacklisted commands to get
- executed anyway. (closes issue #12765) ........ ................
-
-2008-06-02 14:40 +0000 [r119743] Philippe Sultan <philippe.sultan@gmail.com>
-
- * channels/chan_jingle.c, /, channels/chan_gtalk.c,
- res/res_jabber.c: Merged revisions 119741 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r119741 |
- phsultan | 2008-06-02 16:35:24 +0200 (Mon, 02 Jun 2008) | 13
- lines Do not link the guest account with any configured XMPP
- client (in jabber.conf). The actual connection is made when a
- call comes in Asterisk. Apply this fix to Jingle too. Fix the
- ast_aji_get_client function that was not able to retrieve an XMPP
- client from its JID. (closes issue #12085) Reported by: junky
- Tested by: phsultan ........
-
-2008-06-02 12:32 +0000 [r119532-119690] Russell Bryant <russell@digium.com>
-
- * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, /: Merged
- revisions 119586,119637 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r119586 | crichter | 2008-06-02 03:46:23 -0500 (Mon, 02 Jun 2008)
- | 9 lines Merged revisions 119585 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r119585 | crichter | 2008-06-02 10:35:28 +0200 (Mo, 02 Jun 2008)
- | 1 line Added counter for unhandled_bmsg Print, this prevents
- the logs to be flooded to fast and save CPU in this error
- scenario. Added 'last_used' element to bc structure, when a
- bchannel changes from used to free this exact time will be marked
- in last_used. When a new channel is requested the find_free_chan
- function will check if the new empty channel was used within the
- last second, if yes it will search for the next channel, if no it
- will return this channel. This simple mechanism has prooven to
- prevent race conditions where the NT and TE tried to allocate the
- exact same channel at the same time (RELEASE cause: 44). ........
- ................ r119637 | crichter | 2008-06-02 04:35:04 -0500
- (Mon, 02 Jun 2008) | 9 lines Merged revisions 119636 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r119636 | crichter | 2008-06-02 11:29:21 +0200 (Mo, 02
- Jun 2008) | 1 line fixed compile issue when dev-mode is enabled
- ........ ................
-
- * /, channels/chan_iax2.c: Merged revisions 119688 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r119688 | russell | 2008-06-02 07:30:42 -0500
- (Mon, 02 Jun 2008) | 11 lines Merged revisions 119687 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r119687 | russell | 2008-06-02 07:30:17 -0500 (Mon, 02 Jun 2008)
- | 3 lines Even of the first PING or LAGRQ doesn't get sent
- because it comes up too soon, make sure to reschedule so it gets
- sent later. ........ ................
-
- * /, channels/chan_iax2.c: Merged revisions 119534 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r119534 | russell | 2008-06-01 20:08:16 -0500
- (Sun, 01 Jun 2008) | 10 lines Merged revisions 119533 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r119533 | russell | 2008-06-01 20:06:09 -0500 (Sun, 01 Jun 2008)
- | 2 lines Change a debug message to an actual debug message
- ........ ................
-
- * apps/app_dial.c, /: Merged revisions 119531 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r119531 | russell | 2008-06-01 20:04:01 -0500 (Sun, 01 Jun 2008)
- | 10 lines Merged revisions 119530 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r119530 | russell | 2008-06-01 20:03:22 -0500 (Sun, 01 Jun 2008)
- | 2 lines Fix another typo in documentation ........
- ................
-
-2008-06-01 21:59 +0000 [r119529] Michiel van Baak <michiel@vanbaak.info>
-
- * apps/app_dial.c, /: Merged revisions 119479 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r119479 | mvanbaak | 2008-06-01 23:06:27 +0200 (Sun, 01 Jun 2008)
- | 10 lines Merged revisions 119478 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r119478 | mvanbaak | 2008-06-01 22:47:55 +0200 (Sun, 01 Jun 2008)
- | 2 lines small typo fix 'retires' => 'retries' ........
- ................
-
-2008-05-30 21:24 +0000 [r119420] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 119419 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r119419 | tilghman | 2008-05-30 16:23:14 -0500 (Fri, 30 May 2008)
- | 14 lines Merged revisions 119404 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r119404 | tilghman | 2008-05-30 16:17:45 -0500 (Fri, 30 May 2008)
- | 6 lines When joinempty=strict, it only failed on join if there
- were busy members. If all members were logged out OR paused, then
- it (incorrectly) let callers join the queue. (closes issue
- #12451) Reported by: davidw ........ ................
-
-2008-05-30 19:48 +0000 [r119356] Joshua Colp <jcolp@digium.com>
-
- * main/autoservice.c, /: Merged revisions 119355 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r119355 | file | 2008-05-30 16:47:30 -0300 (Fri, 30 May 2008) |
- 10 lines Merged revisions 119354 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r119354 | file | 2008-05-30 16:46:37 -0300 (Fri, 30 May 2008) | 2
- lines Fix a bug I found while testing for another issue. ........
- ................
-
-2008-05-30 17:13 +0000 [r119304] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_stack.c: Oops, broke 1.6 (thanks MattF)
-
-2008-05-30 16:57 +0000 [r119303] Michiel van Baak <michiel@vanbaak.info>
-
- * contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk,
- contrib/init.d/rc.debian.asterisk,
- contrib/init.d/rc.mandrake.asterisk, /,
- contrib/init.d/rc.redhat.asterisk,
- contrib/init.d/rc.gentoo.asterisk,
- contrib/init.d/rc.slackware.asterisk: Merged revisions 119302 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r119302 | mvanbaak | 2008-05-30 18:47:24 +0200
- (Fri, 30 May 2008) | 22 lines Merged revisions 119301 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r119301 | mvanbaak | 2008-05-30 18:44:39 +0200 (Fri, 30 May 2008)
- | 14 lines dont use a bashism way to check the $VERSION variable.
- The rc/init.d scripts, and safe_asterisk work on normal sh now
- again. Tested on: OpenBSD 4.2 (me) Debian etch (me) Ubuntu Hardy
- (me and loloski) FC9 (loloski) (closes issue #12687) Reported by:
- loloski Patches: 20080529-12687-safe_asterisk-fixversion.diff.txt
- uploaded by mvanbaak (license 7) Tested by: loloski, mvanbaak
- ........ ................
-
-2008-05-30 16:40 +0000 [r119297-119300] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_stack.c, /: Merged revisions 119299 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r119299 |
- tilghman | 2008-05-30 11:40:13 -0500 (Fri, 30 May 2008) | 2 lines
- Suppress warning about pbx structure already existing ........
-
- * apps/app_stack.c, apps/app_dial.c, include/asterisk/agi.h, /,
- CHANGES: Merged revisions 119296 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r119296 |
- tilghman | 2008-05-30 11:10:46 -0500 (Fri, 30 May 2008) | 8 lines
- Add native AGI command GOSUB, as invoking Gosub with EXEC does
- not work properly. (closes issue #12760) Reported by: Corydon76
- Patches: 20080530__bug12760.diff.txt uploaded by Corydon76
- (license 14) Tested by: tim_ringenbach, Corydon76 ........
-
-2008-05-30 13:01 +0000 [r119158-119240] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 119239 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r119239 | russell | 2008-05-30 07:59:11 -0500
- (Fri, 30 May 2008) | 23 lines Merged revisions 119238 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r119238 | russell | 2008-05-30 07:55:36 -0500
- (Fri, 30 May 2008) | 15 lines Merged revisions 119237 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r119237 | russell | 2008-05-30 07:49:39 -0500 (Fri, 30 May 2008)
- | 7 lines - Instead of only enforcing destination call number
- checking on an ACK, check all full frames except for PING and
- LAGRQ, which may be sent by older versions too quickly to contain
- the destination call number. (As suggested by Tim Panton on the
- asterisk-dev list) - Merge changes from
- team/russell/iax2-frame-race, which prevents PING and LAGRQ from
- being sent before the destination call number is known. ........
- ................ ................
-
- * main/autoservice.c, /: Merged revisions 119157 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r119157 | russell | 2008-05-29 17:28:50 -0500 (Thu, 29 May 2008)
- | 18 lines Merged revisions 119156 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r119156 | russell | 2008-05-29 17:24:29 -0500 (Thu, 29 May 2008)
- | 10 lines Fix a race condition in channel autoservice. There was
- still a small window of opportunity for a DTMF frame, or some
- other deferred frame type, to come in and get dropped. (closes
- issue #12656) (closes issue #12656) Reported by: dimas Patches:
- v3-12656.patch uploaded by dimas (license 88) -- with some
- modifications by me ........ ................
-
-2008-05-29 20:26 +0000 [r119073] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 119072 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r119072 | tilghman | 2008-05-29 15:25:33 -0500 (Thu, 29 May 2008)
- | 15 lines Merged revisions 119071 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r119071 | tilghman | 2008-05-29 15:24:11 -0500 (Thu, 29 May 2008)
- | 7 lines Call waiting tone occurs too often, because it's
- getting serviced by both subchannels. (closes issue #11354)
- Reported by: cahen Patches: 20080512__bug11354.diff.txt uploaded
- by Corydon76 (license 14) ........ ................
-
-2008-05-29 19:06 +0000 [r118960-119014] Russell Bryant <russell@digium.com>
-
- * apps/app_milliwatt.c, /: Merged revisions 119013 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r119013 | russell | 2008-05-29 14:05:33 -0500
- (Thu, 29 May 2008) | 12 lines Merged revisions 119012 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r119012 | russell | 2008-05-29 14:04:52 -0500 (Thu, 29 May 2008)
- | 4 lines - Fix a typo in the argument to Playtones - use
- ast_safe_sleep() instead of calling the wait application (thanks
- to tilghman for pointing these out!) ........ ................
-
- * /, channels/chan_iax2.c: Merged revisions 119010 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r119010 | russell | 2008-05-29 13:54:11 -0500
- (Thu, 29 May 2008) | 24 lines Merged revisions 119009 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r119009 | russell | 2008-05-29 13:49:12 -0500
- (Thu, 29 May 2008) | 16 lines Merged revisions 119008 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r119008 | russell | 2008-05-29 13:45:21 -0500 (Thu, 29 May 2008)
- | 7 lines Merge changes from
- team/russell/iax2-another-fix-to-the-fix As described in the
- following post to the asterisk-dev mailing list, only enforce
- destination call numbers when processing an ACK.
- http://lists.digium.com/pipermail/asterisk-dev/2008-May/033217.html
- (closes issue #12631) ........ ................ ................
-
- * apps/app_milliwatt.c, /: Merged revisions 118962 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r118962 | russell | 2008-05-29 12:52:00 -0500
- (Thu, 29 May 2008) | 11 lines Merged revisions 118961 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r118961 | russell | 2008-05-29 12:51:29 -0500 (Thu, 29 May 2008)
- | 3 lines - Mark app_milliwatt dependent on res_indications
- (thanks to jsmith) - fix a typo in a log message (thanks to
- qwell) ........ ................
-
- * apps/app_milliwatt.c, /: Merged revisions 118959 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r118959 | russell | 2008-05-29 12:46:04 -0500
- (Thu, 29 May 2008) | 11 lines Merged revisions 118956 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r118956 | russell | 2008-05-29 12:38:38 -0500 (Thu, 29 May 2008)
- | 3 lines Change milliwatt to use the proper tone by default
- (1004 Hz) instead of 1000 Hz. An option is there to use 1000 Hz
- for anyone that might want it. ........ ................
-
-2008-05-29 17:42 +0000 [r118958] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_mgcp.c, channels/chan_zap.c, /,
- channels/chan_agent.c, channels/chan_alsa.c, main/utils.c,
- include/asterisk/lock.h, channels/chan_iax2.c: Merged revisions
- 118955,118957 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r118955 | tilghman | 2008-05-29 12:35:19 -0500 (Thu, 29 May 2008)
- | 11 lines Merged revisions 118953 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r118953 | tilghman | 2008-05-29 12:20:16 -0500 (Thu, 29 May 2008)
- | 3 lines Add some debugging code that ensures that when we do
- deadlock avoidance, we don't lose the information about how a
- lock was originally acquired. ........ ................ r118957 |
- tilghman | 2008-05-29 12:39:50 -0500 (Thu, 29 May 2008) | 10
- lines Merged revisions 118954 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r118954 | tilghman | 2008-05-29 12:33:01 -0500 (Thu, 29 May 2008)
- | 2 lines Define also when not DEBUG_THREADS ........
- ................
-
-2008-05-29 04:11 +0000 [r118909] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, apps/app_forkcdr.c, /: Merged revisions 118880 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r118880 | murf | 2008-05-28 19:29:09 -0600 (Wed,
- 28 May 2008) | 54 lines Merged revisions 118858 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r118858 | murf | 2008-05-28 18:25:28 -0600 (Wed, 28 May 2008) |
- 46 lines (closes issue #10668) (closes issue #11721) (closes
- issue #12726) Reported by: arkadia Tested by: murf These changes:
- 1. revert the changes made via bug 10668; I should have known
- that such changes, even tho they made sense at the time, seemed
- like an omission, etc, were actually integral to the CDR system
- via forkCDR. It makes sense to me now that forkCDR didn't
- natively end any CDR's, but rather depended on natively closing
- them all at hangup time via traversing and closing them all,
- whether locked or not. I still don't completely understand the
- benefits of setvar and answer operating on locked cdrs, but I've
- seen enough to revert those changes also, and stop messing up
- users who depended on that behavior. bug 12726 found reverting
- the changes fixed his changes, and after a long review and
- working on forkCDR, I can see why. 2. Apply the suggested
- enhancements proposed in 10668, but in a completely compatible
- way. ForkCDR will behave exactly as before, but now has new
- options that will allow some actions to be taken that will
- slightly modify the outcome and side-effects of forkCDR. Based on
- conversations I've had with various people, these small tweaks
- will allow some users to get the behavior they need. For
- instance, users executing forkCDR in an AGI script will find the
- answer time set, and DISPOSITION set, a situation not covered
- when the routines were first written. 3. A small problem in the
- cdr serializer would output answer and end times even when they
- were not set. This is now fixed. ........ ................
-
-2008-05-28 18:07 +0000 [r118781] Michiel van Baak <michiel@vanbaak.info>
-
- * /, channels/chan_skinny.c: Merged revisions 118750 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r118750 | mvanbaak | 2008-05-28 19:58:21 +0200 (Wed, 28 May 2008)
- | 2 lines remove unused astobj.h header file from chan_skinny.c
- ........
-
-2008-05-28 14:31 +0000 [r118648] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Merged
- revisions 118647 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r118647 | file | 2008-05-28 11:29:01 -0300 (Wed, 28 May 2008) |
- 12 lines Merged revisions 118646 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r118646 | file | 2008-05-28 11:23:34 -0300 (Wed, 28 May 2008) | 4
- lines Add an option to use the source IP address of RTP as the
- destination IP address of UDPTL when a specific option is
- enabled. If the remote side is properly configured (ports
- forwarded) then UDPTL will flow. (closes issue #10417) Reported
- by: cstadlmann ........ ................
-
-2008-05-28 14:13 +0000 [r118615-118645] Philippe Sultan <philippe.sultan@gmail.com>
-
- * channels/chan_jingle.c, /, include/asterisk/jingle.h: Merged
- revisions 118644 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r118644 |
- phsultan | 2008-05-28 16:10:48 +0200 (Wed, 28 May 2008) | 10
- lines Changed to temporary namespaces to match with latest XEPs.
- As soon as Jingle is completely standardized, we can set those
- namespaces to their final values. Added two attributes to the
- jingle_pvt struct to store the content name attributes. Reported
- by Robert McQueen on Telepathy's framework mailing list :
- http://lists.freedesktop.org/archives/telepathy/2008-May/001971.html
- Keeping working on our Jingle stack! ........
-
- * channels/chan_jingle.c, /: Merged revisions 118614 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r118614 | phsultan | 2008-05-28 10:39:10 +0200 (Wed, 28 May 2008)
- | 1 line Code simplification ........
-
-2008-05-27 19:35 +0000 [r118561] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 118560 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r118560 | file | 2008-05-27 16:34:14 -0300 (Tue, 27 May 2008) |
- 12 lines Merged revisions 118558 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r118558 | file | 2008-05-27 16:32:38 -0300 (Tue, 27 May 2008) | 4
- lines Fix an issue where codec preferences were not set on
- dialogs that were not authenticated via a user or peer and allow
- framing to work without rtpmap in the SDP. (closes issue #12501)
- Reported by: slimey ........ ................
-
-2008-05-27 19:28 +0000 [r118557] Russell Bryant <russell@digium.com>
-
- * /, include/asterisk/compat.h: Merged revisions 118556 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r118556 | russell | 2008-05-27 14:27:48 -0500 (Tue, 27
- May 2008) | 6 lines Add printf format attribute for vasprintf().
- (closes issue #12729) Reported by: snuffy Patches: bug_12729.diff
- uploaded by snuffy (license 35) ........
-
-2008-05-27 19:22 +0000 [r118555] Tilghman Lesher <tlesher@digium.com>
-
- * main/cli.c, /: Merged revisions 118554 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r118554 | tilghman | 2008-05-27 14:21:03 -0500 (Tue, 27 May 2008)
- | 14 lines Merged revisions 118551 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r118551 | tilghman | 2008-05-27 14:15:27 -0500 (Tue, 27 May 2008)
- | 6 lines When showing an error message for a command, don't
- shorten the command output, as it tends to confuse the user (it's
- fine for suggesting other commands, however). Reported by:
- seanbright (on #asterisk-dev) Fixed by: me ........
- ................
-
-2008-05-27 19:09 +0000 [r118518] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_chanspy.c, /: Merged revisions 118514 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r118514 | mmichelson | 2008-05-27 14:08:24 -0500 (Tue, 27 May
- 2008) | 19 lines Merged revisions 118509 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r118509 | mmichelson | 2008-05-27 14:07:26 -0500 (Tue, 27 May
- 2008) | 11 lines Russell noted to me that in the case that
- separate threads use their own addressing system, the fix I made
- for issue 12376 does not guarantee uniqueness to the datastores'
- uids. Though I know of no system that works this way, I am going
- to change this right now to prevent trying to track down some
- future bug that may occur and cause untold hours of debugging
- time to track down. The change involves using a global counter
- which increases with each new chanspy_ds which is created. This
- guarantees uniqueness. ........ ................
-
-2008-05-27 18:59 +0000 [r118471] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c, /: Merged revisions 118466 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r118466 | tilghman | 2008-05-27 13:59:06 -0500 (Tue, 27 May 2008)
- | 16 lines Merged revisions 118465 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r118465 | tilghman | 2008-05-27 13:58:09 -0500 (Tue, 27 May 2008)
- | 8 lines NULL character should terminate only commands back to
- the core, not log messages to the console. (closes issue #12731)
- Reported by: seanbright Patches: 20080527__bug12731.diff.txt
- uploaded by Corydon76 (license 14) Tested by: seanbright ........
- ................
-
-2008-05-27 17:25 +0000 [r118418] Michiel van Baak <michiel@vanbaak.info>
-
- * apps/app_voicemail.c: small update to the g() option of
- app_voicemail to note that gain changes only work on zap channels
- right now. issue #12578 shows it's not clear right now.
-
-2008-05-27 16:48 +0000 [r118378-118382] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_chanspy.c, /: Merged revisions 118371 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r118371 | mmichelson | 2008-05-27 11:43:36 -0500 (Tue, 27 May
- 2008) | 22 lines Merged revisions 118365 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r118365 | mmichelson | 2008-05-27 11:38:38 -0500 (Tue, 27 May
- 2008) | 14 lines Add a unique id to the datastore allocated in
- app_chanspy since it is possible that multiple spies may be
- listening to the same channel. (closes issue #12376) Reported by:
- DougUDI Patches: 12376_chanspy_uid.diff uploaded by putnopvut
- (license 60) Tested by: destiny6628 (closes issue #12243)
- Reported by: atis ........ ................
-
- * /: Hmm, I apparently forgot to commit the block of revision
- 118175. Now I'm doing it.
-
-2008-05-27 15:47 +0000 [r118360] Tilghman Lesher <tlesher@digium.com>
-
- * /, configs/queues.conf.sample: Merged revisions 118359 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r118359 | tilghman | 2008-05-27 10:46:58 -0500
- (Tue, 27 May 2008) | 11 lines Merged revisions 118358 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r118358 | tilghman | 2008-05-27 10:45:37 -0500 (Tue, 27 May 2008)
- | 3 lines Add a note that pbx_config.so is needed for Local
- channels. (Closes issue #12671) ........ ................
-
-2008-05-27 14:51 +0000 [r118331] Russell Bryant <russell@digium.com>
-
- * /, include/asterisk/compat.h: Merged revisions 118328 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r118328 | russell | 2008-05-27 09:51:13 -0500 (Tue, 27
- May 2008) | 2 lines Add printf attribute to asprintf ........
-
-2008-05-27 13:30 +0000 [r118301-118303] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_config_ldap.c: Merged revisions 118302 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r118302 | tilghman | 2008-05-27 08:30:10 -0500 (Tue, 27 May 2008)
- | 6 lines When binding anonymously, credentials are still needed.
- (closes issue #12601) Reported by: suretec Patches:
- res_config_ldap.c.patch uploaded by suretec (license 70) ........
-
- * /, pbx/pbx_realtime.c: Merged revisions 118300 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r118300 |
- tilghman | 2008-05-27 08:13:17 -0500 (Tue, 27 May 2008) | 4 lines
- In compat14 mode, don't translate pipes inside expressions, as
- they aren't argument delimiters, but rather 'or' symbols. (Closes
- issue #12723) ........
-
-2008-05-25 16:20 +0000 [r118253] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 118252 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r118252 | tilghman | 2008-05-25 11:17:05 -0500 (Sun, 25 May 2008)
- | 20 lines Merged revisions 118251 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r118251 | tilghman | 2008-05-25 11:02:04 -0500 (Sun, 25 May 2008)
- | 12 lines Realtime flag affects construction in multiple ways,
- so consulting whether rtcachefriends was set was done too soon
- (needed to be done inside build_peer, not just as a flag to
- build_peer). Also, fullcontact needed to be reconstructed,
- because realtime separates the embedded ';' into multiple fields.
- (closes issue #12722) Reported by: barthpbx Patches:
- 20080525__bug12722.diff.txt uploaded by Corydon76 (license 14)
- Tested by: barthpbx (Much of the discussion happened on
- #asterisk-dev for diagnosing this issue) ........
- ................
-
-2008-05-24 01:15 +0000 [r118177-118179] Jeff Peeler <jpeeler@digium.com>
-
- * doc/api-1.6.0-changes.odt (added), /: Merged revisions 118178 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r118178 | jpeeler | 2008-05-23 20:14:41 -0500 (Fri, 23
- May 2008) | 1 line add document describing API changes from 1.4.0
- to 1.6.0 ........
-
-2008-05-23 21:37 +0000 [r118168] Brett Bryant <bbryant@digium.com>
-
- * main/manager.c, /, main/http.c, include/asterisk/manager.h:
- Merged revisions 118161 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r118161 |
- bbryant | 2008-05-23 16:19:42 -0500 (Fri, 23 May 2008) | 3 lines
- Add new functionality to http server that requires manager
- authentication for any path that includes a directory named
- 'private'. This patch also requires manager authentication for
- any POST's being sent to the server as well to help secure
- uploads. ........
-
-2008-05-23 21:31 +0000 [r118165] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_zap.c: Merged revisions 118164 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r118164 | jpeeler | 2008-05-23 16:26:39 -0500 (Fri, 23 May 2008)
- | 9 lines Merged revisions 118163 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r118163 | jpeeler | 2008-05-23 16:21:35 -0500 (Fri, 23 May 2008)
- | 1 line Fix a few things I missed to ensure zt_chan_conf
- structure is not modified in mkintf ........ ................
-
-2008-05-23 18:15 +0000 [r118130] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_odbc.c, /: Merged revisions 118129 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r118129 |
- tilghman | 2008-05-23 13:09:14 -0500 (Fri, 23 May 2008) | 3 lines
- Protect the object from changing while the 'odbc show' CLI
- command is running (Closes issue #12704) ........
-
-2008-05-23 13:00 +0000 [r118054] Tilghman Lesher <tlesher@digium.com>
-
- * doc/cli.txt (added), /: Merged revisions 118053 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r118053 | tilghman | 2008-05-23 08:00:10 -0500 (Fri, 23 May 2008)
- | 11 lines Merged revisions 118052 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r118052 | tilghman | 2008-05-23 07:59:16 -0500 (Fri, 23 May 2008)
- | 3 lines Add information on using the Asterisk console,
- including tab command line completion. (Closes issue #12681)
- ........ ................
-
-2008-05-23 12:37 +0000 [r118050] Russell Bryant <russell@digium.com>
-
- * include/asterisk/utils.h, /, main/utils.c: Merged revisions
- 118049 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r118049 | russell | 2008-05-23 07:37:31 -0500 (Fri, 23 May 2008)
- | 17 lines Merged revisions 118048 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r118048 | russell | 2008-05-23 07:30:53 -0500 (Fri, 23 May 2008)
- | 9 lines Don't declare a function that takes variable arguments
- as inline, because it's not valid, and on some compilers, will
- emit a warning.
- http://gcc.gnu.org/onlinedocs/gcc/Inline.html#Inline (closes
- issue #12289) Reported by: francesco_r Patches by Tilghman, final
- patch by me ........ ................
-
-2008-05-23 11:02 +0000 [r118021] Philippe Sultan <philippe.sultan@gmail.com>
-
- * /, channels/chan_gtalk.c, res/res_jabber.c: Merged revisions
- 118020 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r118020 |
- phsultan | 2008-05-23 12:33:21 +0200 (Fri, 23 May 2008) | 15
- lines - remove whitespaces between tags in received XML packets
- before giving them to the parser ; - report Gtalk error messages
- from a buddy to the console. This patch makes Asterisk "Google
- Jingle" (chan_gtalk) implementation work with Empathy. Note that
- this is only true for audio streams, not video. Thank you to PH
- for his great help! (closes issue #12647) Reported by: PH
- Patches: trunk-12647-1.diff uploaded by phsultan (license 73)
- Tested by: phsultan, PH ........
-
-2008-05-22 21:43 +0000 [r117984-117987] Tilghman Lesher <tlesher@digium.com>
-
- * /, pbx/pbx_realtime.c, configs/pbx_realtime.conf (added): Merged
- revisions 117986 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r117986 |
- tilghman | 2008-05-22 16:42:50 -0500 (Thu, 22 May 2008) | 2 lines
- Add a compatibility option for upgrading realtime extensions
- ........
-
-2008-05-22 18:55 +0000 [r117901] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c, /: Merged revisions 117900 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r117900 | tilghman | 2008-05-22 13:54:41 -0500 (Thu, 22 May 2008)
- | 10 lines Merged revisions 117899 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r117899 | tilghman | 2008-05-22 13:53:53 -0500 (Thu, 22 May 2008)
- | 2 lines Also remove preamble from asynchronous events (reported
- by jsmith on #asterisk-dev) ........ ................
-
-2008-05-22 15:51 +0000 [r117793] Sean Bright <sean.bright@gmail.com>
-
- * /, configs/jabber.conf.sample: Merged revisions 117792 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r117792 | seanbright | 2008-05-22 11:49:17 -0400 (Thu,
- 22 May 2008) | 1 line Minor text fix. roster -> resource.
- ........
-
-2008-05-22 13:41 +0000 [r117757] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c, /, build_tools/make_buildopts_h: Merged
- revisions 117756 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r117756 |
- russell | 2008-05-22 08:40:52 -0500 (Thu, 22 May 2008) | 5 lines
- Store build-time options as a string in AST_BUILDOPTS in
- buildopts.h. Also, display this information in the "core show
- settings" CLI command. This is useful if you want to verify that
- you're running a build with DONT_OPTIMIZE, DEBUG_THREADS, etc.
- ........
-
-2008-05-21 22:01 +0000 [r117659-117660] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_zap.c: Merged revisions 117658 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r117658 | jpeeler | 2008-05-21 16:31:17 -0500 (Wed, 21 May 2008)
- | 10 lines Merged revisions 117582 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r117582 | jpeeler | 2008-05-21 15:11:14 -0500 (Wed, 21 May 2008)
- | 2 lines Ensure that passed in zt_chan_conf structure is not
- modified in mkintf. ........ ................
-
- * channels/chan_zap.c, /: Merged revisions 117628 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r117628 | jpeeler | 2008-05-21 15:44:04 -0500 (Wed, 21 May 2008)
- | 12 lines Merged revisions 117462 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r117462 | jpeeler | 2008-05-21 11:58:40 -0500 (Wed, 21 May 2008)
- | 3 lines Pass a pointer for the conf parameter to the function
- mkintf rather than the whole zt_chan_conf structure. Another
- commit is following to make sure the zt_chan_conf structure is
- not modified. ........ ................
-
-2008-05-21 19:45 +0000 [r117576] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 117575 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r117575 | file | 2008-05-21 16:39:42 -0300 (Wed, 21 May 2008) |
- 10 lines Merged revisions 117574 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r117574 | file | 2008-05-21 16:38:28 -0300 (Wed, 21 May 2008) | 2
- lines Apply the autoframing setting to dialogs that do not get
- matched against a user or peer. ........ ................
-
-2008-05-21 18:44 +0000 [r117522] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c, /: Merged revisions 117520 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r117520 | tilghman | 2008-05-21 13:43:26 -0500 (Wed, 21 May 2008)
- | 11 lines Merged revisions 117519 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r117519 | tilghman | 2008-05-21 13:40:14 -0500 (Wed, 21 May 2008)
- | 3 lines Strip the preamble from the output also when -rx is not
- being used (Related to issue #12702) ........ ................
-
-2008-05-21 18:29 +0000 [r117486-117516] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c, /: Merged revisions 117515 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r117515 | russell | 2008-05-21 13:29:05 -0500 (Wed, 21 May 2008)
- | 12 lines Merged revisions 117514 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r117514 | russell | 2008-05-21 13:28:46 -0500 (Wed, 21 May 2008)
- | 4 lines Don't filter the magic character in the network
- verboser. It gets filtered once it reaches the client. (related
- to issue #12702, pointed out by tilghman) ........
- ................
-
- * main/asterisk.c, pbx/pbx_gtkconsole.c, /: Merged revisions 117508
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r117508 | russell | 2008-05-21 13:20:11 -0500
- (Wed, 21 May 2008) | 15 lines Merged revisions 117507 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r117507 | russell | 2008-05-21 13:19:34 -0500 (Wed, 21 May 2008)
- | 7 lines 1) Don't print the verbose marker in front of every
- message from ast_verbose() being sent to remote consoles. 2) Fix
- pbx_gtkconsole to filter out the verbose marker. (related to
- issue #12702) ........ ................
-
- * main/asterisk.c, /: Merged revisions 117481 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r117481 | russell | 2008-05-21 13:12:19 -0500 (Wed, 21 May 2008)
- | 14 lines Merged revisions 117479 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r117479 | russell | 2008-05-21 13:11:51 -0500 (Wed, 21 May 2008)
- | 6 lines Don't display the verbose marker for calls to
- ast_verbose() that do not include a VERBOSE_PREFIX in front of
- the message. (closes issue #12702) Reported by: johnlange Patched
- by me ........ ................
-
-2008-05-21 02:21 +0000 [r117368] Mark Michelson <mmichelson@digium.com>
-
- * main/config.c, /: Merged revisions 117367 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r117367 |
- mmichelson | 2008-05-20 21:20:31 -0500 (Tue, 20 May 2008) | 19
- lines Be sure that we cache included files for each source file
- which loads a configuration file. As it was, only the first did
- so. This led to a problem if the included file was changed (but
- not the configuration file which includes it) and the second
- source file attempted to reload the configuration. It would not
- see that the included file had changed. In this particular
- example, res_phoneprov and chan_sip both loaded sip.conf, which
- included a file call sip.peers.conf. Since res_phoneprov was the
- first to load sip.conf, only it cached the fact that sip.conf
- included sip.peers.conf. If sip.peers.conf were changed and
- sip.conf were not and a sip reload were issued (meaning that
- chan_sip attempts to reload sip.conf only if it and its included
- files have changed) the changes made to sip.peers.conf would not
- be seen and therefore no action would be taken. (closes issue
- #12693) Reported by: marsosa ........
-
-2008-05-21 01:20 +0000 [r117365] Steve Murphy <murf@digium.com>
-
- * /, utils/ael_main.c: Merged revisions 117335 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r117335 |
- murf | 2008-05-20 19:00:28 -0600 (Tue, 20 May 2008) | 10 lines
- These changes were made via the comments atis_work made at 4:30am
- (Mountain Time zone- US) in #asterisk-dev on 20 May 2008. He
- noted that a backslash was being inserted before commas in app
- call arguments in the extensions.conf.aeldump file that you get
- from aelparse with the -w arg. This was being generated from code
- left over from 1.4, where commas were substituted with '|', and
- any remaining commas needed to be escaped. Many thanks to atis
- for his comment; please let us know if these changes break
- anything! ........
-
-2008-05-19 16:58 +0000 [r117134-117137] Joshua Colp <jcolp@digium.com>
-
- * res/res_smdi.c, /: Merged revisions 117136 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r117136 | file | 2008-05-19 13:53:33 -0300 (Mon, 19 May 2008) |
- 14 lines Merged revisions 117135 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r117135 | file | 2008-05-19 13:50:52 -0300 (Mon, 19 May 2008) | 6
- lines Use the right pthread lock and condition when waiting.
- (closes issue #12664) Reported by: tomo1657 Patches:
- res_smdi.c.patch uploaded by tomo1657 (license 484) ........
- ................
-
-2008-05-19 16:07 +0000 [r117089] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/utils.h, /: Merged revisions 117088 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r117088 | tilghman | 2008-05-19 11:07:09 -0500
- (Mon, 19 May 2008) | 10 lines Merged revisions 117086 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r117086 | tilghman | 2008-05-19 11:05:05 -0500 (Mon, 19 May 2008)
- | 2 lines The addition of usleep(2) within ast_assert requires
- the inclusion of the unistd.h header ........ ................
-
-2008-05-19 16:05 +0000 [r117083-117087] Joshua Colp <jcolp@digium.com>
-
- * /, main/logger.c: Merged revisions 117085 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r117085 |
- file | 2008-05-19 13:03:33 -0300 (Mon, 19 May 2008) | 4 lines The
- logger closes the files it is logging to when reloading so we
- have to read in the logger configuration even if it has not
- changed so that the logs get opened again. (closes issue #12665)
- Reported by: DennisD ........
-
- * /, channels/h323/ast_h323.cxx: Merged revisions 117082 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r117082 | file | 2008-05-19 12:24:44 -0300 (Mon,
- 19 May 2008) | 14 lines Merged revisions 117081 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r117081 | file | 2008-05-19 12:22:10 -0300 (Mon, 19 May 2008) | 6
- lines Make chan_h323 work with pwlib 1.12.0 (closes issue #12682)
- Reported by: bamby Patches: pwlib_nopipe.diff uploaded by bamby
- (license 430) ........ ................
-
-2008-05-19 03:44 +0000 [r116980] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 116979 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r116979 | russell | 2008-05-18 22:44:28 -0500
- (Sun, 18 May 2008) | 12 lines Merged revisions 116978 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r116978 | russell | 2008-05-18 22:44:04 -0500 (Sun, 18 May 2008)
- | 4 lines Avoid access of uninitialized memory. This caused a
- bunch of crashes for me while doing load testing of development
- branch where I'm working on some performance improvements.
- ........ ................
-
-2008-05-18 21:18 +0000 [r116949] Tilghman Lesher <tlesher@digium.com>
-
- * /, utils/astcanary.c: Merged revisions 116948 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r116948 |
- tilghman | 2008-05-18 16:15:58 -0500 (Sun, 18 May 2008) | 4 lines
- Add a set of text to the file astcanary uses to communicate back
- the main Asterisk process, which explains the purpose for the
- file being there. This should assist people who find the file and
- wonder why it exists. ........
-
-2008-05-18 19:59 +0000 [r116922] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 116919 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r116919 |
- russell | 2008-05-18 14:58:10 -0500 (Sun, 18 May 2008) | 3 lines
- Remove duplicate colon on Reason header (closes issue #12678)
- ........
-
-2008-05-17 19:40 +0000 [r116849-116885] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_skinny.c: Merged revisions 116800 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r116800 | file | 2008-05-16 17:30:24 -0300 (Fri,
- 16 May 2008) | 12 lines 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 ........ ................
-
-2008-05-16 20:03 +0000 [r116798] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 116797 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r116797 |
- mattf | 2008-05-16 15:00:04 -0500 (Fri, 16 May 2008) | 1 line Try
- to see if we can make our ringback situation a little better
- ........
-
-2008-05-15 22:07 +0000 [r116636-116695] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/utils.h, /, include/asterisk/strings.h: Merged
- revisions 116694 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r116694 |
- tilghman | 2008-05-15 17:05:47 -0500 (Thu, 15 May 2008) | 4 lines
- Add an extra check in ast_strlen_zero, and make ast_assert() not
- print the file, line, and function name twice. (Closes issue
- #12650) ........
-
- * cdr/cdr_csv.c, /: Merged revisions 116631 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r116631 |
- tilghman | 2008-05-15 12:58:22 -0500 (Thu, 15 May 2008) | 3 lines
- Don't unload config on reload, when config has not changed.
- (Closes issue #12652) ........
-
-2008-05-14 21:41 +0000 [r116470] Russell Bryant <russell@digium.com>
-
- * main/rtp.c, main/sched.c, main/channel.c, main/udptl.c,
- include/asterisk/utils.h, /, channels/chan_agent.c,
- main/abstract_jb.c, include/asterisk/channel.h: Merged revisions
- 116469 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r116469 | russell | 2008-05-14 16:40:43 -0500 (Wed, 14 May 2008)
- | 12 lines Merged revisions 116463 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r116463 | russell | 2008-05-14 16:32:00 -0500 (Wed, 14 May 2008)
- | 4 lines Add ast_assert(), which can be used to handle fatal
- errors. It is only compiled in if dev-mode is enabled, and only
- aborts if DO_CRASH is defined. (inspired by issue #12650)
- ........ ................
-
-2008-05-14 21:39 +0000 [r116468] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_agi.c: Merged revisions 116467 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r116467 | tilghman | 2008-05-14 16:39:06 -0500 (Wed, 14 May 2008)
- | 15 lines Merged revisions 116466 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r116466 | tilghman | 2008-05-14 16:38:09 -0500 (Wed, 14 May 2008)
- | 7 lines Avoid zombies when the channel exits before the AGI.
- (closes issue #12648) Reported by: gkloepfer Patches:
- 20080514__bug12648.diff.txt uploaded by Corydon76 (license 14)
- Tested by: gkloepfer ........ ................
-
-2008-05-14 20:43 +0000 [r116408-116411] Jason Parker <jparker@digium.com>
-
- * /, configs/voicemail.conf.sample: Merged revisions 116410 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r116410 | qwell | 2008-05-14 15:43:26 -0500
- (Wed, 14 May 2008) | 9 lines Merged revisions 116409 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r116409 | qwell | 2008-05-14 15:43:08 -0500 (Wed, 14 May
- 2008) | 1 line Document exitcontext in app_voicemail sample
- config ........ ................
-
- * apps/app_voicemail.c, /: Merged revisions 116407 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r116407 | qwell | 2008-05-14 15:36:55 -0500 (Wed, 14 May 2008) |
- 9 lines Voicemail "* exit" should not require an exitcontext to
- be specified. The behavior in 1.4 was that it would use the
- current context if an exitcontext existed. (closes issue #12605)
- Reported by: kenjreno Patches: 12605-starexit.diff uploaded by
- qwell (license 4) Tested by: file ........
-
-2008-05-14 18:54 +0000 [r116351-116354] Joshua Colp <jcolp@digium.com>
-
- * /, main/Makefile: Merged revisions 116353 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r116353 | file | 2008-05-14 15:54:16 -0300 (Wed, 14 May 2008) |
- 12 lines Merged revisions 116352 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r116352 | file | 2008-05-14 15:53:39 -0300 (Wed, 14 May 2008) | 4
- lines Add linux-gnueabi in. (closes issue #12529) Reported by:
- tzafrir ........ ................
-
- * /, res/res_config_ldap.c: Merged revisions 116350 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r116350 | file | 2008-05-14 15:25:54 -0300 (Wed, 14 May 2008) | 4
- lines Make the ldap version setting work without having both
- version and protocol set. (closes issue #12613) Reported by:
- suretec ........
-
-2008-05-14 17:01 +0000 [r116319] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_externalivr.c: Merged revisions 116298 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r116298 | tilghman | 2008-05-14 11:53:23 -0500
- (Wed, 14 May 2008) | 15 lines Merged revisions 116296 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r116296 | tilghman | 2008-05-14 11:46:48 -0500 (Wed, 14 May 2008)
- | 2 lines Detect another way for a connection to have gone away.
- (closes issue #12618) Reported by: ctooley Patches:
- 1.4-externalivr-test_fd.diff uploaded by ctooley (license 136)
- trunk-externalivr-test_fd.diff uploaded by ctooley (license 136)
- ........ ................
-
-2008-05-14 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.6.0-beta9 released.
-
-2008-05-14 13:13 +0000 [r116236] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 116234 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r116234 | oej | 2008-05-14 15:05:15 +0200 (Ons, 14 Maj 2008) | 11
- lines Merged revisions 116230 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r116230 | oej | 2008-05-14 14:51:06 +0200 (Ons, 14 Maj 2008) | 3
- lines Accept text messages even with Content-Type:
- text/plain;charset=Södermanländska ........ ................
-
-2008-05-14 00:20 +0000 [r116096-116139] Mark Michelson <mmichelson@digium.com>
-
- * main/channel.c, /, include/asterisk/lock.h: Merged revisions
- 116089 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r116089 | mmichelson | 2008-05-13 18:54:01 -0500 (Tue, 13 May
- 2008) | 20 lines Merged revisions 116088 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r116088 | mmichelson | 2008-05-13 18:47:49 -0500 (Tue, 13 May
- 2008) | 12 lines A change to the way channel locks are handled
- when DEBUG_CHANNEL_LOCKS is defined. After debugging a deadlock,
- it was noticed that when DEBUG_CHANNEL_LOCKS is enabled in
- menuselect, the actual origin of channel locks is obscured by the
- fact that all channel locks appear to happen in the function
- ast_channel_lock(). This code change redefines ast_channel_lock
- to be a macro which maps to __ast_channel_lock(), which then
- relays the proper file name, line number, and function name
- information to the core lock functions so that this information
- will be displayed in the case that there is some sort of locking
- error or core show locks is issued. ........ ................
-
-2008-05-13 21:19 +0000 [r116020-116040] Russell Bryant <russell@digium.com>
-
- * channels/chan_local.c, /: Merged revisions 116039 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r116039 | russell | 2008-05-13 16:18:55 -0500
- (Tue, 13 May 2008) | 32 lines Merged revisions 116038 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r116038 | russell | 2008-05-13 16:17:23 -0500 (Tue, 13 May 2008)
- | 24 lines Fix a deadlock involving channel autoservice and
- chan_local that was debugged and fixed by mmichelson and me. We
- observed a system that had a bunch of threads stuck in
- ast_autoservice_stop(). The reason these threads were waiting
- around is because this function waits to ensure that the channel
- list in the autoservice thread gets rebuilt before the stop()
- function returns. However, the autoservice thread was also
- locked, so the autoservice channel list was never getting
- rebuilt. The autoservice thread was stuck waiting for the channel
- lock on a local channel. However, the local channel was locked by
- a thread that was stuck in the autoservice stop function. It
- turned out that the issue came down to the local_queue_frame()
- function in chan_local. This function assumed that one of the
- channels passed in as an argument was locked when called.
- However, that was not always the case. There were multiple cases
- in which this channel was not locked when the function was
- called. We fixed up chan_local to indicate to this function
- whether this channel was locked or not. The previous assumption
- had caused local_queue_frame() to improperly return with the
- channel locked, where it would then never get unlocked. (closes
- issue #12584) (related to issue #12603) ........ ................
-
- * main/autoservice.c, /: Merged revisions 116001 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r116001 | russell | 2008-05-13 16:07:59 -0500 (Tue, 13 May 2008)
- | 13 lines Merged revisions 115990 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115990 | russell | 2008-05-13 16:05:57 -0500 (Tue, 13 May 2008)
- | 5 lines Fix an issue that I noticed in autoservice while
- mmichelson and I were debugging a different problem. I noticed
- that it was theoretically possible for two threads to attempt to
- start the autoservice thread at the same time. This change makes
- the process of starting the autoservice thread, thread-safe.
- ........ ................
-
-2008-05-13 20:30 +0000 [r115946] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_alsa.c: Merged revisions 115945 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r115945 | file | 2008-05-13 17:29:27 -0300 (Tue,
- 13 May 2008) | 12 lines Merged revisions 115944 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115944 | file | 2008-05-13 17:28:23 -0300 (Tue, 13 May 2008) | 4
- lines Use the right flag to open the audio in non-blocking.
- (closes issue #12616) Reported by: nicklewisdigiumuser ........
- ................
-
-2008-05-13 20:19 +0000 [r115940-115942] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 115941 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r115941 |
- mattf | 2008-05-13 15:18:04 -0500 (Tue, 13 May 2008) | 1 line
- Need to clear calling_party_cat variable after we retrieve it
- ........
-
- * channels/chan_zap.c, /: Merged revisions 115939 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r115939 |
- mattf | 2008-05-13 15:11:20 -0500 (Tue, 13 May 2008) | 1 line Add
- support for receiving calling party category ........
-
-2008-05-13 18:38 +0000 [r115887] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c, /: Merged revisions 115886 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r115886 | tilghman | 2008-05-13 13:38:11 -0500 (Tue, 13 May 2008)
- | 11 lines Merged revisions 115884 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115884 | tilghman | 2008-05-13 13:36:13 -0500 (Tue, 13 May 2008)
- | 3 lines If the socket dies (read returns 0=EOF), return
- immediately. (Closes issue #12637) ........ ................
-
-2008-05-13 17:48 +0000 [r115848-115851] Russell Bryant <russell@digium.com>
-
- * res/res_smdi.c, /: Merged revisions 115847 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r115847 |
- russell | 2008-05-13 12:14:22 -0500 (Tue, 13 May 2008) | 2 lines
- Initialize the start time in smdi_msg_wait. Somehow this code got
- lost in trunk. ........
-
-2008-05-12 17:57 +0000 [r115738] Mark Michelson <mmichelson@digium.com>
-
- * main/utils.c: Merged revisions 115737 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r115737 | mmichelson | 2008-05-12 12:55:08 -0500 (Mon, 12 May
- 2008) | 15 lines Merged revisions 115735 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115735 | mmichelson | 2008-05-12 12:51:14 -0500 (Mon, 12 May
- 2008) | 7 lines If a thread holds no locks, do not print any
- information on the thread when issuing a core show locks command.
- This will help to de-clutter output somewhat. Russell said it
- would be fine to place this improvement in the 1.4 branch, so
- that's why it's going here too. ........ ................
-
-2008-05-12 16:36 +0000 [r115706] Jason Parker <jparker@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 115705 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r115705 |
- qwell | 2008-05-12 11:35:50 -0500 (Mon, 12 May 2008) | 1 line
- Correctly document state interface for AddQueueMember. Discovered
- while looking at issue #12626. ........
-
-2008-05-12 15:18 +0000 [r115672] Brett Bryant <bbryant@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 115669 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r115669 | bbryant | 2008-05-12 10:17:32 -0500 (Mon, 12 May 2008)
- | 3 lines A small change to fix iax2 native bridging. ........
-
-2008-05-11 03:27 +0000 [r115599-115601] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c, /, configure,
- include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
- 115600 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r115600 |
- mattf | 2008-05-10 22:23:05 -0500 (Sat, 10 May 2008) | 1 line Add
- Zap MTP2 support to chan_zap ........
-
- * channels/chan_zap.c, /: Merged revisions 115598 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r115598 |
- mattf | 2008-05-10 21:19:21 -0500 (Sat, 10 May 2008) | 1 line
- Open up audio channel when we get ACM on SS7 event ........
-
-2008-05-10 14:22 +0000 [r115597] Tilghman Lesher <tlesher@digium.com>
-
- * /, cdr/cdr_pgsql.c: Merged revisions 115596 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r115596 |
- tilghman | 2008-05-10 09:19:41 -0500 (Sat, 10 May 2008) | 2 lines
- Ensure that "calldate" is acceptable for a column name. ........
-
-2008-05-09 16:38 +0000 [r115581] Joshua Colp <jcolp@digium.com>
-
- * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
- Merged revisions 115580 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r115580 | file | 2008-05-09 13:36:58 -0300 (Fri, 09 May 2008) |
- 10 lines Merged revisions 115579 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115579 | file | 2008-05-09 13:34:08 -0300 (Fri, 09 May 2008) | 2
- lines Improve res_ninit and res_ndestroy autoconf logic on the
- Darwin platform. ........ ................
-
-2008-05-08 19:21 +0000 [r115553-115570] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 115569 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r115569 | russell | 2008-05-08 14:20:35 -0500
- (Thu, 08 May 2008) | 10 lines Merged revisions 115568 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115568 | russell | 2008-05-08 14:19:50 -0500 (Thu, 08 May 2008)
- | 2 lines Remove debug output. ........ ................
-
- * /, channels/chan_iax2.c: Merged revisions 115566 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r115566 | russell | 2008-05-08 14:17:04 -0500
- (Thu, 08 May 2008) | 41 lines Merged revisions 115565 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r115565 | russell | 2008-05-08 14:15:25 -0500
- (Thu, 08 May 2008) | 33 lines Merged revisions 115564 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r115564 | russell | 2008-05-08 14:14:04 -0500 (Thu, 08 May 2008)
- | 25 lines Fix a race condition that bbryant just found while
- doing some IAX2 testing. He was running Asterisk trunk running
- IAX2 calls through a few Asterisk boxes, however, the audio was
- extremely choppy. We looked at a packet trace and saw a storm of
- INVAL and VNAK frames being sent from one box to another. It
- turned out that what had happened was that one box tried to send
- a CONTROL frame before the 3 way handshake had completed. So,
- that frame did not include the destination call number, because
- it didn't have it yet. Part of our recent work for security
- issues included an additional check to ensure that frames that
- are supposed to include the destination call number have the
- correct one. This caused the frame to be rejected with an INVAL.
- The frame would get retransmitted for forever, rejected every
- time ... This race condition exists in all versions that got the
- security changes, in theory. However, it is really only likely
- that this would cause a problem in Asterisk trunk. There was a
- control frame being sent (SRCUPDATE) at the _very_ beginning of
- the call, which does not exist in 1.2 or 1.4. However, I am
- fixing all versions that could potentially be affected by the
- introduced race condition. These changes are what bbryant and I
- came up with to fix the issue. Instead of simply dropping control
- frames that get sent before the handshake is complete, the code
- attempts to wait a little while, since in most cases, the
- handshake will complete very quickly. If it doesn't complete
- after yielding for a little while, then the frame gets dropped.
- ........ ................ ................
-
- * /, channels/chan_sip.c: Merged revisions 115562 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r115562 | russell | 2008-05-08 11:14:08 -0500 (Thu, 08 May 2008)
- | 11 lines Merged revisions 115561 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115561 | russell | 2008-05-08 11:11:33 -0500 (Thu, 08 May 2008)
- | 3 lines Don't give up on attempting an outbound registration if
- we receive a 408 Timeout. (closes issue #12323) ........
- ................
-
- * /, contrib/scripts/postgres_cdr.sql (removed): Merged revisions
- 115558 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r115558 | russell | 2008-05-08 10:38:27 -0500 (Thu, 08 May 2008)
- | 11 lines Merged revisions 115557 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115557 | russell | 2008-05-08 10:37:49 -0500 (Thu, 08 May 2008)
- | 3 lines remove postgres_cdr.sql, as the CDR schema is in
- realtime_pgsql.sql, as well (closes issue #9676) ........
- ................
-
- * contrib/init.d/rc.debian.asterisk, /: Merged revisions 115555 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r115555 | russell | 2008-05-08 10:32:48 -0500
- (Thu, 08 May 2008) | 11 lines Merged revisions 115554 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115554 | russell | 2008-05-08 10:32:08 -0500 (Thu, 08 May 2008)
- | 3 lines Don't exit the script if Asterisk is not running.
- (closes issue #12611) ........ ................
-
- * main/pbx.c, /: Merged revisions 115552 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r115552 | russell | 2008-05-08 10:26:49 -0500 (Thu, 08 May 2008)
- | 12 lines Merged revisions 115551 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115551 | russell | 2008-05-08 10:24:54 -0500 (Thu, 08 May 2008)
- | 4 lines Don't use a channel before checking for channel
- allocation failure. (closes issue #12609) Reported by: edantie
- ........ ................
-
-2008-05-08 15:08 +0000 [r115549] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 115548 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r115548 |
- mattf | 2008-05-08 10:04:45 -0500 (Thu, 08 May 2008) | 1 line
- Remove unused code as well as demote an error message to a debug
- message ........
-
-2008-05-08 14:41 +0000 [r115538-115547] Russell Bryant <russell@digium.com>
-
- * contrib/init.d/rc.debian.asterisk, /: Merged revisions 115546 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r115546 | russell | 2008-05-08 09:41:12 -0500
- (Thu, 08 May 2008) | 12 lines Merged revisions 115545 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115545 | russell | 2008-05-08 09:40:53 -0500 (Thu, 08 May 2008)
- | 4 lines Use the same method for executing Asterisk as the rest
- of the script. (closes issue #12611) Reported by: b_plessis
- ........ ................
-
-2008-05-07 18:35 +0000 [r115514-115524] Russell Bryant <russell@digium.com>
-
- * /, res/res_config_ldap.c: Merged revisions 115523 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r115523 | russell | 2008-05-07 13:33:50 -0500 (Wed, 07 May 2008)
- | 6 lines Only save a password if a username exists. (closes
- issue #12600) Reported By: suretec Patch by me ........
-
- * /, res/res_config_ldap.c: Merged revisions 115521 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r115521 | russell | 2008-05-07 13:30:12 -0500 (Wed, 07 May 2008)
- | 7 lines Use the default that the log output claims will be used
- for the basedn (closes issue #12599) Reported by: suretec
- Patches: 12599.patch uploaded by juggie (license 24) ........
-
- * /, channels/chan_h323.c: Merged revisions 115519 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r115519 | russell | 2008-05-07 13:24:51 -0500 (Wed, 07 May 2008)
- | 2 lines Let chan_h323 build in dev mode ........
-
- * /, include/asterisk/dlinkedlists.h (removed),
- channels/chan_iax2.c: Merged revisions 115513 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r115513 | russell | 2008-05-07 12:28:19 -0500 (Wed, 07 May 2008)
- | 19 lines Merged revisions 115512 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r115512 | russell | 2008-05-07 11:24:09 -0500
- (Wed, 07 May 2008) | 11 lines Merged revisions 115511 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r115511 | russell | 2008-05-07 11:22:49 -0500 (Wed, 07 May 2008)
- | 3 lines Remove remnants of dlinkedlists. I didn't actually use
- them in the final version of my IAX2 improvements. ........
- ................ ................
-
-2008-05-07 13:49 +0000 [r115510] Tilghman Lesher <tlesher@digium.com>
-
- * contrib/scripts/asterisk.ldap-schema,
- contrib/scripts/asterisk.ldif, /: Merged revisions 115509 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r115509 | tilghman | 2008-05-07 08:49:15 -0500 (Wed, 07
- May 2008) | 2 lines Update typos in description fields (closes
- issue #12598) Reported by: suretec Patches:
- asterisk_schema_changes.patch uploaded by suretec (license 70)
- ........
-
-2008-05-06 19:56 +0000 [r115420-115424] Jason Parker <jparker@digium.com>
-
- * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 115423
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r115423 | qwell | 2008-05-06 14:55:45 -0500
- (Tue, 06 May 2008) | 23 lines Merged revisions 115422 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r115422 | qwell | 2008-05-06 14:55:29 -0500
- (Tue, 06 May 2008) | 15 lines Merged revisions 115421 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r115421 | qwell | 2008-05-06 14:54:57 -0500 (Tue, 06 May 2008) |
- 7 lines read requires an argument on some non-bash shells (closes
- issue #12593) Reported by: bkruse Patches:
- getilbc.sh_12593_v1.diff uploaded by bkruse (license 132)
- ........ ................ ................
-
- * /, res/res_musiconhold.c: Merged revisions 115419 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r115419 | qwell | 2008-05-06 14:38:44 -0500
- (Tue, 06 May 2008) | 15 lines Merged revisions 115418 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115418 | qwell | 2008-05-06 14:34:58 -0500 (Tue, 06 May 2008) |
- 7 lines Switch to using ast_random() rather than just rand().
- This does not fix the bug reported, but I believe it is correct.
- (from issue #12446) Patches: bug_12446.diff uploaded by snuffy
- (license 35) ........ ................
-
-2008-05-06 19:33 +0000 [r115417] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c, /: Merged revisions 115416 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r115416 | tilghman | 2008-05-06 14:32:29 -0500 (Tue, 06 May 2008)
- | 10 lines Merged revisions 115415 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115415 | tilghman | 2008-05-06 14:31:39 -0500 (Tue, 06 May 2008)
- | 2 lines Don't print the terminating NUL. (Closes issue #12589)
- ........ ................
-
-2008-05-06 13:57 +0000 [r115343] Joshua Colp <jcolp@digium.com>
-
- * /, configure, configure.ac: Merged revisions 115342 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r115342 | file | 2008-05-06 10:55:44 -0300 (Tue,
- 06 May 2008) | 10 lines Merged revisions 115341 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115341 | file | 2008-05-06 10:54:15 -0300 (Tue, 06 May 2008) | 2
- lines Add in missing argument. ........ ................
-
-2008-05-05 23:01 +0000 [r115335] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c, /, main/logger.c: Merged revisions 115334 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r115334 | tilghman | 2008-05-05 18:00:31 -0500
- (Mon, 05 May 2008) | 15 lines Merged revisions 115333 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115333 | tilghman | 2008-05-05 17:50:31 -0500 (Mon, 05 May 2008)
- | 7 lines Separate verbose output from CLI output, by using a
- preamble. (closes issue #12402) Reported by: Corydon76 Patches:
- 20080410__no_verbose_in_rx_output.diff.txt uploaded by Corydon76
- (license 14) 20080501__no_verbose_in_rx_output__1.4.diff.txt
- uploaded by Corydon76 (license 14) ........ ................
-
-2008-05-05 22:17 +0000 [r115331] Joshua Colp <jcolp@digium.com>
-
- * /, build_tools/menuselect-deps.in, configure,
- include/asterisk/autoconfig.h.in, codecs/codec_speex.c,
- configure.ac: Merged revisions 115328 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r115328 | file | 2008-05-05 19:13:57 -0300 (Mon, 05 May 2008) |
- 10 lines Merged revisions 115327 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115327 | file | 2008-05-05 19:10:05 -0300 (Mon, 05 May 2008) | 2
- lines Make sure that either the main speex library contains
- preprocess functions or that speexdsp does. If both fail then
- speex stuff can not be built. ........ ................
-
-2008-05-05 22:14 +0000 [r115330] Mark Michelson <mmichelson@digium.com>
-
- * main/config.c, /: Merged revisions 115329 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r115329 |
- mmichelson | 2008-05-05 17:14:06 -0500 (Mon, 05 May 2008) | 15
- lines #execing the same file multiple times led to warning
- messages saying that the same file was being #included twice.
- This was due to the fact that #exec created a temporary file
- which was then #included. The name of the temporary file was the
- name of the #exec'd file, with the Unix timestamp and thread ID
- concatenated. The issue was that if multiple #exec statements of
- the same file were reached in the same second, then the result
- was that the temporary files would have duplicate names. To
- resolve this, the temporary file now has microsecond resolution
- for the timestamp portion. (closes issue #12574) Reported by:
- jmls Patches: 12574.patch uploaded by putnopvut (license 60)
- Tested by: jmls, putnopvut ........
-
-2008-05-05 21:44 +0000 [r115322] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 115321 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r115321 | mmichelson | 2008-05-05 16:43:21 -0500 (Mon, 05 May
- 2008) | 21 lines Merged revisions 115320 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115320 | mmichelson | 2008-05-05 16:41:34 -0500 (Mon, 05 May
- 2008) | 13 lines Don't consider a caller "handled" until the
- caller is bridged with a queue member. There was too much of an
- opportunity for the member to hang up (either during a delay,
- announcement, or overly long agi) between the time that he
- answered the phone and the time when he actually was bridged with
- the caller. The consequence of this was that if the member hung
- up in that interval, then proper abandonment details would not be
- noted in the queue log if the caller were to hang up at any point
- after the member hangup. (closes issue #12561) Reported by:
- ablackthorn ........ ................
-
-2008-05-05 20:28 +0000 [r115316] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 115315 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r115315 | russell | 2008-05-05 15:28:17 -0500 (Mon, 05 May 2008)
- | 2 lines Remove my rant, since I have now replaced the rant with
- code. ........
-
-2008-05-05 19:58 +0000 [r115310] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/res_odbc.h, /: Merged revisions 115309 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r115309 | tilghman | 2008-05-05 14:57:28 -0500
- (Mon, 05 May 2008) | 10 lines Merged revisions 115308 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115308 | tilghman | 2008-05-05 14:55:55 -0500 (Mon, 05 May 2008)
- | 2 lines Err, the documentation on the return value of
- ast_odbc_backslash_is_escape is exactly backwards. ........
- ................
-
-2008-05-05 19:50 +0000 [r115306] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 115305 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r115305 | russell | 2008-05-05 14:50:24 -0500 (Mon, 05 May 2008)
- | 13 lines Merged revisions 115304 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115304 | russell | 2008-05-05 14:49:25 -0500 (Mon, 05 May 2008)
- | 5 lines Avoid putting opaque="" in Digest authentication. This
- patch came from switchvox. It fixes authentication with Primus in
- Canada, and has been in use for a very long time without causing
- problems with any other providers. (closes issue AST-36) ........
- ................
-
-2008-05-05 19:43 +0000 [r115303] Tilghman Lesher <tlesher@digium.com>
-
- * /, UPGRADE.txt: Merged revisions 115302 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r115302 |
- tilghman | 2008-05-05 14:42:36 -0500 (Mon, 05 May 2008) | 2 lines
- Note change for ExecIf syntax (caught by jmls on IRC) ........
-
-2008-05-05 10:55 +0000 [r115289] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, UPGRADE.txt: Merged revisions 115288 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r115288 |
- kpfleming | 2008-05-05 05:55:09 -0500 (Mon, 05 May 2008) | 2
- lines clarify wording ........
-
-2008-05-05 03:26 +0000 [r115287] Tilghman Lesher <tlesher@digium.com>
-
- * contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk,
- contrib/init.d/rc.debian.asterisk,
- contrib/init.d/rc.mandrake.asterisk, /,
- contrib/init.d/rc.redhat.asterisk,
- contrib/init.d/rc.gentoo.asterisk,
- contrib/init.d/rc.slackware.asterisk: Merged revisions 115286 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r115286 | tilghman | 2008-05-04 22:25:35 -0500
- (Sun, 04 May 2008) | 15 lines Merged revisions 115285 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115285 | tilghman | 2008-05-04 22:22:25 -0500 (Sun, 04 May 2008)
- | 7 lines When starting Asterisk, bug out if Asterisk is already
- running. (closes issue #12525) Reported by: explidous Patches:
- 20080428__bug12525.diff.txt uploaded by Corydon76 (license 14)
- Tested by: mvanbaak ........ ................
-
-2008-05-04 02:12 +0000 [r115278-115284] Joshua Colp <jcolp@digium.com>
-
- * /, configure, acinclude.m4: Merged revisions 115283 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r115283 | file | 2008-05-03 23:11:01 -0300 (Sat,
- 03 May 2008) | 10 lines Merged revisions 115282 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115282 | file | 2008-05-03 23:09:44 -0300 (Sat, 03 May 2008) | 2
- lines Expand the test function for GCC attributes so that more
- complex attributes are properly recognized. ........
- ................
-
- * /, include/asterisk/compiler.h: Merged revisions 115280 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r115280 | file | 2008-05-03 22:52:00 -0300 (Sat,
- 03 May 2008) | 10 lines Merged revisions 115279 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115279 | file | 2008-05-03 22:50:59 -0300 (Sat, 03 May 2008) | 2
- lines For my next trick I will make these work with what our
- autoconf header file gives us. ........ ................
-
- * /, configure, acinclude.m4: Merged revisions 115277 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r115277 | file | 2008-05-03 22:45:21 -0300 (Sat,
- 03 May 2008) | 10 lines Merged revisions 115276 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115276 | file | 2008-05-03 22:43:26 -0300 (Sat, 03 May 2008) | 2
- lines Treat warnings as errors when checking if a GCC attribute
- exists. We have to do this as GCC will just ignore the attribute
- and pop up a warning, it won't actually fail to compile. ........
- ................
-
-2008-05-03 04:25 +0000 [r115269-115275] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * /: block voicemail mwi notification subscriptions taskprocessor
-
- * /: block pbx taskprocessor
-
- * /: block app_queue taskprocessor
-
- * /: blocked taskprocessors
-
-2008-05-02 14:55 +0000 [r115198-115200] Mark Michelson <mmichelson@digium.com>
-
- * /, include/asterisk/sched.h: Merged revisions 115197 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r115197 | mmichelson | 2008-05-02 09:28:55 -0500
- (Fri, 02 May 2008) | 14 lines Merged revisions 115196 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115196 | mmichelson | 2008-05-02 09:28:19 -0500 (Fri, 02 May
- 2008) | 6 lines Clarify a comment that was, well, just wrong. It
- turns out that ignoring the way that macros expand. Instead, I
- have clarified in the comment why the macro will work even if the
- scheduler id for the task to be deleted changes during the
- execution of the macro. ........ ................
-
-2008-05-02 02:57 +0000 [r115107-115160] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/res_odbc.h, /: Merged revisions 115104 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r115104 | tilghman | 2008-05-01 18:21:13 -0500
- (Thu, 01 May 2008) | 10 lines Merged revisions 115102 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115102 | tilghman | 2008-05-01 18:20:25 -0500 (Thu, 01 May 2008)
- | 2 lines Change the comment of deprecated to an actual compiler
- deprecation ........ ................
-
-2008-05-01 19:01 +0000 [r115020] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/utils.c: Merged revisions 115018 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r115018 | tilghman | 2008-05-01 14:00:18 -0500 (Thu, 01 May 2008)
- | 14 lines Merged revisions 115017 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115017 | tilghman | 2008-05-01 13:59:08 -0500 (Thu, 01 May 2008)
- | 6 lines '#' is another reserved character for URIs that also
- needs to be escaped. (closes issue #10543) Reported by: blitzrage
- Patches: 20080418__bug10543.diff.txt uploaded by Corydon76
- (license 14) ........ ................
-
-2008-05-01 17:28 +0000 [r114932] Russell Bryant <russell@digium.com>
-
- * /, UPGRADE.txt: Merged revisions 114931 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r114931 |
- russell | 2008-05-01 12:28:25 -0500 (Thu, 01 May 2008) | 4 lines
- Clarify the deprecation notice about Macro() to note that it will
- not be removed for the sake of backwards compatibility, since it
- is a non-trivial task to convert existing large dialplans that
- depend on Macro() to use GoSub(), instead. ........
-
-2008-05-01 16:52 +0000 [r114923] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 114922 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r114922 |
- qwell | 2008-05-01 11:49:24 -0500 (Thu, 01 May 2008) | 10 lines
- Allow dringXrange to properly default to 10, as was done in 1.4.
- dringXrange is a new feature that was added, and it attempted to
- default, but only when the option was specified. (closes issue
- #12536) Reported by: bjm Patches: 12536-dringXrange.diff uploaded
- by qwell (license 4) Tested by: bjm ........
-
-2008-04-30 20:20 +0000 [r114909] Russell Bryant <russell@digium.com>
-
- * include/asterisk/dlinkedlists.h (added): Add the dlinkedlists
- implementation from trunk
-
-2008-04-30 20:17 +0000 [r114907-114908] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: Make 1.6.0 compile
-
-2008-04-30 17:06 +0000 [r114900] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 114899 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114899 | oej | 2008-04-30 18:55:49 +0200 (Ons, 30 Apr 2008) | 15
- lines Merged revisions 114890 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114890 | oej | 2008-04-30 18:23:17 +0200 (Ons, 30 Apr 2008) | 7
- lines Don't crash on bad SIP replys. Fix created in Huntsville
- together with Mark M (putnopvut) (closes issue #12363) Reported
- by: jvandal Tested by: putnopvut, oej ........ ................
-
-2008-04-30 16:41 +0000 [r114893] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_console.c, channels/chan_iax2.c: Merged
- revisions 114892 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114892 | russell | 2008-04-30 11:34:24 -0500 (Wed, 30 Apr 2008)
- | 36 lines Merged revisions 114891 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114891 | russell | 2008-04-30 11:30:01 -0500 (Wed, 30 Apr 2008)
- | 28 lines Merge changes from team/russell/iax2_find_callno and
- iax2_find_callno_1.4 These changes address a critical performance
- issue introduced in the latest release. The fix for the latest
- security issue included a change that made Asterisk randomly
- choose call numbers to make them more difficult to guess by
- attackers. However, due to some inefficient (this is by far, an
- understatement) code, when Asterisk chose high call numbers,
- chan_iax2 became unusable after just a small number of calls. On
- a small embedded platform, it would not be able to handle a
- single call. On my Intel Core 2 Duo @ 2.33 GHz, I couldn't run
- more than about 16 IAX2 channels. Ouch. These changes address
- some performance issues of the find_callno() function that have
- bothered me for a very long time. On every incoming media frame,
- it iterated through every possible call number trying to find a
- matching active call. This involved a mutex lock and unlock for
- each call number checked. So, if the random call number chosen
- was 20000, then every media frame would cause 20000 locks and
- unlocks. Previously, this problem was not as obvious since
- Asterisk always chose the lowest call number it could. A second
- container for IAX2 pvt structs has been added. It is an astobj2
- hash table. When we know the remote side's call number, the pvt
- goes into the hash table with a hash value of the remote side's
- call number. Then, lookups for incoming media frames are a very
- fast hash lookup instead of an absolutely insane array traversal.
- In a quick test, I was able to get more than 3600% more IAX2
- channels on my machine with these changes. ........
- ................
-
-2008-04-30 16:15 +0000 [r114889] Jeff Peeler <jpeeler@digium.com>
-
- * /, channels/chan_console.c: Merged revisions 114888 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r114888 | jpeeler | 2008-04-30 11:14:43 -0500 (Wed, 30 Apr 2008)
- | 3 lines Fixes a bug where if a stream monitor thread was not
- created (caused from failure of opening or starting the stream)
- pthread_cancel was called with an invalid thread ID. ........
-
-2008-04-30 14:55 +0000 [r114877-114886] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, channels/iax2.h, channels/chan_iax2.c: Merged revisions 114884
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114884 | kpfleming | 2008-04-30 09:49:51 -0500
- (Wed, 30 Apr 2008) | 10 lines Merged revisions 114880 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114880 | kpfleming | 2008-04-30 09:46:57 -0500 (Wed, 30 Apr
- 2008) | 2 lines use the ARRAY_LEN macro for indexing through the
- iaxs/iaxsl arrays so that the size of the arrays can be adjusted
- in one place, and change the size of the arrays from 32768 calls
- to 2048 calls when LOW_MEMORY is defined ........
- ................
-
- * /, Makefile.rules: Merged revisions 114876 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114876 | kpfleming | 2008-04-30 07:15:43 -0500 (Wed, 30 Apr
- 2008) | 10 lines Merged revisions 114875 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114875 | kpfleming | 2008-04-30 07:14:07 -0500 (Wed, 30 Apr
- 2008) | 2 lines pay attention to *all* header files for
- dependency tracking, not just the local ones (inspired by r578 of
- asterisk-addons by tilghman) ........ ................
-
-2008-04-29 22:55 +0000 [r114867] Jeff Peeler <jpeeler@digium.com>
-
- * /, channels/iax2-provision.c: Merged revisions 114866 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r114866 | jpeeler | 2008-04-29 17:54:14 -0500 (Tue, 29
- Apr 2008) | 2 lines Fixes a problem where all the templates were
- marked as dead no matter what. The templates should only be
- marked as dead if a configuration file has been successfully
- loaded and has changes. Bug found while making API documentation
- for 1.6.0. ........
-
-2008-04-29 21:09 +0000 [r114850-114858] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 114849 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114849 | mmichelson | 2008-04-29 14:42:04 -0500 (Tue, 29 Apr
- 2008) | 22 lines Merged revisions 114848 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114848 | mmichelson | 2008-04-29 14:40:06 -0500 (Tue, 29 Apr
- 2008) | 14 lines Use the MACRO_CONTEXT and MACRO_EXTEN channel
- variables instead of the channel's macrocontext and macroexten
- fields. This is needed because if macros are daisy-chained, the
- incorrect context and extension are placed on the new channel. I
- also added locking to the channel prior to accessing these
- variables as noted in trunk's janitor project file. (closes issue
- #12549) Reported by: darren1713 Patches:
- app_queue.c.macroextenpatch uploaded by darren1713 (license 116)
- (with modifications from me) Tested by: putnopvut ........
- ................
-
-2008-04-29 19:04 +0000 [r114846] Kevin P. Fleming <kpfleming@digium.com>
-
- * /: bug is not present in this branch
-
-2008-04-29 17:11 +0000 [r114831] Jason Parker <jparker@digium.com>
-
- * res/res_config_pgsql.c, /: Merged revisions 114830 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114830 | qwell | 2008-04-29 12:10:55 -0500
- (Tue, 29 Apr 2008) | 9 lines Merged revisions 114829 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r114829 | qwell | 2008-04-29 12:08:55 -0500 (Tue, 29 Apr
- 2008) | 1 line Change warning message to debug, since there are
- cases where 0 results is perfectly fine. ........
- ................
-
-2008-04-29 12:55 +0000 [r114825] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 114824
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114824 | kpfleming | 2008-04-29 07:54:31 -0500
- (Tue, 29 Apr 2008) | 18 lines Merged revisions 114823 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r114823 | kpfleming | 2008-04-29 07:53:12 -0500
- (Tue, 29 Apr 2008) | 10 lines Merged revisions 114822 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r114822 | kpfleming | 2008-04-29 07:52:32 -0500 (Tue, 29 Apr
- 2008) | 2 lines stop script from appending source code if run
- multiple times ........ ................ ................
-
-2008-04-28 17:04 +0000 [r114777] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 114776 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r114776 |
- mattf | 2008-04-28 12:00:38 -0500 (Mon, 28 Apr 2008) | 1 line Fix
- deadlock issue in chan_zap with libss7 due to channel variables
- being set with the channel pvt lock being held. #12512 ........
-
-2008-04-28 13:44 +0000 [r114714] Joshua Colp <jcolp@digium.com>
-
- * /, configure, configure.ac: Merged revisions 114713 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r114713 | file | 2008-04-28 10:42:13 -0300 (Mon, 28 Apr 2008) | 2
- lines Update autoconf logic with latest API change for libss7.
- ........
-
-2008-04-28 04:54 +0000 [r114707-114710] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_jingle.c, /, channels/chan_gtalk.c: Merged
- revisions 114709 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114709 | tilghman | 2008-04-27 23:53:20 -0500 (Sun, 27 Apr 2008)
- | 13 lines Merged revisions 114708 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114708 | tilghman | 2008-04-27 23:47:39 -0500 (Sun, 27 Apr 2008)
- | 5 lines When modules are embedded, they take on a different
- name, without the ".so" extension. Specifically check for this
- name, when we're checking if a module is loaded. (Closes issue
- #12534) ........ ................
-
-2008-04-27 15:20 +0000 [r114701] Michiel van Baak <michiel@vanbaak.info>
-
- * /, channels/chan_skinny.c: Merged revisions 114700 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk Merged to 1.6
- because it fixes a crash. ........ r114700 | mvanbaak |
- 2008-04-27 17:17:18 +0200 (Sun, 27 Apr 2008) | 8 lines 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 ........
-
-2008-04-27 01:30 +0000 [r114697] Sean Bright <sean.bright@gmail.com>
-
- * /, configure, configure.ac: Merged revisions 114696 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114696 | seanbright | 2008-04-26 21:28:32 -0400
- (Sat, 26 Apr 2008) | 13 lines Merged revisions 114695 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114695 | seanbright | 2008-04-26 21:26:15 -0400 (Sat, 26 Apr
- 2008) | 5 lines When we don't explicitly pass a path to the
- --with-tds configure option, we may end up finding tds.h in
- /usr/local/include instead of /usr/include. If this happens, the
- grep that looks for the version (from tdsver.h) will fail and
- we'll have some problems during the build. ........
- ................
-
-2008-04-26 15:09 +0000 [r114684-114693] Tilghman Lesher <tlesher@digium.com>
-
- * /, contrib/scripts/vmail.cgi: Merged revisions 114690 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114690 | tilghman | 2008-04-26 08:17:19 -0500
- (Sat, 26 Apr 2008) | 14 lines Merged revisions 114689 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114689 | tilghman | 2008-04-26 08:15:21 -0500 (Sat, 26 Apr 2008)
- | 6 lines Clicking forward without selecting a message leaves an
- errant .lock file. (closes issue #12528) Reported by: pukepail
- Patches: patch.diff uploaded by pukepail (license 431) ........
- ................
-
-2008-04-25 22:05 +0000 [r114671-114677] Russell Bryant <russell@digium.com>
-
- * /, pbx/pbx_lua.c: Merged revisions 114676 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r114676 |
- russell | 2008-04-25 17:04:46 -0500 (Fri, 25 Apr 2008) | 7 lines
- Lock the channel around datastore access (closes issue #12527)
- Reported by: mnicholson Patches: pbx_lua4.diff uploaded by
- mnicholson (license 96) ........
-
- * /, channels/chan_iax2.c: Merged revisions 114674 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114674 | russell | 2008-04-25 17:00:35 -0500
- (Fri, 25 Apr 2008) | 11 lines Merged revisions 114673 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114673 | russell | 2008-04-25 16:54:40 -0500 (Fri, 25 Apr 2008)
- | 3 lines Use consistent logic for checking to see if a call
- number has been chosen yet. Also, remove some redundant logic I
- recently added in a fix. ........ ................
-
-2008-04-25 19:34 +0000 [r114664] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_chanspy.c, /: Merged revisions 114663 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114663 | mmichelson | 2008-04-25 14:33:27 -0500 (Fri, 25 Apr
- 2008) | 12 lines Merged revisions 114662 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114662 | mmichelson | 2008-04-25 14:32:02 -0500 (Fri, 25 Apr
- 2008) | 4 lines Move the unlock of the spyee channel to outside
- the start_spying() function so that the channel is not unlocked
- twice when using whisper mode. ........ ................
-
-2008-04-25 16:26 +0000 [r114652] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 114651 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r114651 | mmichelson | 2008-04-25 11:25:17 -0500 (Fri, 25 Apr
- 2008) | 4 lines Fix a memory leak and protect against potential
- dereferences of a NULL pointer. ........
-
-2008-04-24 22:14 +0000 [r114636] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 114635 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r114635 |
- file | 2008-04-24 19:11:46 -0300 (Thu, 24 Apr 2008) | 4 lines Hey
- look, it builds. (closes issue #12519) Reported by: falves11
- ........
-
-2008-04-24 21:36 +0000 [r114626-114634] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 114633 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114633 | mmichelson | 2008-04-24 16:35:39 -0500 (Thu, 24 Apr
- 2008) | 19 lines Merged revisions 114632 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114632 | mmichelson | 2008-04-24 16:35:08 -0500 (Thu, 24 Apr
- 2008) | 11 lines Re-invite RTP during a masquerade so that, for
- instance, an AMI redirect of two channels which are natively
- bridged will preserve audio on both channels. This prevents a
- problem with Asterisk not re-inviting due to one of the channels
- having being a zombie. (closes issue #12513) Reported by:
- mneuhauser Patches:
- asterisk-1.4-114602_restore-RTP-on-fixup.patch uploaded by
- mneuhauser (license 425) ........ ................
-
- * /, apps/app_queue.c: Merged revisions 114629 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114629 | mmichelson | 2008-04-24 15:43:52 -0500 (Thu, 24 Apr
- 2008) | 16 lines Merged revisions 114628 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114628 | mmichelson | 2008-04-24 15:43:03 -0500 (Thu, 24 Apr
- 2008) | 8 lines Output of channel variables when
- eventwhencalled=vars was set was being truncated two characters.
- This patch corrects the problem. (closes issue #12493) Reported
- by: davidw ........ ................
-
- * channels/chan_local.c, /: Merged revisions 114625 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114625 | mmichelson | 2008-04-24 15:06:06 -0500
- (Thu, 24 Apr 2008) | 18 lines Merged revisions 114624 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114624 | mmichelson | 2008-04-24 15:04:24 -0500 (Thu, 24 Apr
- 2008) | 10 lines Resolve a deadlock in chan_local by releasing
- the channel lock temporarily. (closes issue #11712) Reported by:
- callguy Patches: 11712.patch uploaded by putnopvut (license 60)
- Tested by: acunningham ........ ................
-
-2008-04-24 19:55 +0000 [r114619-114623] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_local.c, /: Merged revisions 114622 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114622 | tilghman | 2008-04-24 14:54:57 -0500
- (Thu, 24 Apr 2008) | 12 lines Merged revisions 114621 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114621 | tilghman | 2008-04-24 14:53:36 -0500 (Thu, 24 Apr 2008)
- | 4 lines Ensure that when we set the accountcode, it actually
- shows up in the CDR. (Fix for AMI Originate) (Closes issue
- #12007) ........ ................
-
- * /, apps/app_meetme.c: Merged revisions 114617 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r114617 |
- tilghman | 2008-04-24 14:24:31 -0500 (Thu, 24 Apr 2008) | 6 lines
- Fix DST calculation, and fix bug in calculation of whether conf
- has started yet or not (Closes issue #12292) Reported by: DEA
- Patches: app_meetme-rt-dst-sched-fix.txt uploaded by DEA (license
- 3) ........
-
-2008-04-24 16:48 +0000 [r114613] Jason Parker <jparker@digium.com>
-
- * channels/chan_misdn.c, /: Merged revisions 114612 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114612 | qwell | 2008-04-24 11:47:01 -0500
- (Thu, 24 Apr 2008) | 17 lines Merged revisions 51989 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #12496) Reported by: daniele Patches:
- misdn-moh-1.6.0-beta7.1.patch uploaded by daniele (license 471)
- Tested by: daniele Technically, I didn't use the patch above
- except to find out what revision to merge - but it's the same
- thing as this revision. ........ r51989 | crichter | 2007-01-24
- 06:57:22 -0600 (Wed, 24 Jan 2007) | 1 line added fix from #8899
- ........ ................
-
-2008-04-24 15:57 +0000 [r114610] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 114609 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114609 | russell | 2008-04-24 10:56:55 -0500
- (Thu, 24 Apr 2008) | 12 lines Merged revisions 114608 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114608 | russell | 2008-04-24 10:55:21 -0500 (Thu, 24 Apr 2008)
- | 4 lines Fix a silly mistake in a change I made yesterday that
- caused chan_iax2 to blow up very quickly. (issue #12515) ........
- ................
-
-2008-04-24 15:00 +0000 [r114607] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Merged revisions 114606 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114606 | oej | 2008-04-24 16:59:05 +0200 (Tor, 24 Apr 2008) | 11
- lines Merged revisions 114603 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114603 | oej | 2008-04-24 16:55:18 +0200 (Tor, 24 Apr 2008) | 3
- lines Only have one max-forwards header in outbound REFERs.
- Discovered in the Asterisk SIP Masterclass in Orlando. Thanks
- Joe! ........ ................
-
-2008-04-24 14:56 +0000 [r114599-114605] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 114604 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r114604 |
- russell | 2008-04-24 09:55:21 -0500 (Thu, 24 Apr 2008) | 3 lines
- Change a verbose message to debug. (closes issue #12514) ........
-
- * /, main/http.c: Merged revisions 114601 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114601 | russell | 2008-04-23 17:53:20 -0500 (Wed, 23 Apr 2008)
- | 14 lines Merged revisions 114600 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114600 | russell | 2008-04-23 17:18:12 -0500 (Wed, 23 Apr 2008)
- | 6 lines Improve some broken cookie parsing code. Previously,
- manager login over HTTP would only work if the mansession_id
- cookie was first. Now, the code builds a list of all of the
- cookies in the Cookie header. This fixes a problem observed by
- users of the Asterisk GUI. (closes AST-20) ........
- ................
-
- * apps/app_chanspy.c, /: Merged revisions 114598 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114598 | russell | 2008-04-23 15:53:05 -0500 (Wed, 23 Apr 2008)
- | 18 lines Merged revisions 114597 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114597 | russell | 2008-04-23 15:49:18 -0500 (Wed, 23 Apr 2008)
- | 10 lines Fix an issue that caused getting the correct next
- channel to not always work. Also, remove setting the amount of
- time to wait for a digit from 5 seconds back down to 1/10 of a
- second. I believe this was so the beep didn't get played over and
- over really fast, but a while back I put in another fix for that
- issue. (closes issue #12498) Reported by: jsmith Patches:
- app_chanspy_channel_walk.trunk.patch uploaded by jsmith (license
- 15) ........ ................
-
-2008-04-23 18:34 +0000 [r114596] Jason Parker <jparker@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 114595 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114595 | qwell | 2008-04-23 13:33:28 -0500
- (Wed, 23 Apr 2008) | 16 lines Merged revisions 114594 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114594 | qwell | 2008-04-23 13:28:44 -0500 (Wed, 23 Apr 2008) |
- 8 lines Fix reload/unload for res_musiconhold module. (closes
- issue #11575) Reported by: sunder Patches: M11575_14_rev3.diff
- uploaded by junky (license 177) bug11575_trunk.diff.txt uploaded
- by jamesgolovich (license 176) ........ ................
-
-2008-04-23 18:01 +0000 [r114589-114593] Russell Bryant <russell@digium.com>
-
- * main/manager.c, /, include/asterisk/manager.h: Merged revisions
- 114592 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114592 | russell | 2008-04-23 13:01:00 -0500 (Wed, 23 Apr 2008)
- | 13 lines Merged revisions 114591 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114591 | russell | 2008-04-23 12:55:31 -0500 (Wed, 23 Apr 2008)
- | 5 lines Store the manager session ID explicitly as 4 byte ID
- instead of a ulong. The mansession_id cookie is coded to be
- limited to 8 characters of hex, and this could break logins from
- 64-bit machines in some cases. (inspired by AST-20) ........
- ................
-
- * /, channels/chan_iax2.c: Merged revisions 114588 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114588 | russell | 2008-04-23 12:18:29 -0500
- (Wed, 23 Apr 2008) | 10 lines Merged revisions 114587 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114587 | russell | 2008-04-23 12:16:32 -0500 (Wed, 23 Apr 2008)
- | 2 lines Fix find_callno_locked() to actually return the callno
- locked in some more cases. ........ ................
-
-2008-04-23 16:57 +0000 [r114586] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Merged revisions 114585 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114585 | oej | 2008-04-23 18:53:34 +0200 (Ons, 23 Apr 2008) | 10
- lines Merged revisions 114584 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114584 | oej | 2008-04-23 18:51:41 +0200 (Ons, 23 Apr 2008) | 2
- lines Add 502 support for both directions, not only one... (see
- r114571) ........ ................
-
-2008-04-23 14:56 +0000 [r114581] Joshua Colp <jcolp@digium.com>
-
- * main/pbx.c, /: Merged revisions 114580 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114580 | file | 2008-04-23 11:55:03 -0300 (Wed, 23 Apr 2008) |
- 12 lines Merged revisions 114579 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114579 | file | 2008-04-23 11:54:11 -0300 (Wed, 23 Apr 2008) | 4
- lines Instead of stopping dialplan execution when SayNumber
- attempts to say a large number that it can not print out a
- message informing the user and continue on. (closes issue #12502)
- Reported by: bcnit ........ ................
-
-2008-04-23 01:00 +0000 [r114576-114578] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 114575 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r114575 | mmichelson | 2008-04-22 19:40:30 -0500 (Tue, 22 Apr
- 2008) | 10 lines Round 1 of IMAP_STORAGE-related app_voicemail
- changes This makes IMAP_STORAGE include the proper headers if you
- have specified the "system" option for --with-imap when running
- the configure script and your IMAP-related headers exist in
- /usr/include/c-client. This change is due to a hasty merge of a
- 1.4 change I made. ........
-
-2008-04-22 23:59 +0000 [r114573] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 114572 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114572 | tilghman | 2008-04-22 18:58:19 -0500 (Tue, 22 Apr 2008)
- | 10 lines Merged revisions 114571 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114571 | tilghman | 2008-04-22 18:51:44 -0500 (Tue, 22 Apr 2008)
- | 2 lines Treat a 502 just like a 503, when it comes to
- processing a response code ........ ................
-
-2008-04-22 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.6.0-beta8 released.
-
-2008-04-22 22:18 +0000 [r114560] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 114559 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114559 | russell | 2008-04-22 17:17:31 -0500
- (Tue, 22 Apr 2008) | 13 lines Merged revisions 114558 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114558 | russell | 2008-04-22 17:15:36 -0500 (Tue, 22 Apr 2008)
- | 5 lines When we receive a full frame that is supposed to
- contain our call number, ensure that it has the correct one.
- (closes issue #10078) (AST-2008-006) ........ ................
-
-2008-04-22 22:04 +0000 [r114556] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, /: Merged revisions 114553 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r114553 |
- murf | 2008-04-22 15:57:57 -0600 (Tue, 22 Apr 2008) | 14 lines
- (closes issue #12469) Reported by: triccyx I had a bit a problem
- reproducing this in my setup (trying not to disturb my other
- stuff) but finally, I got it. The problem appears to be that the
- extension is being added in replace mode, which kinda assumes
- that the pattern trie has been formed, when in fact, in this
- case, it was not. The checks being done are not nec. when the
- tree is not yet formed, as changes like this will be summarized
- when the trie is formed in the future. I tested the fix, and the
- crash no longer happens. Feel free to open the bug again if this
- fix doesn't cure the problem. ........
-
-2008-04-22 21:16 +0000 [r114544-114552] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /: Merged revisions 114548 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r114548 |
- russell | 2008-04-22 15:25:56 -0500 (Tue, 22 Apr 2008) | 2 lines
- re-add a fix that got lost with a recent change ........
-
-2008-04-22 18:14 +0000 [r114541] Jason Parker <jparker@digium.com>
-
- * main/pbx.c, /, include/asterisk/pbx.h, apps/app_queue.c: Merged
- revisions 114540 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r114540 |
- qwell | 2008-04-22 13:14:09 -0500 (Tue, 22 Apr 2008) | 8 lines
- Allow setqueuevar=yes (et al) to work, after changes to
- pbx_builtin_setvar() (closes issue #12490) Reported by: bcnit
- Patches: 12490-queuevars-3.diff uploaded by qwell (license 4)
- Tested by: qwell ........
-
-2008-04-22 18:06 +0000 [r114534-114539] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 114538 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114538 | russell | 2008-04-22 13:04:39 -0500
- (Tue, 22 Apr 2008) | 17 lines Merged revisions 114537 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114537 | russell | 2008-04-22 13:03:33 -0500 (Tue, 22 Apr 2008)
- | 9 lines If the dial string passed to the call channel callback
- does not indicate an extension, then consider the extension on
- the channel before falling back to the default. (closes issue
- #12479) Reported by: darren1713 Patches:
- exten_dial_fix_chan_iax2.c.patch uploaded by darren1713 (license
- 116) ........ ................
-
-2008-04-22 15:46 +0000 [r114524-114528] Russell Bryant <russell@digium.com>
-
- * main/manager.c, /: Merged revisions 114527 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r114527 |
- russell | 2008-04-22 10:46:01 -0500 (Tue, 22 Apr 2008) | 8 lines
- Correct action_ping() and action_events() with regards to Manager
- 1.1 documentation. Also, fix a bug in xml_translate(). (closes
- issue #11649) Reported by: ys Patches: trunk_manager.c.diff
- uploaded by ys (license 281) ........
-
-2008-04-21 20:23 +0000 [r114422] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 114389 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r114389 |
- mattf | 2008-04-21 13:44:35 -0500 (Mon, 21 Apr 2008) | 1 line Add
- support for generic name transmission (#12484) on SS7 in chan_zap
- ........
-
-2008-04-21 15:38 +0000 [r114328] Jeff Peeler <jpeeler@digium.com>
-
- * /, apps/app_authenticate.c: Merged revisions 114327 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r114327 | jpeeler | 2008-04-21 10:34:37 -0500 (Mon, 21 Apr 2008)
- | 2 lines This removes an invalid warning message for an
- incorrectly entered pin, but more importantly removes an
- inapplicable check. If the first argument passed to
- app_authenticate does not contain a '/', the argument should be
- treated as the sole fixed "password" to match against and that is
- all. (Previous behavior was attempting to open a file based on
- the pin.) ........
-
-2008-04-21 14:42 +0000 [r114321-114324] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 114323 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114323 | file | 2008-04-21 11:40:33 -0300 (Mon, 21 Apr 2008) |
- 12 lines Merged revisions 114322 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114322 | file | 2008-04-21 11:39:32 -0300 (Mon, 21 Apr 2008) | 4
- lines Only drop audio if we receive it without a progress
- indication. We allow other frames through such as DTMF because
- they may be needed to complete the call. (closes issue #12440)
- Reported by: aragon ........ ................
-
- * /, res/res_config_ldap.c: Merged revisions 114320 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r114320 | file | 2008-04-21 11:34:06 -0300 (Mon, 21 Apr 2008) | 6
- lines Only print out the error message if ldap_modify_ext_s
- actually returns an error, and not success. (closes issue #12438)
- Reported by: gservat Patches: res_config_ldap.c-patch-code
- uploaded by gservat (license 466) ........
-
-2008-04-19 17:00 +0000 [r114304] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: SS7:Added - Generic Name / Access Transport
- / Redirecting Number handling. #12425
-
-2008-04-18 21:51 +0000 [r114277-114286] Russell Bryant <russell@digium.com>
-
- * main/manager.c, /: Merged revisions 114285 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114285 | russell | 2008-04-18 16:51:05 -0500 (Fri, 18 Apr 2008)
- | 10 lines Merged revisions 114284 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114284 | russell | 2008-04-18 16:48:06 -0500 (Fri, 18 Apr 2008)
- | 2 lines Don't destroy a manager session if poll() returns an
- error of EAGAIN. ........ ................
-
- * Makefile, /: Merged revisions 114279 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114279 | russell | 2008-04-18 15:01:47 -0500 (Fri, 18 Apr 2008)
- | 10 lines Merged revisions 114278 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114278 | russell | 2008-04-18 15:01:09 -0500 (Fri, 18 Apr 2008)
- | 2 lines ensure directories are created before we try to install
- stuff into them ........ ................
-
- * Makefile, /: Merged revisions 114276 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114276 | russell | 2008-04-18 14:59:17 -0500 (Fri, 18 Apr 2008)
- | 10 lines Merged revisions 114275 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114275 | russell | 2008-04-18 14:58:55 -0500 (Fri, 18 Apr 2008)
- | 2 lines SUBDIRS_INSTALL is already listed as a subtarget for
- bininstall ........ ................
-
-2008-04-18 19:36 +0000 [r114262-114272] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_unistim.c, /: Merged revisions 114271 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r114271 | file | 2008-04-18 16:35:33 -0300 (Fri, 18 Apr 2008) | 4
- lines Make sure ADSI is marked as unavailable on Unistim channels
- so voicemail does not try to do some ADSI jazz. (closes issue
- #12460) Reported by: PerryB ........
-
-2008-04-18 18:04 +0000 [r114260] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_zap.c, /, main/callerid.c: Merged revisions 114259
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114259 | mmichelson | 2008-04-18 13:03:06 -0500
- (Fri, 18 Apr 2008) | 14 lines Merged revisions 114257 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114257 | mmichelson | 2008-04-18 12:44:29 -0500 (Fri, 18 Apr
- 2008) | 6 lines Clearing up error messages so they make a bit
- more sense. Also removing a redundant error message. Issue AST-15
- ........ ................
-
-2008-04-18 16:12 +0000 [r114255] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_config_ldap.c: Merged revisions 114254 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r114254 | file | 2008-04-18 13:11:27 -0300 (Fri, 18 Apr 2008) | 4
- lines If the parsing of the config file fails make sure we unlock
- ldap_lock. (closes issue #12477) Reported by: IgorG ........
-
-2008-04-18 13:40 +0000 [r114247] Sean Bright <sean.bright@gmail.com>
-
- * channels/chan_sip.c: Merged revisions 114246 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114246 | seanbright | 2008-04-18 09:38:07 -0400 (Fri, 18 Apr
- 2008) | 9 lines Merged revisions 114245 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114245 | seanbright | 2008-04-18 09:33:32 -0400 (Fri, 18 Apr
- 2008) | 1 line Only complete the SIP channel name once for 'sip
- show channel <channel>' ........ ................
-
-2008-04-18 06:54 +0000 [r114244] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_setcallerid.c, /: Merged revisions 114243 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114243 | tilghman | 2008-04-18 01:53:47 -0500
- (Fri, 18 Apr 2008) | 11 lines Merged revisions 114242 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114242 | tilghman | 2008-04-18 01:49:16 -0500 (Fri, 18 Apr 2008)
- | 3 lines For consistency sake, ensure that the values that
- ${CALLINGPRES} returns are valid as an input to SetCallingPres.
- (Closes issue #12472) ........ ................
-
-2008-04-17 23:09 +0000 [r114232-114241] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 114151 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114151 | oej | 2008-04-15 15:39:29 -0500 (Tue, 15 Apr 2008) | 10
- lines Merged revisions 114148 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114148 | oej | 2008-04-15 22:26:05 +0200 (Tis, 15 Apr 2008) | 2
- lines Handle subscribe queues in all situations... Thanks to
- festr_ on irc for telling me about this bug. ........
- ................
-
- * /, channels/chan_sip.c: Merged revisions 114150 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r114150 |
- oej | 2008-04-15 15:31:08 -0500 (Tue, 15 Apr 2008) | 2 lines
- Adding chanvar to SIPPEER from 1.4 branch ........
-
- * main/autoservice.c, /: Merged revisions 114233 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114233 | russell | 2008-04-17 17:24:00 -0500 (Thu, 17 Apr 2008)
- | 14 lines Merged revisions 114230 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114230 | russell | 2008-04-17 17:15:43 -0500 (Thu, 17 Apr 2008)
- | 6 lines Remove redundant safety net. The check for the
- autoservice channel list state accomplishes the same goal in a
- better way. (issue #12470) Reported By: atis ........
- ................
-
-2008-04-17 21:05 +0000 [r114228] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_chanspy.c, /: Merged revisions 114227 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114227 | mmichelson | 2008-04-17 16:04:40 -0500 (Thu, 17 Apr
- 2008) | 17 lines Merged revisions 114226 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114226 | mmichelson | 2008-04-17 16:03:29 -0500 (Thu, 17 Apr
- 2008) | 9 lines Declaration of the peer channel in this scope was
- making it so the peer variable defined in the outer scope was
- never set properly, therefore making iterating through the
- channel list always restart from the beginning. This bug would
- have affected anyone who called chanspy without specifying a
- first argument. (closes issue #12461) Reported by: stever28
- ........ ................
-
-2008-04-17 16:51 +0000 [r114210-114213] Mark Michelson <mmichelson@digium.com>
-
- * main/dsp.c, main/frame.c, /, include/asterisk/dsp.h,
- include/asterisk/frame.h: Merged revisions 114208 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114208 | mmichelson | 2008-04-17 11:40:12 -0500
- (Thu, 17 Apr 2008) | 20 lines Merged revisions 114207 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114207 | mmichelson | 2008-04-17 11:28:03 -0500 (Thu, 17 Apr
- 2008) | 12 lines It was possible for a reference to a frame which
- was part of a freed DSP to still be referenced, leading to memory
- corruption and eventual crashes. This code change ensures that
- the dsp is freed when we are finished with the frame. This change
- is very similar to a change Russell made with translators back a
- month or so ago. (closes issue #11999) Reported by: destiny6628
- Patches: 11999.patch uploaded by putnopvut (license 60) Tested
- by: destiny6628, victoryure ........ ................
-
-2008-04-17 16:26 +0000 [r114206] Russell Bryant <russell@digium.com>
-
- * Makefile, /: Merged revisions 114205 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114205 | russell | 2008-04-17 11:25:29 -0500 (Thu, 17 Apr 2008)
- | 11 lines Merged revisions 114204 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114204 | russell | 2008-04-17 11:23:45 -0500 (Thu, 17 Apr 2008)
- | 3 lines Fix the bininstall target to install from subdirs, as
- well. (closes issue AST-8, patch from bmd at switchvox) ........
- ................
-
-2008-04-17 15:17 +0000 [r114203] Tilghman Lesher <tlesher@digium.com>
-
- * doc/CODING-GUIDELINES, /: Merged revisions 114202 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r114202 | tilghman | 2008-04-17 10:12:52 -0500 (Thu, 17 Apr 2008)
- | 2 lines fileio.h does not exist; io.h does, though. ........
-
-2008-04-17 13:55 +0000 [r114200] Philippe Sultan <philippe.sultan@gmail.com>
-
- * /, res/res_jabber.c: Merged revisions 114199 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114199 | phsultan | 2008-04-17 15:46:17 +0200 (Thu, 17 Apr 2008)
- | 10 lines Merged revisions 114198 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114198 | phsultan | 2008-04-17 15:42:23 +0200 (Thu, 17 Apr 2008)
- | 2 lines Use keepalives effectively in order diagnose bug
- #12432. ........ ................
-
-2008-04-17 12:59 +0000 [r114197] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_agi.c: Merged revisions 114196 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114196 | tilghman | 2008-04-17 07:59:04 -0500 (Thu, 17 Apr 2008)
- | 16 lines Merged revisions 114195 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114195 | tilghman | 2008-04-17 07:56:38 -0500 (Thu, 17 Apr 2008)
- | 8 lines Add special case for when the agi cannot be executed,
- to comply with the documentation that we return failure in that
- case. (closes issue #12462) Reported by: fmueller Patches:
- 20080416__bug12462.diff.txt uploaded by Corydon76 (license 14)
- Tested by: fmueller ........ ................
-
-2008-04-17 10:56 +0000 [r114193] Sean Bright <sean.bright@gmail.com>
-
- * apps/app_chanspy.c, /: Merged revisions 114192 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114192 | seanbright | 2008-04-17 06:55:05 -0400 (Thu, 17 Apr
- 2008) | 9 lines Merged revisions 114191 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114191 | seanbright | 2008-04-17 06:51:20 -0400 (Thu, 17 Apr
- 2008) | 1 line Make sure we have enough room for the recording's
- filename. ........ ................
-
-2008-04-16 20:48 +0000 [r114186] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 114185 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114185 | kpfleming | 2008-04-16 15:47:30 -0500 (Wed, 16 Apr
- 2008) | 14 lines Merged revisions 114184 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114184 | kpfleming | 2008-04-16 15:46:38 -0500 (Wed, 16 Apr
- 2008) | 6 lines use the ZT_SET_DIALPARAMS ioctl properly by
- initializing the structure to all zeroes in case it contains
- fields that we don't write values into (which it does as of
- Zaptel 1.4.10) (closes issue #12456) Reported by: fnordian
- ........ ................
-
-2008-04-15 20:53 +0000 [r114153] Tilghman Lesher <tlesher@digium.com>
-
- * /, cdr/cdr_pgsql.c: Merged revisions 114152 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r114152 |
- tilghman | 2008-04-15 15:51:08 -0500 (Tue, 15 Apr 2008) | 2 lines
- Oops, buffer wasn't long enough for query ........
-
-2008-04-15 20:09 +0000 [r114147] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, /: Merged revisions 114146 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r114146 |
- murf | 2008-04-15 13:59:50 -0600 (Tue, 15 Apr 2008) | 8 lines
- These changes: a. fix a self-found problem with SPAWN-ing an
- extension, where matches were not being found b. correct some
- wording in a comment c. Add some debug for future debugging.
- ........
-
-2008-04-15 17:22 +0000 [r114132-114142] Jason Parker <jparker@digium.com>
-
- * channels/chan_unistim.c, /: Merged revisions 114141 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r114141 | qwell | 2008-04-15 12:21:58 -0500 (Tue, 15 Apr 2008) |
- 8 lines Shorten the mac address pattern, since some phones use
- different identifiers (such as the i2050 softphone). (closes
- issue #12398) Reported by: c_hans Patches: chan_unistim_svn.diff
- uploaded by c (license 460) Tested by: c_hans ........
-
- * contrib/scripts/autosupport, /: Merged revisions 114139 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114139 | qwell | 2008-04-15 12:17:37 -0500
- (Tue, 15 Apr 2008) | 15 lines Merged revisions 114138 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114138 | qwell | 2008-04-15 12:17:18 -0500 (Tue, 15 Apr 2008) |
- 7 lines Update Digium autosupport script, for more useful
- information. (closes issue #12452) Reported by: angler Patches:
- autosupport.diff uploaded by angler (license 106) ........
- ................
-
- * /, apps/app_queue.c: Merged revisions 114134 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114134 | qwell | 2008-04-15 11:18:38 -0500 (Tue, 15 Apr 2008) |
- 16 lines Merged revisions 114133 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114133 | qwell | 2008-04-15 11:18:08 -0500 (Tue, 15 Apr 2008) |
- 8 lines Allow autofill to work in the general section of
- queues.conf. Additionally, don't try to (re)set options when they
- have empty values in realtime (all unset columns would have an
- empty value). (closes issue #12445) Reported by: atis Patches:
- 12445-autofill.diff uploaded by qwell (license 4) ........
- ................
-
-2008-04-14 18:34 +0000 [r114122] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_h323.c: Merged revisions 114121 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114121 | qwell | 2008-04-14 13:34:17 -0500
- (Mon, 14 Apr 2008) | 15 lines Merged revisions 114120 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114120 | qwell | 2008-04-14 13:31:57 -0500 (Mon, 14 Apr 2008) |
- 7 lines The call_token on the pvt can occasionally be NULL,
- causing a crash. If it is NULL, we can skip this channel, since
- it can't the one we're looking for. (closes issue #9299) Reported
- by: vazir ........ ................
-
-2008-04-14 17:42 +0000 [r114119] Mark Michelson <mmichelson@digium.com>
-
- * main/channel.c, /: Merged revisions 114118 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114118 | mmichelson | 2008-04-14 12:42:20 -0500 (Mon, 14 Apr
- 2008) | 19 lines Merged revisions 114117 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114117 | mmichelson | 2008-04-14 12:41:03 -0500 (Mon, 14 Apr
- 2008) | 11 lines Increase the retry count when attempting to show
- channels. This apparently cleared an issue someone was seeing
- when attempting to show channels when the load was high. (closes
- issue #11667) Reported by: falves11 Patches: 11677.txt uploaded
- by russell (license 2) Tested by: falves11 ........
- ................
-
-2008-04-14 16:33 +0000 [r114116] Tilghman Lesher <tlesher@digium.com>
-
- * /, contrib/scripts/astcli: Merged revisions 114115 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r114115 | tilghman | 2008-04-14 11:32:59 -0500 (Mon, 14 Apr 2008)
- | 2 lines Make tab-completion work for all cases ........
-
-2008-04-14 16:25 +0000 [r114114] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_dial.c, /, apps/app_queue.c: Merged revisions 114113 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114113 | mmichelson | 2008-04-14 11:25:09 -0500
- (Mon, 14 Apr 2008) | 17 lines Merged revisions 114112 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114112 | mmichelson | 2008-04-14 11:24:22 -0500 (Mon, 14 Apr
- 2008) | 9 lines If the datastore has been moved to another
- channel due to a masquerade, then freeing the datastore here
- causes an eventual double free when the new channel hangs up. We
- should only free the datastore if we were able to successfully
- remove it from the channel we are referencing (i.e. the datastore
- was not moved). (closes issue #12359) Reported by: pguido
- ........ ................
-
-2008-04-14 15:02 +0000 [r114108] Mark Michelson <mmichelson@digium.com>
-
- * main/channel.c, /: Merged revisions 114107 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114107 | mmichelson | 2008-04-14 10:01:36 -0500 (Mon, 14 Apr
- 2008) | 13 lines Merged revisions 114106 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114106 | mmichelson | 2008-04-14 09:58:02 -0500 (Mon, 14 Apr
- 2008) | 5 lines Save a local copy of the generate callback prior
- to unlocking the channel in case the generate callback goes NULL
- on us after the channel is unlocked. Thanks to Russell for
- pointing this need out to me. ........ ................
-
-2008-04-14 14:54 +0000 [r114102-114105] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 114104 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114104 | file | 2008-04-14 11:53:33 -0300 (Mon, 14 Apr 2008) |
- 12 lines Merged revisions 114103 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114103 | file | 2008-04-14 11:52:46 -0300 (Mon, 14 Apr 2008) | 4
- lines It is possible for the remote side to say they want T38 but
- not give any capabilities. (closes issue #12414) Reported by: MVF
- ........ ................
-
- * main/rtp.c, /: Merged revisions 114101 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114101 | file | 2008-04-14 10:53:33 -0300 (Mon, 14 Apr 2008) |
- 12 lines Merged revisions 114100 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114100 | file | 2008-04-14 10:52:49 -0300 (Mon, 14 Apr 2008) | 4
- lines Don't change the SSRC when a new source comes into play,
- this might happen quite often and depending on the remote side...
- they might not like this. (closes issue #12353) Reported by:
- dimas ........ ................
-
-2008-04-14 02:59 +0000 [r114097-114099] Tilghman Lesher <tlesher@digium.com>
-
- * /, contrib/scripts/astcli: Merged revisions 114098 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r114098 | tilghman | 2008-04-13 21:55:41 -0500 (Sun, 13 Apr 2008)
- | 3 lines Add tab command-line completion (Closes issue #12428)
- ........
-
- * /, apps/app_meetme.c: Merged revisions 114096 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r114096 |
- tilghman | 2008-04-13 09:35:43 -0500 (Sun, 13 Apr 2008) | 3 lines
- Use ast_mkdir instead of mkdir (Closes issue #12430) ........
-
-2008-04-12 16:22 +0000 [r114094-114095] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Make sure linkset is locked exiting
- ss7_start_call
-
- * channels/chan_zap.c: Make sure we start incoming calls on SS7
- with echo cancellation enabled. Also make sure when completing a
- COT we call ss7_start_call with the proper locks held. Lastly,
- make sure if we fail to get a channel from zt_new that we don't
- assume it's there.
-
-2008-04-11 23:27 +0000 [r114089-114091] Tilghman Lesher <tlesher@digium.com>
-
- * /, cdr/cdr_pgsql.c: Merged revisions 114090 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r114090 |
- tilghman | 2008-04-11 18:26:56 -0500 (Fri, 11 Apr 2008) | 3 lines
- If any field is not null, but has no default, then it must be set
- or the insert will fail. (Closes issue #12285) ........
-
- * /, configs/res_ldap.conf.sample: Merged revisions 114088 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r114088 | tilghman | 2008-04-11 18:21:54 -0500 (Fri, 11
- Apr 2008) | 3 lines Make the sample config match the contributed
- LDAP schema (Closes issue #12421) ........
-
-2008-04-11 23:21 +0000 [r114087] Terry Wilson <twilson@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 114084 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114084 | twilson | 2008-04-11 17:48:52 -0500
- (Fri, 11 Apr 2008) | 15 lines Merged revisions 114083 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114083 | twilson | 2008-04-11 17:32:51 -0500 (Fri, 11 Apr 2008)
- | 7 lines Several places in the code called find_callno() (which
- releases the lock on the pvt structure) and then immediately
- locked the call and did things with it. Unfortunately, the call
- can disappear between the find_callno and the lock, causing Bad
- Stuff(tm) to happen. Added find_callno_locked() function to
- return the callno withtout unlocking for instances that it is
- needed. (issue #12400) Reported by: ztel ........
- ................
-
-2008-04-11 23:13 +0000 [r114086] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_config_ldap.c: Merged revisions 114085 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r114085 | tilghman | 2008-04-11 18:12:16 -0500 (Fri, 11 Apr 2008)
- | 7 lines Use the correct function for free'ing objects, and
- maybe we won't crash. (closes issue #12163) Reported by: gservat
- Patches: 20080411__bug12163.diff.txt uploaded by Corydon76
- (license 14) Tested by: gservat ........
-
-2008-04-11 15:51 +0000 [r114065] Mark Michelson <mmichelson@digium.com>
-
- * /, main/features.c: Merged revisions 114064 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114064 | mmichelson | 2008-04-11 10:49:35 -0500 (Fri, 11 Apr
- 2008) | 19 lines Merged revisions 114063 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114063 | mmichelson | 2008-04-11 10:44:28 -0500 (Fri, 11 Apr
- 2008) | 11 lines Fix a race condition that may happen between a
- sip hangup and a "core show channel" command. This patch adds
- locking to prevent the resulting crash. (closes issue #12155)
- Reported by: tsearle Patches: show_channels_crash2.patch uploaded
- by tsearle (license 373) Tested by: tsearle ........
- ................
-
-2008-04-11 14:56 +0000 [r114062] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_config_ldap.c: Merged revisions 114061 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r114061 | tilghman | 2008-04-11 09:54:22 -0500 (Fri, 11 Apr 2008)
- | 6 lines Errors are all greater than 0 (closes issue #12422)
- Reported by: nito Patches:
- res_config_ldap_result_check_patch.diff uploaded by nito (license
- 340) ........
-
-2008-04-10 22:23 +0000 [r114056] Mark Michelson <mmichelson@digium.com>
-
- * utils/conf2ael.c, utils/check_expr.c, utils/Makefile,
- main/manager.c, /, utils/astman.c, utils/hashtest.c,
- main/utils.c, include/asterisk/lock.h, utils/ael_main.c,
- utils/hashtest2.c: Merged revisions 114052 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114052 | mmichelson | 2008-04-10 17:02:32 -0500 (Thu, 10 Apr
- 2008) | 11 lines Merged revisions 114051 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114051 | mmichelson | 2008-04-10 15:59:49 -0500 (Thu, 10 Apr
- 2008) | 3 lines Fix 1.4 build when LOW_MEMORY is enabled.
- ........ ................
-
-2008-04-10 19:59 +0000 [r114047] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 114046 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114046 | mmichelson | 2008-04-10 14:58:36 -0500 (Thu, 10 Apr
- 2008) | 14 lines Merged revisions 114045 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114045 | mmichelson | 2008-04-10 14:55:33 -0500 (Thu, 10 Apr
- 2008) | 6 lines Be sure that we're not about to set bridgepvt
- NULL prior to dereferencing it. (closes issue #11775) Reported
- by: fujin ........ ................
-
-2008-04-10 19:09 +0000 [r114043] Tilghman Lesher <tlesher@digium.com>
-
- * /, contrib/scripts/astcli: Merged revisions 114042 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r114042 | tilghman | 2008-04-10 14:04:29 -0500 (Thu, 10 Apr 2008)
- | 7 lines The hydra grows yet another head... (closes issue
- #12401) Reported by: davevg Patches: astcli.diff2 uploaded by
- davevg (license 209) Tested by: davevg, Corydon76 ........
-
-2008-04-10 17:27 +0000 [r114037] Jason Parker <jparker@digium.com>
-
- * /, main/file.c: Merged revisions 114036 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114036 | qwell | 2008-04-10 12:27:16 -0500 (Thu, 10 Apr 2008) |
- 18 lines Merged revisions 114035 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114035 | qwell | 2008-04-10 12:26:10 -0500 (Thu, 10 Apr 2008) |
- 10 lines Only try to prefix language if we are not using an
- absolute path (suffix it otherwise).
- en/var/lib/asterisk/sounds/blah.gsm is a very silly path. (closes
- issue #12379) Reported by: kuj Patches: 12379-absolutepath.diff
- uploaded by qwell (license 4) Tested by: kuj, qwell ........
- ................
-
-2008-04-10 16:00 +0000 [r114023-114034] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 114030 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114030 | file | 2008-04-10 12:10:47 -0300 (Thu, 10 Apr 2008) |
- 14 lines Merged revisions 114029 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114029 | file | 2008-04-10 12:09:04 -0300 (Thu, 10 Apr 2008) | 6
- lines Create the directory where name recordings will go if it
- does not exist. (closes issue #12311) Reported by: rkeene
- Patches: 12311-mkdir.diff uploaded by qwell (license 4) ........
- ................
-
- * apps/app_voicemail.c, /: Merged revisions 114027 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r114027 | file | 2008-04-10 11:53:19 -0300 (Thu, 10 Apr 2008) | 6
- lines Don't hardcode ru into the digits filename so that
- languageprefix can work. (closes issue #12404) Reported by: IgorG
- Patches: voicemail_ru_hardcoded-v1.patch uploaded by IgorG
- (license 20) ........
-
- * main/rtp.c, channels/chan_unistim.c, /, channels/chan_skinny.c:
- Merged revisions 114024 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r114024 |
- file | 2008-04-10 10:45:45 -0300 (Thu, 10 Apr 2008) | 4 lines Fix
- spelling of existent in a few places. (closes issue #12409)
- Reported by: candlerb ........
-
- * /, channels/chan_sip.c: Merged revisions 114022 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114022 | file | 2008-04-10 10:28:30 -0300 (Thu, 10 Apr 2008) |
- 14 lines Merged revisions 114021 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114021 | file | 2008-04-10 10:27:11 -0300 (Thu, 10 Apr 2008) | 6
- lines Don't add custom URI options if they don't exist OR they
- are empty. (closes issue #12407) Reported by: homesick Patches:
- uri_options-1.4.diff uploaded by homesick (license 91) ........
- ................
-
-2008-04-09 22:34 +0000 [r113929-113982] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 113980 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r113980 |
- mmichelson | 2008-04-09 17:32:32 -0500 (Wed, 09 Apr 2008) | 8
- lines Fix a crash that happened due to accessing free'd memory
- (closes issue #12396) Reported by: tcalosi Patches: 12396.patch
- uploaded by putnopvut (license 60) Tested by: tcalosi ........
-
- * /, channels/chan_sip.c: Merged revisions 113928 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r113928 | mmichelson | 2008-04-09 15:56:14 -0500 (Wed, 09 Apr
- 2008) | 16 lines Merged revisions 113927 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r113927 | mmichelson | 2008-04-09 15:54:31 -0500 (Wed, 09 Apr
- 2008) | 8 lines We need to set the persistant_route [sic]
- parameter for the sip_pvt during the initial INVITE, no matter if
- we're building the route set from an INVITE request or response.
- (closes issue #12391) Reported by: benjaminbohlmann Tested by:
- benjaminbohlmann ........ ................
-
-2008-04-09 19:02 +0000 [r113876] Tilghman Lesher <tlesher@digium.com>
-
- * cdr/cdr_csv.c, /, configs/cdr.conf.sample: Merged revisions
- 113875 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r113875 | tilghman | 2008-04-09 14:00:40 -0500 (Wed, 09 Apr 2008)
- | 12 lines Merged revisions 113874 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r113874 | tilghman | 2008-04-09 13:57:33 -0500 (Wed, 09 Apr 2008)
- | 4 lines If the [csv] section does not exist in cdr.conf, then
- an unload/load sequence is needed to correct the problem. Track
- whether the load succeeded with a variable, so we can fix this
- with a simple reload event, instead. ........ ................
-
-2008-04-09 17:56 +0000 [r113839] Jason Parker <jparker@digium.com>
-
- * /, contrib/scripts/astcli: Merged revisions 113838 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r113838 | qwell | 2008-04-09 12:56:07 -0500 (Wed, 09 Apr 2008) |
- 2 lines Fix a small file handle "leak" pointed out by jjshoe on
- #asterisk. ........
-
-2008-04-09 17:50 +0000 [r113837] Mark Michelson <mmichelson@digium.com>
-
- * main/pbx.c, /: Merged revisions 113836 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r113836 |
- mmichelson | 2008-04-09 12:48:33 -0500 (Wed, 09 Apr 2008) | 14
- lines There was a subtle logical difference between 1.4 and trunk
- with regards to how timeouts were handled. In 1.4, if the
- absolute timeout were reached on a call, no matter what the
- return value of ast_spawn_extension was, the pbx would attempt to
- go to the 'T' extension or hangup otherwise. The rearrangement of
- this function in trunk made this check only happen in the case
- that ast_spawn_extension returned 0. If ast_spawn_extension
- returned 1, then the fact that the timeout expired resulted in a
- no-op, and would cause an infinite loop to occur in
- __ast_pbx_run. This change fixes this problem. Now timeouts will
- behave as they did in 1.4 (closes issue #11550) Reported by: pj
- Tested by: putnopvut ........
-
-2008-04-09 16:53 +0000 [r113786] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 113785 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r113785 | file | 2008-04-09 13:52:04 -0300 (Wed,
- 09 Apr 2008) | 12 lines Merged revisions 113784 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r113784 | file | 2008-04-09 13:50:45 -0300 (Wed, 09 Apr 2008) | 4
- lines If we receive an AUTHREQ from the remote server and we are
- unable to reply (for example they have a secret configured, but
- we do not) then queue a hangup frame on the Asterisk channel.
- This will cause the channel to hangup and a HANGUP to be sent via
- IAX2 to the remote side which is the proper thing to do in this
- scenario. (closes issue #12385) Reported by: viraptor ........
- ................
-
-2008-04-09 14:42 +0000 [r113683] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 113682 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r113682 | mmichelson | 2008-04-09 09:41:58 -0500 (Wed, 09 Apr
- 2008) | 17 lines Merged revisions 113681 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r113681 | mmichelson | 2008-04-09 09:40:05 -0500 (Wed, 09 Apr
- 2008) | 9 lines If Asterisk receives a 488 on an INVITE (not a
- reinvite), then we should not send a BYE. (closes issue #12392)
- Reported by: fnordian Patches: chan_sip.patch uploaded by
- fnordian (license 110) with small modification from me ........
- ................
-
-2008-04-09 13:56 +0000 [r113648-113650] Tilghman Lesher <tlesher@digium.com>
-
- * /, contrib/scripts/astcli: Merged revisions 113647 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r113647 | tilghman | 2008-04-09 08:23:44 -0500 (Wed, 09 Apr 2008)
- | 6 lines Additional enhancements (closes issue #12390) Reported
- by: tzafrir Patches: astcli_fixes.diff uploaded by tzafrir
- (license 46) ........
-
-2008-04-09 01:40 +0000 [r113598] Terry Wilson <twilson@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 113597 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r113597 | twilson | 2008-04-08 20:36:58 -0500
- (Tue, 08 Apr 2008) | 10 lines Merged revisions 113596 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r113596 | twilson | 2008-04-08 20:34:25 -0500 (Tue, 08 Apr 2008)
- | 2 lines Initialize fr->cacheable to make valgrind happy
- ........ ................
-
-2008-04-08 21:34 +0000 [r113560] Tilghman Lesher <tlesher@digium.com>
-
- * /, contrib/scripts/astcli (added): Merged revisions 113559 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r113559 | tilghman | 2008-04-08 16:33:11 -0500 (Tue, 08
- Apr 2008) | 6 lines Add commandline tool for doing CLI commands
- through AMI (instead of using asterisk -rx) (closes issue #12389)
- Reported by: davevg Patches: astcli uploaded by davevg (license
- 209) ........
-
-2008-04-08 18:49 +0000 [r113404-113506] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_skinny.c: Merged revisions 113505 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r113505 | qwell | 2008-04-08 13:49:21 -0500
- (Tue, 08 Apr 2008) | 9 lines 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. ........ ................
-
- * /, channels/chan_skinny.c: Merged revisions 113455 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r113455 | qwell | 2008-04-08 13:08:35 -0500
- (Tue, 08 Apr 2008) | 12 lines 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. ........ ................
-
- * main/asterisk.c, /: Merged revisions 113403 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r113403 | qwell | 2008-04-08 12:00:55 -0500 (Tue, 08 Apr 2008) |
- 9 lines Merged revisions 113402 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r113402 | qwell | 2008-04-08 11:56:52 -0500 (Tue, 08 Apr 2008) |
- 1 line Work around some silliness caused by sys/capability.h -
- this should fix compile errors a number of users have been
- experiencing. ........ ................
-
-2008-04-08 16:56 +0000 [r113350-113401] Tilghman Lesher <tlesher@digium.com>
-
- * /, contrib/scripts/astgenkey.8: Merged revisions 113400 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r113400 | tilghman | 2008-04-08 11:54:21 -0500
- (Tue, 08 Apr 2008) | 14 lines Merged revisions 113399 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r113399 | tilghman | 2008-04-08 11:51:28 -0500 (Tue, 08 Apr 2008)
- | 6 lines Add security note on astgenkey's manpage. (closes issue
- #12373) Reported by: lmamane Patches: 20080406__bug12373.diff.txt
- uploaded by Corydon76 (license 14) ........ ................
-
- * /, channels/chan_sip.c: Merged revisions 113349 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r113349 | tilghman | 2008-04-08 10:48:58 -0500 (Tue, 08 Apr 2008)
- | 15 lines Merged revisions 113348 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r113348 | tilghman | 2008-04-08 10:39:16 -0500 (Tue, 08 Apr 2008)
- | 7 lines Move check for still-bridged channels out a little
- further, to avoid possible deadlocks. (Closes issue #12252)
- Reported by: callguy Patches: 20080319__bug12252.diff.txt
- uploaded by Corydon76 (license 14) Tested by: callguy ........
- ................
-
-2008-04-08 15:10 +0000 [r113298-113299] Joshua Colp <jcolp@digium.com>
-
- * /, main/audiohook.c: Merged revisions 113297 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r113297 | file | 2008-04-08 12:05:35 -0300 (Tue, 08 Apr 2008) |
- 12 lines Merged revisions 113296 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r113296 | file | 2008-04-08 12:03:43 -0300 (Tue, 08 Apr 2008) | 4
- lines If audio suddenly gets fed into one side of a channel after
- a lapse of frames flush the other factory so that old audio does
- not remain in the factory causing the sync code to not execute.
- (closes issue #12296) Reported by: jvandal ........
- ................
-
-2008-04-07 22:17 +0000 [r113246] Tilghman Lesher <tlesher@digium.com>
-
- * /, configs/manager.conf.sample: Merged revisions 113245 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r113245 | tilghman | 2008-04-07 17:16:46 -0500 (Mon, 07
- Apr 2008) | 2 lines Additional note ........
-
-2008-04-07 21:49 +0000 [r113244] Jason Parker <jparker@digium.com>
-
- * /, configs/manager.conf.sample: Merged revisions 113243 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r113243 | qwell | 2008-04-07 16:49:27 -0500 (Mon, 07 Apr
- 2008) | 1 line Document 'originate' permission in manager sample
- config. ........
-
-2008-04-07 21:36 +0000 [r113242] Jeff Peeler <jpeeler@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 113241 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r113241 | jpeeler | 2008-04-07 16:35:48 -0500 (Mon, 07 Apr 2008)
- | 23 lines Merged revisions 113013 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r113013 | jpeeler | 2008-04-07 10:18:10 -0500 (Mon, 07 Apr 2008)
- | 15 lines Merged revisions 113012 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r113012 | jpeeler | 2008-04-07 10:16:44 -0500 (Mon, 07 Apr 2008)
- | 7 lines (closes issue #12362) (closes issue #12372) Reported
- by: vinsik Tested by: tecnoxarxa This one line change makes an if
- inside a for loop (in realtime_peer) check all the ast_variables
- the loop was intending to test rather than just the first one.
- ........ ................ ................
-
-2008-04-07 19:10 +0000 [r113174] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_skinny.c, configs/skinny.conf.sample: Merged
- revisions 113119 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r113119 | qwell | 2008-04-07 13:02:51 -0500 (Mon, 07 Apr 2008) |
- 16 lines 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 ........ ................
-
-2008-04-07 19:08 +0000 [r113173] Tilghman Lesher <tlesher@digium.com>
-
- * /, funcs/func_strings.c: Merged revisions 113172 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r113172 | tilghman | 2008-04-07 14:06:46 -0500
- (Mon, 07 Apr 2008) | 11 lines Merged revisions 113117 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r113117 | tilghman | 2008-04-07 12:51:49 -0500 (Mon, 07 Apr 2008)
- | 3 lines Force ast_mktime() to check for DST, since strptime(3)
- does not. (Closes issue #12374) ........ ................
-
-2008-04-07 16:13 +0000 [r113067] Mark Michelson <mmichelson@digium.com>
-
- * main/channel.c, /: Merged revisions 113066 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r113066 | mmichelson | 2008-04-07 11:12:30 -0500 (Mon, 07 Apr
- 2008) | 21 lines Merged revisions 113065 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r113065 | mmichelson | 2008-04-07 11:08:45 -0500 (Mon, 07 Apr
- 2008) | 13 lines This fix prevents a deadlock that was
- experienced in chan_local. There was deadlock prevention in place
- in chan_local, but it would not work in a specific case because
- the channel was recursively locked. By unlocking the channel
- prior to calling the generator's generate callback in
- ast_read_generator_actions(), we prevent the recursive locking,
- and therefore the deadlock. (closes issue #12307) Reported by:
- callguy Patches: 12307.patch uploaded by putnopvut (license 60)
- Tested by: callguy ........ ................
-
-2008-04-07 15:28 +0000 [r113042] Jeff Peeler <jpeeler@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 113013 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r113013 | jpeeler | 2008-04-07 10:18:10 -0500 (Mon, 07 Apr 2008)
- | 15 lines Merged revisions 113012 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r113012 | jpeeler | 2008-04-07 10:16:44 -0500 (Mon, 07 Apr 2008)
- | 7 lines (closes issue #12362) (closes issue #12372) Reported
- by: vinsik Tested by: tecnoxarxa This one line change makes an if
- inside a for loop (in realtime_peer) check all the ast_variables
- the loop was intending to test rather than just the first one.
- ........ ................
-
-2008-04-05 13:30 +0000 [r112973-112975] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_agi.c: Merged revisions 112972 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r112972 |
- tilghman | 2008-04-05 08:24:12 -0500 (Sat, 05 Apr 2008) | 6 lines
- AsyncAGI should not close the manager session on error. (closes
- issue #12370) Reported by: srt Patches: asterisk-12370.diff
- uploaded by srt (license 378) ........
-
-2008-04-04 19:30 +0000 [r112786-112822] Philippe Sultan <philippe.sultan@gmail.com>
-
- * /, channels/chan_gtalk.c: Merged revisions 112821 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r112821 | phsultan | 2008-04-04 21:28:49 +0200
- (Fri, 04 Apr 2008) | 9 lines Merged revisions 112820 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r112820 | phsultan | 2008-04-04 21:26:15 +0200 (Fri, 04
- Apr 2008) | 1 line Free newly allocated channel before returning
- ........ ................
-
- * /, channels/chan_gtalk.c: Merged revisions 112785 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r112785 | phsultan | 2008-04-04 19:32:46 +0200
- (Fri, 04 Apr 2008) | 15 lines Merged revisions 112766 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r112766 | phsultan | 2008-04-04 19:16:59 +0200 (Fri, 04 Apr 2008)
- | 7 lines Prevent call connections when codecs don't match.
- (closes issue #10604) Reported by: keepitcool Patches:
- branch-1.4-10604-2.diff uploaded by phsultan (license 73) Tested
- by: phsultan ........ ................
-
-2008-04-04 01:08 +0000 [r112715] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * main/asterisk.c, /: Merged revisions 112653,112656,112714 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r112653 | dhubbard | 2008-04-03 17:13:11 -0500 (Thu, 03
- Apr 2008) | 1 line add a Zaptel timer check to verify the timer
- is responding when Zaptel support is compiled into Asterisk and
- Zaptel drivers are loaded. This will help people not waste their
- valuable time debugging side effects. ........ r112656 | dhubbard
- | 2008-04-03 17:19:43 -0500 (Thu, 03 Apr 2008) | 1 line satisfy
- buildbot ........ r112714 | dhubbard | 2008-04-03 19:57:33 -0500
- (Thu, 03 Apr 2008) | 1 line sleep long enough for the zaptel
- timer error message to display before exit ........
-
-2008-04-04 00:54 +0000 [r112713] Joshua Colp <jcolp@digium.com>
-
- * /, main/Makefile: Merged revisions 112712 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r112712 | file | 2008-04-03 21:53:19 -0300 (Thu, 03 Apr 2008) |
- 10 lines Merged revisions 112711 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r112711 | file | 2008-04-03 21:52:36 -0300 (Thu, 03 Apr 2008) | 2
- lines Pass in the path to Zaptel for systems that install Zaptel
- headers in a separate location. ........ ................
-
-2008-04-03 14:42 +0000 [r112601] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 112600 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r112600 | mmichelson | 2008-04-03 09:35:47 -0500 (Thu, 03 Apr
- 2008) | 17 lines Merged revisions 112599 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r112599 | mmichelson | 2008-04-03 09:32:20 -0500 (Thu, 03 Apr
- 2008) | 9 lines Fix the testing of the "res" variable so that it
- is more logically correct and makes the correct warning and debug
- messages print. (closes issue #12361) Reported by: one47 Patches:
- chan_zap_deferred_digit.patch uploaded by one47 (license 23)
- ........ ................
-
-2008-04-02 17:37 +0000 [r112470] Mark Michelson <mmichelson@digium.com>
-
- * main/manager.c, /: Merged revisions 112469 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r112469 | mmichelson | 2008-04-02 12:36:49 -0500 (Wed, 02 Apr
- 2008) | 21 lines Merged revisions 112468 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r112468 | mmichelson | 2008-04-02 12:36:04 -0500 (Wed, 02 Apr
- 2008) | 13 lines Fix a race condition in the manager. It is
- possible that a new manager event could be appended during a
- brief time when the manager is not waiting for input. If an event
- comes during this period, we need to set an indicator that there
- is an event pending so that the manager doesn't attempt to wait
- forever for an event that already happened. (closes issue #12354)
- Reported by: bamby Patches: manager_race_condition.diff uploaded
- by bamby (license 430) (comments added by me) ........
- ................
-
-2008-04-02 15:27 +0000 [r112436] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 112431 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r112431 |
- file | 2008-04-02 12:26:51 -0300 (Wed, 02 Apr 2008) | 7 lines
- Since the SIP request structure gets reused multiple times with
- TCP handling we have to clear the debug state or else we will
- keep spitting out debug even after it has been turned off.
- (closes issue #12169) Reported by: pj Patches:
- 12169-debugoff-2.diff uploaded by qwell (license 4) Tested by: pj
- ........
-
-2008-04-02 14:33 +0000 [r112395] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 112394 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r112394 | mmichelson | 2008-04-02 09:32:43 -0500 (Wed, 02 Apr
- 2008) | 14 lines Merged revisions 112393 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r112393 | mmichelson | 2008-04-02 09:32:00 -0500 (Wed, 02 Apr
- 2008) | 6 lines Ensure that there is no timeout if none is
- specified. (closes issue #12349) Reported by: johnlange ........
- ................
-
-2008-04-01 22:48 +0000 [r112359] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, /: Merged revisions 112357 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r112357 |
- murf | 2008-04-01 16:45:10 -0600 (Tue, 01 Apr 2008) | 1 line
- Bumped across another test set for the new exten pattern matcher,
- which revealed a problem with the CANMATCH/MATCHMORE modes.
- Direct matches were getting in the way. Fixed. ........
-
-2008-04-01 20:20 +0000 [r112299] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, /: Merged revisions 112289 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r112289 |
- murf | 2008-04-01 14:02:19 -0600 (Tue, 01 Apr 2008) | 21 lines
- (closes issue #12298) Reported by: falves11 Patches: 12298.patch1
- uploaded by murf (license 17) Tested by: murf I have hopes that
- the changes made over the last few days will finalize and
- solidify this code. While there are bound to be small tweaks
- still needed, I feel that the job (at last) is somewhat
- completed. Finally, I had a chance to comprehend how the scoring
- of extension patterns was done in the previous version, and I've
- come very close to using the exact same criteria in the new
- pattern matching code. The left-right sorting is now replicated
- in the trie structure itself, such that the first match found
- will the 'best' match. Compared the results against 1.4 for
- several extensions. Replicated falves11's setup and it works.
- Used some devious patterns provided by jsmith, supplemented with
- a few of my own. Looks good. ........
-
-2008-04-01 18:09 +0000 [r112211] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c, /: Merged revisions 112210 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r112210 | file | 2008-04-01 15:06:13 -0300 (Tue, 01 Apr 2008) |
- 12 lines Merged revisions 112209 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r112209 | file | 2008-04-01 15:02:43 -0300 (Tue, 01 Apr 2008) | 4
- lines Disable Packet2Packet bridging when we need to feed DTMF
- frames into the core. Some implementations do not like how we
- switch between things. (closes issue #12212) Reported by: bamby
- ........ ................
-
-2008-04-01 17:52 +0000 [r112170-112206] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 112205 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r112205 | file | 2008-04-01 14:48:52 -0300 (Tue, 01 Apr 2008) |
- 12 lines Merged revisions 112204 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r112204 | file | 2008-04-01 14:43:46 -0300 (Tue, 01 Apr 2008) | 4
- lines Do not pass audio until the remote side has indicated they
- are providing early media, or if the channel has been answered.
- (closes issue #11823) Reported by: SDamm ........
- ................
-
-2008-04-01 17:25 +0000 [r112157] Mark Michelson <mmichelson@digium.com>
-
- * main/dns.c, /: Merged revisions 112148 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r112148 | mmichelson | 2008-04-01 12:23:19 -0500 (Tue, 01 Apr
- 2008) | 18 lines Merged revisions 112138 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r112138 | mmichelson | 2008-04-01 12:21:21 -0500 (Tue, 01 Apr
- 2008) | 10 lines Initialize the __res_state structure used for
- dns purposes to all 0's prior to using it. This is due to
- valgrind's complaints on issue #12284 as well as an excerpt found
- in "Description" portion of the online man page found here:
- http://www.iti.cs.tu-bs.de/cgi-bin/UNIXhelp/man-cgi?res_nquery+3RESOLV
- (pertains to issue #12284 but does not necessarily close it)
- ........ ................
-
-2008-04-01 16:57 +0000 [r112127] Joshua Colp <jcolp@digium.com>
-
- * include/asterisk/slinfactory.h, /, main/slinfactory.c: Merged
- revisions 112126 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r112126 | file | 2008-04-01 13:50:37 -0300 (Tue, 01 Apr 2008) |
- 13 lines Merged revisions 112125 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r112125 | file | 2008-04-01 13:45:14 -0300 (Tue, 01 Apr 2008) | 5
- lines Ensure that we do not exceed the hold's maximum size with a
- single frame. (closes issue #12047) Reported by: fabianoheringer
- Tested by: fabianoheringer ........ ................
-
-2008-03-31 22:17 +0000 [r112070-112072] Jason Parker <jparker@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 112069 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r112069 | qwell | 2008-03-31 16:48:30 -0500
- (Mon, 31 Mar 2008) | 13 lines Merged revisions 112068 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r112068 | qwell | 2008-03-31 16:48:05 -0500 (Mon, 31 Mar 2008) |
- 5 lines Fix a silly infinite loop when choosing an invalid
- option. (closes issue #12315) Reported by: jmls ........
- ................
-
-2008-03-31 21:03 +0000 [r112034-112036] Terry Wilson <twilson@digium.com>
-
- * /, main/http.c: Merged revisions 112033 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r112033 |
- twilson | 2008-03-31 15:45:05 -0500 (Mon, 31 Mar 2008) | 2 lines
- Handle blank prefix= in http.conf ........
-
-2008-03-31 17:15 +0000 [r111997-111999] Russell Bryant <russell@digium.com>
-
- * Makefile, /: Merged revisions 111998 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r111998 |
- russell | 2008-03-31 12:14:58 -0500 (Mon, 31 Mar 2008) | 7 lines
- Ensure configure gets run on a clean checkout. (closes issue
- #12197) Reported by: juggie Patches: 12197.diff uploaded by
- juggie (license 24) ........
-
-2008-03-31 14:22 +0000 [r111962] Joshua Colp <jcolp@digium.com>
-
- * res/res_config_sqlite.c, /: Merged revisions 111961 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r111961 | file | 2008-03-31 11:20:39 -0300 (Mon, 31 Mar 2008) | 4
- lines Initialize all these here tmp pointers at declaration. They
- confused some compilers a wee bit. (closes issue #12333) Reported
- by: ovi ........
-
-2008-03-29 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.6.0-beta7.1 released.
-
- Asterisk 1.6.0-beta7 was tagged against trunk, instead of the 1.6.0 branch.
-
-2008-03-28 21:46 +0000 [r111858] Jason Parker <jparker@digium.com>
-
- * codecs/gsm/inc/private.h, /: Merged revisions 111857 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r111857 | qwell | 2008-03-28 16:46:02 -0500
- (Fri, 28 Mar 2008) | 20 lines Merged revisions 111856 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r111856 | qwell | 2008-03-28 16:45:35 -0500 (Fri, 28 Mar 2008) |
- 12 lines Allow gsm to compile correctly on x86 with gcc4
- optimizations. (closes issue #11243) Reported by: whiskerp
- Patches: 11243-maybe-asm.diff uploaded by qwell (license 4)
- Tested by: Seggy (IRC) Note: While I did write this patch, I
- would not have found this if fossil had not reported and fixed
- issue #12253. A huge thanks to him for helping to (indirectly)
- find the problem here. ........ ................
-
-2008-03-28 19:11 +0000 [r111722-111776] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_skinny.c: Merged revisions 111721 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r111721 | qwell | 2008-03-28 12:57:12 -0500
- (Fri, 28 Mar 2008) | 9 lines 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. ........ ................
-
-2008-03-28 16:21 +0000 [r111660] Jason Parker <jparker@digium.com>
-
- * /, formats/format_wav_gsm.c: Merged revisions 111659 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r111659 | qwell | 2008-03-28 11:20:59 -0500
- (Fri, 28 Mar 2008) | 16 lines Merged revisions 111658 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r111658 | qwell | 2008-03-28 11:19:56 -0500 (Fri, 28 Mar 2008) |
- 8 lines The file size of WAV49 does not need to be an even
- number. (closes issue #12128) Reported by: mdu113 Patches:
- 12128-noevenlength.diff uploaded by qwell (license 4) Tested by:
- qwell, mdu113 ........ ................
-
-2008-03-28 14:43 +0000 [r111607-111608] Tilghman Lesher <tlesher@digium.com>
-
- * doc/valgrind.txt, /: Merged revisions 111606 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r111606 | tilghman | 2008-03-28 09:37:28 -0500 (Fri, 28 Mar 2008)
- | 11 lines Merged revisions 111605 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r111605 | tilghman | 2008-03-28 09:35:45 -0500 (Fri, 28 Mar 2008)
- | 3 lines Update debugging text, since Valgrind eliminated the
- --log-file-exactly option. (Closes issue #12320) ........
- ................
-
-2008-03-28 00:56 +0000 [r111566] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 111565 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r111565 |
- file | 2008-03-27 21:55:47 -0300 (Thu, 27 Mar 2008) | 2 lines
- Forgetting to unregister a manager action is bad, mmmk? ........
-
-2008-03-28 00:17 +0000 [r111534] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 111533 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r111533 |
- mmichelson | 2008-03-27 19:12:52 -0500 (Thu, 27 Mar 2008) | 10
- lines Fix a crash that would happen when attempting to unload the
- app_queue module. The problem was that when the refcount on the
- queue hit 0, the destructor was called, and inside the
- destructor, another function was called which would increase the
- refcount back to 1 again and then decrease it again back to 0 for
- every member in the queue. This meant that the destructor was
- being recursively called, leading to a double free of the queue.
- This is now fixed by making sure to unlink the queue from the
- queues container prior to the final unref of the queue. ........
-
-2008-03-27 21:28 +0000 [r111498] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, /: Merged revisions 111497 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r111497 |
- murf | 2008-03-27 15:25:55 -0600 (Thu, 27 Mar 2008) | 1 line
- comment cleanup and iron out a really dumb mistake in handling
- the '.'-wildcard in the new exten pattern matcher. ........
-
-2008-03-27 19:30 +0000 [r111444] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/acl.c: Merged revisions 111443 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r111443 | tilghman | 2008-03-27 14:26:45 -0500 (Thu, 27 Mar 2008)
- | 14 lines Merged revisions 111442 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r111442 | tilghman | 2008-03-27 14:23:12 -0500 (Thu, 27 Mar 2008)
- | 6 lines For FreeBSD, at least, the ifa_addr element could be
- NULL. (closes issue #12300) Reported by: festr Patches:
- acl.c.patch uploaded by festr (license 443) ........
- ................
-
-2008-03-27 13:42 +0000 [r111361-111411] Steve Murphy <murf@digium.com>
-
- * apps/app_playback.c, main/pbx.c, /: Merged revisions 111410 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r111410 | murf | 2008-03-27 07:29:41 -0600 (Thu,
- 27 Mar 2008) | 17 lines Merged revisions 111391 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r111391 | murf | 2008-03-27 07:03:28 -0600 (Thu, 27 Mar 2008) | 9
- lines These small documentation updates made in response to a
- query in asterisk-users, where a user was using Playback, but
- needed the features of Background, and had no idea that
- Background existed, or that it might provide the features he
- needed. I thought the best way to avert these kinds of queries
- was to provide "See Also" references in all three of
- "Background", "Playback", "WaitExten". Perhaps a project to do
- this with all related apps is in order. ........ ................
-
- * res/ael/pval.c, /, include/asterisk/pval.h, pbx/pbx_ael.c,
- include/asterisk/ael_structs.h: Merged revisions 111360 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r111360 | murf | 2008-03-26 22:47:12 -0600 (Wed,
- 26 Mar 2008) | 23 lines Merged revisions 111341 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r111341 | murf | 2008-03-26 21:21:05 -0600 (Wed, 26 Mar 2008) |
- 15 lines (closes issue #12302) Reported by: pj Tested by: murf
- These changes will set a channel variable ~~EXTEN~~ just before
- generating code for a switch, with the value of ${EXTEN}. The
- exten is marked as having a switch, and ever after that, till the
- end of the exten, we substitute any ${EXTEN} with ${~~EXTEN~~}
- instead in application arguments; (and the ${EXTEN: also). The
- reason for this, is that because switches are coded using
- separate extensions to provide pattern matching, and jumping
- to/from these switch extensions messes up the ${EXTEN} value,
- which blows the minds of users. ........ ................
-
-2008-03-27 00:36 +0000 [r111247-111339] Jason Parker <jparker@digium.com>
-
- * main/frame.c, /: Merged revisions 111285 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r111285 | qwell | 2008-03-26 19:25:56 -0500 (Wed, 26 Mar 2008) |
- 9 lines Merged revisions 111280 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r111280 | qwell | 2008-03-26 19:25:13 -0500 (Wed, 26 Mar 2008) |
- 1 line Put this flag back so we don't change the API. ........
- ................
-
- * main/frame.c, /: Merged revisions 111246 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r111246 | qwell | 2008-03-26 18:27:33 -0500 (Wed, 26 Mar 2008) |
- 17 lines Merged revisions 111245 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r111245 | qwell | 2008-03-26 18:26:33 -0500 (Wed, 26 Mar 2008) |
- 9 lines Remove excessive smoother optimization that was causing
- audio glitches (small "pops") after (about 200ms later) an
- "incorrectly" sized frame was received. While it would be very
- nice to keep this as optimized as possible, it makes no sense for
- the smoother to be dropping random bits of audio like this. Isn't
- that the whole point of a smoother? Closes issue #12093. ........
- ................
-
-2008-03-26 19:57 +0000 [r111131] Joshua Colp <jcolp@digium.com>
-
- * contrib/scripts/autosupport, /: Merged revisions 111130 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r111130 | file | 2008-03-26 16:56:40 -0300 (Wed,
- 26 Mar 2008) | 14 lines Merged revisions 111129 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r111129 | file | 2008-03-26 16:55:08 -0300 (Wed, 26 Mar 2008) | 6
- lines Update autosupport script. (closes issue #12310) Reported
- by: angler Patches: autosupport.diff uploaded by angler (license
- 106) ........ ................
-
-2008-03-26 19:53 +0000 [r111128] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, UPGRADE.txt: Merged revisions 111127 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r111127 | kpfleming | 2008-03-26 14:52:27 -0500 (Wed, 26 Mar
- 2008) | 18 lines Merged revisions 111126 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r111126 | kpfleming | 2008-03-26 14:51:24 -0500
- (Wed, 26 Mar 2008) | 10 lines Merged revisions 111125 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r111125 | kpfleming | 2008-03-26 14:49:30 -0500 (Wed, 26 Mar
- 2008) | 2 lines update UPGRADE notes to document usage of the
- script ........ ................ ................
-
-2008-03-26 19:41 +0000 [r111124] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 111123 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r111123 | mmichelson | 2008-03-26 14:39:23 -0500
- (Wed, 26 Mar 2008) | 12 lines Merged revisions 111121 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r111121 | mmichelson | 2008-03-26 14:37:36 -0500 (Wed, 26 Mar
- 2008) | 4 lines This code change is made just for clarification.
- It does exactly the same thing as before. It just doesn't look as
- wrong. ........ ................
-
-2008-03-26 19:27 +0000 [r111072] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 111067 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r111067 | mmichelson | 2008-03-26 14:26:23 -0500
- (Wed, 26 Mar 2008) | 17 lines Merged revisions 111049 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r111049 | mmichelson | 2008-03-26 14:22:16 -0500 (Wed, 26 Mar
- 2008) | 9 lines Add a lock to the vm_state structure and use the
- lock around mail_open calls to prevent concurrent access of the
- same mailstream. This, along with trunk's ability to configure
- TCP timeouts for IMAP storage will help to prevent crashes and
- hangs when using voicemail with IMAP storage. (closes issue
- #10487) Reported by: ewilhelmsen ........ ................
-
-2008-03-26 19:08 +0000 [r111026] Kevin P. Fleming <kpfleming@digium.com>
-
- * codecs/ilbc, /, contrib/scripts/get_ilbc_source.sh (added):
- Merged revisions 111025 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r111025 | kpfleming | 2008-03-26 14:08:00 -0500 (Wed, 26 Mar
- 2008) | 18 lines Merged revisions 111024 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r111024 | kpfleming | 2008-03-26 14:06:56 -0500
- (Wed, 26 Mar 2008) | 10 lines Merged revisions 111019 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r111019 | kpfleming | 2008-03-26 13:58:37 -0500 (Wed, 26 Mar
- 2008) | 2 lines add a script to make getting the iLBC source code
- simple for end users ........ ................ ................
-
-2008-03-26 19:06 +0000 [r111018-111023] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 111021 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r111021 | file | 2008-03-26 16:05:42 -0300 (Wed, 26 Mar 2008) |
- 12 lines Merged revisions 111020 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r111020 | file | 2008-03-26 16:04:35 -0300 (Wed, 26 Mar 2008) | 4
- lines If we are requested to authenticate a reinvite make sure
- that it contains T38 SDP if need be. (closes issue #11995)
- Reported by: fall ........ ................
-
- * /, channels/chan_iax2.c: Merged revisions 111017 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r111017 | file | 2008-03-26 15:42:52 -0300 (Wed,
- 26 Mar 2008) | 12 lines Merged revisions 110628 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4
- lines Add an option (transmit_silence) which transmits silence
- during both Record() and DTMF generation. The reason this is an
- option is that in order to transmit silence we have to setup a
- translation path. This may not be needed/wanted in all cases.
- (closes issue #10058) Reported by: tracinet ........
- ................
-
-2008-03-26 17:44 +0000 [r110964] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, UPGRADE.txt: Merged revisions 110963 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r110963 | kpfleming | 2008-03-26 12:44:09 -0500 (Wed, 26 Mar
- 2008) | 10 lines Merged revisions 110962 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r110962 | kpfleming | 2008-03-26 12:43:02 -0500 (Wed, 26 Mar
- 2008) | 2 lines add note that the user will need to enable
- codec_ilbc to get it to build ........ ................
-
-2008-03-26 17:35 +0000 [r110959] Donny Kavanagh <donnyk@gmail.com>
-
- * /, doc/snmp.txt: Merged revisions 110911 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r110911 |
- juggie | 2008-03-26 13:24:54 -0400 (Wed, 26 Mar 2008) | 8 lines
- update documentation to reflect the changes in the way configure
- detects net-snmp. (closes issue #12067) Reported by: juggie
- Patches: 12067_snmp_doc.patch uploaded by juggie (license 24)
- Tested by: juggie ........
-
-2008-03-26 17:15 +0000 [r110882] Kevin P. Fleming <kpfleming@digium.com>
-
- * codecs/ilbc/constants.h (removed), codecs/ilbc/iLBC_decode.h
- (removed), codecs/ilbc/iCBSearch.c (removed), codecs/Makefile,
- codecs/ilbc/filter.c (removed), codecs/ilbc/hpInput.c (removed),
- codecs/ilbc/gainquant.c (removed), codecs/ilbc/hpOutput.c
- (removed), codecs/ilbc/iCBSearch.h (removed),
- codecs/ilbc/filter.h (removed), codecs/ilbc/hpInput.h (removed),
- codecs/ilbc/gainquant.h (removed), codecs/ilbc/LPCencode.c
- (removed), codecs/ilbc/hpOutput.h (removed),
- codecs/ilbc/StateSearchW.c (removed), codecs/codec_ilbc.c,
- codecs/ilbc/LPCencode.h (removed), codecs/ilbc/iCBConstruct.c
- (removed), codecs/ilbc/StateSearchW.h (removed),
- codecs/ilbc/syntFilter.c (removed), /, codecs/ilbc/iCBConstruct.h
- (removed), codecs/ilbc/syntFilter.h (removed),
- codecs/ilbc/packing.c (removed), codecs/ilbc/StateConstructW.c
- (removed), codecs/ilbc/packing.h (removed),
- codecs/ilbc/libilbc.vcproj (removed),
- codecs/ilbc/StateConstructW.h (removed), codecs/ilbc/LPCdecode.c
- (removed), codecs/ilbc/getCBvec.c (removed),
- codecs/ilbc/enhancer.c (removed), codecs/ilbc/lsf.c (removed),
- codecs/ilbc/iLBC_encode.c (removed), codecs/ilbc/getCBvec.h
- (removed), codecs/ilbc/LPCdecode.h (removed),
- codecs/ilbc/enhancer.h (removed), codecs/ilbc/FrameClassify.c
- (removed), codecs/ilbc/iLBC_define.h (removed), codecs/ilbc/lsf.h
- (removed), codecs/ilbc/iLBC_encode.h (removed),
- codecs/ilbc/FrameClassify.h (removed), codecs/ilbc/helpfun.c
- (removed), codecs/ilbc/doCPLC.c (removed),
- codecs/ilbc/anaFilter.c (removed), codecs/ilbc/helpfun.h
- (removed), codecs/ilbc/createCB.c (removed), codecs/ilbc/doCPLC.h
- (removed), codecs/ilbc/anaFilter.h (removed), UPGRADE.txt,
- codecs/ilbc/constants.c (removed), codecs/ilbc/iLBC_decode.c
- (removed), codecs/ilbc/createCB.h (removed), CHANGES: Merged
- revisions 110881 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r110881 | kpfleming | 2008-03-26 10:10:28 -0700 (Wed, 26 Mar
- 2008) | 18 lines Merged revisions 110880 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r110880 | kpfleming | 2008-03-26 09:42:35 -0700
- (Wed, 26 Mar 2008) | 10 lines Merged revisions 110869 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r110869 | kpfleming | 2008-03-26 08:53:46 -0700 (Wed, 26 Mar
- 2008) | 2 lines due to licensing restrictions, we cannot
- distribute the source code for iLBC encoding and decoding... so
- remove it, and add instructions on how the user can obtain it
- themselves ........ ................ ................
-
-2008-03-26 15:33 +0000 [r110866-110868] Joshua Colp <jcolp@digium.com>
-
- * /: Merged revisions 110726 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r110726 |
- jpeeler | 2008-03-25 17:02:57 -0300 (Tue, 25 Mar 2008) | 2 lines
- This one line change makes an if inside a for loop (in
- realtime_peer) check all the ast_variables the loop was intending
- to test rather than just the first one. ........
-
-2008-03-26 00:03 +0000 [r110832] Mark Michelson <mmichelson@digium.com>
-
- * main/manager.c, /: Merged revisions 110831 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r110831 |
- mmichelson | 2008-03-25 19:02:31 -0500 (Tue, 25 Mar 2008) | 6
- lines This ensures that the manager interface is not enabled by
- default. Prior to this change, it was possible to start Asterisk
- with the manager interface enabled, then either comment out the
- enabled option or make manager.conf unopenable and the manager
- interface would still be enabled. ........
-
-2008-03-25 22:52 +0000 [r110781] Jason Parker <jparker@digium.com>
-
- * cdr/cdr_custom.c, /: Merged revisions 110780 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r110780 | qwell | 2008-03-25 17:51:55 -0500 (Tue, 25 Mar 2008) |
- 14 lines Merged revisions 110779 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r110779 | qwell | 2008-03-25 17:51:17 -0500 (Tue, 25 Mar 2008) |
- 6 lines Make file access in cdr_custom similar to cdr_csv. Fixes
- issue #12268. Patch borrowed from r82344 ........
- ................
-
-2008-03-25 22:11 +0000 [r110778] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_sip.c: This one line change makes an if inside a
- for loop (in realtime_peer) check all the ast_variables the loop
- was intending to test rather than just the first one.
-
-2008-03-25 17:47 +0000 [r110690-110692] Tilghman Lesher <tlesher@digium.com>
-
- * configs/extensions.conf.sample, /, configs/voicemail.conf.sample:
- Merged revisions 110691 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r110691 |
- tilghman | 2008-03-25 12:46:34 -0500 (Tue, 25 Mar 2008) | 6 lines
- Update sample configurations to make virtual hosting more
- obvious. (closes issue #11969) Reported by: pprindeville Patches:
- acme-virtualpbx.1.6.patch uploaded by pprindeville (license 347)
- ........
-
- * configs/extensions.conf.sample, /: Merged revisions 110689 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r110689 | tilghman | 2008-03-25 12:40:28 -0500 (Tue, 25
- Mar 2008) | 6 lines Update the sample configuration, to use Macro
- less (since it's now deprecated). (closes issue #12293) Reported
- by: pprindeville Patches: bugid-0012293.1.6.patch uploaded by
- pprindeville (license 347) ........
-
-2008-03-25 15:43 +0000 [r110637-110638] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: Oops.
-
- * /, channels/chan_sip.c: Merged revisions 110636 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r110636 | mmichelson | 2008-03-25 10:41:33 -0500 (Tue, 25 Mar
- 2008) | 15 lines Merged revisions 110635 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r110635 | mmichelson | 2008-03-25 10:40:33 -0500 (Tue, 25 Mar
- 2008) | 7 lines When reverting a commit, I accidentally left in
- this bit which was an experiment to see what would happen. It
- passed the compile test, and I didn't notice I had left this
- change in too. So this is a revert of a revert...sort of.
- ........ ................
-
-2008-03-25 15:39 +0000 [r110630-110634] Joshua Colp <jcolp@digium.com>
-
- * include/asterisk/options.h, main/asterisk.c, Makefile, /,
- main/app.c: Merged revisions 110629 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r110629 | file | 2008-03-25 11:39:45 -0300 (Tue, 25 Mar 2008) |
- 12 lines Merged revisions 110628 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4
- lines Add an option (transmit_silence) which transmits silence
- during both Record() and DTMF generation. The reason this is an
- option is that in order to transmit silence we have to setup a
- translation path. This may not be needed/wanted in all cases.
- (closes issue #10058) Reported by: tracinet ........
- ................
-
-2008-03-24 20:14 +0000 [r110620-110622] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 110619 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r110619 | mmichelson | 2008-03-24 14:19:37 -0500 (Mon, 24 Mar
- 2008) | 23 lines Merged revisions 110618 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r110618 | mmichelson | 2008-03-24 14:17:41 -0500 (Mon, 24 Mar
- 2008) | 15 lines This is a revert for revision 108288. The reason
- is that that revision was not for an actual bug fix per se, and
- so it really should not have been in 1.4 in the first place.
- Plus, people who compile with DO_CRASH are more likely to
- encounter a crash due to this change. While I think the usage of
- DO_CRASH in ast_sched_del is a bit absurd, this sort of change is
- beyond the scope of 1.4 and should be done instead in a developer
- branch based on trunk so that all scheduler functions are fixed
- at once. I also am reverting the change to trunk and 1.6 since
- they also suffer from the DO_CRASH potential. (closes issue
- #12272) Reported by: qq12345 ........ ................
-
-2008-03-24 17:36 +0000 [r110616] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 110615 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r110615 | russell | 2008-03-24 12:36:04 -0500
- (Mon, 24 Mar 2008) | 10 lines Merged revisions 110614 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r110614 | russell | 2008-03-24 12:34:56 -0500 (Mon, 24 Mar 2008)
- | 2 lines Turn a NOTICE into a DEBUG message. ........
- ................
-
-2008-03-24 15:29 +0000 [r110611] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 110610 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r110610 |
- file | 2008-03-24 12:28:25 -0300 (Mon, 24 Mar 2008) | 6 lines
- Only print out the set_address_from_contact host verbose message
- if debugging is enabled on the dialog. (closes issue #12280)
- Reported by: rjain Patches: chan_sip.c.diff uploaded by rjain
- (license 226) ........
-
-2008-03-21 21:52 +0000 [r110579] Jason Parker <jparker@digium.com>
-
- * /, sounds/Makefile: Merged revisions 110578 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r110578 |
- qwell | 2008-03-21 16:52:06 -0500 (Fri, 21 Mar 2008) | 1 line
- Update to 1.4.11 core sounds. ........
-
-2008-03-21 15:25 +0000 [r110501] Russell Bryant <russell@digium.com>
-
- * /, configs/sip.conf.sample, CHANGES: Merged revisions 110499 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r110499 | russell | 2008-03-21 10:24:43 -0500 (Fri, 21
- Mar 2008) | 3 lines Note that the TCP and TLS support is
- currently considered experimental and is subject to change while
- we work out the remaining issues. ........
-
-2008-03-21 14:36 +0000 [r110476] Jason Parker <jparker@digium.com>
-
- * /, codecs/gsm/Makefile: Merged revisions 110475 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r110475 | qwell | 2008-03-21 09:36:17 -0500 (Fri, 21 Mar 2008) |
- 15 lines Merged revisions 110474 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r110474 | qwell | 2008-03-21 09:32:52 -0500 (Fri, 21 Mar 2008) |
- 7 lines Don't attempt to do optimizations of gsm on mips
- platforms either. (closes issue #12270) Reported by: zandbelt
- Patches: 026-gsm-mips.patch uploaded by zandbelt (license 33)
- ........ ................
-
-2008-03-20 23:14 +0000 [r110304-110397] Russell Bryant <russell@digium.com>
-
- * main/autoservice.c, /: Merged revisions 110396 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r110396 | russell | 2008-03-20 18:14:13 -0500 (Thu, 20 Mar 2008)
- | 17 lines Merged revisions 110395 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r110395 | russell | 2008-03-20 18:13:56 -0500 (Thu, 20 Mar 2008)
- | 9 lines Shorten the ast_waitfor() timeout from 500 ms to 50 ms
- in the autoservice thread. This really should not make a
- difference except in very rare cases. That case would be that all
- of the channels in autoservice are not generating any frames. In
- that case, this change reduces the potential amount of time that
- a thread waits in ast_autoservice_stop() for the autoservice
- thread to wrap back around to the beginning of its loop. (closes
- issue #12266, reported by dimas) ........ ................
-
- * codecs/codec_g722.c, /: Merged revisions 110339 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r110339 |
- russell | 2008-03-20 17:02:20 -0500 (Thu, 20 Mar 2008) | 3 lines
- Use the correct buffer for g722tolin16_sample. This shouldn't
- have caused any problems, but Qwell noticed the typo here.
- ........
-
- * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions
- 110337 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r110337 | russell | 2008-03-20 16:55:50 -0500 (Thu, 20 Mar 2008)
- | 22 lines Merged revisions 110336 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r110336 | russell | 2008-03-20 16:54:58 -0500
- (Thu, 20 Mar 2008) | 14 lines Merged revisions 110335 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r110335 | russell | 2008-03-20 16:53:27 -0500 (Thu, 20 Mar 2008)
- | 6 lines Fix some very broken code that was introduced in 1.2.26
- as a part of the security fix. The dnsmgr is not appropriate
- here. The dnsmgr takes a pointer to an address structure that a
- background thread continuously updates. However, in these cases,
- a stack variable was passed. That means that the dnsmgr thread
- would be continuously writing to bogus memory. ........
- ................ ................
-
- * /, main/file.c: Merged revisions 110303 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r110303 |
- russell | 2008-03-20 15:08:26 -0500 (Thu, 20 Mar 2008) | 8 lines
- Fix a bug when using zaptel timing for playing back files that
- have a sample rate other than 8 kHz. The issue here is that
- format modules give a "whennext" sample value, which is used to
- calculate when to set a timer for to retrieve the next frame.
- However, the zaptel timer operates on 8 kHz samples, so this must
- be taken into account. (another part of issue #12164, reported by
- milazzo and jsmith, patch by me) ........
-
-2008-03-20 18:02 +0000 [r110273] Mark Michelson <mmichelson@digium.com>
-
- * main/dial.c, /: Merged revisions 110272 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r110272 |
- mmichelson | 2008-03-20 13:01:36 -0500 (Thu, 20 Mar 2008) | 3
- lines Add missing unlock ........
-
-2008-03-20 17:45 +0000 [r110269-110271] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /, res/res_musiconhold.c: Merged revisions 110268
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r110268 | russell | 2008-03-20 12:41:22 -0500 (Thu, 20
- Mar 2008) | 27 lines Add some fixes that I made in regards to
- wideband codec handling to get G.722 music on hold working for
- me. (issue #12164, reported by milazzo and jsmith, patches by me)
- res/res_musiconhold.c: - I moved a single line so that the sample
- queue update happened before ast_write(). The reason that this
- was a bug is that the G.722 frame originally says it has 320
- samples in it (which is correct). However, when the frame is
- written to a channel that uses RTP, main/rtp.c modifies the frame
- to cut the number of samples in half before it sends it on the
- wire. This is to account for the stupid incorrect G.722 spec that
- makes it so we have to lie about the number of samples with RTP.
- I should probably go and re-work the RTP code so it doesn't
- modify the frame so that a bug like this won't happen in the
- future. However, this change to MOH is harmless. main/channel.c:
- - I made two fixes in regards to generator timing. Generators use
- samples for timing. However, this code assumed 8 kHz samples. In
- one case, it was a hard coded 160 samples, that is now written as
- the sample rate / 50. The other place was dealing with timing a
- generator based on frames coming from the other direction.
- However, that would have only worked if the sample rates for the
- formats in both directions were the same. The code now takes into
- account that the sample rates may differ, and scales the
- generator samples accordingly. ........
-
-2008-03-19 23:00 +0000 [r110165] Russell Bryant <russell@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 110164 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r110164 | russell | 2008-03-19 17:58:33 -0500 (Wed, 19 Mar 2008)
- | 13 lines Merged revisions 110163 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r110163 | russell | 2008-03-19 17:57:59 -0500 (Wed, 19 Mar 2008)
- | 5 lines Fix a bug where when calls on the trunk side hang up
- while on hold, the state is not properly reflected. (closes issue
- #11990, reported by anakaoka, patched by me) ........
- ................
-
-2008-03-19 21:06 +0000 [r110088] Jeff Peeler <jpeeler@digium.com>
-
- * /: marking rev 110087 from trunk as not applying
-
-2008-03-19 20:37 +0000 [r110085] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_chanspy.c, /: Merged revisions 110084 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r110084 | mmichelson | 2008-03-19 15:34:13 -0500 (Wed, 19 Mar
- 2008) | 12 lines Merged revisions 110083 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r110083 | mmichelson | 2008-03-19 15:33:03 -0500 (Wed, 19 Mar
- 2008) | 4 lines Add a missing unlock in the case that memory
- allocation fails in app_chanspy. Thanks to Russell for confirming
- that this was an issue. ........ ................
-
-2008-03-19 19:14 +0000 [r110037] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 110036 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r110036 | file | 2008-03-19 16:13:39 -0300 (Wed,
- 19 Mar 2008) | 12 lines Merged revisions 110035 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r110035 | file | 2008-03-19 16:11:33 -0300 (Wed, 19 Mar 2008) | 4
- lines Add sanity checking for position resuming. We *have* to
- make sure that the position does not exceed the total number of
- files present, and we have to make sure that the position's
- filename is the same as previous. These values can change if a
- music class is reloaded and give unpredictable behavior. (closes
- issue #11663) Reported by: junky ........ ................
-
-2008-03-19 19:00 +0000 [r110024-110032] Russell Bryant <russell@digium.com>
-
- * Makefile, build_tools/cflags.xml, build_tools/cflags-devmode.xml
- (added), /: Merged revisions 109974 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r109974 | qwell | 2008-03-19 12:15:14 -0500 (Wed, 19 Mar 2008) |
- 13 lines Merged revisions 109973 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r109973 | qwell | 2008-03-19 12:12:52 -0500 (Wed, 19 Mar 2008) |
- 5 lines People report bugs about Asterisk crashing with DO_CRASH
- enabled was getting a little silly... Now we only show certain
- cflags when you run configure with --enable-dev-mode
- (corresponding menuselect change to follow) ........
- ................
-
-2008-03-19 18:26 +0000 [r109971-110021] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c, /: Merged revisions 110020 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r110020 | file | 2008-03-19 15:25:33 -0300 (Wed, 19 Mar 2008) |
- 14 lines Merged revisions 110019 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r110019 | file | 2008-03-19 15:20:28 -0300 (Wed, 19 Mar 2008) | 6
- lines Make sure that the mark bit does not incorrectly cause
- video frame timestamps to be calculated as if they are audio
- frames. (closes issue #11429) Reported by: sperreault Patches:
- 11429-frametype.diff uploaded by qwell (license 4) ........
- ................
-
-2008-03-19 16:46 +0000 [r109969] Steve Murphy <murf@digium.com>
-
- * main/config.c, /: Merged revisions 109942 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r109942 | murf | 2008-03-19 10:24:51 -0600 (Wed, 19 Mar 2008) |
- 80 lines Merged revisions 109908 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r109908 | murf | 2008-03-19 09:41:13 -0600 (Wed, 19 Mar 2008) |
- 72 lines (closes issue #11442) Reported by: tzafrir Patches:
- 11442.patch uploaded by murf (license 17) Tested by: murf I
- didn't give tzafrir very much time to test this, but if he does
- still have remaining issues, he is welcome to re-open this bug,
- and we'll do what is called for. I reproduced the problem, and
- tested the fix, so I hope I am not jumping by just going ahead
- and committing the fix. The problem was with what file_save does
- with templates; firstly, it tended to print out multiple options:
- [my_category](!)(templateref) instead of
- [my_category](!,templateref) which is fixed by this patch.
- Nextly, the code to suppress output of duplicate declarations
- that would occur because the reader copies inherited declarations
- down the hierarchy, was not working. Thus: [master-template](!)
- mastervar = bar [template](!,master-template) tvar = value
- [cat](template) catvar = val would be rewritten as: ;! ;!
- Automatically generated configuration file ;! Filename:
- experiment.conf (/etc/asterisk/experiment.conf) ;! Generator:
- Manager ;! Creation Date: Tue Mar 18 23:17:46 2008 ;!
- [master-template](!) mastervar = bar
- [template](!,master-template) mastervar = bar tvar = value
- [cat](template) mastervar = bar tvar = value catvar = val This
- has been fixed. Since the config reader 'explodes' inherited vars
- into the category, users may, in certain circumstances, see
- output different from what they originally entered, but it should
- be both correct and equivalent. ........ ................
-
-2008-03-19 04:06 +0000 [r109834-109840] Russell Bryant <russell@digium.com>
-
- * /, main/utils.c: Merged revisions 109839 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r109839 | russell | 2008-03-18 23:06:31 -0500 (Tue, 18 Mar 2008)
- | 10 lines Merged revisions 109838 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r109838 | russell | 2008-03-18 23:06:05 -0500 (Tue, 18 Mar 2008)
- | 2 lines Tweak spacing in a recent change because I'm very
- picky. ........ ................
-
- * apps/app_chanspy.c, /: Merged revisions 109764 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r109764 | russell | 2008-03-18 17:36:02 -0500 (Tue, 18 Mar 2008)
- | 11 lines Merged revisions 109763 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r109763 | russell | 2008-03-18 17:34:42 -0500 (Tue, 18 Mar 2008)
- | 3 lines Fix one place where the chanspy datastore isn't removed
- from a channel. (issue #12243, reported by atis, patch by me)
- ........ ................
-
-2008-03-18 23:23 +0000 [r109779] Tilghman Lesher <tlesher@digium.com>
-
- * /, configs/res_ldap.conf.sample, res/res_config_ldap.c: Merged
- revisions 109775 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r109775 |
- tilghman | 2008-03-18 18:22:25 -0500 (Tue, 18 Mar 2008) | 3 lines
- Change back to using ldap_initialize() and let the user specify a
- URL directly, instead of trying to piece it together, badly.
- ........
-
-2008-03-18 21:03 +0000 [r109716] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 109714 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r109714 | mmichelson | 2008-03-18 15:59:02 -0500 (Tue, 18 Mar
- 2008) | 20 lines Merged revisions 109713 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r109713 | mmichelson | 2008-03-18 15:52:15 -0500 (Tue, 18 Mar
- 2008) | 12 lines This patch makes it so that all queue member
- status changes are handled through device state code. This
- removes several problems people were seeing where their queue
- members would get into an "unknown" state. Huge props go to atis
- on this one since he was the one who found the code section that
- was causing the problem and proposed the solution. I just wrote
- what he suggested :) (closes issue #12127) Reported by: atis
- Patches: 12127v3.patch uploaded by putnopvut (license 60) Tested
- by: atis, jvandal ........ ................
-
-2008-03-18 20:14 +0000 [r109684] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_config_ldap.c: Merged revisions 109683 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r109683 | tilghman | 2008-03-18 15:13:40 -0500 (Tue, 18 Mar 2008)
- | 4 lines Set protocol version, port number correctly. (closes
- issue #12211, closes issue #12209) Reported by: sylvain ........
-
-2008-03-18 19:24 +0000 [r109654] Jason Parker <jparker@digium.com>
-
- * /, codecs/log2comp.h: Merged revisions 109651 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r109651 | qwell | 2008-03-18 14:24:15 -0500 (Tue, 18 Mar 2008) |
- 15 lines Merged revisions 109648 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r109648 | qwell | 2008-03-18 14:23:44 -0500 (Tue, 18 Mar 2008) |
- 7 lines Allow codecs that use log2comp (g726) to compile
- correctly on x86 with gcc4 optimizations. (closes issue #12253)
- Reported by: fossil Patches: log2comp.patch uploaded by fossil
- (license 140) ........ ................
-
-2008-03-18 19:00 +0000 [r109546-109622] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 109576 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r109576 | mmichelson | 2008-03-18 12:59:18 -0500
- (Tue, 18 Mar 2008) | 14 lines Merged revisions 109575 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r109575 | mmichelson | 2008-03-18 12:58:11 -0500 (Tue, 18 Mar
- 2008) | 6 lines Make sure an agent doesn't try to send dtmf to a
- NULL channel closes issue #12242 Reported by Yourname ........
- ................
-
- * include/asterisk/astmm.h, /: Merged revisions 109545 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r109545 | mmichelson | 2008-03-18 12:00:53 -0500 (Tue, 18 Mar
- 2008) | 3 lines Add format attribute to printf-style functions in
- astmm.h ........
-
-2008-03-18 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.6.0-beta6 released.
-
-2008-03-18 17:01 +0000 [r109546] Mark Michelson <mmichelson@digium.com>
-
- * include/asterisk/astmm.h, /: Merged revisions 109545 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r109545 | mmichelson | 2008-03-18 12:00:53 -0500 (Tue, 18 Mar
- 2008) | 3 lines Add format attribute to printf-style functions in
- astmm.h ........
-
-2008-03-18 16:26 +0000 [r109487] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
- channels/chan_misdn.c, /: Merged revisions 109475 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r109475 | kpfleming | 2008-03-18 11:23:05 -0500 (Tue, 18 Mar
- 2008) | 2 lines fix up various warnings found via the addition of
- format string checking... some of these were really, really bad
- code ........
-
-2008-03-18 15:58 +0000 [r109454-109459] Russell Bryant <russell@digium.com>
-
- * Makefile, channels/chan_misdn.c, include/asterisk/strings.h,
- res/res_indications.c, utils/extconf.c, main/asterisk.c,
- apps/app_voicemail.c, utils/check_expr.c,
- cdr/cdr_sqlite3_custom.c, apps/app_meetme.c, /,
- res/res_phoneprov.c, main/utils.c, channels/chan_iax2.c,
- utils/frame.c, main/cli.c, funcs/func_enum.c, main/manager.c,
- include/asterisk/astobj.h, res/res_agi.c, main/features.c,
- apps/app_minivm.c, res/res_realtime.c, res/res_config_ldap.c,
- include/asterisk/utils.h, channels/chan_sip.c,
- apps/app_festival.c, main/translate.c, main/jitterbuf.c,
- utils/astman.c, include/jitterbuf.h, apps/app_queue.c: Merged
- revisions 109447 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r109447 |
- twilson | 2008-03-18 10:43:34 -0500 (Tue, 18 Mar 2008) | 3 lines
- Go through and fix a bunch of places where character strings were
- being interpreted as format strings. Most of these changes are
- solely to make compiling with -Wsecurity and -Wformat=2 happy,
- and were not actual problems, per se. I also added format
- attributes to any printf wrapper functions I found that didn't
- have them. -Wsecurity and -Wmissing-format-attribute added to
- --enable-dev-mode. ........
-
- * configs/sip_notify.conf.sample, /: Merged revisions 109111 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r109111 | qwell | 2008-03-17 11:37:31 -0500 (Mon, 17 Mar
- 2008) | 10 lines Add sample events for aastra phones.
- aastra-check-cfg is the same as the other check-cfg entries, and
- aastra-xml is to load a pre-configured xml script. (closes issue
- #12229) Reported by: gowen72 Patches: aastra.patch uploaded by
- gowen72 (license 432) ........
-
-2008-03-18 15:50 +0000 [r109453] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, configure, include/asterisk/autoconfig.h.in, acinclude.m4:
- Merged revisions 109451 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r109451 |
- kpfleming | 2008-03-18 10:50:29 -0500 (Tue, 18 Mar 2008) | 2
- lines ensure that dependencies on AST_C_DEFINE_CHECK symbols work
- properly ........
-
-2008-03-18 15:50 +0000 [r109448-109452] Russell Bryant <russell@digium.com>
-
- * main/dial.c, /: Merged revisions 108962 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r108962 | mvanbaak | 2008-03-16 16:50:58 -0500 (Sun, 16 Mar 2008)
- | 15 lines Merged revisions 108961 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r108961 | mvanbaak | 2008-03-16 22:47:10 +0100 (Sun, 16 Mar 2008)
- | 7 lines add missing break to case AST_CONTROL_SRCUPDATE (closes
- issue #12228) Reported by: andrew Patches: SRC.patch uploaded by
- andrew (license 240) ........ ................
-
-2008-03-18 15:16 +0000 [r109398] Joshua Colp <jcolp@digium.com>
-
- * main/manager.c, /, main/logger.c: Merged revisions 109396 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r109396 | file | 2008-03-18 12:13:07 -0300 (Tue, 18 Mar
- 2008) | 3 lines Make sure values are interpreted as character
- strings and not format strings. (AST-2008-004) ........
-
-2008-03-18 15:14 +0000 [r109397] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael-test/ael-ntest23 (added),
- pbx/ael/ael-test/ael-ntest23/t1/a.ael,
- pbx/ael/ael-test/ael-ntest23/t1/b.ael,
- pbx/ael/ael-test/ael-ntest23/t1/c.ael,
- pbx/ael/ael-test/ael-ntest23/t2/d.ael,
- pbx/ael/ael-test/ael-ntest23/t2/e.ael,
- pbx/ael/ael-test/ael-ntest23/t2/f.ael, res/ael/ael_lex.c,
- pbx/ael/ael-test/ref.ael-ntest23 (added),
- pbx/ael/ael-test/ael-ntest23/t3/g.ael,
- pbx/ael/ael-test/ael-ntest23/t3/h.ael,
- pbx/ael/ael-test/ael-ntest23/t3/i.ael, res/ael/ael.flex,
- pbx/ael/ael-test/ael-ntest23/t3/j.ael,
- pbx/ael/ael-test/ael-ntest23/qq.ael,
- pbx/ael/ael-test/ael-ntest23/t1, pbx/ael/ael-test/ael-ntest23/t2,
- pbx/ael/ael-test/ael-ntest23/t3, /,
- pbx/ael/ael-test/ael-ntest23/extensions.ael: Merged revisions
- 109357 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r109357 | murf | 2008-03-18 08:09:50 -0600 (Tue, 18 Mar 2008) |
- 25 lines Merged revisions 109309 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r109309 | murf | 2008-03-18 00:37:15 -0600 (Tue, 18 Mar 2008) |
- 17 lines (closes issue #11903) Reported by: atis Many thanks to
- atis for spotting this problem and reporting it. The fix was to
- straighten out how items are placed on and removed from the file
- stack. Regressions as well as the provided test case helped to
- straighten out all code paths. valgrind was used to make sure all
- memory allocated was freed. Sorry for not solving this earlier. I
- got distracted. Added the ntest23 regression test, which is
- mainly a copy of ntest22, but with a few juicy errors thrown in,
- to replicate the kind of error that atis spotted. ........
- ................
-
-2008-03-18 15:11 +0000 [r109395] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 109389 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r109389 |
- qwell | 2008-03-18 10:07:04 -0500 (Tue, 18 Mar 2008) | 3 lines Do
- not return with a successful authentication if the From header
- ends up empty. (AST-2008-003) ........
-
-2008-03-18 15:09 +0000 [r109392] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c, /, channels/chan_sip.c: Merged revisions 109390 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r109390 | file | 2008-03-18 12:08:09 -0300 (Tue,
- 18 Mar 2008) | 11 lines Merged revisions 109386 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r109386 | file | 2008-03-18 11:58:39 -0300 (Tue, 18 Mar 2008) | 3
- lines Put a maximum limit on the number of payloads accepted, and
- also make sure a given payload does not exceed our maximum value.
- (AST-2008-002) ........ ................
-
-2008-03-18 00:40 +0000 [r109283] Sean Bright <sean.bright@gmail.com>
-
- * /, configure, configure.ac: Merged revisions 109282 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r109282 | seanbright | 2008-03-17 20:28:39 -0400 (Mon, 17 Mar
- 2008) | 1 line Fix a typo ........
-
-2008-03-17 22:24 +0000 [r109254] Terry Wilson <twilson@digium.com>
-
- * build_tools/cflags.xml, /, build_tools/menuselect-deps.in,
- configure, include/asterisk/autoconfig.h.in, main/Makefile,
- configure.ac, main/http.c, main/minimime (removed),
- build_tools/make_buildopts_h, makeopts.in: Merged revisions
- 109229 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r109229 |
- twilson | 2008-03-17 17:10:06 -0500 (Mon, 17 Mar 2008) | 5 lines
- Replace minimime with superior GMime library so that the entire
- contents of an http post are not read into memory. This does
- introduce a dependency on the GMime library for handling HTTP
- POSTs, but it is available in most distros. If the library is
- present, then the compile flag for ENABLE_UPLOADS is enabled by
- default in menuselect. ........
-
-2008-03-17 22:07 +0000 [r109228] Mark Michelson <mmichelson@digium.com>
-
- * /, main/utils.c: Merged revisions 109227 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r109227 | mmichelson | 2008-03-17 17:06:44 -0500 (Mon, 17 Mar
- 2008) | 20 lines Merged revisions 109226 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r109226 | mmichelson | 2008-03-17 17:05:49 -0500 (Mon, 17 Mar
- 2008) | 12 lines Fix a logic flaw in the code that stores lock
- info which is displayed via the "core show locks" command. The
- idea behind this section of code was to remove the previous lock
- from the list if it was a trylock that had failed. Unfortunately,
- instead of checking the status of the previous lock, we were
- referencing the index immediately following the previous lock in
- the lock_info->locks array. The result of this problem, under the
- right circumstances, was that the lock which we currently in the
- process of attempting to acquire could "overwrite" the previous
- lock which was acquired. While this does not in any way affect
- typical operation, it *could* lead to misleading "core show
- locks" output. ........ ................
-
-2008-03-17 18:11 +0000 [r109175] Michiel van Baak <michiel@vanbaak.info>
-
- * /, channels/chan_skinny.c: Merged revisions 109168 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r109168 | mvanbaak | 2008-03-17 18:43:46 +0100 (Mon, 17 Mar 2008)
- | 11 lines 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) ........
-
-2008-03-17 17:42 +0000 [r109167] Kevin P. Fleming <kpfleming@digium.com>
-
- * Makefile, /, configure, configure.ac, acinclude.m4: Merged
- revisions 109166 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r109166 |
- kpfleming | 2008-03-17 12:31:46 -0500 (Mon, 17 Mar 2008) | 3
- lines don't define Zaptel features as libraries, they aren't, and
- we don't want '--with-zaptel-<foo>' configure options for them
- also some minor cleanups ........
-
-2008-03-17 16:47 +0000 [r109109-109114] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 109108 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r109108 | file | 2008-03-17 13:26:36 -0300 (Mon, 17 Mar 2008) |
- 12 lines Merged revisions 109107 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r109107 | file | 2008-03-17 13:24:29 -0300 (Mon, 17 Mar 2008) | 4
- lines 200 OKs in response to a reinvite need to be sent reliably.
- If the remote side does not receive one the dialog will be torn
- down. (closes issue #12208) Reported by: atrash ........
- ................
-
-2008-03-17 14:21 +0000 [r109027] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_chanspy.c, /: Merged revisions 109024 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r109024 | mmichelson | 2008-03-17 09:21:14 -0500 (Mon, 17 Mar
- 2008) | 14 lines Merged revisions 109012 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r109012 | mmichelson | 2008-03-17 09:18:26 -0500 (Mon, 17 Mar
- 2008) | 6 lines Make sure that we release the lock on the spyee
- channel if the spyee or spy has hung up (closes issue #12232)
- Reported by: atis ........ ................
-
-2008-03-16 17:56 +0000 [r108928-108930] Russell Bryant <russell@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 108927 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r108927 | russell | 2008-03-16 12:53:46 -0500 (Sun, 16 Mar 2008)
- | 7 lines Fix polling for mailbox changes in mailboxes that are
- not in the default vm context. (closes issue #12223) Reported by:
- DEA Patches: vm-polled-imap.txt uploaded by DEA (license 3)
- ........
-
-2008-03-15 16:21 +0000 [r108741-108895] Russell Bryant <russell@digium.com>
-
- * Makefile, /: Merged revisions 108799 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r108799 |
- russell | 2008-03-14 15:14:06 -0500 (Fri, 14 Mar 2008) | 8 lines
- Make sure configure is run before menuselect on a clean checkout
- (closes issue #12197) Reported by: juggie Patches: 12197.diff
- uploaded by juggie (license 24) ........
-
- * channels/chan_oss.c, /: Merged revisions 108797 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r108797 | russell | 2008-03-14 15:09:37 -0500 (Fri, 14 Mar 2008)
- | 13 lines Merged revisions 108796 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r108796 | russell | 2008-03-14 15:09:22 -0500 (Fri, 14 Mar 2008)
- | 5 lines Fix a channel name issue. chan_oss registers the
- "Console" channel type, but it created channels with an "OSS"
- prefix. (closes issue #12194, reported by davidw, patched by me)
- ........ ................
-
- * contrib/init.d/rc.suse.asterisk, /: Merged revisions 108793 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r108793 | russell | 2008-03-14 15:04:56 -0500
- (Fri, 14 Mar 2008) | 12 lines Merged revisions 108792 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r108792 | russell | 2008-03-14 15:04:35 -0500 (Fri, 14 Mar 2008)
- | 4 lines Update the SuSE init script to start networking before
- asterisk, as well. (closes issue #12200, reported by and change
- suggested by reinerotto) ........ ................
-
- * /, configure, acinclude.m4: Merged revisions 108740 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r108740 | russell | 2008-03-14 12:05:11 -0500 (Fri, 14 Mar 2008)
- | 5 lines Do a link test in AST_EXT_TOOL_CHECK() to ensure we
- have all the required libs reported by the tool. (closes issue
- #12067, reported by Juggie, patched by me) ........
-
-2008-03-14 16:54 +0000 [r108739] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 108738 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r108738 | mmichelson | 2008-03-14 11:52:51 -0500 (Fri, 14 Mar
- 2008) | 41 lines Merged revisions 108737 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r108737 | mmichelson | 2008-03-14 11:44:08 -0500 (Fri, 14 Mar
- 2008) | 33 lines Fix a race condition in the SIP packet scheduler
- which could cause a crash. chan_sip uses the scheduler API in
- order to schedule retransmission of reliable packets (such as
- INVITES). If a retransmission of a packet is occurring, then the
- packet is removed from the scheduler and retrans_pkt is called.
- Meanwhile, if a response is received from the packet as
- previously transmitted, then when we ACK the response, we will
- remove the packet from the scheduler and free the packet. The
- problem is that both the ACK function and retrans_pkt attempt to
- acquire the same lock at the beginning of the function call. This
- means that if the ACK function acquires the lock first, then it
- will free the packet which retrans_pkt is about to read from and
- write to. The result is a crash. The solution: 1. If the ACK
- function fails to remove the packet from the scheduler and the
- retransmit id of the packet is not -1 (meaning that we have not
- reached the maximum number of retransmissions) then release the
- lock and yield so that retrans_pkt may acquire the lock and
- operate. 2. Make absolutely certain that the ACK function does
- not recursively lock the lock in question. If it does, then
- releasing the lock will do no good, since retrans_pkt will still
- be unable to acquire the lock. (closes issue #12098) Reported by:
- wegbert (closes issue #12089) Reported by: PTorres Patches:
- 12098-putnopvutv3.patch uploaded by putnopvut (license 60) Tested
- by: jvandal ........ ................
-
-2008-03-14 14:33 +0000 [r108684] Jason Parker <jparker@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 108683 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r108683 | qwell | 2008-03-14 09:32:55 -0500
- (Fri, 14 Mar 2008) | 12 lines Merged revisions 108682 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r108682 | qwell | 2008-03-14 09:29:05 -0500 (Fri, 14 Mar 2008) |
- 4 lines Fix a potential segfault if chan (or chan->music_state)
- is NULL. Closes issue #12210, credit to edantie for pointing this
- out. ........ ................
-
-2008-03-13 21:48 +0000 [r108587] Mark Michelson <mmichelson@digium.com>
-
- * main/manager.c, /: Merged revisions 108586 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r108586 |
- mmichelson | 2008-03-13 16:47:55 -0500 (Thu, 13 Mar 2008) | 3
- lines Make this compile ........
-
-2008-03-13 21:41 +0000 [r108585] Russell Bryant <russell@digium.com>
-
- * apps/app_chanspy.c, main/channel.c, /,
- include/asterisk/channel.h: Merged revisions 108584 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r108584 | russell | 2008-03-13 16:40:43 -0500
- (Thu, 13 Mar 2008) | 19 lines Merged revisions 108583 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r108583 | russell | 2008-03-13 16:38:16 -0500 (Thu, 13 Mar 2008)
- | 11 lines Fix another issue that was causing crashes in chanspy.
- This introduces a new datastore callback, called chan_fixup().
- The concept is exactly like the fixup callback that is used in
- the channel technology interface. This callback gets called when
- the owning channel changes due to a masquerade. Before this was
- introduced, if a masquerade happened on a channel being spyed on,
- the channel pointer in the datastore became invalid. (closes
- issue #12187) (reported by, and lots of testing from atis) (props
- to file for the help with ideas) ........ ................
-
-2008-03-13 21:31 +0000 [r108582] Mark Michelson <mmichelson@digium.com>
-
- * main/manager.c, /: Merged revisions 108529 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r108529 |
- mmichelson | 2008-03-13 15:59:00 -0500 (Thu, 13 Mar 2008) | 11
- lines Fixing a potential buffer overflow in the manager command
- ModuleCheck. Though this overflow is exploitable remotely, we are
- NOT issuing a security advisory for this since in order to
- exploit the overflow, the attacker would have to establish an
- authenticated manager session AND have the system privilege. By
- gaining this privilege, the attacker already has more powerful
- weapons at his disposal than overflowing a buffer with a
- malformed manager header, so the vulnerability in this case
- really lies with the authentication method that allowed the
- attacker to gain the system privilege in the first place.
- ........
-
-2008-03-13 21:07 +0000 [r108347-108532] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 108531 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r108531 | russell | 2008-03-13 16:06:52 -0500 (Thu, 13 Mar 2008)
- | 18 lines Merged revisions 108530 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r108530 | russell | 2008-03-13 16:06:33 -0500 (Thu, 13 Mar 2008)
- | 10 lines Make a tweak that gets the LEDs on polycom phones to
- blink when an extension that has been subscribed to goes on hold.
- Otherwise, they just stay on like it does when an extension is in
- use. (closes issue #11263) Reported by: russell Patches:
- notify_hold.rev1.txt uploaded by russell (license 2) Tested by:
- russell ........ ................
-
- * apps/app_voicemail.c, /: Merged revisions 108508 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r108508 | russell | 2008-03-13 15:35:28 -0500 (Thu, 13 Mar 2008)
- | 2 lines Fix a place where configuration values could cause an
- overflow of a buffer. ........
-
- * /, apps/app_followme.c: Merged revisions 108472 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r108472 | russell | 2008-03-13 15:26:59 -0500 (Thu, 13 Mar 2008)
- | 12 lines Merged revisions 108469 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r108469 | russell | 2008-03-13 15:26:28 -0500 (Thu, 13 Mar 2008)
- | 4 lines Fix a couple uses of sprintf. The second one could
- actually cause an overflow of a stack buffer. It's not a security
- issue though, it only depends on your configuration. ........
- ................
-
- * /, main/features.c: Merged revisions 107465 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r107465 |
- file | 2008-03-11 10:05:17 -0500 (Tue, 11 Mar 2008) | 4 lines
- Clarify comment about masquerading and playback of the parking
- slot. (closes issue #12180) Reported by: davidw ........
-
- * /, channels/chan_sip.c: Merged revisions 107157 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r107157 |
- file | 2008-03-10 15:00:21 -0500 (Mon, 10 Mar 2008) | 4 lines If
- we receive a 488 on a T38 request reinvite back to audio. As well
- reinvite across a bridge back to audio if one side doesn't
- negotiate to T38. (closes issue #8677) Reported by: alex-911
- ........
-
- * /: Merged revisions 106892 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r106892 |
- mattf | 2008-03-07 16:36:49 -0600 (Fri, 07 Mar 2008) | 1 line
- Make sure we don't start a call when we have already done so in
- response to a COT message ........
-
- * /, main/editline/Makefile.in: Merged revisions 106843 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r106843 | qwell | 2008-03-07 16:15:20 -0600
- (Fri, 07 Mar 2008) | 13 lines Merged revisions 106842 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r106842 | qwell | 2008-03-07 16:14:45 -0600 (Fri, 07 Mar 2008) |
- 5 lines Fix hardcoded grep in editline, were GNU grep is
- required. (closes issue #12124) Reported by: dmartin ........
- ................
-
- * include/asterisk/http.h, main/tcptls.c, main/manager.c, /,
- channels/chan_sip.c, res/res_phoneprov.c, main/http.c,
- include/asterisk/tcptls.h: Merged revisions 108295 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r108295 | russell | 2008-03-12 17:13:18 -0500 (Wed, 12 Mar 2008)
- | 3 lines Rename ast_tcptls_server_instance to session_instance,
- since this pertains to server and client usage. ........
-
- * /, main/http.c: Merged revisions 108346 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r108346 |
- russell | 2008-03-12 17:49:26 -0500 (Wed, 12 Mar 2008) | 4 lines
- Make the default prefix empty, like it was in Asterisk 1.4.
- (closes issue #12198, reported by bkruse, patched by me) ........
-
-2008-03-12 22:10 +0000 [r108246-108294] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 108293 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r108293 |
- mmichelson | 2008-03-12 17:09:52 -0500 (Wed, 12 Mar 2008) | 3
- lines Let's get this to compile ........
-
- * /, channels/chan_sip.c: Merged revisions 108289 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r108289 | mmichelson | 2008-03-12 16:57:41 -0500 (Wed, 12 Mar
- 2008) | 22 lines Merged revisions 108288 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r108288 | mmichelson | 2008-03-12 16:53:46 -0500 (Wed, 12 Mar
- 2008) | 14 lines Change AST_SCHED_DEL use to ast_sched_del for
- autocongestion in chan_sip. The scheduler callback will always
- return 0. This means that this id is never rescheduled, so it
- makes no sense to loop trying to delete the id from the scheduler
- queue. If we fail to remove the item from the queue once, it will
- fail every single time. (Yes I realize that in this case, the
- macro would exit early because the id is set to -1 in the
- callback, but it still makes no sense to use that macro in favor
- of calling ast_sched_del once and being done with it) This is the
- first of potentially several such fixes. ........
- ................
-
- * /, include/asterisk/sched.h: Merged revisions 108238 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r108238 | mmichelson | 2008-03-12 16:19:30 -0500
- (Wed, 12 Mar 2008) | 20 lines Merged revisions 108227 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r108227 | mmichelson | 2008-03-12 16:16:28 -0500 (Wed, 12 Mar
- 2008) | 12 lines Added a large comment before the AST_SCHED_DEL
- macro to explain its purpose as well as when it is appropriate
- and when it is not appropriate to use it. I also removed the part
- of the debug message that mentions that this is probably a bug
- because there are some perfectly legitimate places where
- ast_sched_del may fail to delete an entry (e.g. when the
- scheduler callback manually reschedules with a new id instead of
- returning non-zero to tell the scheduler to reschedule with the
- same idea). I also raised the debug level of the debug message in
- AST_SCHED_DEL since it seems like it could come up quite
- frequently since the macro is probably being used in several
- places where it shouldn't be. Also removed the redundant line,
- file, and function information since that is provided by ast_log.
- ........ ................
-
-2008-03-12 20:29 +0000 [r108205] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 108191 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r108191 | kpfleming | 2008-03-12 15:27:01 -0500 (Wed, 12 Mar
- 2008) | 14 lines Merged revisions 108086 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r108086 | kpfleming | 2008-03-12 14:16:07 -0500 (Wed, 12 Mar
- 2008) | 6 lines if we receive an INVITE with a Content-Length
- that is not a valid number, or is zero, then don't process the
- rest of the message body looking for an SDP closes issue #11475
- Reported by: andrebarbosa ........ ................
-
-2008-03-12 19:59 +0000 [r108138] Russell Bryant <russell@digium.com>
-
- * apps/app_chanspy.c, main/channel.c, /: Merged revisions 108137
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r108137 | russell | 2008-03-12 14:59:05 -0500
- (Wed, 12 Mar 2008) | 48 lines Merged revisions 108135 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r108135 | russell | 2008-03-12 14:57:42 -0500 (Wed, 12 Mar 2008)
- | 40 lines (closes issue #12187, reported by atis, fixed by me
- after some brainstorming on the issue with mmichelson) - Update
- copyright info on app_chanspy. - Fix a race condition that caused
- app_chanspy to crash. The issue was that the chanspy datastore
- magic that was used to ensure that spyee channels did not
- disappear out from under the code did not completely solve the
- problem. It was actually possible for chanspy to acquire a
- channel reference out of its datastore to a channel that was in
- the middle of being destroyed. That was because datastore
- destruction in ast_channel_free() was done near the end. So, this
- left the code in app_chanspy accessing a channel that was
- partially, or completely invalid because it was in the process of
- being free'd by another thread. The following sort of shows the
- code path where the race occurred:
- =============================================================================
- Thread 1 (PBX thread for spyee chan) || Thread 2 (chanspy)
- --------------------------------------||-------------------------------------
- ast_channel_free() || - remove channel from channel list || -
- lock/unlock the channel to ensure || that no references retrieved
- from || the channel list exist. ||
- --------------------------------------||-------------------------------------
- || channel_spy() - destroy some channel data || - Lock chanspy
- datastore || - Retrieve reference to channel || - lock channel ||
- - Unlock chanspy datastore
- --------------------------------------||-------------------------------------
- - destroy channel datastores || - call chanspy datastore d'tor ||
- which NULL's out the ds' || - Operate on the channel ...
- reference to the channel || || - free the channel || || || -
- unlock the channel
- --------------------------------------||-------------------------------------
- =============================================================================
- ........ ................
-
-2008-03-12 18:31 +0000 [r108085] Joshua Colp <jcolp@digium.com>
-
- * apps/app_mixmonitor.c, /, include/asterisk/audiohook.h,
- main/audiohook.c: Merged revisions 108084 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r108084 | file | 2008-03-12 15:29:33 -0300 (Wed, 12 Mar 2008) |
- 12 lines Merged revisions 108083 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r108083 | file | 2008-03-12 15:26:37 -0300 (Wed, 12 Mar 2008) | 4
- lines Add a trigger mode that triggers on both read and write.
- The actual function that returns the combined audio frame though
- will wait until both sides have fed in audio, or until one side
- stops (such as the case when you call Wait). (closes issue
- #11945) Reported by: xheliox ........ ................
-
-2008-03-12 17:03 +0000 [r108033] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /: Merged revisions 108032 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r108032 | russell | 2008-03-12 12:02:57 -0500 (Wed, 12 Mar 2008)
- | 12 lines Merged revisions 108031 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r108031 | russell | 2008-03-12 11:59:07 -0500 (Wed, 12 Mar 2008)
- | 4 lines Destroy the channel lock after the channel datastores.
- (inspired by issue #12187) ........ ................
-
-2008-03-12 07:44 +0000 [r107879-107999] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 107998 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r107998 |
- tilghman | 2008-03-12 02:43:03 -0500 (Wed, 12 Mar 2008) | 7 lines
- Deadlock fixes (closes issue #12143) Reported by: kactus Patches:
- 20080312__bug12143__2.diff.txt uploaded by Corydon76 (license 14)
- Tested by: kactus ........
-
- * main/loader.c, /, apps/app_dumpchan.c, apps/app_zapras.c: Merged
- revisions 107960 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r107960 |
- tilghman | 2008-03-12 00:46:39 -0500 (Wed, 12 Mar 2008) | 4 lines
- Revert several changes from revision 102525, as the changes were
- not compatible, and, in fact, introduced regressions. (Closes
- issue #12190) ........
-
- * contrib/scripts/iax-friends.sql, /,
- contrib/scripts/sip-friends.sql: Merged revisions 107878 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r107878 | tilghman | 2008-03-11 20:54:00 -0500
- (Tue, 11 Mar 2008) | 10 lines Merged revisions 107877 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r107877 | tilghman | 2008-03-11 20:52:40 -0500 (Tue, 11 Mar 2008)
- | 2 lines Document all of the possible realtime fields ........
- ................
-
-2008-03-11 23:38 +0000 [r107828] Jason Parker <jparker@digium.com>
-
- * /, doc/voicemail_odbc_postgresql.txt: Merged revisions 107827 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r107827 | qwell | 2008-03-11 18:38:00 -0500
- (Tue, 11 Mar 2008) | 15 lines Merged revisions 107826 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r107826 | qwell | 2008-03-11 18:37:05 -0500 (Tue, 11 Mar 2008) |
- 7 lines Update documentation for pgsql ODBC voicemail. (closes
- issue #12186) Reported by: jsmith Patches:
- vm_pgsql_doc_update.patch uploaded by jsmith (license 15)
- ........ ................
-
-2008-03-11 22:59 +0000 [r107723-107793] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_config_sqlite.c, main/config.c, res/res_config_curl.c,
- res/res_config_pgsql.c, res/res_config_odbc.c, /,
- include/asterisk/config.h, res/res_config_ldap.c: Merged
- revisions 107791 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r107791 |
- tilghman | 2008-03-11 17:55:16 -0500 (Tue, 11 Mar 2008) | 5 lines
- An offhand comment from Russell made me realize that the
- configuration file caching would not work properly for users.conf
- and any other file read from more than one place. I needed to add
- the filename which requested the config file to get it to work
- properly. ........
-
-2008-03-11 20:54 +0000 [r107720] Jason Parker <jparker@digium.com>
-
- * channels/chan_jingle.c, /, channels/chan_gtalk.c: Merged
- revisions 107718 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r107718 | qwell | 2008-03-11 15:53:48 -0500 (Tue, 11 Mar 2008) |
- 13 lines Merged revisions 107714 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r107714 | qwell | 2008-03-11 15:49:56 -0500 (Tue, 11 Mar 2008) |
- 5 lines Copy voicemail dependency logic for res_adsi to
- chan_gtalk and chan_jingle (for jabber). (closes issue #12014)
- Reported by: junky ........ ................
-
-2008-03-11 20:51 +0000 [r107716] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, Makefile.rules, channels/Makefile: Merged revisions 107715 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r107715 | kpfleming | 2008-03-11 15:50:57 -0500
- (Tue, 11 Mar 2008) | 10 lines Merged revisions 107713 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r107713 | kpfleming | 2008-03-11 15:48:58 -0500 (Tue, 11 Mar
- 2008) | 2 lines get chan_vpb to build properly in dev mode
- ........ ................
-
-2008-03-11 20:37 +0000 [r107584-107711] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_page.c: Merged revisions 107710 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r107710 |
- file | 2008-03-11 17:36:14 -0300 (Tue, 11 Mar 2008) | 6 lines
- Dial a device even if it's state is unknown. (closes issue
- #12184) Reported by: bluecrow76 Patches:
- asterisk-svn-app_page.c.devicestate_unknown.diff uploaded by
- bluecrow76 (license 270) ........
-
- * /, main/features.c: Merged revisions 107659 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r107659 | file | 2008-03-11 16:23:28 -0300 (Tue, 11 Mar 2008) |
- 12 lines Merged revisions 107646 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r107646 | file | 2008-03-11 16:20:01 -0300 (Tue, 11 Mar 2008) | 4
- lines Make sure the visible indication is on the right channel so
- when the masquerade happens the proper indication is enacted.
- (closes issue #11707) Reported by: iam ........ ................
-
- * /, apps/app_meetme.c: Merged revisions 107638 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r107638 | file | 2008-03-11 15:48:59 -0300 (Tue, 11 Mar 2008) |
- 12 lines Merged revisions 107637 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r107637 | file | 2008-03-11 15:47:33 -0300 (Tue, 11 Mar 2008) | 4
- lines Add an additional check for setting conference parameter
- when using the marked user options. It was possible for it to
- return to a no listen/no talk state if a masquerade happened.
- (closes issue #12136) Reported by: aragon ........
- ................
-
-2008-03-11 15:39 +0000 [r107374-107526] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_vpb.cc, /: Merged revisions 107525 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r107525 | kpfleming | 2008-03-11 10:39:37 -0500 (Tue, 11 Mar
- 2008) | 2 lines fix another potential bug found by gcc 4.3
- ........
-
- * apps/app_rpt.c, channels/misdn/isdn_lib.c, codecs/Makefile, /,
- apps/app_sms.c: Merged revisions 107466 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r107466 | kpfleming | 2008-03-11 10:13:38 -0500 (Tue, 11 Mar
- 2008) | 10 lines Merged revisions 107464 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r107464 | kpfleming | 2008-03-11 09:53:03 -0500 (Tue, 11 Mar
- 2008) | 2 lines fix various other problems found by gcc 4.3
- ........ ................
-
- * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
- apps/app_sms.c: Merged revisions 107462 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r107462 | kpfleming | 2008-03-11 09:37:03 -0500 (Tue, 11 Mar
- 2008) | 10 lines Merged revisions 107461 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r107461 | kpfleming | 2008-03-11 09:33:45 -0500 (Tue, 11 Mar
- 2008) | 2 lines stop checking for mktime() in the configure
- script... we don't use it, and the test is buggy under gcc 4.3
- ........ ................
-
- * /, configure, main/Makefile, configure.ac, makeopts.in: Merged
- revisions 107409 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r107409 | kpfleming | 2008-03-11 09:09:49 -0500 (Tue, 11 Mar
- 2008) | 13 lines Merged revisions 107408 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r107408 | kpfleming | 2008-03-11 09:07:59 -0500 (Tue, 11 Mar
- 2008) | 5 lines check for compiler support for
- -fno-strict-overflow before using it (tested with Debian's gcc
- 4.3, 4.1 and 3.4) (closes issue #12179) Reported by: Netview
- ........ ................
-
- * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
- Merged revisions 107406 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r107406 | kpfleming | 2008-03-11 08:58:37 -0500 (Tue, 11 Mar
- 2008) | 10 lines Merged revisions 107405 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r107405 | kpfleming | 2008-03-11 08:57:08 -0500 (Tue, 11 Mar
- 2008) | 2 lines fix small bug in IMAP toolkit testing ........
- ................
-
- * main/udptl.c, utils/Makefile, /, main/Makefile,
- main/editline/readline.c, res/Makefile: Merged revisions 107373
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r107373 | kpfleming | 2008-03-11 06:36:51 -0500
- (Tue, 11 Mar 2008) | 19 lines Merged revisions 107352 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r107352 | kpfleming | 2008-03-11 06:04:29 -0500 (Tue, 11 Mar
- 2008) | 11 lines fix up various compiler warnings found with
- gcc-4.3: - the output of flex includes a static function called
- 'input' that is not used, so for the moment we'll stop having the
- compiler tell us about unused variables in the flex source files
- (a better fix would be to improve our flex post-processing to
- remove the unused function) - main/stdtime/localtime.c makes
- assumptions about signed integer overflow, and gcc-4.3's improved
- optimizer tries to take advantage of handling potential overflow
- conditions at compile time; for now, suppress these optimizations
- until we can fiure out if the code needs improvement -
- main/udptl.c has some references to uninitialized variables; in
- one case there was no bug, but in the other it was certainly
- possibly for unexpected behavior to occur -
- main/editline/readline.c had an unused variable ........
- ................
-
-2008-03-11 01:27 +0000 [r107336] Terry Wilson <twilson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 107292 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r107292 | twilson | 2008-03-10 20:09:46 -0500 (Mon, 10 Mar 2008)
- | 10 lines Merged revisions 107290 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r107290 | twilson | 2008-03-10 19:59:18 -0500 (Mon, 10 Mar 2008)
- | 2 lines If we fail to alloc a channel, we should re-lock the
- pvt structure before returning. ........ ................
-
-2008-03-10 23:46 +0000 [r107289] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, /: Merged revisions 107019 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r107019 |
- murf | 2008-03-10 08:55:21 -0600 (Mon, 10 Mar 2008) | 1 line way
- back in July, in r.75706, a fix was made ot the strftime usages,
- which was good, but in this case, the check for a nil time was
- accidentally removed, and now it is restored, to keep timevals
- like '1969-12-31 17:00:00' from showing up in the cdrs. No idea
- what databases will do with this. No bugs filed as yet, but it
- felt like a bug. ........
-
-2008-03-10 20:29 +0000 [r107180] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 107177 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r107177 | qwell | 2008-03-10 15:28:33 -0500 (Mon, 10 Mar 2008) |
- 13 lines Merged revisions 107173 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r107173 | qwell | 2008-03-10 15:27:08 -0500 (Mon, 10 Mar 2008) |
- 5 lines Make sure to reenable echo can after a "failed"
- (canceled, etc) three-way call. (closes issue #11335) Reported
- by: rebuild ........ ................
-
-2008-03-10 20:18 +0000 [r107101-107163] Russell Bryant <russell@digium.com>
-
- * main/pbx.c, /: Merged revisions 107162 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r107162 | russell | 2008-03-10 15:17:37 -0500 (Mon, 10 Mar 2008)
- | 16 lines Merged revisions 107161 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r107161 | russell | 2008-03-10 15:17:11 -0500 (Mon, 10 Mar 2008)
- | 8 lines Fix another bug specifically related to asynchronous
- call origination. Once the PBX is started on the channel using
- ast_pbx_start(), then the ownership of the channel has been
- passed on to another thread. We can no longer access it in this
- code. If the channel gets hung up very quickly, it is possible
- that we could access a channel that has been free'd. (inspired by
- BE-386) ........ ................
-
- * main/pbx.c, /: Merged revisions 107159 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r107159 | russell | 2008-03-10 15:05:12 -0500 (Mon, 10 Mar 2008)
- | 17 lines Merged revisions 107158 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r107158 | russell | 2008-03-10 15:04:27 -0500 (Mon, 10 Mar 2008)
- | 9 lines Fix some bugs related to originating calls. If the code
- failed to start a PBX on the channel (such as if you set a call
- limit based on the system's load average), then there were cases
- where a channel that has already been free'd using ast_hangup()
- got accessed. This caused weird memory corruption and crashes to
- occur. (fixes issue BE-386) (much debugging credit goes to
- twilson, final patch written by me) ........ ................
-
- * main/channel.c, /: Merged revisions 107103 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r107103 | russell | 2008-03-10 12:13:34 -0500 (Mon, 10 Mar 2008)
- | 10 lines Merged revisions 107102 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r107102 | russell | 2008-03-10 12:13:17 -0500 (Mon, 10 Mar 2008)
- | 2 lines Resolve a compiler warning. ........ ................
-
- * main/channel.c, /: Merged revisions 107100 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r107100 | russell | 2008-03-10 11:59:13 -0500 (Mon, 10 Mar 2008)
- | 11 lines Merged revisions 107099 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r107099 | russell | 2008-03-10 11:58:57 -0500 (Mon, 10 Mar 2008)
- | 3 lines Fix a race condition where the generator can go away
- (closes issue #12175, reported by edantie, patched by me)
- ........ ................
-
-2008-03-10 15:46 +0000 [r107069] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 107068 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r107068 |
- mmichelson | 2008-03-10 10:45:13 -0500 (Mon, 10 Mar 2008) | 10
- lines app_queue has now been doxygenified thanks to snuffy! The
- ony thing I changed was the way that locks are referenced, since
- the old 1.2 names were still used in the comments. (closes issue
- #11997) Reported by: snuffy Patches: bug_11997_queue_doxy.diff
- uploaded by snuffy (license 35) ........
-
-2008-03-10 14:38 +0000 [r107018] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c, main/cdr.c, /, include/asterisk/cdr.h: Merged
- revisions 107017 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r107017 | file | 2008-03-10 11:36:16 -0300 (Mon, 10 Mar 2008) |
- 15 lines Merged revisions 107016 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r107016 | file | 2008-03-10 11:33:02 -0300 (Mon, 10 Mar 2008) | 7
- lines Move where unanswered CDRs are dropped to the CDR core, not
- everything uses app_dial. (closes issue #11516) Reported by: ys
- Patches: branch_1.4_cdr.diff uploaded by ys (license 281) Tested
- by: anest, jcapp, dartvader ........ ................
-
-2008-03-08 17:54 +0000 [r106997] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Make sure we don't start a call on a channel
- that has already started a call
-
-2008-03-08 16:14 +0000 [r106947] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 106946 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r106946 | kpfleming | 2008-03-08 10:03:48 -0600 (Sat, 08 Mar
- 2008) | 10 lines Merged revisions 106945 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r106945 | kpfleming | 2008-03-08 09:59:42 -0600 (Sat, 08 Mar
- 2008) | 2 lines don't generate D-Channel "up" and "down" messages
- unless the channel state is actually changing; also, generate the
- "up" message when an implicit "up" occurs due to reception of a
- normal event when we thought the channel was "down" ........
- ................
-
-2008-03-07 22:53 +0000 [r106897] Russell Bryant <russell@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 106896 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r106896 | russell | 2008-03-07 16:52:46 -0600 (Fri, 07 Mar 2008)
- | 10 lines Merged revisions 106895 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r106895 | russell | 2008-03-07 16:51:23 -0600 (Fri, 07 Mar 2008)
- | 2 lines Only start the SLA thread if SLA has actually been
- configured. ........ ................
-
-2008-03-07 19:34 +0000 [r106790] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /: Merged revisions 106789 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r106789 | file | 2008-03-07 15:33:09 -0400 (Fri, 07 Mar 2008) |
- 12 lines Merged revisions 106788 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r106788 | file | 2008-03-07 15:32:00 -0400 (Fri, 07 Mar 2008) | 4
- lines Ignore source update control frame. (closes issue #12168)
- Reported by: plack ........ ................
-
-2008-03-07 17:18 +0000 [r106686-106713] Russell Bryant <russell@digium.com>
-
- * /, include/asterisk/sched.h: Merged revisions 106707 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r106707 | russell | 2008-03-07 11:17:30 -0600
- (Fri, 07 Mar 2008) | 16 lines Merged revisions 106704 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r106704 | russell | 2008-03-07 11:16:58 -0600 (Fri, 07 Mar 2008)
- | 8 lines Change a warning message to a debug message. This is
- happening quite frequently, and it is not worth spamming users
- with these messages unless we are pretty confident that it should
- never happen. As it stands today, it _will_ and _does_ happen and
- until that gets cleaned up a reasonable amount on the development
- side, let's not spam the logs of everyone else. (closes issue
- #12154) ........ ................
-
- * doc/smdi.txt, /: Merged revisions 106684 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r106684 |
- russell | 2008-03-07 10:31:48 -0600 (Fri, 07 Mar 2008) | 2 lines
- fix example usage ........
-
-2008-03-07 16:27 +0000 [r106554-106662] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 106654 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r106654 | tilghman | 2008-03-07 10:26:07 -0600
- (Fri, 07 Mar 2008) | 11 lines Merged revisions 106635 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r106635 | tilghman | 2008-03-07 10:22:11 -0600 (Fri, 07 Mar 2008)
- | 3 lines Warn the user when a temporary greeting exists (Closes
- issue #11409) ........ ................
-
- * main/rtp.c, /: Merged revisions 106607 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r106607 | tilghman | 2008-03-07 09:22:34 -0600 (Fri, 07 Mar 2008)
- | 11 lines Merged revisions 106606 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r106606 | tilghman | 2008-03-07 09:20:52 -0600 (Fri, 07 Mar 2008)
- | 3 lines Properly initialize rtp->schedid (Closes issue #12154)
- ........ ................
-
- * apps/app_chanspy.c, apps/app_rpt.c, main/asterisk.c,
- apps/app_speech_utils.c, apps/app_voicemail.c, main/channel.c,
- funcs/func_enum.c, channels/chan_misdn.c, main/frame.c, /,
- channels/chan_sip.c, funcs/func_odbc.c, funcs/func_strings.c,
- utils/extconf.c: Merged revisions 106553 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r106553 | tilghman | 2008-03-07 00:54:47 -0600 (Fri, 07 Mar 2008)
- | 14 lines Merged revisions 106552 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r106552 | tilghman | 2008-03-07 00:36:33 -0600 (Fri, 07 Mar 2008)
- | 6 lines Safely use the strncat() function. (closes issue
- #11958) Reported by: norman Patches: 20080209__bug11958.diff.txt
- uploaded by Corydon76 (license 14) ........ ................
-
-2008-03-07 01:19 +0000 [r106502-106520] Russell Bryant <russell@digium.com>
-
- * doc/smdi.txt, /: Merged revisions 106518 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r106518 |
- russell | 2008-03-06 19:19:02 -0600 (Thu, 06 Mar 2008) | 1 line
- minor text changes ........
-
- * doc/smdi.txt, /: Merged revisions 106507 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r106507 |
- russell | 2008-03-06 19:15:36 -0600 (Thu, 06 Mar 2008) | 2 lines
- Add updated SMDI documentation that I had only sitting in my
- email ... oops ........
-
- * main/rtp.c, codecs/codec_g722.c, /, formats/format_pcm.c,
- main/file.c: Merged revisions 106501 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r106501 |
- russell | 2008-03-06 18:24:58 -0600 (Thu, 06 Mar 2008) | 28 lines
- Merge changes from team/russell/g722-sillyness ... Fix a number
- of other places where the number of samples in a G722 frame was
- not properly handled because of various reasons. main/rtp.c: -
- When a G722 frame is read from the smoother, the number of
- samples in the frame must be divided by 2 before being sent out
- over the network. Even though G722 is 16 kHz, an error in some
- previous spec has made it so that we have to list the number of
- samples such as if it was 8 kHz. main/file.c: - When scheduling
- the next time to expect a frame, take into account that the
- format of the file we're reading from may not be 8 kHz.
- codecs/codec_g722.c: - When converting from G722 to slinear,
- g722_decode() expects its samples parameter to be in the silly
- (real samples / 2) format. Make it so. - When converting from
- slinear to G722, properly set the number of samples in the frame
- to be the number of bytes of output * 2. formats/format_pcm.c: -
- This format module handles G722, among a number of other formats.
- However, the read() and seek() functions did not account for the
- fact that G722 has 2 samples per byte. (closes issue #12130,
- reported by rickross, patched by me) ........
-
-2008-03-06 22:16 +0000 [r106442] Mark Michelson <mmichelson@digium.com>
-
- * main/pbx.c, /: Merged revisions 106438 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r106438 | mmichelson | 2008-03-06 16:11:26 -0600 (Thu, 06 Mar
- 2008) | 16 lines Merged revisions 106437 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r106437 | mmichelson | 2008-03-06 16:10:07 -0600 (Thu, 06 Mar
- 2008) | 8 lines Quell an annoying message that is likely to print
- every single time that ast_pbx_outgoing_app is called. The reason
- is that __ast_request_and_dial allocates the cdr for the channel,
- so it should be expected that the channel will have a cdr on it.
- Thanks to joetester on IRC for pointing this out ........
- ................
-
-2008-03-06 22:15 +0000 [r106440] Jason Parker <jparker@digium.com>
-
- * /, main/file.c: Merged revisions 106439 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r106439 |
- qwell | 2008-03-06 16:11:30 -0600 (Thu, 06 Mar 2008) | 8 lines
- Fix file playback in many cases. (closes issue #12115) Reported
- by: pj Patches: v2-fileexists.patch uploaded by dimas (license
- 88) (with modifications by me) Tested by: dimas, qwell, russell
- ........
-
-2008-03-06 20:39 +0000 [r106433] Donny Kavanagh <donnyk@gmail.com>
-
- * /, res/res_agi.c: Merged revisions 106399 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r106399 |
- juggie | 2008-03-06 14:31:50 -0500 (Thu, 06 Mar 2008) | 9 lines
- trivial fix for an agi error when attempting to use EAGI on a
- dead/hungup channel, we now print an error that makes sense given
- our removal of deadagi as an actual application. (closes issue
- #12161) Reported by: explidous Patches: res_agi_12161.patch
- uploaded by juggie (license 24) Tested by: juggie ........
-
-2008-03-06 05:25 +0000 [r106330-106359] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_config_ldap.c: Merged revisions 106346 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r106346 | tilghman | 2008-03-05 23:21:39 -0600 (Wed, 05 Mar 2008)
- | 7 lines Missing braces, fix parsing (closes issue #12112)
- Reported by: cyrenity Patches: res_config_ldap.patch-03-03-2008
- uploaded by cyrenity (license 416) Tested by: cyrenity, Corydon76
- ........
-
- * /, sounds/sounds.xml, sounds/Makefile: Merged revisions 106329
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r106329 | tilghman | 2008-03-05 22:45:16 -0600
- (Wed, 05 Mar 2008) | 10 lines Merged revisions 106328 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r106328 | tilghman | 2008-03-05 22:40:06 -0600 (Wed, 05 Mar 2008)
- | 2 lines Upgrade to the next release of sounds ........
- ................
-
-2008-03-06 00:23 +0000 [r106299-106320] Russell Bryant <russell@digium.com>
-
- * channels/chan_oss.c, main/rtp.c, main/channel.c,
- channels/chan_phone.c, main/dial.c, channels/chan_skinny.c,
- main/file.c, channels/chan_h323.c, channels/chan_alsa.c,
- include/asterisk/frame.h, channels/chan_mgcp.c,
- channels/chan_unistim.c, apps/app_dial.c, channels/chan_zap.c, /,
- channels/chan_sip.c, channels/chan_console.c,
- apps/app_followme.c: Merged revisions 106239 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r106239 | file | 2008-03-05 16:43:22 -0600 (Wed, 05 Mar 2008) |
- 12 lines 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
- ........ ................
-
- * /, channels/chan_iax2.c: Merged revisions 106238 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r106238 | russell | 2008-03-05 16:40:58 -0600
- (Wed, 05 Mar 2008) | 11 lines Merged revisions 106237 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r106237 | russell | 2008-03-05 16:37:09 -0600 (Wed, 05 Mar 2008)
- | 3 lines Fix a potential deadlock and a few different potential
- crashes. (closes issue #12145, reported by thiagarcia, patched by
- me) ........ ................
-
- * /, doc/tex/realtime.tex: Merged revisions 106186 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r106186 | mvanbaak | 2008-03-05 15:19:06 -0600 (Wed, 05 Mar 2008)
- | 7 lines document var_metric usage to prevent bugreports that
- are actually configuration issues (closes issue #12151) Reported
- by: caio1982 Patches: DB_metric3.diff uploaded by caio1982
- (license 22) ........
-
- * main/rtp.c, /, main/translate.c, include/asterisk/frame.h: Merged
- revisions 105933 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r105933 | russell | 2008-03-04 19:54:16 -0600 (Tue, 04 Mar 2008)
- | 13 lines Merged revisions 105932 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r105932 | russell | 2008-03-04 19:52:18 -0600 (Tue, 04 Mar 2008)
- | 5 lines Fix a bug that I just noticed in the RTP code. The
- calculation for setting the len field in an ast_frame of audio
- was wrong when G.722 is in use. The len field represents the
- number of ms of audio that the frame contains. It would have set
- the value to be twice what it should be. ........
- ................
-
- * funcs/func_global.c, /: Merged revisions 105899 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r105899 |
- russell | 2008-03-04 18:45:39 -0600 (Tue, 04 Mar 2008) | 3 lines
- Fix the SHARED() read callback to properly unlock the channel.
- This function could not have worked, as it left the channel
- locked in all cases. ........
-
- * main/manager.c, /: Merged revisions 105864 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r105864 |
- mmichelson | 2008-03-04 17:24:56 -0600 (Tue, 04 Mar 2008) | 5
- lines There are several places in manager.c where BUFSIZ is used
- for a buffer which will contain nowhere near that amount of data.
- This makes these buffers more reasonably sized. ........
-
- * main/asterisk.c, channels/chan_zap.c, /, channels/console_gui.c,
- apps/app_queue.c: Merged revisions 105841 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r105841 |
- tilghman | 2008-03-04 17:10:45 -0600 (Tue, 04 Mar 2008) | 2 lines
- Fix minor misuses of snprintf ........
-
- * main/rtp.c, main/netsock.c, main/cryptostub.c, main/file.c,
- main/callerid.c, main/alaw.c, main/dsp.c, main/dlfcn.c,
- main/frame.c, /, main/say.c, main/utils.c, main/enum.c,
- main/astobj2.c, main/config.c, main/fskmodem.c, main/poll.c,
- main/loader.c, main/term.c, main/cli.c, main/channel.c,
- main/dial.c, main/manager.c, main/tdd.c, main/strcompat.c,
- main/features.c, main/logger.c, main/app.c, main/image.c,
- main/dns.c, main/pbx.c, main/translate.c, main/jitterbuf.c:
- Merged revisions 105840 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r105840 |
- tilghman | 2008-03-04 17:04:29 -0600 (Tue, 04 Mar 2008) | 2 lines
- Whitespace changes only ........
-
- * main/tcptls.c, main/manager.c, /, channels/chan_sip.c,
- main/http.c, include/asterisk/tcptls.h: Merged revisions 105804
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r105804 | russell | 2008-03-04 16:28:03 -0600 (Tue, 04
- Mar 2008) | 2 lines add a destroy API call for a server instance
- ........
-
- * main/tcptls.c, main/manager.c, /, channels/chan_sip.c,
- main/http.c, include/asterisk/tcptls.h: Merged revisions 105785
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r105785 | russell | 2008-03-04 16:23:21 -0600 (Tue, 04
- Mar 2008) | 2 lines More public API name changes to use an
- appropriate ast_ prefix ........
-
- * include/asterisk/http.h, main/tcptls.c, main/manager.c, /,
- channels/chan_sip.c, res/res_phoneprov.c, main/http.c,
- include/asterisk/tcptls.h: Merged revisions 105773 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r105773 | russell | 2008-03-04 16:15:18 -0600 (Tue, 04 Mar 2008)
- | 2 lines Rename public object server_instance to
- ast_tcptls_server_instance ........
-
- * /, channels/chan_sip.c: Merged revisions 105734 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r105734 |
- russell | 2008-03-04 14:36:16 -0600 (Tue, 04 Mar 2008) | 6 lines
- Fix some bugs in the SIP tcp helper thread. - fix a spot where a
- lock wouldn't get unlocked in an error condition - call
- ast_mutex_destroy() on the lock before freeing its memory
- (related to issue #11972) ........
-
- * /, res/res_phoneprov.c: Merged revisions 105733 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r105733 |
- twilson | 2008-03-04 14:32:55 -0600 (Tue, 04 Mar 2008) | 2 lines
- Set username to default to the category name if it isn't
- overridden by a usernmae= setting in users.conf ........
-
- * main/rtp.c, /: Merged revisions 105677 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r105677 | file | 2008-03-04 12:11:38 -0600 (Tue, 04 Mar 2008) |
- 10 lines Merged revisions 105676 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r105676 | file | 2008-03-04 14:10:34 -0400 (Tue, 04 Mar 2008) | 2
- lines In addition to setting the marker bit let's change our ssrc
- so they know for sure it is a different source. ........
- ................
-
- * main/rtp.c, /, channels/chan_sip.c, include/asterisk/rtp.h:
- Merged revisions 105675 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r105675 | file | 2008-03-04 12:08:42 -0600 (Tue, 04 Mar 2008) |
- 16 lines Merged revisions 105674 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r105674 | file | 2008-03-04 14:05:28 -0400 (Tue, 04 Mar 2008) | 8
- lines When a new source of audio comes in (such as music on hold)
- make sure the marker bit gets set. (closes issue #10355) Reported
- by: wdecarne Patches: 10355.diff uploaded by file (license 11)
- (closes issue #11491) Reported by: kanderson ........
- ................
-
-2008-03-05 17:42 +0000 [r106140] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_talkdetect.c: Merged revisions 106139 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r106139 | tilghman | 2008-03-05 11:40:42 -0600 (Wed, 05 Mar 2008)
- | 3 lines Should check these values for non-NULL before scanning.
- (Closes issue #12147) ........
-
-2008-03-05 15:43 +0000 [r106041] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 106040 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r106040 | kpfleming | 2008-03-05 09:40:40 -0600 (Wed, 05 Mar
- 2008) | 15 lines Merged revisions 106038 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r106038 | kpfleming | 2008-03-05 09:32:35 -0600 (Wed, 05 Mar
- 2008) | 7 lines when a PRI call must be moved to a different B
- channel at the request of the other endpoint, ensure that any DSP
- active on the original channel is moved to the new one (closes
- issue #11917) Reported by: mavetju Tested by: mavetju ........
- ................
-
-2008-03-05 15:31 +0000 [r106037] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c, include/asterisk/sched.h: Merged
- revisions 106036 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r106036 | tilghman | 2008-03-05 09:23:32 -0600 (Wed, 05 Mar 2008)
- | 15 lines Merged revisions 106015 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r106015 | tilghman | 2008-03-05 09:17:16 -0600 (Wed, 05 Mar 2008)
- | 7 lines Correctly initialize retransid in SIP, and ensure that
- the warning when failing to delete a schedule entry can actually
- hit the log. (closes issue #12140) Reported by: slavon Patches:
- sch2.patch uploaded by slavon (license 288) (Patch slightly
- modified by me) ........ ................
-
-2008-03-04 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.6.0-beta5 released.
-
-2008-03-04 16:55 +0000 [r105574-105597] Russell Bryant <russell@digium.com>
-
- * CHANGES: Update CHANGES heading
-
- * funcs/func_version.c: Simplify a trivial snprintf() with
- ast_copy_string()
-
- * main/hashtab.c: Make it so you don't have to cast away const in a
- couple places
-
- * main/hashtab.c: remove unnecessary casts
-
- * main/pbx.c: - Add curly braces around the while loop - Properly
- break out of the loop on error when an included context is not
- found
-
- * main/pbx.c: Use ast_copy_string() instead of strncpy(), and use
- sizeof() instead of a magic number
-
- * channels/chan_zap.c: Fix some code that was improperly changed in
- revision 104866 from issue #12079. (closes issue #12129, reported
- by elguero, patched by me)
-
-2008-03-03 18:08 +0000 [r105573] Jason Parker <jparker@digium.com>
-
- * /, res/snmp/agent.c: Merged revisions 105572 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r105572 | qwell | 2008-03-03 12:06:52 -0600 (Mon, 03 Mar 2008) |
- 7 lines Fix types for astNumChannels and astConfigCallsProcessed.
- (closes issue #12114) Reported by: jeffg Patches: 12114.patch
- uploaded by jeffg (license 192) ........
-
-2008-03-03 17:17 +0000 [r105564-105571] Russell Bryant <russell@digium.com>
-
- * channels/chan_local.c, /: Merged revisions 105570 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r105570 | russell | 2008-03-03 11:16:53 -0600 (Mon, 03
- Mar 2008) | 3 lines In the case of an ast_channel allocation
- failure, take the local_pvt out of the pvt list before destroying
- it. ........
-
- * channels/chan_local.c, /: Merged revisions 105568 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r105568 | russell | 2008-03-03 11:05:16 -0600 (Mon, 03
- Mar 2008) | 3 lines Fix a potential memory leak of the local_pvt
- struct when ast_channel allocation fails. Also, in passing,
- centralize the code necessary to destroy a local_pvt. ........
-
- * main/autoservice.c, /: Merged revisions 105565 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r105565 | russell | 2008-03-03 10:01:50 -0600 (Mon, 03 Mar 2008)
- | 3 lines Update the copyright information for autoservice. Most
- of the code in this file now is stuff that I have written
- recently ... ........
-
- * main/channel.c, main/autoservice.c, /,
- include/asterisk/_private.h, main/asterisk.c: 3) In addition to
- merging the changes below, change trunk back to a regular LIST
- instead of an RWLIST. The way this list works makes it such that
- a RWLIST provides no additional benefit. Also, a mutex is needed
- for use with the thread condition. Merged revisions 105563 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r105563 | russell | 2008-03-03 09:50:43 -0600 (Mon, 03 Mar 2008)
- | 24 lines Merge in some changes from
- team/russell/autoservice-nochans-1.4 These changes fix up some
- dubious code that I came across while auditing what happens in
- the autoservice thread when there are no channels currently in
- autoservice. 1) Change it so that autoservice thread doesn't keep
- looping around calling ast_waitfor_n() on 0 channels twice a
- second. Instead, use a thread condition so that the thread
- properly goes to sleep and does not wake up until a channel is
- put into autoservice. This actually fixes an interesting bug, as
- well. If the autoservice thread is already running (almost always
- is the case), then when the thread goes from having 0 channels to
- have 1 channel to autoservice, that channel would have to wait
- for up to 1/2 of a second to have the first frame read from it.
- 2) Fix up the code in ast_waitfor_nandfds() for when it gets
- called with no channels and no fds to poll() on, such as was the
- case with the previous code for the autoservice thread. In this
- case, the code would call alloca(0), and pass the result as the
- first argument to poll(). In this case, the 2nd argument to
- poll() specified that there were no fds, so this invalid pointer
- shouldn't actually get dereferenced, but, this code makes it
- explicit and ensures the pointers are NULL unless we have valid
- data to put there. (related to issue #12116) ........
-
-2008-03-03 15:30 +0000 [r105558-105561] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /: Merged revisions 105560 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r105560 | file | 2008-03-03 11:28:59 -0400 (Mon, 03 Mar 2008) | 7
- lines It is possible for no audio to pass between the current
- digit and next digit so expand logic that clears emulation to
- AST_FRAME_NULL. (closes issue #11911) Reported by: edgreenberg
- Patches: v1-11911.patch uploaded by dimas (license 88) Tested by:
- tbsky ........
-
- * /, channels/chan_sip.c: Merged revisions 105557 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r105557 | file | 2008-03-03 11:15:39 -0400 (Mon, 03 Mar 2008) | 6
- lines Add a comment to describe some logic. (closes issue #12120)
- Reported by: flefoll Patches:
- chan_sip.c.br14.patch-just-a-comment uploaded by flefoll (license
- 244) ........
-
-2008-03-01 03:59 +0000 [r105509] Joshua Colp <jcolp@digium.com>
-
- * main/slinfactory.c: Add support for 16KHz signed linear.
-
-2008-03-01 02:03 +0000 [r105479] Tilghman Lesher <tlesher@digium.com>
-
- * /: Drop bad property
-
-2008-03-01 01:30 +0000 [r105477] Terry Wilson <twilson@digium.com>
-
- * apps/app_dial.c, include/asterisk/app.h,
- main/global_datastores.c, /, main/features.c, main/app.c,
- include/asterisk/global_datastores.h: Asterisk, when parking can
- drop rights a caller when a parking timeout occurs. Also, when
- doing built-in attended transfers, sometimes incorrectly passes
- rights from the transferrer to the transferee. This patch tries
- to fixes the parking issue and lays some groundwork for later
- fixing the transfer issue. (closes issue #11520) Reported by:
- pliew Tested by: otherwiseguy
-
-2008-03-01 00:53 +0000 [r105461] Russell Bryant <russell@digium.com>
-
- * CHANGES, funcs/func_devstate.c: Add a "devstate change" CLI
- command to control custom device states. Also, do some additional
- code cleanup and improvement in passing. (closes issue #12106)
- Reported by: nizon Patches: devstate-patch.txt uploaded by nizon
- (license 415) -- Updated to trunk, and tab completion added by me
-
-2008-02-29 23:53 +0000 [r105411] Tilghman Lesher <tlesher@digium.com>
-
- * cdr/cdr_adaptive_odbc.c: Convert to use ast_str
-
-2008-02-29 23:36 +0000 [r105410] Russell Bryant <russell@digium.com>
-
- * main/autoservice.c, /: Merged revisions 105409 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r105409 | russell | 2008-02-29 17:34:32 -0600 (Fri, 29 Feb 2008)
- | 23 lines Fix a major bug in autoservice. There was a race
- condition in the handling of the list of channels in autoservice.
- The problem was that it was possible for a channel to get removed
- from autoservice and destroyed, while the autoservice thread was
- still messing with the channel. This led to memory corruption,
- and caused crashes. This explains multiple backtraces I have seen
- that have references to autoservice, but do to the nature of the
- issue (memory corruption), could cause crashes in a number of
- areas. (fixes the crash in BE-386) (closes issue #11694) (closes
- issue #11940) The following issues could be related. If you are
- the reporter of one of these, please update to include this fix
- and try again. (potentially fixes issue #11189) (potentially
- fixes issue #12107) (potentially fixes issue #11573) (potentially
- fixes issue #12008) (potentially fixes issue #11189) (potentially
- fixes issue #11993) (potentially fixes issue #11791) ........
-
-2008-02-29 18:34 +0000 [r105378] Joshua Colp <jcolp@digium.com>
-
- * configs/sip.conf.sample: Add documentation for setting
- username/password in SIP dial string. (closes issue #11587)
- Reported by: sobomax Patches: dialstring_doc.diff uploaded by
- sobomax (license 359)
-
-2008-02-29 14:50 +0000 [r105263-105327] Philippe Sultan <philippe.sultan@gmail.com>
-
- * /, res/res_jabber.c: Merged revisions 105326 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r105326 | phsultan | 2008-02-29 15:47:10 +0100 (Fri, 29 Feb 2008)
- | 1 line Fix a potential memory leak ........
-
- * channels/chan_jingle.c, channels/chan_gtalk.c, res/res_jabber.c:
- Remove unnecessary if statements before calling iks_delete
- (redundant check is done inside iks_delete), thus making the code
- conform with coding guidelines.
-
-2008-02-29 13:55 +0000 [r105262] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 105261 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r105261 | file | 2008-02-29 09:48:13 -0400 (Fri, 29 Feb
- 2008) | 4 lines Bump up the size of the uniqueid variable.
- (closes issue #12107) Reported by: asgaroth ........
-
-2008-02-29 13:12 +0000 [r105210] Philippe Sultan <philippe.sultan@gmail.com>
-
- * res/res_jabber.c: Automatically create new buddy upon reception
- of a presence stanza of type subscribed. (closes issue #12066)
- Reported by: ffadaie Patches: branch-1.4-12066-1.diff uploaded by
- phsultan (license 73) trunk-12066-1.diff uploaded by phsultan
- (license 73) Tested by: ffadaie, phsultan
-
-2008-02-29 01:15 +0000 [r105176] Tilghman Lesher <tlesher@digium.com>
-
- * contrib/init.d/rc.debian.asterisk, /: Merged revisions 105113 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r105113 | tilghman | 2008-02-28 15:56:54 -0600 (Thu, 28 Feb 2008)
- | 7 lines Update init script for LSB compat (closes issue #9843)
- Reported by: ibc Patches: rc.debian.asterisk.patch uploaded by
- ibc (license 211) Tested by: paravoid ........
-
-2008-02-28 22:39 +0000 [r105144] Russell Bryant <russell@digium.com>
-
- * /, main/utils.c, include/asterisk/lock.h, utils/check_expr.c:
- Merged revisions 105116 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r105116 | russell | 2008-02-28 16:23:05 -0600 (Thu, 28 Feb 2008)
- | 8 lines Fix a bug in the lock tracking code that was discovered
- by mmichelson. The issue is that if the lock history array was
- full, then the functions to mark a lock as acquired or not would
- adjust the stats for whatever lock is at the end of the array,
- which may not be itself. So, do a sanity check to make sure that
- we're updating lock info for the proper lock. (This explains the
- bizarre stats on lock #63 in BE-396, thanks Mark!) ........
-
-2008-02-28 20:14 +0000 [r105060-105061] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 105059 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r105059 | mmichelson | 2008-02-28 14:11:57 -0600 (Thu, 28 Feb
- 2008) | 6 lines When using autofill, members who are in use
- should be counted towards the number of available members to call
- if ringinuse is set to yes. Thanks to jmls who brought this issue
- up on IRC ........
-
- * main/dial.c, /: Merged revisions 104841 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104841 | mmichelson | 2008-02-27 15:49:20 -0600 (Wed, 27 Feb
- 2008) | 17 lines Two fixes: 1. Make the list of ast_dial_channels
- a lockable list. This is because in some cases, the ast_dial may
- exist in multiple threads due to asynchronous execution of its
- application, and I found some cases where race conditions could
- exist. 2. Check in ast_dial_join to be sure that the channel
- still exists before attempting to lock it, since it could have
- gotten hung up but the is_running_app flag on the
- ast_dial_channel may not have been cleared yet. (closes issue
- #12038) Reported by: jvandal Patches: 12038v2.patch uploaded by
- putnopvut (license 60) Tested by: jvandal ........
-
-2008-02-28 19:21 +0000 [r105006] Jason Parker <jparker@digium.com>
-
- * main/cdr.c, main/pbx.c, /: Merged revisions 105005 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r105005 | qwell | 2008-02-28 13:20:10 -0600 (Thu, 28 Feb
- 2008) | 9 lines Make pbx_exec pass an empty string into
- applications, if we get NULL. This protects against possible
- segfaults in applications that may try to use data before
- checking length (ast_strdupa'ing it, for example) (closes issue
- #12100) Reported by: foxfire Patches: 12100-nullappargs.diff
- uploaded by qwell (license 4) ........
-
-2008-02-28 14:42 +0000 [r104974] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_vpb.cc: Fix crash when configuration does not match
- hardware detection. (closes issue #12096) Reported by: mmickan
- Patches: chan_vpb.cc.diff uploaded by mmickan (license 400)
-
-2008-02-28 04:37 +0000 [r104921] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_skinny.c: 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. ........
-
-2008-02-28 00:11 +0000 [r104869] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/Makefile, build_tools/strip_nonapi: Merged revisions
- 104868 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104868 | tilghman | 2008-02-27 18:05:06 -0600 (Wed, 27 Feb 2008)
- | 7 lines Compatibility fix for PPC64 (closes issue #12081)
- Reported by: jcollie Patches: asterisk-1.4.18-funcdesc.patch
- uploaded by jcollie (license 412) Tested by: jcollie, Corydon76
- ........
-
-2008-02-27 23:58 +0000 [r104866] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c: reduce indentation in alloc_sub (issue
- #12079) Reported by: tzafrir Patches: alloc_sub uploaded by
- tzafrir (license 46)
-
-2008-02-27 21:02 +0000 [r104788] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_chanspy.c: Merged revisions 104787 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104787 | file | 2008-02-27 16:56:23 -0400 (Wed, 27 Feb 2008) | 2
- lines Don't loop around infinitely trying to spy on our own
- channel, and don't forget to free/detach the datastore upon
- hangup of the spy. ........
-
-2008-02-27 20:37 +0000 [r104784] Mark Michelson <mmichelson@digium.com>
-
- * /, main/file.c: Merged revisions 104783 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104783 | mmichelson | 2008-02-27 14:36:26 -0600 (Wed, 27 Feb
- 2008) | 4 lines Bump a couple of more buffers up by 2 so that
- annoying warnings aren't generated like crazy on every
- fileexists_core call. ........
-
-2008-02-27 19:36 +0000 [r104756] Jason Parker <jparker@digium.com>
-
- * apps/app_voicemail.c: Remove useless 's' and 'key' variables, in
- favor of 'val', which serves the exact same purpose.
-
-2008-02-27 18:20 +0000 [r104705] Tilghman Lesher <tlesher@digium.com>
-
- * main/manager.c, /: Merged revisions 104704 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104704 | tilghman | 2008-02-27 12:15:10 -0600 (Wed, 27 Feb 2008)
- | 2 lines Ensure the session ID can't be 0. ........
-
-2008-02-27 17:45 +0000 [r104687] Joshua Colp <jcolp@digium.com>
-
- * /, main/file.c: Merged revisions 104665 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104665 | file | 2008-02-27 13:41:40 -0400 (Wed, 27 Feb 2008) | 2
- lines Bump up the buffer by 2. ........
-
-2008-02-27 17:36 +0000 [r104643] Russell Bryant <russell@digium.com>
-
- * /, apps/app_chanspy.c: Merged revisions 104625 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104625 | russell | 2008-02-27 11:33:04 -0600 (Wed, 27 Feb 2008)
- | 4 lines Fix a problem in ChanSpy where it could get stuck in an
- infinite loop without being able to detect that the calling
- channel hung up. (closes issue #12076, reported by junky, patched
- by me) ........
-
-2008-02-27 17:31 +0000 [r104617] Jason Parker <jparker@digium.com>
-
- * /, main/features.c: Merged revisions 104598 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104598 | qwell | 2008-02-27 11:26:55 -0600 (Wed, 27 Feb 2008) |
- 8 lines Inherit language from the transfering channel on a blind
- transfer. (closes issue #11682) Reported by: caio1982 Patches:
- local_atxfer_lang3-1.4.diff uploaded by caio1982 (license 22)
- Tested by: caio1982, victoryure ........
-
-2008-02-27 17:12 +0000 [r104595-104597] Joshua Colp <jcolp@digium.com>
-
- * /, main/loader.c: Merged revisions 104596 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104596 | file | 2008-02-27 13:07:33 -0400 (Wed, 27 Feb 2008) | 4
- lines Use the lock (which already existed, it just wasn't used)
- on the updaters list to protect the contents instead of the
- overall module list lock. (closes issue #12080) Reported by:
- ChaseVenters ........
-
- * channels/chan_sip.c: After further discussion revert my previous
- commit for this. Currently in order to ensure devicestate is the
- expected value in another module (such as app_queue) then
- chan_sip must be loaded before hand.
-
-2008-02-27 16:54 +0000 [r104594] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, main/file.c: Merged revisions 104593 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104593 | kpfleming | 2008-02-27 10:53:06 -0600 (Wed, 27 Feb
- 2008) | 8 lines fallback to standard English prompts properly
- when using new prompt directory layout (closes issue #11831)
- Reported by: IgorG Patches: fallbacken.v1.diff uploaded by IgorG
- (license 20) (modified by me to improve code and conform rest of
- function to coding guidelines) ........
-
-2008-02-27 16:26 +0000 [r104537-104539] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: When queueing up a device state change when
- the peer is loaded from the configuration give it a state of not
- in use. We have to do this because the channel technology may not
- yet be registered so the state could not be queried and would be
- considered invalid. (closes issue #12087) Reported by: liorm
-
- * res/res_smdi.c, /: Merged revisions 104536 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104536 | file | 2008-02-27 11:52:02 -0400 (Wed, 27 Feb 2008) | 4
- lines Only stop the MWI monitor thread if it was actually
- started. (closes issue #12086) Reported by: francesco_r ........
-
-2008-02-27 15:34 +0000 [r104534] Tilghman Lesher <tlesher@digium.com>
-
- * utils/astcanary.c: open(2) needs a mode argument when O_CREAT is
- specified. (Closes issue #12083)
-
-2008-02-27 15:31 +0000 [r104533] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c, main/rtp.c: Fix T38 passthrough regression
- introduced by state changes. (closes issue #12078) Reported by:
- dimas Patches: v1-12078.patch uploaded by dimas (license 88)
- (closes issue #12074) Reported by: Ivan
-
-2008-02-27 08:20 +0000 [r104502] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_vpb.cc, configs/vpb.conf.sample,
- include/asterisk/module.h: Bring Voicetronix driver up to date
- with current drivers (closes issue #12084) Reported by: mmickan
- Patches: chan_vpb.cc.diff uploaded by mmickan (license 400)
- module.h.diff uploaded by mmickan (license 400) vpb.conf.sample
- uploaded by mmickan (license 400)
-
-2008-02-27 04:42 +0000 [r104419-104473] Russell Bryant <russell@digium.com>
-
- * doc/janitor-projects.txt: note that the chan_sip conversion is
- already in progress
-
- * doc/janitor-projects.txt: add another janitor project
-
- * doc/janitor-projects.txt: Add the stuff from the janitor projects
- page that is still relevant. I figure that if we keep this in the
- tree, it will be much easier to keep up to date. The page on
- asterisk.org just links to this on svn.digium.com/view
-
-2008-02-27 03:52 +0000 [r104418] Jason Parker <jparker@digium.com>
-
- * doc/janitor-projects.txt (added): Create placeholder file...for
- now.
-
-2008-02-27 02:05 +0000 [r104388] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: Whitespace changes only
-
-2008-02-27 01:16 +0000 [r104333-104335] Russell Bryant <russell@digium.com>
-
- * /, apps/app_chanspy.c: Merged revisions 104334 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104334 | russell | 2008-02-26 19:15:02 -0600 (Tue, 26 Feb 2008)
- | 3 lines Avoid some recursion in the cleanup code for the
- chanspy datastore (closes issue #12076, reported by junky,
- patched by me) ........
-
-2008-02-26 22:14 +0000 [r104301] Steve Murphy <murf@digium.com>
-
- * res/snmp/agent.c: small change to allow this file to compile. No
- problem if you don't install the libsnmp package.
-
-2008-02-26 20:33 +0000 [r104244-104270] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c: I swear I compiled this ... *cough*
-
- * res/res_phoneprov.c: fix this module, too
-
- * funcs/func_version.c: fix this module
-
- * Makefile, include/asterisk, build_tools/make_version_h (added):
- Re-add the automatically generated version.h, so that modules can
- include for making build time decisions for cross asterisk
- version compatibility
-
- * main/manager.c, channels/chan_sip.c, include/asterisk/version.h
- (removed), build_tools/make_version_c, res/res_agi.c,
- main/http.c, include/asterisk/ast_version.h (added): Rename
- version.h to ast_version.h. Next, I will be re-adding version.h
- as an automatically generated file like it used to be. This still
- needs to be there for modules that have to check it to compile
- against multiple asterisk versions.
-
-2008-02-26 19:14 +0000 [r104215] Joshua Colp <jcolp@digium.com>
-
- * main/cdr.c, main/pbx.c, include/asterisk/cdr.h, CHANGES: Add an
- 'e' option to ResetCDR which re-enables a CDR that has been
- disabled. (closes issue #11170) Reported by: kratzers Patches:
- ResetCDR.1.diff uploaded by kratzers (license 307)
-
-2008-02-26 18:40 +0000 [r104176] Tilghman Lesher <tlesher@digium.com>
-
- * doc/CODING-GUIDELINES: 1) Make braces mandatory for if/for/while,
- even around single statements. 2) Revise the argument parsing
- section, showing use of the standard macros. 3) Fix a typo.
-
-2008-02-26 18:27 +0000 [r104140-104142] Jason Parker <jparker@digium.com>
-
- * Makefile, /: Merged revisions 104141 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104141 | qwell | 2008-02-26 12:26:12 -0600 (Tue, 26 Feb 2008) |
- 1 line Add badshell to .PHONY target (thanks Kevin) ........
-
- * Makefile, /: Merged revisions 104139 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104139 | qwell | 2008-02-26 12:09:13 -0600 (Tue, 26 Feb 2008) |
- 2 lines Since all shells aren't as awesome as bash, we have to
- fail if somebody tries to use a literal "~" in DESTDIR. ........
-
-2008-02-26 16:51 +0000 [r104137] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Formatting and doxygen while waiting on an
- airport...
-
-2008-02-26 16:36 +0000 [r104133-104136] Jason Parker <jparker@digium.com>
-
- * /, sounds/Makefile: Merged revisions 104135 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104135 | qwell | 2008-02-26 10:35:06 -0600 (Tue, 26 Feb 2008) |
- 5 lines Revert previous abspath change. ...abspath is new in GNU
- make 3.81. I feel so...defeated. Must find new fix! ........
-
- * /, sounds/Makefile: Merged revisions 104132 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104132 | qwell | 2008-02-26 10:08:44 -0600 (Tue, 26 Feb 2008) |
- 9 lines Fix a very bizarre issue we were seeing with our buildbot
- when using a DESTDIR that wasn't an absolute path (such as
- DESTDIR=~/asterisk-1.4). Apparently what was happening, was that
- some of the targets were being expanded to the full path, so $@
- ended up being /root/asterisk-1.4/[...]/ rather than
- ~/asterisk-1.4/[...]/ It appears that this may be a new "feature"
- in GNU make. (*cough*
- http://en.wikipedia.org/wiki/Principle_of_least_surprise *cough*)
- ........
-
-2008-02-26 14:51 +0000 [r104127] Mark Michelson <mmichelson@digium.com>
-
- * main/features.c: Remove more hardcoded pipe symbols and replace
- with commas. (closes issue #12072) Reported by: SimonSharman
- Patches: features.patch uploaded by SimonSharman (license 410)
- Tested by: SimonSharman
-
-2008-02-26 06:43 +0000 [r104125] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_odbc.c: Use the readhandle for reads (closes issue
- #12069)
-
-2008-02-26 00:38 +0000 [r104120-104124] Russell Bryant <russell@digium.com>
-
- * res/res_smdi.c: Add a \todo to convert this module to the event
- system
-
- * CHANGES: Update CHANGES for SMDI stuff
-
- * channels/chan_zap.c, res/res_smdi.c, /, configs/smdi.conf.sample,
- include/asterisk/smdi.h, apps/app_voicemail.c: Merged revisions
- 104119 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104119 | russell | 2008-02-25 18:25:29 -0600 (Mon, 25 Feb 2008)
- | 33 lines Merge changes from team/russell/smdi-1.4 This commit
- brings in a significant set of changes to the SMDI support in
- Asterisk. There were a number of bugs in the current
- implementation, most notably being that it was very likely on
- busy systems to pop off the wrong message from the SMDI message
- queue. So, this set of changes fixes the issues discovered as
- well as introducing some new ways to use the SMDI support which
- are required to avoid the bugs with grabbing the wrong message
- off of the queue. This code introduces a new interface to SMDI,
- with two dialplan functions. First, you get an SMDI message in
- the dialplan using SMDI_MSG_RETRIEVE() and then you access
- details in the message using the SMDI_MSG() function. A side
- benefit of this is that it now supports more than just chan_zap.
- For example, with this implementation, you can have some FXO
- lines being terminated on a SIP gateway, but the SMDI link in
- Asterisk. Another issue with the current implementation is that
- it is quite common that the station ID that comes in on the SMDI
- link is not necessarily the same as the Asterisk voicemail box.
- There are now additional directives in the smdi.conf
- configuration file which let you map SMDI station IDs to Asterisk
- voicemail boxes. Yet another issue with the current SMDI support
- was related to MWI reporting over the SMDI link. The current code
- could only report a MWI change when the change was made by
- someone calling into voicemail. If the change was made by some
- other entity (such as with IMAP storage, or with a web interface
- of some kind), then the MWI change would never be sent. The SMDI
- module can now poll for MWI changes if configured to do so. This
- work was inspired by and primarily done for the University of
- Pennsylvania. (also related to issue #9260) ........
-
-2008-02-25 23:56 +0000 [r104103-104110] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c, UPGRADE.txt: Deprecate the "stripmsd" option
- in favor of dialplan substring variable syntax. (closes issue
- #12060)
-
- * /, apps/app_chanspy.c: Merged revisions 104106 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104106 | russell | 2008-02-25 17:42:42 -0600 (Mon, 25 Feb 2008)
- | 10 lines This patch fixes some pretty significant problems with
- how app_chanspy handles pointers to channels that are being spied
- upon. It was very likely that a crash would occur if the channel
- being spied upon hung up. This was because the current
- ast_channel handling _requires_ that the object is locked or else
- it could disappear at any time (except in the owning channel
- thread). So, this patch uses some channel datastore magic on the
- spied upon channel to be able to detect if and when the channel
- goes away. (closes issue #11877) (patch written by me, but thanks
- to kpfleming for the idea, and to file for review) ........
-
- * /, main/utils.c: Merged revisions 104102 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104102 | russell | 2008-02-25 17:19:05 -0600 (Mon, 25 Feb 2008)
- | 7 lines Improve the lock tracking code a bit so that a bunch of
- old locks that threads failed to lock don't sit around in the
- history. When a lock is first locked, this checks to see if the
- last lock in the list was one that was failed to be locked. If it
- is, then that was a lock that we're no longer sitting in a
- trylock loop trying to lock, so just remove it. (inspired by
- issue #11712) ........
-
-2008-02-25 23:04 +0000 [r104097-104101] Tilghman Lesher <tlesher@digium.com>
-
- * cdr/cdr_pgsql.c, CHANGES: Permit additional CDR columns to be
- saved in Postgres. Note that these changes are
- backward-compatible, so no changes to UPGRADE.txt are necessary.
- (closes issue #9279) Reported by: rottenroddy Patches:
- 20080125__bug9279.diff.txt uploaded by Corydon76 (license 14)
- Tested by: Corydon76
-
- * funcs/func_global.c: Shared space for variables (instead of
- letting other channels muck with your own) (closes issue #11943)
- Reported by: ramonpeek Patches: 20080208__bug11943__2.diff.txt
- uploaded by Corydon76 (license 14) Tested by: jmls
-
- * /, apps/app_voicemail.c: Merged revisions 104094 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r104094 | tilghman | 2008-02-25 15:31:47 -0600 (Mon, 25
- Feb 2008) | 5 lines If the destination folder is full, don't
- delete a message when exiting. (closes issue #12065) Reported by:
- selsky Patch by: (myself) ........
-
-2008-02-25 21:40 +0000 [r104096] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 104095 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104095 | file | 2008-02-25 17:37:20 -0400 (Mon, 25 Feb 2008) | 6
- lines Make it so a users.conf user creates both a SIP peer and a
- SIP user. The user will be used for inbound authentication for
- the device, and peer will be used for placing calls to the
- device. (closes issue #9044) Reported by: queuetue Patches:
- sip-gui-friend.diff uploaded by qwell (license 4) ........
-
-2008-02-25 20:50 +0000 [r104093] Jason Parker <jparker@digium.com>
-
- * /, main/config.c: Merged revisions 104092 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104092 | qwell | 2008-02-25 14:49:42 -0600 (Mon, 25 Feb 2008) |
- 11 lines Allow the use of #include and #exec in situations where
- the max include depth was only 1. Specifically, this fixes using
- #include and #exec in extconfig.conf. This was basically caused
- because the config file itself raises the include level to 1. I
- opted not to raise the include limit, because recursion here
- could cause very bizarre behavior. Pointed out, and tested by
- jmls (closes issue #12064) ........
-
-2008-02-25 19:02 +0000 [r104089] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: Instead of outputting a verbose message
- every so often let's make it a debug message.
-
-2008-02-25 19:00 +0000 [r104088] Brett Bryant <bbryant@digium.com>
-
- * doc/siptls.txt, configs/sip.conf.sample: Adding more tls
- configuration details to sip.conf sample, with a list of valid
- ciphers provided in both files. .. First commit since July, woot
-
-2008-02-25 18:38 +0000 [r104087] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 104086 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r104086 | russell | 2008-02-25 12:38:10 -0600 (Mon, 25
- Feb 2008) | 4 lines Ensure that the channel doesn't disappear in
- agent_logoff(). If it does, it could cause a crash. (fixes the
- crash reported in BE-396) ........
-
-2008-02-25 16:18 +0000 [r104081-104085] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 104084 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104084 | file | 2008-02-25 12:16:13 -0400 (Mon, 25 Feb 2008) | 6
- lines If a resubscription comes in for a dialog we no longer know
- about tell the remote side that the dialog does not exist so they
- subscribe again using a new dialog. (closes issue #10727)
- Reported by: s0l4rb03 Patches: 10727-2.diff uploaded by file
- (license 11) ........
-
- * /, channels/chan_sip.c: Merged revisions 104082 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104082 | file | 2008-02-25 11:17:18 -0400 (Mon, 25 Feb 2008) | 6
- lines Due to recent changes tag will no longer be NULL if not
- present so we have to use ast_strlen_zero to see if it's actually
- blank. (closes issue #12061) Reported by: flefoll Patches:
- chan_sip.c.br14.patch_pedantic_no_totag uploaded by flefoll
- (license 244) ........
-
- * res/res_config_pgsql.c: Fix building of trunk. dbpass is always
- going to exist.
-
-2008-02-24 02:37 +0000 [r104073-104074] Steve Murphy <murf@digium.com>
-
- * channels/chan_sip.c: Enforce a space between function args as per
- code review.
-
- * res/res_config_pgsql.c: On a 64-bit machine, with dev-mode turned
- on, and pgsql installed, I get warnings that stops the compile.
- They are fixed now.
-
-2008-02-22 23:56 +0000 [r104045] Doug Bailey <dbailey@digium.com>
-
- * channels/chan_zap.c, configure, configure.ac: Add protection to
- chan_zap build when NEONMWI events are not defined
-
-2008-02-22 22:55 +0000 [r104036-104039] Tilghman Lesher <tlesher@digium.com>
-
- * doc/manager_1_1.txt, main/manager.c, UPGRADE.txt, CHANGES,
- include/asterisk/manager.h: Move Originate to a separate
- privilege and require the additional System privilege to call out
- to a subshell.
-
- * /, channels/chan_sip.c: Merged revisions 104037 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104037 | tilghman | 2008-02-22 16:45:14 -0600 (Fri, 22 Feb 2008)
- | 6 lines Backwards debug message. (closes issue #12052) Reported
- by: flefoll Patches: chan_sip.c.br14.patch_found-notfound
- uploaded by flefoll (license 244) ........
-
- * res/res_config_pgsql.c: Allow database password to be NULL and
- several other cleanups. (closes issue #12048) Reported by: bukaj
- Patches: 20080222__bug12048.diff.txt uploaded by Corydon76
- (license 14) Tested by: bukaj
-
-2008-02-21 21:27 +0000 [r104031] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: fix a typo
-
-2008-02-21 21:09 +0000 [r104025-104029] Mark Michelson <mmichelson@digium.com>
-
- * res/res_agi.c: Instead of a notice, make the message about a
- hung-up channel a debug message, and revert the original logic on
- the if statement. Thanks to Juggie for bringing this to my
- attention.
-
-2008-02-21 17:38 +0000 [r104024] Doug Bailey <dbailey@digium.com>
-
- * channels/chan_zap.c: Added configuration distinction between neon
- and fsk mwi detection Add the detection for neon MWI events got
- rid of extraneous handle_init_event call in monitor loop
-
-2008-02-21 16:46 +0000 [r104020] Mark Michelson <mmichelson@digium.com>
-
- * res/res_agi.c: Don't print the fact that we are using dead mode
- in AGI if called from the 'h' extension since it is well-known
- that it will be running in dead mode. (closes issue #12046)
- Reported by: explidous
-
-2008-02-21 16:44 +0000 [r104019] Joshua Colp <jcolp@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac:
- Disable epoll as it has caused more obscure issues then any of my
- previous code. I will continue to work on it in a separate branch
- to make it stable for a release and test it against the following
- issues. (closes issue #11253) Reported by: falves11 (closes issue
- #11657) Reported by: davevg (closes issue #11033) Reported by:
- falves11
-
-2008-02-21 14:44 +0000 [r104016] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/manager.c, /: Merged revisions 104015 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104015 | kpfleming | 2008-02-21 08:33:51 -0600 (Thu, 21 Feb
- 2008) | 2 lines reduce the likelihood that HTTP Manager session
- ids will consist of primarily '1' bits ........
-
-2008-02-21 05:21 +0000 [r104014] Tilghman Lesher <tlesher@digium.com>
-
- * utils/astman.c: Ignore some more unused generated events. (closes
- issue #12042) Reported by: junky Patches: astman_events.diff
- uploaded by junky (license 177)
-
-2008-02-20 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.6.0-beta4 released.
-
-2008-02-20 22:34 +0000 [r103957] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 103956 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r103956 | mmichelson | 2008-02-20 16:32:22 -0600 (Wed, 20 Feb
- 2008) | 8 lines Clear up confusion when viewing the
- QUEUE_WAITING_COUNT of a "dead" realtime queue. Since from the
- user's perspective, the queue does exist, we shouldn't tell them
- we couldn't find the queue. Instead since it is a dead queue,
- report a 0 waiting count This issue was brought up on IRC by jmls
- ........
-
-2008-02-20 22:29 +0000 [r103954-103955] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_h323.c: Try to do Packet2Packet bridging with
- chan_h323 if reinviting isn't enabled. (closes issue #11901)
- Reported by: pj
-
- * channels/chan_zap.c, /: Merged revisions 103953 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r103953 | file | 2008-02-20 18:06:59 -0400 (Wed, 20 Feb 2008) | 6
- lines Don't wait for additional digits when overlap dialing is
- enabled if the setup message contains the sending_complete
- information element. (closes issue #11785) Reported by: klaus3000
- Patches: sending_complete_overlap_asterisk-1.4.17.patch.txt
- uploaded by klaus3000 (license 65) ........
-
-2008-02-20 21:41 +0000 [r103908] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_local.c, /: Merged revisions 103904 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r103904 | mmichelson | 2008-02-20 15:40:08 -0600 (Wed,
- 20 Feb 2008) | 6 lines Fix a crash if the channel becomes NULL
- while attempting to lock it. (closes issue #12039) Reported by:
- danpwi ........
-
-2008-02-20 21:36 +0000 [r103903] Jason Parker <jparker@digium.com>
-
- * include/asterisk/dsp.h, main/dsp.c: Largely refactor DSP tone
- detection routines. Separate fax detection from digit detected.
- Added CED (called) tone detection for fax (previously, only CNG
- (calling) was supported). Separate DTMF/MF code paths where
- appropriate. Allow detection of arbitary tones. (closes issue
- #11796) Reported by: dimas Patches: v6-dsp-faxtones.patch
- uploaded by dimas (license 88) Tested by: dimas, IgorG, Cache
-
-2008-02-20 21:08 +0000 [r103902] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Fix a crash due to the wrong variable being
- used when building a directory string. (closes issue #12027)
- Reported by: jaroth Patches: forward.patch uploaded by jaroth
- (license 50) Tested by: jaroth
-
-2008-02-20 18:29 +0000 [r103846-103847] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/sched.h: Add some documentation fixups
-
- * /, main/stdtime/localtime.c: Merged revisions 103845 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r103845 | tilghman | 2008-02-20 11:53:00 -0600 (Wed, 20
- Feb 2008) | 7 lines Compat fix for Solaris (closes issue #12022)
- Reported by: asgaroth Patches: 20080219__bug12022.diff.txt
- uploaded by Corydon76 (license 14) Tested by: asgaroth ........
-
-2008-02-20 15:21 +0000 [r103844] Mark Michelson <mmichelson@digium.com>
-
- * res/res_monitor.c: Fix another spot where a hard-coded '|' hadn't
- been converted to ',' (closes issue #12034) Reported by: kowalma
-
-2008-02-20 03:52 +0000 [r103838-103842] Joshua Colp <jcolp@digium.com>
-
- * main/audiohook.c: *mumble*
-
- * main/audiohook.c: file not found.
-
- * main/audiohook.c: Minor test...
-
-2008-02-20 00:49 +0000 [r103833] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: When using IMAP storage, if the folder you
- attempt to save to does not exist, create it first. (closes issue
- #12032) Reported by: jaroth Patches: createfolder.patch uploaded
- by jaroth (license 50) Tested by: jaroth
-
-2008-02-19 22:35 +0000 [r103831-103832] Jason Parker <jparker@digium.com>
-
- * main/channel.c: Make sure to mask out non-audio first as well
-
- * main/channel.c: Maybe we should set the value before we test it?
- Fixes an issue people have been seeing (unreported?) with file
- playback not working.
-
-2008-02-19 21:54 +0000 [r103824-103828] Joshua Colp <jcolp@digium.com>
-
- * main/loader.c: Add a log message that appears when you try to
- unload a module that isn't loaded. (closes issue #12033) Reported
- by: jamesgolovich Patches: asterisk-loader.diff.txt uploaded by
- jamesgolovich (license 176)
-
- * main/file.c: Only output a log message saying the format does not
- exist if it actually does not exist, not if the file itself could
- not be opened. (closes issue #11828) Reported by: IgorG Patches:
- readfile.v1.diff uploaded by IgorG (license 20)
-
- * /, channels/h323/ast_h323.cxx: Merged revisions 103823 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r103823 | file | 2008-02-19 16:28:08 -0400 (Tue, 19 Feb 2008) | 6
- lines Send CallerID Name in setup message. (closes issue #11241)
- Reported by: tusar Patches: h323id_as_callerid_name.patch
- uploaded by tusar (license 344) ........
-
-2008-02-19 20:06 +0000 [r103822] Russell Bryant <russell@digium.com>
-
- * channels/chan_local.c, /: Merged revisions 103821 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r103821 | russell | 2008-02-19 14:02:49 -0600 (Tue, 19
- Feb 2008) | 8 lines Account for the fact that the "other" channel
- can disappear while the local pvt is not locked. (fixes a problem
- introduced in rev 100581) (closes issue #12012) Reported by:
- stevedavies Patch by me ........
-
-2008-02-19 19:27 +0000 [r103819-103820] Joshua Colp <jcolp@digium.com>
-
- * apps/app_authenticate.c: len already contains the position we
- want to examine, if we move one left again we'll actually
- probably be looking at a digit. (issue #12030) Reported by:
- alligosh
-
- * apps/app_channelredirect.c, UPGRADE.txt, CHANGES: Add
- CHANNELREDIRECT_STATUS variable to ChannelRedirect() dialplan
- application. This will either be set to NOCHANNEL if the given
- channel was not found or SUCCESS if it worked. (closes issue
- #11553) Reported by: johan Patches:
- UPGRADE.txt.channelredirect.patch uploaded by johan (license 334)
- CHANGES.channelredirect.patch uploaded by johan (license 334)
- app_channelredirect-20080219.patch uploaded by johan (license
- 334)
-
-2008-02-19 18:14 +0000 [r103818] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_zap.c: (closes issue #11864) Reported by: julianjm
- Patches: chan_zap.c-1.4-devicestate-v1.diff uploaded by julianjm
- (license 99) Patch fixes problem of device state incorrectly
- reporting idle before PBX answers incoming call on FXO channel.
- Device status is updated now during new channel creation.
-
-2008-02-19 17:33 +0000 [r103808-103813] Joshua Colp <jcolp@digium.com>
-
- * /, configure, configure.ac: Merged revisions 103812 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r103812 | file | 2008-02-19 13:31:32 -0400 (Tue, 19 Feb
- 2008) | 4 lines Don't look for launchd when cross compiling.
- (closes issue #12029) Reported by: ovi ........
-
-2008-02-19 00:59 +0000 [r103805] Tilghman Lesher <tlesher@digium.com>
-
- * main/say.c: Change verbosity into debug for Hebrew (and various
- whitespace fixes) (Closes issue #12011)
-
-2008-02-18 23:58 +0000 [r103798-103802] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /: Merged revisions 103801 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r103801 | file | 2008-02-18 19:56:48 -0400 (Mon, 18 Feb 2008) |
- 10 lines Ensure that emulated DTMFs do not get interrupted by
- another begin frame. (closes issue #11740) Reported by: gserra
- Patches: v1-11740.patch uploaded by dimas (license 88) (closes
- issue #11955) Reported by: tsearle (closes issue #10530) Reported
- by: xmarksthespot ........
-
- * main/channel.c, main/frame.c, channels/chan_sip.c,
- include/asterisk/channel.h, include/asterisk/frame.h: Add a
- non-invasive API for application level manipulation of T38 on a
- channel. This uses control frames (so they can even pass across
- IAX2) to negotiate T38 and provided a way of getting the current
- status of T38 using queryoption. This should by no means cause
- any issues and if it does I will take responsibility for it.
- (closes issue #11873) Reported by: dimas Patches:
- v4-t38-api.patch uploaded by dimas (license 88)
-
- * main/frame.c: Add some missing control frames.
-
-2008-02-18 22:33 +0000 [r103796] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 103795 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r103795 | qwell | 2008-02-18 16:28:56 -0600 (Mon, 18 Feb 2008) |
- 1 line Fix previous commit so that we actually disable
- echocanbridged if echocancel is off. ........
-
-2008-02-18 21:57 +0000 [r103794] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Commit chan_zap portion of #11964: add the
- ability to get ORIG_CALLED_NUM
-
-2008-02-18 21:30 +0000 [r103791] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 103790 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r103790 | qwell | 2008-02-18 15:23:32 -0600 (Mon, 18 Feb 2008) |
- 4 lines Correct a message when echocancelwhenbridged is on, but
- echocancel is not. Closes issue #12019 ........
-
-2008-02-18 20:58 +0000 [r103788] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Make sure EC is enabled when SS7 call comes
- in. Also add support for multiple DPCs per linkset. #11779
-
-2008-02-18 20:53 +0000 [r103787] Mark Michelson <mmichelson@digium.com>
-
- * /, main/app.c: Merged revisions 103786 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r103786 | mmichelson | 2008-02-18 14:52:09 -0600 (Mon, 18 Feb
- 2008) | 10 lines There was an invalid assumption when calculating
- the duration of a file that the filestream in question was
- created properly. Unfortunately this led to a segfault in the
- situation where an unknown format was specified in voicemail.conf
- and a voicemail was recorded. Now, we first check to be sure that
- the stream was written correctly or else assume a zero duration.
- (closes issue #12021) Reported by: jakep Tested by: putnopvut
- ........
-
-2008-02-18 19:47 +0000 [r103783] Michiel van Baak <michiel@vanbaak.info>
-
- * main/asterisk.c: make the output of 'core show settings' a bit
- nicer. (closes issue #12020) Reported by: seanbright Patches:
- asterisk.c.patch uploaded by seanbright (license 71)
-
-2008-02-18 17:45 +0000 [r103781] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c, main/rtp.c: Merged revisions 103780 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r103780 | tilghman | 2008-02-18 11:31:52 -0600 (Mon, 18 Feb 2008)
- | 9 lines When a SIP channel is being auto-destroyed, it's
- possible for it to still be in bridge code. When that happens, we
- crash. Delay the RTP destruction until the bridge is ended.
- (closes issue #11960) Reported by: norman Patches:
- 20080215__bug11960__2.diff.txt uploaded by Corydon76 (license 14)
- Tested by: norman ........
-
-2008-02-18 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.6.0-beta3 released.
-
-2008-02-18 17:12 +0000 [r103772] Olle Johansson <oej@edvina.net>
-
- * main/channel.c, channels/chan_sip.c: Make sure we can set up
- calls without audio (text+video). And ... it works!
-
-2008-02-18 16:40 +0000 [r103771] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 103770 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r103770 | mmichelson | 2008-02-18 10:37:31 -0600 (Mon, 18 Feb
- 2008) | 10 lines Fix a linked list corruption that under the
- right circumstances could lead to a looped list, meaning it will
- traverse forever. (closes issue #11818) Reported by: michael-fig
- Patches: 11818.patch uploaded by putnopvut (license 60) Tested
- by: michael-fig ........
-
-2008-02-18 16:13 +0000 [r103764-103769] Joshua Colp <jcolp@digium.com>
-
- * apps/app_channelredirect.c, main/pbx.c, include/asterisk/pbx.h:
- Add an API call (ast_async_parseable_goto) which parses a goto
- string and does an async goto instead of an explicit goto.
- (closes issue #11753) Reported by: johan
-
- * /, channels/chan_sip.c: Merged revisions 103763 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r103763 | file | 2008-02-18 11:33:14 -0400 (Mon, 18 Feb 2008) | 2
- lines Don't care if the extension given doesn't exist for
- subscription based MWI. ........
-
-2008-02-18 10:10 +0000 [r103755] Olle Johansson <oej@edvina.net>
-
- * CHANGES, channels/chan_iax2.c: - No space in manager event names,
- please - Add new event to CHANGES
-
-2008-02-18 04:43 +0000 [r103754] Tilghman Lesher <tlesher@digium.com>
-
- * build_tools/cflags.xml, main/channel.c, main/pbx.c,
- funcs/func_channel.c, include/asterisk/channel.h, CHANGES,
- main/cli.c: Context tracing for channels (closes issue #11268)
- Reported by: moy Patches:
- chantrace-datastored-encapsulated-rev94934.patch uploaded by moy
- (license 222)
-
-2008-02-16 21:22 +0000 [r103750] Michiel van Baak <michiel@vanbaak.info>
-
- * channels/chan_skinny.c: 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
-
-2008-02-15 23:32 +0000 [r103742] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 103741 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r103741 | russell | 2008-02-15 17:31:39 -0600 (Fri, 15
- Feb 2008) | 8 lines Fix a crash in chan_iax2 due to a race
- condition (closes issue #11780) Reported by: guillecabeza
- Patches: bug_iax2_jb_1.4.patch uploaded by guillecabeza (license
- 380) bug_iax2_jb_trunk.patch uploaded by guillecabeza (license
- 380) ........
-
-2008-02-15 23:20 +0000 [r103740] Mark Michelson <mmichelson@digium.com>
-
- * CHANGES: Document GotoIfTime change from svn revision 103738
-
-2008-02-15 23:14 +0000 [r103739] Russell Bryant <russell@digium.com>
-
- * include/asterisk/aes.h: Fix a regression in Asterisk 1.6 related
- to the use of AES encryption. 1024 was used instead of 128 when
- using AES from OpenSSL. Many thanks to d1mas for figuring this
- one out! (closes issue #11946) Reported by: bbhoss Patches:
- v1-11946.patch uploaded by dimas (license 88)
-
-2008-02-15 23:07 +0000 [r103737-103738] Mark Michelson <mmichelson@digium.com>
-
- * main/pbx.c: Add proper "false" case behavior to GotoIfTime
- (closes issue #11719) Reported by: kshumard Patches:
- gotoiftime.twobranches.patch uploaded by kshumard (license 92)
- Tested by: kshumard
-
- * apps/app_voicemail.c: Fix redeclaration of variables when using
- IMAP storage (closes issue #11988) Reported by: jaroth Patches:
- variable_cleanup.patch uploaded by jaroth (license 50)
-
-2008-02-15 19:50 +0000 [r103727-103729] Russell Bryant <russell@digium.com>
-
- * /, main/loader.c: Merged revisions 103728 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r103728 | russell | 2008-02-15 13:50:11 -0600 (Fri, 15 Feb 2008)
- | 4 lines In the case that you try to directly reload a module
- has returned AST_MODULE_LOAD_DECLINE, log a message indicating
- that the module is not fully initialized and must be initialized
- using "module load". ........
-
- * /, main/loader.c: Merged revisions 103726 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r103726 | russell | 2008-02-15 12:33:29 -0600 (Fri, 15 Feb 2008)
- | 6 lines Don't attempt to execute the reload callback for a
- module that returned AST_MODULE_LOAD_DECLINE. This fixes a crash
- that was reported against chan_console in trunk. (closes issue
- #11953, reported by junky, fixed by me) ........
-
-2008-02-15 17:32 +0000 [r103725] Mark Michelson <mmichelson@digium.com>
-
- * doc/tex/imapstorage.tex, /, configure, configure.ac: Merged
- revisions 103722 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r103722 | mmichelson | 2008-02-15 11:26:37 -0600 (Fri, 15 Feb
- 2008) | 8 lines Final round of changes for configure script logic
- for IMAP Now if a directory is specified, then we will search
- that directory for a source installation of the IMAP toolkit. If
- none is found, then we will use that directory as the basis for
- detecting a package installation of the IMAP c-client. If that
- check fails, then configure will fail. ........
-
-2008-02-15 17:29 +0000 [r103723] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c, channels/chan_sip.c, res/res_phoneprov.c,
- include/asterisk/extconf.h, channels/misdn/isdn_msg_parser.c,
- apps/app_queue.c, channels/misdn/isdn_lib.c, main/config.c,
- main/channel.c, res/res_config_curl.c, channels/misdn/isdn_lib.h,
- main/ast_expr2f.c, channels/misdn/ie.c,
- channels/misdn/chan_misdn_config.h, channels/misdn/portinfo.c,
- include/asterisk/strings.h, res/res_config_ldap.c,
- include/asterisk/time.h: Fix up some doxygen issues. (closes
- issue #11996) Patches: bug_11996_doxygen.diff uploaded by snuffy
- (license 35)
-
-2008-02-15 15:45 +0000 [r103716] Tilghman Lesher <tlesher@digium.com>
-
- * utils/conf2ael.c: Remove extraneous copy (closes issue #12002)
- Reported by: junky Patches: conf2ael.diff uploaded by junky
- (license 177)
-
-2008-02-15 15:11 +0000 [r103699-103715] Mark Michelson <mmichelson@digium.com>
-
- * configure, configure.ac: Merging of changes from 1.4 revision
- 103713.
-
- * doc/tex/imapstorage.tex, configure, configure.ac: Same changes as
- made to 1.4 in revision 103710
-
- * doc/tex/imapstorage.tex: Trunk version of 1.4's imap
- documentation updates
-
- * configure, configure.ac: See commit message for svn revision
- 103698. This behavior is the same as what is described there. The
- difference is that trunk already had the --with-imap=system
- option, but it only checked the include path for headers in the
- imap directory and not also the c-client directory.
-
-2008-02-14 21:21 +0000 [r103694] Jason Parker <jparker@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac: Modify
- ldap autoconf function, so that an incorrect ldap library is not
- found on Solaris (it is incompatible). Also removes second check
- for awk, which causes Solaris to find an incompatible version of
- awk. (closes issue #11829) Reported by: snuffy Patches:
- bug-11829.diff uploaded by snuffy (license 35)
-
-2008-02-14 21:04 +0000 [r103687-103691] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 103690 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r103690 | mmichelson | 2008-02-14 15:03:02 -0600 (Thu,
- 14 Feb 2008) | 3 lines Fix build for non-IMAP builds ........
-
- * /, apps/app_voicemail.c: Merged revisions 103688 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r103688 | mmichelson | 2008-02-14 14:55:48 -0600 (Thu,
- 14 Feb 2008) | 9 lines Fix the new message count if delete=yes
- when using IMAP storage. (closes issue #11406) Reported by:
- jaroth Patches: deleteflag_v2.patch uploaded by jaroth (license
- 50) Tested by: jaroth ........
-
- * configs/queues.conf.sample, UPGRADE.txt, apps/app_queue.c: Change
- the queue holdtime announcement to happen at any interval (not
- just greater than two minutes). Remove the saying of less-than
- for holdtime announcements since it can lead to awkward holdtime
- announcements. Using '1' as a queue-round-seconds value is no
- longer valid. (closes issue #9736) Reported by: caio1982 Patches:
- queue_announce5.diff uploaded by caio1982 (license 22) Tested by:
- caio1982, putnopvut
-
-2008-02-14 19:52 +0000 [r103685] Jason Parker <jparker@digium.com>
-
- * /, funcs/func_cdr.c: Merged revisions 103683 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r103683 | qwell | 2008-02-14 13:51:10 -0600 (Thu, 14 Feb 2008) |
- 5 lines Document the 'l' option to the CDR() function. (Thanks
- voipgate for pointing out the option, and Leif for providing text
- for it.) Closes issue #11695. ........
-
-2008-02-14 19:47 +0000 [r103682] Jeff Peeler <jpeeler@digium.com>
-
- * apps/app_externalivr.c: a few syntax changes and safer code
-
-2008-02-14 18:39 +0000 [r103677] Jason Parker <jparker@digium.com>
-
- * channels/chan_iax2.c: Add periodic jitter stats to CLI and
- manager. (closes issue #8188) Reported by: stevedavies Patches:
- jblogging-trunk.patch uploaded by stevedavies
- jblogging-trunk_wmgrevent.patch uploaded by johann8384
- updated_jbloggin-trunk_mgrevent.patch uploaded by johann8384
- (license 190) (with additional changes by me) Tested by:
- stevedavies, johann8384
-
-2008-02-14 10:19 +0000 [r103668] Olle Johansson <oej@edvina.net>
-
- * res/res_agi.c, apps/app_externalivr.c: Formatting fixes
-
-2008-02-13 21:04 +0000 [r103662] Jeff Peeler <jpeeler@digium.com>
-
- * apps/app_externalivr.c: (closes issue #11825) Reported by:
- ctooley Patches: additional_eivr_commands.patch uploaded by
- ctooley (license 136) Tested by: ctooley
-
-2008-02-13 15:47 +0000 [r103658] Mark Michelson <mmichelson@digium.com>
-
- * UPGRADE.txt, res/res_musiconhold.c: 1. Deprecate SetMusicOnHold
- and WaitMusicOnHold. 2. Add a duration parameter to MusicOnHold
- (closes issue #11904) Reported by: dimas Patches: v2-moh.patch
- uploaded by dimas (license 88) Tested by: dimas
-
-2008-02-13 00:55 +0000 [r103559] Mark Michelson <mmichelson@digium.com>
-
- * main/event.c: Fix a small logic error in ast_event_iterator_next.
- The previous logic allowed for the iterator to indicate there was
- more data than there really was, causing the iterator read beyond
- the end of the event structure. This led to invalid memory reads
- and potential crashes.
-
-2008-02-12 22:26 +0000 [r103447-103506] Jason Parker <jparker@digium.com>
-
- * main/manager.c: Even more sane permissions. This should be
- handled via a umask, like in many other places.
-
- * main/manager.c: Use slight more sane permissions
-
-2008-02-12 15:39 +0000 [r103387-103388] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c: Remove development version notice.
-
- * main/manager.c: Fix build on *BSD. These permissions constants
- are not available there.
-
-2008-02-12 15:13 +0000 [r103386] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 103385 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r103385 | file | 2008-02-12 11:09:24 -0400 (Tue, 12 Feb 2008) | 4
- lines Even if no CallerID name or number has been provided by the
- remote party still use the configured sip.conf ones. (closes
- issue #11977) Reported by: pj ........
-
-2008-02-12 14:08 +0000 [r103341] Philippe Sultan <philippe.sultan@gmail.com>
-
- * include/asterisk/jabber.h, res/res_jabber.c: Use an ast_flags
- structure in aji_client and aji_buddy rather than an integer.
- Modify calls to various ast_*_flag macros accordingly.
-
-2008-02-12 00:24 +0000 [r103331] Jeff Peeler <jpeeler@digium.com>
-
- * main/manager.c, include/asterisk/config.h, CHANGES,
- main/config.c: Requested changes from Pari, reviewed by Russell.
- Added ability to retrieve list of categories in a config file.
- Added ability to retrieve the content of a particular category.
- Added ability to empty a context. Created new action to create a
- new file. Updated delete action to allow deletion by line number
- with respect to category. Added new action insert to add new
- variable to category at specified line. Updated action newcat to
- allow new category to be inserted in file above another existing
- category.
-
-2008-02-11 22:10 +0000 [r103317-103325] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 103324 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r103324 | file | 2008-02-11 18:09:07 -0400 (Mon, 11 Feb 2008) | 4
- lines If entering a conference with the 'w' option ensure that we
- can't listen or speak until the marked user appears. (closes
- issue #11835) Reported by: alanmcmillan ........
-
- * res/res_agi.c: Remove ast_module_user usage from res_agi. This is
- taken care of in the core.
-
- * main/pbx.c, main/manager.c, main/translate.c, main/logger.c,
- main/app.c, main/utils.c, main/indications.c, main/asterisk.c,
- main/rtp.c: Just some minor coding style cleanup...
-
- * main/pbx.c: Fix Manager Redirect while in an AGI. (closes issue
- #10661) Reported by: junky
-
-2008-02-11 17:09 +0000 [r103316] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, configs/zapata.conf.sample: Merged revisions 103315 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r103315 | kpfleming | 2008-02-11 11:05:22 -0600 (Mon, 11 Feb
- 2008) | 2 lines improve 2BCT documentation a bit (thanks Jared)
- ........
-
-2008-02-11 16:17 +0000 [r103313-103314] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, channels/chan_iax2.c: Add support for allowing a
- native bridge to happen when the L option is enabled. The RTP
- bridging could already handle this, it just needed to be enabled
- in the main bridging code. (issue #10647) Reported by: samdell3
-
- * channels/chan_skinny.c: 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)
-
-2008-02-11 06:05 +0000 [r103306] James Golovich <james@gnuinter.net>
-
- * channels/chan_sip.c: Don't wipe out transport and fd in chan_sip
- on reload (issue #11930)
-
-2008-02-11 03:03 +0000 [r103282-103284] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Fix improper indentation. Thanks again to
- snuffy for pointing it out.
-
- * apps/app_queue.c: Add a couple of comments to clarify the
- unreffing of queues. Thanks to snuffy for the idea.
-
- * main/event.c: Fix a problem regarding network vs. host byte order
- in the event API. ast_event_iterator_get_ie_type should return
- the ie type in host byte order. Furthermore, ast_event_get_ie_raw
- should already have its ie type argument in host byte order since
- it could be called externally (and it in fact is called in this
- way by ast_event_get_cached).
-
-2008-02-09 11:27 +0000 [r103249] Michiel van Baak <michiel@vanbaak.info>
-
- * apps/app_dial.c, apps/app_dictate.c, apps/app_echo.c,
- apps/app_authenticate.c, apps/app_disa.c, apps/app_chanisavail.c,
- apps/app_exec.c, apps/app_db.c, apps/app_controlplayback.c,
- apps/app_channelredirect.c, apps/app_directed_pickup.c,
- apps/app_dumpchan.c, apps/app_amd.c, apps/app_externalivr.c,
- apps/app_directory.c, apps/app_chanspy.c, apps/app_cdr.c:
- whitespace fixes only.
-
-2008-02-09 06:33 +0000 [r103198] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 103197 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r103197 | tilghman | 2008-02-09 00:23:49 -0600 (Sat, 09
- Feb 2008) | 4 lines Commit fix for being unable to send voicemail
- from VoiceMailMain Reported by: William F Acker (via the -users
- mailing list) Patch by: Corydon76 (license 14) ........
-
-2008-02-08 21:26 +0000 [r103171] Russell Bryant <russell@digium.com>
-
- * main/udptl.c, main/pbx.c, channels/chan_sip.c,
- channels/chan_iax2.c, res/res_jabber.c, apps/app_playback.c,
- main/rtp.c, channels/chan_usbradio.c, main/cdr.c,
- channels/chan_skinny.c, apps/app_minivm.c, res/res_agi.c,
- pbx/pbx_ael.c, pbx/pbx_dundi.c, funcs/func_devstate.c,
- apps/app_rpt.c, main/asterisk.c, channels/chan_mgcp.c,
- apps/app_voicemail.c: 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!
-
-2008-02-08 18:58 +0000 [r103071-103122] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Forgot that AST_LIST_REMOVE_CURRENT takes
- different arguments in trunk than 1.4.
-
- * /, apps/app_queue.c: Merged revisions 103120 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r103120 | mmichelson | 2008-02-08 12:48:17 -0600 (Fri, 08 Feb
- 2008) | 10 lines Prevent a potential three-thread deadlock. Also
- added a comment block to explicitly state the locking order
- necessary inside app_queue. (closes issue #11862) Reported by:
- flujan Patches: 11862.patch uploaded by putnopvut (license 60)
- Tested by: flujan ........
-
- * /, channels/chan_iax2.c: Merged revisions 103070 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r103070 | mmichelson | 2008-02-08 12:00:38 -0600 (Fri,
- 08 Feb 2008) | 6 lines Yield the thread and return -1 if the
- ioctl fails for Zaptel timing device. (closes issue #11891)
- Reported by: tzafrir ........
-
-2008-02-08 16:49 +0000 [r103044] Russell Bryant <russell@digium.com>
-
- * UPGRADE-1.2.txt (added), UPGRADE-1.4.txt (added), UPGRADE.txt: At
- the request of ManxPower, include the UPGRADE.txt from 1.2 and
- 1.4, as well. This way, if people need to go back and review what
- was deprecated in previous major releases, it is readily
- available to them. Thanks for the suggestion!
-
-2008-02-08 15:31 +0000 [r102969-103018] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Fix a network byte order issue and ensure
- when creating an outgoing dialog that the socket always contains
- information such as type and port. (closes issue #11916) Reported
- by: mnnojd
-
- * /, channels/chan_iax2.c: Merged revisions 102968 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r102968 | file | 2008-02-08 11:08:20 -0400 (Fri, 08 Feb
- 2008) | 4 lines Make sure the presence of dbsecret is factored
- into user scoring. (closes issue #11952) Reported by: bbhoss
- ........
-
-2008-02-07 21:37 +0000 [r102933] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_chanspy.c: This is a combination new feature/bug fix for
- app_chanspy. New feature: Add the 'e' option, which takes as an
- argument a list of interfaces separated by colons. This way, you
- will only be able to spy on this limited list of interfaces. Bug
- fix: change some pointer checks to ast_strlen_zero so that spying
- would work properly even if no channel was specified as the first
- argument to chanspy. (closes issue #10072) Reported by:
- xmarksthespot Patches:
- bugfix+newfeature10072patchtotrunkrev102726.diff uploaded by
- xmarksthespot (license 16) Tested by: xmarksthespot, mvanbaak
-
-2008-02-07 21:08 +0000 [r102906-102908] Michiel van Baak <michiel@vanbaak.info>
-
- * apps/app_adsiprog.c: whitespace fixes only
-
- * apps/app_alarmreceiver.c: There she goes! First commit from me to
- trunk \o/ Make app_alarmreceiver honor code guidelines and fix
- whitespace errors. No functional changes.
-
-2008-02-07 20:02 +0000 [r102859] Jason Parker <jparker@digium.com>
-
- * /, main/features.c: Merged revisions 102858 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r102858 | qwell | 2008-02-07 13:53:55 -0600 (Thu, 07 Feb 2008) |
- 7 lines Specify which digit string was matched in debug message.
- (closes issue #11949) Reported by: dimas Patches:
- v1-feature-debug.patch uploaded by dimas (license 88) ........
-
-2008-02-07 16:47 +0000 [r102808] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, configs/zapata.conf.sample: Merged revisions 102807 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r102807 | kpfleming | 2008-02-07 10:41:55 -0600 (Thu, 07 Feb
- 2008) | 2 lines document usage of 'transfer' configuration option
- for ISDN PRI switch-side transfers ........
-
-2008-02-06 20:12 +0000 [r102777] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Add the channel's unique id to the AgentCalled
- manager event to make it more consistent with other manager
- events.
-
-2008-02-06 18:01 +0000 [r102726] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 102725 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r102725 | file | 2008-02-06 13:59:23 -0400 (Wed, 06 Feb 2008) | 2
- lines Only consider a T.38-only INVITE compatible if we have both
- a joint capability between us and them and if they provided T.38.
- ........
-
-2008-02-06 16:23 +0000 [r102700] Terry Wilson <twilson@digium.com>
-
- * funcs/func_realtime.c: Add REALTIME_STORE and REALTIME_DESTROY
- dialplan functions provided by sergee. I just added the ability
- to set multiple fields at once after discussions with Tilghman
- and Russell. Currently limited to 30 fields. (closes issue
- #11887) Reported by: sergee Patches:
- rt-func-store-destroy-multivalue.diff uploaded by otherwiseguy
- (license 396) Tested by: sergee, otherwiseguy
-
-2008-02-06 15:20 +0000 [r102652] Russell Bryant <russell@digium.com>
-
- * /, configs/features.conf.sample: Merged revisions 102651 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r102651 | russell | 2008-02-06 09:19:41 -0600 (Wed, 06 Feb 2008)
- | 3 lines Clarify setting DYNAMIC_FEATURES so that it gets
- inherited by outbound channels. (due to a discussion between me
- and a user via email) ........
-
-2008-02-06 03:05 +0000 [r102602] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 102576 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r102576 | tilghman | 2008-02-05 18:26:02 -0600 (Tue, 05
- Feb 2008) | 4 lines Move around some defines to unbreak ODBC
- storage. (closes issue #11932) Reported by: snuffy ........
-
-2008-02-06 00:08 +0000 [r102501-102550] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Remove an extra debug message I left in
-
- * channels/chan_unistim.c, apps/app_dial.c, main/pbx.c,
- apps/app_privacy.c, apps/app_alarmreceiver.c, res/res_jabber.c,
- apps/app_followme.c, main/loader.c, channels/chan_usbradio.c,
- main/tcptls.c, res/res_agi.c, apps/app_minivm.c,
- apps/app_dumpchan.c, main/logger.c, apps/app_zapras.c,
- main/astmm.c: Get rid of any remaining ast_verbose calls in the
- code in favor of ast_verb (closes issue #11934) Reported by:
- mvanbaak Patches: 20080205_astverb-2.diff.txt uploaded by
- mvanbaak (license 7)
-
- * apps/app_voicemail.c: Change verbose messages to use the ast_verb
- macro. (closes issue #11931) Reported by: snuffy Patches:
- bug-11931.diff uploaded by snuffy (license 35)
-
-2008-02-05 20:51 +0000 [r102500] Jason Parker <jparker@digium.com>
-
- * main/pbx.c: Change where priority of a goto is adjusted.
- Partially reverts 102272. Closes issue #11929 (credit to file for
- fix suggestion - we still <3 you)
-
-2008-02-05 20:03 +0000 [r102454] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_mgcp.c: Merged revisions 102453 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r102453 | mmichelson | 2008-02-05 14:02:44 -0600 (Tue,
- 05 Feb 2008) | 8 lines Clear the DTMF buffer on hangup. (closes
- issue #11919) Reported by: eferro Patches:
- mgcp_dtmfclean_on_hangup.diff uploaded by eferro (license 337)
- Tested by: eferro ........
-
-2008-02-05 19:58 +0000 [r102379-102452] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Yeah yeah, I broke building on trunk. Shoot
- me.
-
- * /, channels/chan_sip.c: Merged revisions 102450 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r102450 | file | 2008-02-05 15:52:30 -0400 (Tue, 05 Feb 2008) | 3
- lines If a REGISTER attempt comes in that is a retransmission of
- a previous REGISTER do not create a new nonce value. (issue
- #BE-381) ........
-
- * /, res/res_clioriginate.c: Merged revisions 102378 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r102378 | file | 2008-02-05 11:09:29 -0400 (Tue, 05 Feb
- 2008) | 4 lines Perform dialing asynchronously when using the
- originate CLI command so the CLI does not appear to block.
- (closes issue #11927) Reported by: bbhoss ........
-
-2008-02-04 21:15 +0000 [r102329] Tilghman Lesher <tlesher@digium.com>
-
- * utils/muted.c, /, configure, include/asterisk/autoconfig.h.in,
- configure.ac, main/asterisk.c: Merged revisions 102323 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r102323 | tilghman | 2008-02-04 15:06:09 -0600 (Mon, 04 Feb 2008)
- | 7 lines Cross-platform fix: OS X now deprecates the use of the
- daemon(3) API. (closes issue #11908) Reported by: oej Patches:
- 20080204__bug11908.diff.txt uploaded by Corydon76 (license 14)
- Tested by: Corydon76 ........
-
-2008-02-04 18:39 +0000 [r102297] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c: Add line numbers to warning/error messages
- (and pretty up some existing ones). (closes issue #11894)
- Reported by: jmls Patches: chan_zap.patch uploaded by jmls
- (license 141)
-
-2008-02-04 15:16 +0000 [r102272] Joshua Colp <jcolp@digium.com>
-
- * main/pbx.c: Update handling of asyncgoto so it properly works on
- channels that are currently executing a PBX. (closes issue
- #11914) Reported by: arnd (closes issue #11753) Reported by:
- johan
-
-2008-02-04 14:37 +0000 [r102262] Jason Parker <jparker@digium.com>
-
- * configs/extensions.ael.sample, configs/extensions.lua.sample:
- Change examples to use G here also. Closes issue #11875
-
-2008-02-04 05:32 +0000 [r102190-102238] Tilghman Lesher <tlesher@digium.com>
-
- * /, funcs/func_strings.c: Merged revisions 102214 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r102214 | tilghman | 2008-02-03 23:10:02 -0600 (Sun, 03
- Feb 2008) | 6 lines Missing braces. (closes issue #11912)
- Reported by: dimas Patches: sprintf.patch uploaded by dimas
- (license 88) ........
-
- * main/manager.c: CoreSettings and CoreStatus are missing the
- terminating "\r\n". Also, some miscellaneous spacing and
- initialization issues. (closes issue #11909) Reported by: srt
- Patches: patch-11909-2.diff uploaded by srt (license 378) Tested
- by: srt
-
-2008-02-03 16:46 +0000 [r102091-102143] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 102142 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r102142 | oej | 2008-02-03 17:38:12 +0100 (Sön, 03 Feb 2008) | 8
- lines Use the same CSEQ on CANCEL as on INVITE (according to RFC
- 3261) (closes issue #9492) Reported by: kryptolus Patches:
- bug9492.txt uploaded by oej (license 306) Tested by: oej ........
-
- * /, channels/chan_sip.c: Merged revisions 102090 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r102090 | oej | 2008-02-03 11:37:32 +0100 (Sön, 03 Feb 2008) | 8
- lines Handle ACK and CANCEL in an invite transaction - even if we
- get INFO transactions during the actual call setup. (closes issue
- #10567) Reported by: jacksch Tested by: oej Patch by: oej
- inspired by suggestions from neutrino88 in the bug tracker
- ........
-
-2008-02-03 06:43 +0000 [r102064] Russell Bryant <russell@digium.com>
-
- * configure, configure.ac: Change the version number in the
- configure script from 1.4 to 1.6
-
-2008-02-02 06:10 +0000 [r101990-102037] Russell Bryant <russell@digium.com>
-
- * include/asterisk/event.h: The documentation page has to be in its
- own comment block to work, apparently. Fix it up!
-
- * /, channels/chan_sip.c: Merged revisions 101989 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r101989 | russell | 2008-02-01 17:06:32 -0600 (Fri, 01 Feb 2008)
- | 5 lines Change the SDP_SAMPLE_RATE macro. It turns out that
- even though G.722 is 16 kHz, it is supposed to specified as 8 kHz
- in the RTP, and RTP timestamps are supposed to be calculated
- based on 8 kHz. (Apparently this is due to a bug in a spec, but
- people follow it anyway, because it's the spec ...) ........
-
-2008-02-01 22:12 +0000 [r101873-101943] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 101942 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r101942 | tilghman | 2008-02-01 15:54:28 -0600 (Fri, 01
- Feb 2008) | 8 lines Fix the VM_DUR variable for forwarded
- voicemail, and fixed several other bugs while I'm in the area.
- (closes issue #11615) Reported by: jamessan Patches:
- 20071226__bug11615__2.diff.txt uploaded by Corydon76 (license 14)
- Tested by: Corydon76, jamessan ........
-
- * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
- Merged revisions 101894 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r101894 | tilghman | 2008-02-01 13:36:12 -0600 (Fri, 01 Feb 2008)
- | 2 lines Change detection of getifaddrs to use
- AST_C_COMPILE_CHECK, backported from trunk (as suggested by
- kpfleming) ........
-
- * res/res_config_curl.c: Fix multi, when using the LIKE query.
- (closes issue #11889) Reported by: jmls Patches:
- res_config_curl.patch uploaded by jmls (license 141) Tested by:
- jmls
-
-2008-02-01 18:24 +0000 [r101869] Jason Parker <jparker@digium.com>
-
- * apps/app_authenticate.c: Comparison, not set :) Thanks mvanbaak.
-
-2008-02-01 18:08 +0000 [r101824] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_odbc.c, configs/res_odbc.conf.sample: Clarify the pooling
- functionality by changing the config file keyword
-
-2008-02-01 17:44 +0000 [r101823] Jason Parker <jparker@digium.com>
-
- * /, apps/app_authenticate.c: Move an feof() call to before the
- fgets(). This would have exited the loop early if you had an
- authentication file with no newline at the end.
-
-2008-02-01 17:28 +0000 [r101819-101821] Russell Bryant <russell@digium.com>
-
- * /, apps/app_authenticate.c: Merged revisions 101818 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r101818 | russell | 2008-02-01 11:23:47 -0600 (Fri, 01
- Feb 2008) | 4 lines Don't overwrite the last character of a line
- if it's not a newline. This would happen if the last line in the
- file doesn't have a newline. (pointed out by Qwell) ........
-
-2008-02-01 16:01 +0000 [r101773] Tilghman Lesher <tlesher@digium.com>
-
- * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
- main/acl.c: Merged revisions 101772 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r101772 | tilghman | 2008-02-01 09:55:58 -0600 (Fri, 01 Feb 2008)
- | 2 lines Compatibility fix for OpenWRT (reported by Brian
- Capouch via the mailing list) ........
-
-2008-02-01 06:27 +0000 [r101694-101746] Russell Bryant <russell@digium.com>
-
- * apps/app_authenticate.c: simplify some code, tweak formatting,
- and reduce indentation
-
- * apps/app_authenticate.c: reduce a level of indentation
-
- * apps/app_channelredirect.c: Get rid of a goto where there was no
- extra cleanup happening at the exit point
-
- * /, channels/chan_iax2.c: Merged revisions 101693 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r101693 | russell | 2008-01-31 18:32:49 -0600 (Thu, 31
- Jan 2008) | 8 lines Add some more sanity checking on IAX2 dial
- strings for the case that no peer or hostname was provided, which
- is the one part of the dial string that is absolutely required.
- If it's not there, bail out. (closes issue #11897) Reported by
- sokhapkin Patch by me ........
-
-2008-02-01 00:08 +0000 [r101650] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_amd.c: Merged revisions 101649 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r101649 | mmichelson | 2008-01-31 18:06:37 -0600 (Thu, 31 Jan
- 2008) | 9 lines From bugtracker: "fix totalAnalysisTime to handle
- periods of no channel activity" (closes issue #9256) Reported by:
- cmaj Patches: amd-dont-wait-too-long-for-frames-take3.diff.txt
- uploaded by cmaj (license 111) Tested by: cmaj, skygreg, ZX81,
- rjain ........
-
-2008-01-31 23:14 +0000 [r101611] Russell Bryant <russell@digium.com>
-
- * /, main/translate.c, main/file.c: Merged revisions 101601 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r101601 | russell | 2008-01-31 17:10:06 -0600 (Thu, 31 Jan 2008)
- | 12 lines Fix a couple of places where ast_frfree() was not
- called on a frame that came from a translator. This showed itself
- by g729 decoders not getting released. Since the flag inside the
- translator frame never got unset by freeing the frame to indicate
- it was no longer in use, the translators never got destroyed, and
- thus the g729 licenses were not released. (closes issue #11892)
- Reported by: xrg Patches: 11892.diff uploaded by russell (license
- 2) Tested by: xrg, russell ........
-
-2008-01-31 22:12 +0000 [r101578-101580] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Forgot an !
-
- * apps/app_queue.c: A change I made to accommodate the "linear"
- strategy in trunk caused queue strategies to not be loaded from
- realtime queues. This commit fixes that. Thanks to jmls for
- pointing this problem out to me on IRC. This also contains some
- changes to S_OR where it should be used. Thanks to Qwell for
- pointing these out.
-
-2008-01-31 21:33 +0000 [r101577] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Fix a simple deadlock that was introduced
- _right_ before this code got merged into trunk. (closes issue
- #11895, reported by pj, patched by me)
-
-2008-01-31 21:31 +0000 [r101532-101576] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Handle the case of a NULL state_interface when
- checking a realtime member. Thanks to jmls for finding this
- issue.
-
- * /, res/res_monitor.c: Merged revisions 101531 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r101531 | mmichelson | 2008-01-31 15:00:24 -0600 (Thu, 31 Jan
- 2008) | 10 lines 1. Prevent the addition of an extra '/' to the
- beginning of an absolute pathname. 2. If ast_monitor_change_fname
- is called and the new filename is the same as the old, then exit
- early and don't set the filename_changed field in the monitor
- structure. Setting it in this case was causing ast_monitor_stop
- to erroneously delete them. (closes issue #11741) Reported by:
- garlew Tested by: putnopvut ........
-
-2008-01-31 19:54 +0000 [r101483] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions
- 101482 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r101482 | qwell | 2008-01-31 13:52:49 -0600 (Thu, 31 Jan 2008) |
- 4 lines Solaris compat fixes for struct in_addr funkiness. Issue
- #11885, patch by snuffy. ........
-
-2008-01-31 19:43 +0000 [r101481] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, /: Merged revisions 101480 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r101480 | murf | 2008-01-31 12:30:37 -0700 (Thu, 31 Jan 2008) | 1
- line closes issue #11845; that's the one where there's a 1004
- byte cdr leak with every AMI Redirect to a zap channel ........
-
-2008-01-31 19:20 +0000 [r101416-101449] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 101433 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r101433 | russell | 2008-01-31 13:17:05 -0600 (Thu, 31
- Jan 2008) | 2 lines Add more missing locking of the agents list
- ... ........
-
- * /, channels/chan_agent.c: Merged revisions 101413-101414 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r101413 | russell | 2008-01-31 13:04:52 -0600 (Thu, 31 Jan 2008)
- | 2 lines Add missing locking to the find_agent() function.
- ........ r101414 | russell | 2008-01-31 13:07:46 -0600 (Thu, 31
- Jan 2008) | 3 lines Move the locking from find_agent() into the
- agent dialplan function handler to ensure that the agent doesn't
- disappear while we're looking at it. ........
-
-2008-01-31 15:36 +0000 [r101393] Joshua Colp <jcolp@digium.com>
-
- * funcs/func_realtime.c: Add missing braces. (closes issue #11886)
- Reported by: sergee Patches: func_realtime_fix-r101392.diff
- uploaded by sergee (license 138)
-
-2008-01-31 05:28 +0000 [r101373] Russell Bryant <russell@digium.com>
-
- * CHANGES: remove entry that is no longer in the tree
-
-2008-01-30 23:10 +0000 [r101344] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: The deprecation of "username" in favor of
- "defaultuser" for SIP peers unfortunately broke realtime
- configurations which still used the "username" field. This was
- taken care of properly when reading from realtime but was not
- handled properly when updating a realtime peer. This change also
- adds a deprecation NOTICE CLI message that will print if using
- the deprecated "username" field. (closes issue #11880) Reported
- by: cabal95 Patches: 11880.patch uploaded by putnopvut (license
- 60) Tested by: cabal95
-
-2008-01-30 20:08 +0000 [r101322] Olle Johansson <oej@edvina.net>
-
- * configs/cli.conf.sample: Clarify configuration file that can be
- misunderstood
-
-2008-01-30 19:03 +0000 [r101296] Jason Parker <jparker@digium.com>
-
- * apps/app_controlplayback.c: Allow disabling the default
- ffwd/rewind keys in the ControlPlayback application. This is done
- in a backward compat way. If the "default" key for ffwd/rew is
- used for another option (such as stop), the "default" is removed.
- (closes issue #11754) Reported by: johan Patches:
- app_controlplayback.c.option3.patch uploaded by johan (license
- 334) Tested by: johan, qwell
-
-2008-01-30 17:12 +0000 [r101271] Olle Johansson <oej@edvina.net>
-
- * configs/rtppage.conf.sample (removed), apps/app_rtppage.c
- (removed): Removing applications that wasn't ready for svn trunk,
- as trunk now has pre-release status.
-
-2008-01-30 16:54 +0000 [r101269] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: Make the VoicemailUsersList AMI command
- consistent with other manager list functions. (closes issue
- #11874) Reported by: srt Patches: voicemail_ami-11847.patch
- uploaded by srt (license 378)
-
-2008-01-30 16:39 +0000 [r101267-101268] Olle Johansson <oej@edvina.net>
-
- * include/asterisk/rtp.h, main/rtp.c: - doxygen fixes - change
- function to void because it always returned the same value and no
- one read it.
-
- * main/rtp.c: Formatting fixes
-
-2008-01-30 15:42 +0000 [r101224] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_rtppage.c: Get trunk to compile
-
-2008-01-30 15:42 +0000 [r101223] Joshua Colp <jcolp@digium.com>
-
- * /, main/slinfactory.c: Merged revisions 101222 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r101222 | file | 2008-01-30 11:41:04 -0400 (Wed, 30 Jan 2008) | 4
- lines Fix an issue where if a frame of higher sample size
- preceeded a frame of lower sample size and ast_slinfactory_read
- was called with a sample size of the combined values or higher a
- crash would happen. (closes issue #11878) Reported by: stuarth
- ........
-
-2008-01-30 15:36 +0000 [r101221] Olle Johansson <oej@edvina.net>
-
- * CHANGES: Update CHANGES with rtppage
-
-2008-01-30 15:35 +0000 [r101220] Jason Parker <jparker@digium.com>
-
- * /, configs/extensions.conf.sample: Merged revisions 101219 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #11875) ........ r101219 | qwell | 2008-01-30 09:34:37
- -0600 (Wed, 30 Jan 2008) | 5 lines Change default config to use
- descending channel order of groups, rather than ascending. Fixes
- a potential source of confusion in glare-type situations. Issue
- 11875, reported by JimVanM. ........
-
-2008-01-30 15:30 +0000 [r101218] Olle Johansson <oej@edvina.net>
-
- * configs/rtppage.conf.sample (added), apps/app_rtppage.c (added):
- Add rtppage() application to do multicast or unicast RTP paging
- to SIP phones. (closes issue #11797) Reported by: macbrody
- Patches: app_rtppage-20080130.c uploaded by macbrody (license
- 352)
-
-2008-01-30 15:27 +0000 [r101217] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 101216 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r101216 | mmichelson | 2008-01-30 09:23:00 -0600 (Wed, 30 Jan
- 2008) | 5 lines Fix a logic error with regards to autofill. Prior
- to this change, it was possible for a caller to go out of turn if
- autofill were enabled and callers ahead in the queue were
- attempting to call a member. This change fixes this. ........
-
-2008-01-30 12:48 +0000 [r101196] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_sip.c: simplify this code and eliminate the return
- value cast that is no longer necessary
-
-2008-01-30 11:27 +0000 [r101153-101154] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c, include/asterisk/channel.h: Constifying the
- interface to get pvt_ids in the bridge, based on suggestion from
- (const char *) Kevin. Thanks!
-
- * /, channels/chan_sip.c: Merged revisions 101152 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r101152 | oej | 2008-01-30 12:20:31 +0100 (Ons, 30 Jan 2008) | 7
- lines Stop musiconhold on attended transfer. (closes issue
- #11872) Reported by: gareth Patches: svn-101018.patch uploaded by
- gareth (license 208) ........
-
-2008-01-30 00:58 +0000 [r101126] Jason Parker <jparker@digium.com>
-
- * CHANGES: Fix a typo
-
-2008-01-30 00:04 +0000 [r101082] Russell Bryant <russell@digium.com>
-
- * CHANGES, apps/app_speech_utils.c: Add the 'n' option to
- SpeechBackground, which has the application not answer the
- channel if it has not already been answered. (closes SPD-51)
-
-2008-01-29 23:59 +0000 [r101081] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * /, build_tools/make_version: Merged revisions 101080 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r101080 | dhubbard | 2008-01-29 17:50:42 -0600 (Tue, 29
- Jan 2008) | 1 line updated build_tools to handle the autotag
- directory structure changes; changes related to BE-353. Patch by
- The Russell and reviewed by The Me. ........
-
-2008-01-29 23:02 +0000 [r101036] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 101035 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r101035 | mmichelson | 2008-01-29 17:02:03 -0600 (Tue, 29 Jan
- 2008) | 3 lines Remove a memory leak from updating realtime
- queues ........
-
-2008-01-29 22:04 +0000 [r101018] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_config_curl.c: Oops, a sizeof error
-
-2008-01-29 19:41 +0000 [r100974] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 100973 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r100973 | mmichelson | 2008-01-29 13:39:00 -0600 (Tue, 29 Jan
- 2008) | 6 lines Fixing an erroneous return value returned when
- attempting to pause or unpause a queue member fails. Fixes
- BE-366, thanks to John Bigelow for writing the patch. ........
-
-2008-01-29 17:44 +0000 [r100933] Russell Bryant <russell@digium.com>
-
- * /, main/Makefile: Merged revisions 100932 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r100932 | russell | 2008-01-29 11:43:41 -0600 (Tue, 29 Jan 2008)
- | 4 lines Fix the last couple of issues related to building from
- a path that contains spaces. (closes issue #11834) ........
-
-2008-01-29 17:42 +0000 [r100931] Jason Parker <jparker@digium.com>
-
- * /, channels/misdn_config.c: Merged revisions 100930 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r100930 | qwell | 2008-01-29 11:41:43 -0600 (Tue, 29 Jan
- 2008) | 6 lines Initialize an array to 0s if config option not
- specified. (closes issue #11860) Patches:
- misdn_get_config.v1.diff uploaded by IgorG (license 20) ........
-
-2008-01-29 17:22 +0000 [r100900-100928] Russell Bryant <russell@digium.com>
-
- * Makefile, /: Merged revisions 100922 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r100922 | russell | 2008-01-29 11:21:33 -0600 (Tue, 29 Jan 2008)
- | 3 lines Use GNU make magic instead of shell magic to escape
- spaces in the working directory. (related to issue #11834)
- ........
-
- * Makefile, /: Merged revisions 100882 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r100882 | russell | 2008-01-29 11:06:43 -0600 (Tue, 29 Jan 2008)
- | 6 lines Fix building Asterisk when the working path has spaces
- in it. (closes issue #11834) Reported by: spendergrass Patched
- by: me ........
-
-2008-01-29 16:14 +0000 [r100843] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 100835 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r100835 | qwell | 2008-01-29 10:10:00 -0600 (Tue, 29 Jan 2008) |
- 5 lines Allow zap groups above 30 to work properly. (closes issue
- #11590) Reported by: tbsky ........
-
-2008-01-29 15:30 +0000 [r100833] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Make externip work as documented. If no port
- is specified it will use the value of bindport instead of always
- being 5060. (closes issue #11858) Reported by: hmodes
-
-2008-01-29 10:50 +0000 [r100794-100795] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /: Merged revisions 100793 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r100793 | crichter | 2008-01-29 11:36:19 +0100 (Di, 29
- Jan 2008) | 1 line fixed potential segfault in misdn show
- channels CLI command ........
-
- * channels/chan_misdn.c, /: Merged revisions 96199 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r96199 | crichter | 2008-01-03 13:12:27 +0100 (Do, 03
- Jan 2008) | 1 line make sure frame is completely clean, before we
- send it to asterisk as DTMF. If we don't make it clean, it
- happens that one way audio occurs.. ........
-
-2008-01-29 09:18 +0000 [r100741-100767] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 100740 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r100740 | oej | 2008-01-29 09:26:48 +0100 (Tis, 29 Jan 2008) | 8
- lines (closes issue #11736) Reported by: MVF Patches:
- bug11736-2.diff uploaded by oej (license 306) Tested by: oej,
- MVF, revolution (russellb: This was the showstopper for the
- release.) ........
-
- * channels/chan_sip.c: Removing code that wasn't supposed to be
- there at all, only at an experimental stage before I found
- another solution. Thanks Kevin, for reminding me.
-
-2008-01-28 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.6.0-beta2 released.
-
-2008-01-28 21:11 +0000 [r100679] Jason Parker <jparker@digium.com>
-
- * build_tools/menuselect-deps.in, configs/vpb.conf.sample (added),
- doc/tex/channelvariables.tex, makeopts.in: Reintroduce more
- chan_vpb stuff that was removed in r100421 and r100422
-
-2008-01-28 21:07 +0000 [r100678] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_vpb.cc (added), configure,
- include/asterisk/autoconfig.h.in, configure.ac,
- channels/Makefile: Re-inserting chan_vpb into trunk.
-
-2008-01-28 21:05 +0000 [r100677] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c, /: Merged revisions 100675 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r100675 | tilghman | 2008-01-28 15:02:02 -0600 (Mon, 28 Jan 2008)
- | 2 lines WaitExten didn't handle AbsoluteTimeout properly (went
- to 't' instead of 'T') ........
-
-2008-01-28 21:02 +0000 [r100676] Jason Parker <jparker@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 100672 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #11795) ........ r100672 | qwell | 2008-01-28 14:42:43
- -0600 (Mon, 28 Jan 2008) | 7 lines When using ODBC_STORAGE, make
- sure we put greeting files into the database like we do with the
- others. Issue #11795 Reported by: dimas Patches: vmgreet.patch
- uploaded by dimas (license 88) ........
-
-2008-01-28 20:40 +0000 [r100632-100671] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Fix up some T38 state change issues. (closes
- issue #11630) Reported by: dimas Patches: v2-sip-t38state.patch
- uploaded by dimas (license 88)
-
- * channels/chan_sip.c: Fix up two scheduling issues. In one
- instance a scheduled item was not deleted when it should have
- been and in the other it was scheduled again when it shouldn't
- have been.
-
-2008-01-28 18:41 +0000 [r100630-100631] Russell Bryant <russell@digium.com>
-
- * main/features.c: Merge rev 100626 from Asterisk 1.4. The svnmerge
- of this commit was a NoOp, since res_features doesn't exist in
- trunk. Thanks to qwell for pointing it out!
-
- * /, channels/chan_sip.c: Merged revisions 100629 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r100629 | russell | 2008-01-28 12:34:20 -0600 (Mon, 28 Jan 2008)
- | 5 lines For some reason, the use of this strdupa() is leading
- to memory corruption on freebsd sparc64. This trivial workaround
- fixes it. (closes issue #10300, closes issue #11857, reported by
- mattias04 and Home-of-the-Brave) ........
-
-2008-01-28 18:27 +0000 [r100628] Tilghman Lesher <tlesher@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac,
- main/logger.c: Normalize the detection for execinfo, so that
- Linux (glibc) and other platforms with libexecinfo will generate
- inline stack backtraces correctly.
-
-2008-01-28 18:27 +0000 [r100627] Russell Bryant <russell@digium.com>
-
- * /: Merged revisions 100626 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r100626 | russell | 2008-01-28 12:26:31 -0600 (Mon, 28 Jan 2008)
- | 7 lines Fix a crash in ast_masq_park_call() (issue #11342)
- Reported by: DEA Patches: res_features-park.txt uploaded by DEA
- (license 3) ........
-
-2008-01-28 18:24 +0000 [r100625] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 100624 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r100624 | qwell | 2008-01-28 12:23:09 -0600 (Mon, 28 Jan 2008) |
- 1 line Correct a comment which made little/no sense. ........
-
-2008-01-28 17:21 +0000 [r100565-100582] Russell Bryant <russell@digium.com>
-
- * main/channel.c, channels/chan_local.c, /,
- include/asterisk/channel.h: Merged revisions 100581 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r100581 | russell | 2008-01-28 11:15:41 -0600 (Mon, 28
- Jan 2008) | 9 lines Make some deadlock related fixes. These bugs
- were discovered and reported internally at Digium by Steve Pitts.
- - Fix up chan_local to ensure that the channel lock is held
- before the local pvt lock. - Don't hold the channel lock when
- executing the timing function, as it can cause a deadlock when
- using chan_local. This actually changes the code back to be how
- it was before the change for issue #10765. But, I added some
- other locking that I think will prevent the problem reported
- there, as well. ........
-
- * main/pbx.c: Clean up some formatting, and simplify a bit of code
- using ast_str
-
-2008-01-28 13:57 +0000 [r100549] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Don't do a network byte order conversion
- when setting the socket's port variable to that of bindaddr's. It
- is already in the correct network byte order. (closes issue
- #11800) Reported by: hmodes
-
-2008-01-28 04:43 +0000 [r100514-100533] Russell Bryant <russell@digium.com>
-
- * main/channel.c: Make a couple more uses of ARRAY_LEN, and convert
- some spaces to tabs
-
- * main/channel.c: - Simplify a line with ARRAY_LEN() - Make a few
- little formatting changes
-
- * main/channel.c: These readlocks always fail for me on my mac, and
- I saw it happen again today on another mac. We ignore the return
- value of locking operations almost everywhere in Asterisk. So,
- ignore these, as well, so Asterisk will actually work on systems
- where this is occurring while I look into what the issue is.
-
-2008-01-27 23:14 +0000 [r100488-100497] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_sip.c, include/asterisk/sched.h,
- channels/chan_iax2.c: With the switch to the ast_sched_replace*
- API in trunk, we lose the correction that was just merged from
- 1.4, so this is a changeover to those APIs to use the macro
- versions, so that we properly detect errors from ast_sched_del,
- instead of simply ignoring the return values.
-
- * main/cdr.c, channels/chan_misdn.c, main/dnsmgr.c, /,
- channels/chan_sip.c, channels/chan_h323.c,
- include/asterisk/sched.h, main/file.c, pbx/pbx_dundi.c,
- channels/chan_iax2.c, main/rtp.c, channels/chan_mgcp.c: Merged
- revisions 100465 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r100465 | tilghman | 2008-01-27 15:59:53 -0600 (Sun, 27 Jan 2008)
- | 11 lines When deleting a task from the scheduler, ignoring the
- return value could possibly cause memory to be accessed after it
- is freed, which causes all sorts of random memory corruption.
- Instead, if a deletion fails, wait a bit and try again (noting
- that another thread could change our taskid value). (closes issue
- #11386) Reported by: flujan Patches: 20080124__bug11386.diff.txt
- uploaded by Corydon76 (license 14) Tested by: Corydon76, flujan,
- stuarth` ........
-
-2008-01-25 22:54 +0000 [r100421-100422] Jason Parker <jparker@digium.com>
-
- * doc/tex/channelvariables.tex: Get rid of that last little bit.
-
- * build_tools/menuselect-deps.in, configs/vpb.conf.sample
- (removed), makeopts.in: Remove more remnants of chan_vpb
-
-2008-01-25 22:39 +0000 [r100419-100420] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_vpb.cc (removed), configure,
- include/asterisk/autoconfig.h.in, configure.ac,
- channels/Makefile, .cleancount: Removing chan_vpb from the tree
-
-2008-01-25 21:26 +0000 [r100379] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 100378 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r100378 | qwell | 2008-01-25 15:24:49 -0600 (Fri, 25 Jan 2008) |
- 2 lines This would have never been true, since we're passing
- (sizeof(req.data) - 1) as the len to recvfrom(). ........
-
-2008-01-25 20:51 +0000 [r100361] Kevin P. Fleming <kpfleming@digium.com>
-
- * apps/app_rpt.c: correct a real problem and silence an annoying
- compiler warning
-
-2008-01-25 14:53 +0000 [r100344] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Insure that we are not going to pass a NULL
- pointer to add_to_interfaces. (closes issue #11840) Reported by:
- junky
-
-2008-01-25 02:52 +0000 [r100325] Joshua Colp <jcolp@digium.com>
-
- * main/dial.c, include/asterisk/dial.h: Add an API call that steals
- the answered channel so that a destruction of the dialing
- structure does not hang it up.
-
-2008-01-24 22:58 +0000 [r100307] Tilghman Lesher <tlesher@digium.com>
-
- * Makefile, build_tools/make_defaults_h: Use the set ASTDBDIR as
- the default, too
-
-2008-01-24 22:36 +0000 [r100305-100306] Kevin P. Fleming <kpfleming@digium.com>
-
- * include/asterisk/app.h: ummm... might be good if this macro
- argument was actually used :-)
-
- * include/asterisk/app.h: add the ability to define a structure
- type for argument parsing when it would be useful to be able to
- pass it between functions
-
-2008-01-24 22:02 +0000 [r100266] James Golovich <james@gnuinter.net>
-
- * channels/chan_sip.c: Fix simple whitespace issue
-
-2008-01-24 22:01 +0000 [r100265] Kevin P. Fleming <kpfleming@digium.com>
-
- * include/asterisk/app.h, /: Merged revisions 100264 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r100264 | kpfleming | 2008-01-24 15:57:41 -0600 (Thu, 24
- Jan 2008) | 2 lines make these macros not assume that the only
- other field in the structure is 'argc'... this is true when
- someone uses AST_DECLARE_APP_ARGS, but it's perfectly reasonable
- to define your own structure as long as it has the right fields
- ........
-
-2008-01-24 20:32 +0000 [r100245] Joshua Colp <jcolp@digium.com>
-
- * main/features.c: Minor cosmetic change...
-
-2008-01-24 18:35 +0000 [r100224] James Golovich <james@gnuinter.net>
-
- * main/astmm.c: Increase the size of filenames stored when astmm is
- used. If the path length was long they would be truncated and
- grouped together with whatever matches
-
-2008-01-24 17:47 +0000 [r100206] Joshua Colp <jcolp@digium.com>
-
- * configs/rtp.conf.sample, CHANGES, main/rtp.c: Merge in strictrtp
- branch. This adds a strictrtp option to rtp.conf which drops
- packets that do not come from the remote party. (closes issue
- #8952) Reported by: amorsen
-
-2008-01-24 17:24 +0000 [r100169] Russell Bryant <russell@digium.com>
-
- * /, main/asterisk.c: Merged revisions 100164 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r100164 | russell | 2008-01-24 11:22:09 -0600 (Thu, 24 Jan 2008)
- | 2 lines Update main Asterisk copyright info to 2008 ........
-
-2008-01-24 16:47 +0000 [r100121-100139] Jason Parker <jparker@digium.com>
-
- * /, res/res_phoneprov.c, main/acl.c: Merged revisions 100138 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r100138 | qwell | 2008-01-24 10:41:29 -0600 (Thu, 24 Jan 2008) |
- 6 lines Fix compilation on Solaris. (closes issue #11832)
- Patches: bug-11832.diff uploaded by snuffy (license 35) ........
-
- * channels/chan_sip.c, main/features.c: Move chan_local dependency
- into places (only one) that previously depended on res_features,
- and used local channels
-
-2008-01-24 15:54 +0000 [r100076-100112] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_zap.c, channels/chan_sip.c, channels/chan_iax2.c,
- channels/chan_mgcp.c: Remove dependency on res_features from some
- channel drivers. It is now part of the core and no longer exists
- as a module.
-
- * main/channel.c: Some more cosmetic changes.
-
- * main/channel.c: Add some spacing.
-
- * main/dial.c: Test hopefully over.
-
- * main/dial.c: Testing something...
-
-2008-01-24 00:04 +0000 [r100057] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_sip.c: fix flag bit definitions to make code from
- issue #11049 actually work; along the way, clarify comments and
- add some dummy flag definitions for other multi-bit flags to
- hopefully stop this from happening in the future (closes issue
- #11049)
-
-2008-01-23 23:09 +0000 [r100039] Jason Parker <jparker@digium.com>
-
- * res/res_features.c (removed), main/Makefile, main/features.c
- (added), include/asterisk/_private.h, CHANGES, .cleancount,
- main/asterisk.c, main/loader.c, include/asterisk/features.h: Move
- code from res_features into (new file) main/features.c
-
-2008-01-23 22:00 +0000 [r100021] Russell Bryant <russell@digium.com>
-
- * CREDITS: Add Sergey Tamkovich to CREDITS. Thank you for your
- contributions!
-
-2008-01-23 21:11 +0000 [r99979-99980] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 99978 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r99978 | oej | 2008-01-23 22:07:16 +0100 (Ons, 23 Jan 2008) | 7
- lines Second attempt. Don't change invitestate when receiving 18x
- messages in CANCEL state. (issue #11736) Reported by: MVF Patch
- by oej. ........
-
- * /, channels/chan_sip.c: Merged revisions 99977 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r99977 | oej | 2008-01-23 21:58:20 +0100 (Ons, 23 Jan 2008) | 9
- lines Make sure we don't cancel destruction on calls in CANCEL
- state, even if we get 183 while waiting for answer on our CANCEL.
- (issue #11736) Reported by: MVF Patches: bug11736.txt uploaded by
- oej (license 306) Tested by: MVF ........
-
-2008-01-23 20:26 +0000 [r99976] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_externalivr.c: Merged revisions 99975 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r99975 | mmichelson | 2008-01-23 14:25:00 -0600 (Wed, 23
- Jan 2008) | 3 lines Fixing a typo. ........
-
-2008-01-23 17:48 +0000 [r99922-99924] Russell Bryant <russell@digium.com>
-
- * /, apps/app_chanspy.c: Merged revisions 99923 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r99923 | russell | 2008-01-23 11:46:55 -0600 (Wed, 23 Jan 2008) |
- 8 lines ChanSpy issues a beep when it starts at the beginning of
- a list of channels to potentially spy on. However, if there were
- no matching channels, it would beep at you over and over, which
- is pretty annoying. Now, it will only beep once in the case that
- there are no channels to spy on, but it will still beep again
- once it reaches the beginning of the channel list again. (closes
- issue #11738, patched by me) ........
-
- * main/tcptls.c: Fix tcptls build when openssl isn't installed
- (closes issue #11813) Reported by: tzafrir Patches:
- asterisk-tcptls.diff.txt uploaded by jamesgolovich (license 176)
-
-2008-01-23 17:27 +0000 [r99920] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c: since echo canceler parameters in Zaptel are
- now signed integers, allow them during parsing
-
-2008-01-23 15:23 +0000 [r99860] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_h323.c: Progress messages don't work (closes issue
- #10497) Reported by: pj Patches: h323-announces-r99483.diff
- uploaded by sergee (license 138) Tested by: pj
-
-2008-01-23 10:18 +0000 [r99839] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: - Add a few comments to sip_xmit - Make sure
- that we are aware of a pending INVITE even if we're using TCP
-
-2008-01-23 05:29 +0000 [r99696-99818] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: Coding guidelines fixups
-
- * /, apps/app_voicemail.c: Merged revisions 99777 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r99777 | tilghman | 2008-01-22 22:31:51 -0600 (Tue, 22 Jan 2008)
- | 8 lines When we reset the password via an external command, we
- should also reset the password stored in the in-memory list, too
- (otherwise it doesn't really take effect). (closes issue #11809)
- Reported by: davetroy Patches: fix_externpass.diff uploaded by
- davetroy (license 384) ........
-
- * /, res/res_odbc.c: Merged revisions 99775 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r99775 | tilghman | 2008-01-22 22:20:15 -0600 (Tue, 22 Jan 2008)
- | 2 lines Oops, should have checked for a NULL obj, here, too
- ........
-
- * res/res_config_ldap.c: Coding guidelines cleanup
-
- * /, main/acl.c: Merged revisions 99718 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r99718 | tilghman | 2008-01-22 18:56:06 -0600 (Tue, 22 Jan 2008)
- | 2 lines Just confirmed that all current platforms need this
- header file ........
-
- * /: Oops
-
- * /, build_tools/menuselect-deps.in, configure,
- include/asterisk/autoconfig.h.in, doc/ldap.txt (added),
- configure.ac, configs/res_ldap.conf.sample (added),
- res/res_config_ldap.c (added), CHANGES, makeopts.in,
- contrib/scripts/asterisk.ldap-schema (added),
- contrib/scripts/asterisk.ldif (added): Add res_config_ldap for
- realtime LDAP engine. (closes issue #5768) Reported by: mguesdon
- Patches: res_config_ldap-v0.7.tar.gz uploaded by mguesdon
- (license 121) res_ldap.conf.sample uploaded by suretec (license
- 70) asterisk-v3.1.4.ldif uploaded by suretec (license 70)
- asterisk-v3.1.4.schema uploaded by suretec (license 70) Tested
- by: oej, mguesdon, suretec, cthorner
-
-2008-01-22 21:09 +0000 [r99647-99653] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 99652 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r99652 | oej | 2008-01-22 21:56:09 +0100 (Tis, 22 Jan 2008) | 4
- lines Thanks to Russell's education I realize that BUFSIZ has
- changed since I learned the C language over 20 years ago...
- Resetting chan_sip to the size of BUFSIZ that I expected in my
- old head to avoid too heavy memory allocations on some systems.
- ........
-
- * doc/tex/channelvariables.tex, CHANGES: Documentation updates for
- BRIDGEPVTCALLID
-
-2008-01-22 20:42 +0000 [r99646] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/acl.c: Merged revisions 99643 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r99643 | tilghman | 2008-01-22 14:34:55 -0600 (Tue, 22 Jan 2008)
- | 2 lines Fix the defines for OS X (and Solaris, too) ........
-
-2008-01-22 20:41 +0000 [r99645] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c: Make sure the command is not just present but is
- also configured to be executed
-
-2008-01-22 20:35 +0000 [r99644] Olle Johansson <oej@edvina.net>
-
- * main/channel.c, channels/chan_sip.c, include/asterisk/channel.h:
- Add a generic function to set the bridged call PVT unique id
- string as a channel variable BRIDGEPVTCALLID This is important
- for call tracing in log files and CDRs, so that the SIP callID
- can be traced along servers. The CHANNEL dialplan function won't
- work here, since the outbound channel is gone when we need the
- Call-ID. Other channel drivers may now implement the same
- function :-), but this patch only supports chan_sip.so. Inspired
- by (issue #11816) Reported by: ctooley Patch by oej
-
-2008-01-22 20:33 +0000 [r99642] Russell Bryant <russell@digium.com>
-
- * configs/cli.conf.sample (added), CHANGES, main/asterisk.c: Change
- the Asterisk CLI startup commands feature to read commands to run
- from cli.conf after a discussion on the -dev list.
-
-2008-01-22 17:46 +0000 [r99595-99596] Olle Johansson <oej@edvina.net>
-
- * channels/chan_local.c, /, res/res_features.c,
- channels/chan_agent.c, apps/app_followme.c: Merged revisions
- 99594 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r99594 | oej | 2008-01-22 18:41:57 +0100 (Tis, 22 Jan 2008) | 3
- lines Add more dependencies on chan_local and add a note to the
- description of chan_local so that people don't disable it in
- menuselect just to clean up. ........
-
- * apps/app_dial.c, /: Merged revisions 99592 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r99592 | oej | 2008-01-22 18:31:17 +0100 (Tis, 22 Jan 2008) | 5
- lines Add dependency on chan_local to app_dial. Dial still runs
- without chan_local, but will be missing forwarding functionality.
- ........
-
-2008-01-22 17:15 +0000 [r99559] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/acl.c: Merged revisions 99540 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r99540 | tilghman | 2008-01-22 10:54:06 -0600 (Tue, 22 Jan 2008)
- | 7 lines Ensure that we can get an address even when we don't
- have a default route. (closes issue #9225) Reported by: junky
- Patches: 20080122__bug9225.diff.txt uploaded by Corydon76
- (license 14) Tested by: oej, loloski, sergee ........
-
-2008-01-22 16:55 +0000 [r99542] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Point out a bug in some debug counter
- handling
-
-2008-01-22 15:25 +0000 [r99464-99521] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Add authentication options to the SIP
- dialstring. Documentation follows separately (issue #11587)
- Reported by: sobomax Patches: chan_sip.c-trunk.diff uploaded by
- sobomax (license 359)
-
- * configs/sip.conf.sample: Documentation updates
-
- * doc/siptls.txt: Small fixes
-
- * main/tcptls.c, channels/chan_zap.c, main/abstract_jb.c,
- include/asterisk/tcptls.h: Doxygen updates
-
-2008-01-21 23:56 +0000 [r99427] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_local.c, /: Merged revisions 99426 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r99426 | mmichelson | 2008-01-21 17:55:26 -0600 (Mon, 21
- Jan 2008) | 12 lines Fixing an issue wherein monitoring local
- channels was not possible. During a channel masquerade, the
- monitors on the two channels involved are swapped. In 99% of the
- cases this results in the desired effect. However, if monitoring
- a local channel, this caused the monitor which was on the local
- channel to get moved onto a channel which is immediately hung up
- after the masquerade has completed. By swapping the monitors
- prior to the masquerade, we avoid the problem by tricking the
- masquerade into placing the monitor back onto the channel where
- we want it. During the investigation of the issue, the channel's
- monitor was the only thing that was swapped in such a manner
- which did not make sense to have done. All other variable
- swapping made sense. ........
-
-2008-01-21 23:25 +0000 [r99424] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c: Fix distinctive ring detection. Reported by:
- milazzo Patches: drings.diff uploaded by milazzo (license 383)
- Closes issue #11799
-
-2008-01-21 22:32 +0000 [r99406] Mark Michelson <mmichelson@digium.com>
-
- * configs/queues.conf.sample, apps/app_queue.c: Adding the
- QUEUENAME variable to the variables set using the setqueuevar
- option in queues.conf. Suggestion comes from Shaun2222 on IRC.
-
-2008-01-21 21:11 +0000 [r99382-99384] Olle Johansson <oej@edvina.net>
-
- * channels/chan_console.c: Remove compiler warning for
- uninitialized variable
-
- * channels/chan_sip.c: Doxygen updates. The TCP/TLS code was
- committed without any doxygen obviously. Tss tss.
-
- * channels/chan_sip.c: Updating doxygen
-
-2008-01-21 18:15 +0000 [r99350] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/res_odbc.h, /, res/res_odbc.c,
- configs/res_odbc.conf.sample: Merged revisions 99341 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r99341 | tilghman | 2008-01-21 12:11:07 -0600 (Mon, 21
- Jan 2008) | 8 lines Permit the user to specify number of seconds
- that a connection may remain idle, which fixes a crash on
- reconnect with the MyODBC driver. (closes issue #11798) Reported
- by: Corydon76 Patches: 20080119__res_odbc__idlecheck.diff.txt
- uploaded by Corydon76 (license 14) Tested by: mvanbaak ........
-
-2008-01-21 16:02 +0000 [r99302] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 99301 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r99301 | file | 2008-01-21 12:01:00 -0400 (Mon, 21 Jan 2008) | 4
- lines Bump the buffer size for Via headers up to 512. There are
- some exceptionally large Via headers out there. (closes issue
- #11783) Reported by: ofirroval ........
-
-2008-01-21 07:02 +0000 [r99280] Olle Johansson <oej@edvina.net>
-
- * CREDITS: Update
-
-2008-01-21 03:54 +0000 [r99265] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Change over to using ast_debug so these
- debug messages don't always show up.
-
-2008-01-20 07:28 +0000 [r99166-99248] Russell Bryant <russell@digium.com>
-
- * channels/chan_console.c: Add a "console active" CLI command,
- which lets you find out which console device is currently active
- for the Asterisk CLI, or to set it. Also, knock multiple device
- support off of the to-do list.
-
- * configs/console.conf.sample: correct the name of a CLI command
- for getting available device names
-
- * configs/console.conf.sample, channels/chan_console.c: Merge
- changes from team/russell/console_devices - Add support for
- multiple devices. All devices are configured in console.conf. -
- Add "console list devices" CLI command to show configured
- devices. Also, changed the old "list devices" to be "list
- available", which queries PortAudio for all audio devices that
- are available for use.
-
- * /, main/slinfactory.c: Merged revisions 99187 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r99187 | russell | 2008-01-19 04:05:27 -0600 (Sat, 19 Jan 2008) |
- 4 lines Fix a couple of memory leaks with frame handling.
- Specifically, ast_frame_free() needed to be called on the frame
- that came from the translator to signed linear. ........
-
- * README: Add Cygwin as an "other" platform that is supported
-
- * README: Various README updates
-
-2008-01-18 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.6.0-beta1 released.
-
-2008-01-18 22:04 +0000 [r99080-99085] Russell Bryant <russell@digium.com>
-
- * CREDITS, include/asterisk/http.h, main/tcptls.c (added),
- main/manager.c, channels/chan_sip.c, doc/siptls.txt (added),
- main/Makefile, main/http.c, include/asterisk/tcptls.h (added),
- configs/sip.conf.sample, CHANGES: Merge changes from
- team/group/sip-tcptls This set of changes introduces TCP and TLS
- support for chan_sip. There are various new options in
- configs/sip.conf.sample that are used to enable these features.
- Also, there is a document, doc/siptls.txt that describes some
- things in more detail. This code was implemented by Brett Bryant
- and James Golovich. It was reviewed by Joshua Colp and myself. A
- number of other people participated in the testing of this code,
- but since it was done outside of the bug tracker, I do not have
- their names. If you were one of them, thanks a lot for the help!
- (closes issue #4903, but with completely different code that what
- exists there.)
-
- * main/frame.c, /, include/asterisk/translate.h: Merged revisions
- 99081 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r99081 | russell | 2008-01-18 15:37:21 -0600 (Fri, 18 Jan 2008) |
- 9 lines Revert adding the packed attribute, as it really doesn't
- make sense why that would do any good. Fix the real bug, which is
- to do the check to see if the frame came from a translator at the
- beginning of ast_frame_free(), instead of at the end. This
- ensures that it always gets checked, even if none of the parts of
- the frame are malloc'd, and also ensures that we aren't looking
- at free'd memory in the case that it is a malloc'd frame. (closes
- issue #11792, reported by explidous, patched by me) ........
-
- * /, include/asterisk/translate.h: Merged revisions 99079 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r99079 | russell | 2008-01-18 15:22:21 -0600 (Fri, 18 Jan 2008) |
- 4 lines Since we're relying on the offset between the frame and
- the beginning of the translator pvt struct, set the packed
- attribute to make sure we get to the right place. (potential fix
- for issue #11792) ........
-
-2008-01-18 16:58 +0000 [r99026] Terry Wilson <twilson@digium.com>
-
- * res/res_features.c: This should at least temporarily fix a
- problem where the 't' Dial option is incorrectly passed to the
- transferee when built-in attended transfers are used. There is
- still a problem with 'T', but better to fix some problems than no
- problems while we work on it. (closes issue #7904) Reported by:
- k-egg Patches: transfer-fix-trunk-r97657.diff uploaded by sergee
- (license 138) Tested by: sergee, otherwiseguy
-
-2008-01-18 06:58 +0000 [r99015-99018] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_odbc.c: Convert func_odbc to use SQLExecDirect for
- speed (closes issue #10723) Reported by: mnicholson Patches:
- func-odbc-direct-execute1.diff uploaded by mnicholson (license
- 96) Tested by: Corydon76, mnicholson, falves11
-
- * res/res_odbc.c: Permit username and password to be NULL (which
- enables pass-through from the layer above). Reported by: lurcher
- Patch by: tilghman (Closes issue #11739)
-
- * funcs/func_cut.c: Reset default CUT delimiter back to '-'
-
-2008-01-17 23:28 +0000 [r99006-99011] Russell Bryant <russell@digium.com>
-
- * channels/chan_console.c: Make the output of "console list
- devices" a bit prettier.
-
- * channels/chan_console.c: List which devices are inputs and
- outputs in "console list devices"
-
- * main/channel.c: Add AST_FORMAT_SLINEAR16 to the list for
- ast_best_codec()
-
- * main/frame.c, /, channels/chan_iax2.c, include/asterisk/frame.h:
- Merged revisions 99004 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r99004 | russell | 2008-01-17 16:37:22 -0600 (Thu, 17 Jan 2008) |
- 10 lines Have IAX2 optimize the codec translation path just like
- chan_sip does it. If the caller's codec is in our codec list,
- move it to the top to avoid transcoding. (closes issue #10500)
- Reported by: stevedavies Patches: iax-prefer-current-codec.patch
- uploaded by stevedavies (license 184)
- iax-prefer-current-codec.1.4.patch uploaded by stevedavies
- (license 184) Tested by: stevedavies, pj, sheldonh ........
-
-2008-01-17 22:22 +0000 [r99002] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Fixing trunk IMAP build (closes issue
- #11788) Reported by: DEA Patches: vm-imap-build-fix.txt uploaded
- by DEA (license 3)
-
-2008-01-17 20:51 +0000 [r98998] Jason Parker <jparker@digium.com>
-
- * Makefile, build_tools/cflags.xml, channels/chan_zap.c,
- main/dsp.c, configs/zapata.conf.sample: Add several busy
- detection related defines to menuselect. Allow better busy detect
- debugging (with BUSYDETECT_DEBUG). Remove very old BUSYDETECT and
- BUSY_DETECT_MARTIN defines. (closes issue #11107) Patches:
- busydetect_enhancement.patch uploaded by agx (license 298)
- busydetect-r94975.diff uploaded by sergee (license 138)
- Additional changes/cleanup by me.
-
-2008-01-17 16:33 +0000 [r98993-98994] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: state_interface could be NULL, so use the
- never-NULL cur->state_interface for this check
-
- * apps/app_queue.c: Get the device state of the state interface
- instead of the interface when creating a new queue member. Thanks
- to Atis Lezdins for bringing this up on the Asterisk-Dev mailing
- list.
-
-2008-01-17 16:21 +0000 [r98992] Jason Parker <jparker@digium.com>
-
- * /, configs/zapata.conf.sample: Merged revisions 98991 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes
- issue #11784) ........ r98991 | qwell | 2008-01-17 10:19:46 -0600
- (Thu, 17 Jan 2008) | 4 lines Add a clarification about the
- immediate= option of zapata.conf Issue 11784, patch by klaus3000.
- ........
-
-2008-01-17 16:17 +0000 [r98989-98990] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c, configs/zapata.conf.sample: major
- reliability and performance improvement in VWMI monitoring for
- FXO ports (code by markster, me and dbailey)
-
- * res/res_config_curl.c: resolve (valid) compiler warning about
- variable that could be used before being initialized
-
-2008-01-17 03:09 +0000 [r98988] Terry Wilson <twilson@digium.com>
-
- * res/res_phoneprov.c, doc/tex/phoneprov.tex,
- configs/phoneprov.conf.sample: Update res_phoneprov to default to
- setting the SERVER variable to the IP the HTTP request for the
- config came in on and the SERVER_PORT to the bindport setting in
- sip.conf. I've left in the ability to override these options,
- because I can't always guess how someone might decide to do
- something weird with what is available to them--although needing
- to is pretty unlikely. Documentation was updated to reflect
- preference for not setting serveraddr, serveriface, or
- serverport. Tested on Linux and OS X.
-
-2008-01-17 00:13 +0000 [r98987] Tilghman Lesher <tlesher@digium.com>
-
- * cdr/cdr_adaptive_odbc.c: Change the way the new filter feature
- works, by allowing it to be a column NOT logged into the
- database. This will allow more granularity of a decision
- evaluated in the dialplan, then takes effect when posting the
- CDR.
-
-2008-01-17 00:05 +0000 [r98986] Russell Bryant <russell@digium.com>
-
- * CHANGES, main/asterisk.c: Add support for an easy way to
- automatically execute some Asterisk CLI commands immediately at
- startup. Any commands in the startup_commands file in the
- Asterisk config diretory will get executed. (closes issue #11781)
- Reported by: jamesgolovich Patches: asterisk-startupcmds.diff.txt
- uploaded by jamesgolovich (license 176) -- With some changes by
- me.
-
-2008-01-16 23:08 +0000 [r98985] Jason Parker <jparker@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac,
- acinclude.m4: Change AST_EXT_TOOL_CHECK to attempt to build
- against <package>_LIB, per recommendations from Russell.
-
-2008-01-16 22:36 +0000 [r98984] Tilghman Lesher <tlesher@digium.com>
-
- * CHANGES: Info about res_config_curl
-
-2008-01-16 22:20 +0000 [r98981] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_config_curl.c (added), main/utils.c: New module
- res_config_curl (closes issue #11747) Reported by: Corydon76
- Patches: res_config_curl.c uploaded by Corydon76 (license 14)
- 20080116__bug11747.diff.txt uploaded by Corydon76 (license 14)
- Tested by: jmls
-
-2008-01-16 21:53 +0000 [r98978] Russell Bryant <russell@digium.com>
-
- * CREDITS, channels/chan_sip.c, configs/sip.conf.sample: Merge the
- changes from issue #10665 from the team/group/sip_session_timers
- branch. This set of changes introduces SIP session timers support
- (RFC 4028). In short, this prevents stuck SIP sessions that were
- not properly torn down due to network or endpoint failures during
- an established SIP session. To quote some of the documentation
- supplied with the patch: "The SIP Session-Timers is an extension
- of the SIP protocol that allows end-points and proxies to refresh
- a session periodically. The sessions are kept alive by sending a
- RE-INVITE or UPDATE request at a negotiated interval. If a
- session refresh fails then all the entities that support Session-
- Timers clear their internal session state. In addition, UAs
- generate a BYE request in order to clear the state in the proxies
- and the remote UA (this is done for the benefit of SIP entities
- in the path that do not support Session-Timers)." (closes issue
- #10665) Reported by: rjain Patches: chan_sip.c.1.diff uploaded by
- rjain (license 226) chan_sip.c.diff uploaded by rjain (license
- 226) sip.conf.sample.diff uploaded by rjain (license 226)
- proc_422_rsp_comment.diff uploaded by rjain (license 226)
- chan_sip.c.cache.diff uploaded by rjain (license 226)
- chan_sip.memalloc uploaded by rjain (license 226)
- chan_sip.memalloc.bugfix uploaded by rjain (license 226) Patches
- tracked in team/group/sip_session_timers, with some additional
- fixes by russell and oej. Tested by: jtodd, rjain, loloski
-
-2008-01-16 19:41 +0000 [r98968-98971] Jason Parker <jparker@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac:
- Partially revert r93898, because it broke the way netsnmp was
- being detected. rizzo, do you want to discuss so we can rethink
- this, or do you have another way?
-
- * CHANGES: Add note about new update.log to CHANGES, by request of
- jmls and further prodding by jsmith.
-
- * Makefile, /: Add logging for 'make update' command (also fixes
- updates in some places). Issue #11766, initial patch by jmls.
-
-2008-01-16 17:51 +0000 [r98967] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 98966 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r98966 | file | 2008-01-16 13:50:10 -0400 (Wed, 16 Jan 2008) | 6
- lines Add missing NULLs at end of two ast_load_realtimes. (closes
- issue #11769) Reported by: tequ Patches: chaniax.patch uploaded
- by dimas (license 88) ........
-
-2008-01-16 17:21 +0000 [r98965] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_local.c, /: Merged revisions 98964 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r98964 | mmichelson | 2008-01-16 11:20:11 -0600 (Wed, 16
- Jan 2008) | 10 lines Fix a deadlock in chan_local in
- local_hangup. There was contention because the local_pvt was held
- and it was attempting to lock a channel, which is the incorrect
- locking order. (closes issue #11730) Reported by: UDI-Doug
- Patches: 11730.patch uploaded by putnopvut (license 60) Tested
- by: UDI-Doug ........
-
-2008-01-16 16:06 +0000 [r98962] Terry Wilson <twilson@digium.com>
-
- * res/res_phoneprov.c: Make users list static
-
-2008-01-16 15:09 +0000 [r98954-98961] Joshua Colp <jcolp@digium.com>
-
- * main/dial.c, /: Merged revisions 98960 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r98960 | file | 2008-01-16 11:08:24 -0400 (Wed, 16 Jan 2008) | 6
- lines Introduce a lock into the dialing API that protects it when
- destroying the structure. (closes issue #11687) Reported by:
- callguy Patches: 11687.diff uploaded by file (license 11)
- ........
-
- * /, main/rtp.c: Merged revisions 98958 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r98958 | file | 2008-01-16 11:03:14 -0400 (Wed, 16 Jan 2008) | 4
- lines Add two more SDP names for ulaw and alaw. (closes issue
- #11777) Reported by: tootai ........
-
- * /, channels/chan_sip.c: Merged revisions 98955 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r98955 | file | 2008-01-15 23:07:24 -0400 (Tue, 15 Jan 2008) | 6
- lines Don't drop the old record route information when dealing
- with packets related to a reinvite. (closes issue #11545)
- Reported by: kebl0155 Patches: reinvite-patch.txt uploaded by
- kebl0155 (license 356) ........
-
- * channels/chan_sip.c: Remove DNS lookup from sip_devicestate. This
- seems to come from way back when and I can't think of a reason
- for it being here, plus it could cause needless DNS lookups.
- (closes issue #10983) Reported by: jtodd
-
-2008-01-16 01:35 +0000 [r98953] Steve Murphy <murf@digium.com>
-
- * main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2.c: Terry found
- this problem with running the expr2 parser on OSX. Make the
- #defines come out the same between the parser & lexer.
-
-2008-01-16 01:17 +0000 [r98952] Joshua Colp <jcolp@digium.com>
-
- * /, build_tools/menuselect-deps.in, configure,
- include/asterisk/autoconfig.h.in, codecs/codec_speex.c,
- configure.ac, makeopts.in: Merged revisions 98951 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r98951 | file | 2008-01-15 21:13:27 -0400 (Tue, 15 Jan
- 2008) | 4 lines Add autoconf logic for speexdsp. Later versions
- use a separate library for some things so we need to use it if
- present in codec_speex. (closes issue #11693) Reported by: yzg
- ........
-
-2008-01-15 23:53 +0000 [r98948] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 98946 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r98946 | russell | 2008-01-15 17:50:10 -0600 (Tue, 15 Jan 2008) |
- 11 lines Change a buffer in check_auth() to be a thread local
- dynamically allocated buffer, instead of a massive buffer on the
- stack. This fixes a crash reported by Qwell due to running out of
- stack space when building with LOW_MEMORY defined. On a very
- related note, the usage of BUFSIZ in various places in chan_sip
- is arbitrary and careless. BUFSIZ is a system specific define. On
- my machine, it is 8192, but by definition (according to google)
- could be as small as 256. So, this buffer in check_auth was 16
- kB. We don't even support SIP messages larger than 4 kB! Further
- usage of this define should be avoided, unless it is used in the
- proper context. ........
-
-2008-01-15 23:52 +0000 [r98947] Tilghman Lesher <tlesher@digium.com>
-
- * cdr/cdr_adaptive_odbc.c, configs/cdr_adaptive_odbc.conf.sample:
- Add the "filter" keyword
-
-2008-01-15 23:35 +0000 [r98944-98945] Russell Bryant <russell@digium.com>
-
- * main/translate.c, include/asterisk/translate.h: Clean up
- something I did for ABI compatability in 1.4
-
- * main/frame.c, /, main/translate.c, main/abstract_jb.c,
- channels/chan_iax2.c, codecs/codec_zap.c,
- include/asterisk/frame.h, main/rtp.c,
- include/asterisk/translate.h: Merged revisions 98943 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r98943 | russell | 2008-01-15 17:26:52 -0600 (Tue, 15
- Jan 2008) | 25 lines Commit a fix for some memory access errors
- pointed out by the valgrind2.txt output on issue #11698. The
- issue here is that it is possible for an instance of a translator
- to get destroyed while the frame allocated as a part of the
- translator is still being processed. Specifically, this is
- possible anywhere between a call to ast_read() and
- ast_frame_free(), which is _a lot_ of places in the code. The
- reason this happens is that the channel might get masqueraded
- during this time. During a masquerade, existing translation paths
- get destroyed. So, this patch fixes the issue in an API and ABI
- compatible way. (This one is for you, paravoid!) It changes an
- int in ast_frame to be used as flag bits. The 1 bit is still used
- to indicate that the frame contains timing information. Also, a
- second flag has been added to indicate that the frame came from a
- translator. When a frame with this flag gets released and has
- this flag, a function is called in translate.c to let it know
- that this frame is doing being processed. At this point, the flag
- gets cleared. Also, if the translator was requested to be
- destroyed while its internal frame still had this flag set, its
- destruction has been deffered until it finds out that the frame
- is no longer being processed. Admittedly, this feels like a hack.
- But, it does fix the issue, and I was not able to think of a
- better solution ... ........
-
-2008-01-15 20:10 +0000 [r98895-98935] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 98934 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r98934 | file | 2008-01-15 16:08:43 -0400 (Tue, 15 Jan 2008) | 4
- lines Based on the boundary found move over the correct amount.
- (closes issue #11750) Reported by: tasker ........
-
- * /, channels/chan_sip.c: Merged revisions 98894 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r98894 | file | 2008-01-14 18:41:55 -0400 (Mon, 14 Jan 2008) | 4
- lines Accept "; boundary=" not just ";boundary=" in the multipart
- mixed content type. (closes issue #11750) Reported by: tasker
- ........
-
-2008-01-14 22:19 +0000 [r98889] Jason Parker <jparker@digium.com>
-
- * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c: Add
- backupdeleted option to app_voicemail (closes issue #10740)
- Reported by: ruffle Patches: app_voicemail.diff uploaded by
- ruffle (license 201) 10740-voicemail.diff uploaded by qwell
- (license 4) 20080113_bug10740.diff.txt uploaded by mvanbaak
- (license 7) Tested by: blitzrage, mvanbaak, qwell
-
-2008-01-14 22:11 +0000 [r98850-98888] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_directory.c: Big improvement for app_directory. This
- patch breaks the do_directory function up so that it is more
- easily parsed by the human brain. It also fixes some errors. I'll
- quote dimas from the original bug description: "app_directory
- contained some duplicate code even before addition of 'm' option.
- Addition of that option doubled amount of that code. Worst of
- all, there are minor differences between these code block and
- bugs caused by these differences. 1. There is a memory leak. In
- the 'menu' mode, result of the convert(pos) function is not freed
- while it should be. 2. In the 'menu' mode check for
- OPT_LISTBYFIRSTNAME flag ('f' option) is not negated as result,
- application works in the mode opposite to what user expect
- (checking last name when user wants the first nd vice versa). 3.
- select_item function plays message for user using res = func1()
- || func2() || func3()... construct. This construct loses the
- actual value returned by ast_waitstream() for example so at the
- end, res does not contain digit user dialed while listening to
- the message. 4. (also in 1.4) application announces entries from
- voicemail.conf/realtime separately from entries from users.conf.
- I see no reason why doing so instead of building combined list.
- 5. Alot of duplicated code as already mentioned." This was tested
- by dimas and I (I tested under valgrind). A word of caution: any
- bug fixes that happen in app_directory in 1.4 will almost
- certainly not merge cleanly into trunk as a result of this, but
- it is well worth it. Huge thanks to dimas for this wonderful
- submission. (closes issue #11744) Reported by: dimas Patches:
- dir3.patch uploaded by dimas (license 88) Tested by: putnopvut,
- dimas
-
-2008-01-14 20:01 +0000 [r98830] Joshua Colp <jcolp@digium.com>
-
- * main/manager.c: Make sure the user's manager secret exists, even
- if it is blank. (closes issue #11749) Reported by: srt
-
-2008-01-14 18:42 +0000 [r98811] Terry Wilson <twilson@digium.com>
-
- * CHANGES: Add description of TOUPPER and TOLOWER dialplan
- functions to CHANGES.
-
-2008-01-14 17:40 +0000 [r98776] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: 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.
-
-2008-01-14 17:39 +0000 [r98775] Russell Bryant <russell@digium.com>
-
- * /, main/translate.c: Merged revisions 98774 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r98774 | russell | 2008-01-14 11:38:38 -0600 (Mon, 14 Jan 2008) |
- 3 lines Revert a change that introduces an unacceptable
- performance hit and is causing memory leaks ... (from rev 97973)
- ........
-
-2008-01-14 17:18 +0000 [r98773] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Fix for potential crash with vmexten
-
-2008-01-14 16:36 +0000 [r98735-98738] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Merged revisions 98737 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r98737 | mmichelson | 2008-01-14 10:35:12 -0600 (Mon, 14 Jan
- 2008) | 3 lines Fixing another compilation error. I'm a bit off
- today :( ........
-
- * /, apps/app_queue.c: Merged revisions 98733 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r98733 | mmichelson | 2008-01-14 10:21:28 -0600 (Mon, 14 Jan
- 2008) | 8 lines Adding explicit defaults for missing options to
- init_queue. This is necessary because if a user either removes or
- comments one of these options and reloads their queues, the
- option will not reset to its default, instead maintaining the
- value from prior to the reload. Thanks to John Bigelow for
- pointing this error out to me. ........
-
-2008-01-14 15:07 +0000 [r98695-98714] Joshua Colp <jcolp@digium.com>
-
- * main/pbx.c: Print out a warning when spaces are used in the
- variable name in Set and MSet. It is extremely hard to debug this
- issue so this should make it easier. (closes issue #11759)
- Reported by: caio1982 Patches: setvar_space_warning1.diff
- uploaded by caio1982 (license 22)
-
- * apps/app_meetme.c, doc/tex/qos.tex, doc/tex/realtime.tex: Update
- documentation. (closes issue #11763) Reported by: IgorG Patches:
- docupd.v1.diff uploaded by IgorG (license 20)
-
-2008-01-14 04:53 +0000 [r98558-98676] Russell Bryant <russell@digium.com>
-
- * apps/app_jack.c: Add another small option for the JACK app and
- JACK_HOOK function. The 'n' option tells JACK not to start jackd
- automatically if it is not already running. Otherwise, the
- default is that jackd will get started for you if it isn't
- running already.
-
- * CHANGES: - Break up the Misc. section a bit with a new section
- for Misc. New Modules - Change spacing a bit in some places for
- consistent indentation
-
- * CHANGES, apps/app_jack.c (added): Bring in the code from
- team/russell/jack/. Add a new module, app_jack, which provides
- interfaces to JACK, the Jack Audio Connection Kit
- (http://www.jackaudio.org/). Two interfaces are provided; there
- is a JACK() application, and a JACK_HOOK() function. Both
- interfaces create an input and output JACK port. The application
- makes these ports the endpoint of the call. The audio coming from
- the channel goes out the output port and whatever comes back in
- on the input port is what gets sent to the channel. The
- JACK_HOOK() function turns on a JACK audiohook on the channel.
- This lets you run the audio coming from a channel through JACK,
- and whatever comes back in is what gets forwarded on as the
- channel's audio. This is very useful for building custom vocoders
- or doing recording or analysis of the channel's audio in another
- application. In case anyone is curious, the platform that
- inspired me to write this is PureData (http://puredata.info/). I
- wrote these JACK interfaces so that I could use Pd to do
- interesting things with the audio of phone calls ...
-
- * build_tools/menuselect-deps.in, configure,
- include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add
- configure script check for JACK.
-
- * build_tools/menuselect-deps.in, configure,
- include/asterisk/autoconfig.h.in, configure.ac, makeopts.in:
- Remove KDE configure script check that isn't used
-
- * main/audiohook.c: Remove a duplicate lock of the audiohook lock
- when destroying manipulate audiohooks. This causes an error when
- we attempt to destroy the lock later when freeing the audiohook.
-
- * main/pbx.c, CHANGES: Add a new CLI command, "core set chanvar",
- which allows you to set a channel variable (or function) on an
- active channel from the CLI.
-
-2008-01-12 18:12 +0000 [r98536] Tilghman Lesher <tlesher@digium.com>
-
- * main/manager.c: Conversion to load manager.conf into memory did
- not convert the password functions correctly. (Closes issue
- #11749)
-
-2008-01-12 05:13 +0000 [r98514] Pari Nannapaneni <paripurnachand@digium.com>
-
- * /, main/http.c: merging a comment added in 1.4
-
-2008-01-12 00:20 +0000 [r98488] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c, CHANGES: Add 'zap set dnd' CLI command, and
- ensure that the AMI DNDState event always gets generated. (closes
- issue #11212) Reported by: tzafrir Patches: zap_dnd.diff uploaded
- by tzafrir (modified by me) (license 46)
-
-2008-01-12 00:17 +0000 [r98487] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_odbc.c: Merged revisions 98467 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r98467 | tilghman | 2008-01-11 18:05:08 -0600 (Fri, 11 Jan 2008)
- | 4 lines Add a connection timeout attribute, as that was what
- was intended with the login timeout, but ODBC divides it up into
- 2 different timeouts. (Closes issue #11745) ........
-
-2008-01-11 23:57 +0000 [r98454] Russell Bryant <russell@digium.com>
-
- * configure, doc/tex/Makefile, configure.ac, makeopts.in: Add some
- extra checking to help out with a potential error when trying to
- run "make asterisk.pdf" when not all of the right packages are
- installed. (closes issue #10763) Reported by: Corydon76 Patches:
- 20070919__bug10763.diff.txt uploaded by Corydon76 (license 14)
- Tested by: Corydon76
-
-2008-01-11 23:10 +0000 [r98436] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c, CHANGES, configs/zapata.conf.sample: Add
- 'auto' signalling mode for Zaptel channels. (closes issue #11690)
- Reported by: tzafrir Patches: signaling_to_signalling.diff
- uploaded by tzafrir (license 46) signalling_cleanup.diff uploaded
- by tzafrir (license 46) zap_auto_default.diff uploaded by tzafrir
- (license 46) zap_no_default_sig.diff uploaded by tzafrir (license
- 46) zap_signal_auto.diff uploaded by tzafrir (license 46)
-
-2008-01-11 23:09 +0000 [r98424-98435] Joshua Colp <jcolp@digium.com>
-
- * main/event.c: Goodbye again drumkilla.
-
- * main/event.c: drumkilla ftw.
-
- * main/audiohook.c: I am no longer Rockin'
-
- * main/audiohook.c: Testing something...
-
-2008-01-11 22:52 +0000 [r98400] Russell Bryant <russell@digium.com>
-
- * /, pbx/pbx_dundi.c: Merged revisions 98390 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r98390 | russell | 2008-01-11 16:46:21 -0600 (Fri, 11 Jan 2008) |
- 9 lines Fix up setting the EID on BSD based systems. (closes
- issue #11646) Reported by: caio1982 Patches:
- dundi_osx_eid6.diff.txt uploaded by caio1982 (license 22)
- dundi_osx_eid6-1.4.diff uploaded by caio1982 (license 22) Tested
- by: caio1982, mvanbaak ........
-
-2008-01-11 19:53 +0000 [r98318-98334] Joshua Colp <jcolp@digium.com>
-
- * /, main/rtp.c: Merged revisions 98325 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r98325 | file | 2008-01-11 15:51:10 -0400 (Fri, 11 Jan 2008) | 6
- lines If the incoming RTP stream changes codec force the bridge
- to break if the other side does not support it. (closes issue
- #11729) Reported by: tsearle Patches: new_codec_patch_udiff.patch
- uploaded by tsearle (license 373) ........
-
- * /, res/res_agi.c: Merged revisions 98317 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r98317 | file | 2008-01-11 15:28:30 -0400 (Fri, 11 Jan 2008) | 6
- lines If the channel is hungup during RECORD FILE send a result
- code of -1 to be uniform with everything else. (closes issue
- #11743) Reported by: davevg Patches: res_agi.diff uploaded by
- davevg (license 209) ........
-
-2008-01-11 19:12 +0000 [r98316] Mark Michelson <mmichelson@digium.com>
-
- * main/channel.c, /: Merged revisions 98315 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r98315 | mmichelson | 2008-01-11 13:10:57 -0600 (Fri, 11 Jan
- 2008) | 5 lines Properly report the hangup cause as no answer
- when someone does not answer (closes issue #10574, reported by
- boch, patched by moy) ........
-
-2008-01-11 19:05 +0000 [r98270-98308] Russell Bryant <russell@digium.com>
-
- * codecs/codec_resample.c: Kevin noted that the thing that I
- _actually_ changed here was that I converted a value from a
- double, to a float, back to a double. Sure enough, when I changed
- my interim variable back to a double, it still blows up.
- Switching all of these to a float fixes the problem. This seems
- like a compiler bug where a double passed as an argument isn't
- getting properly aligned, so I'll have to see if I can replicate
- it with a small test program. (related to issue #11725)
-
- * codecs/codec_resample.c: Fix a bus error that happened when
- asterisk was built with optimizations on with platforms that
- explode on unaligned access. I'm not exactly sure why this fixes
- it, but it fixed it on the machine I was testing on. If it makes
- sense to you, feel free to enlighten me. :) (closes issue #11725,
- patched by me)
-
-2008-01-11 18:35 +0000 [r98268-98269] Tilghman Lesher <tlesher@digium.com>
-
- * cdr/cdr_adaptive_odbc.c: Port Nick Gorham's timestamp patch to
- adaptive_odbc, too
-
- * cdr/cdr_odbc.c: Commit Nick Gorham's suggestion for timestamp fix
-
-2008-01-11 17:27 +0000 [r98220] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_followme.c: Merged revisions 98219 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r98219 | file | 2008-01-11 13:22:53 -0400 (Fri, 11 Jan 2008) | 4
- lines Ensure the return value of ast_bridge_call is passed back
- up as the application return value. This is needed for transfers
- to function so the PBX core knows to continue execution. (closes
- issue #10327) Reported by: kkiely ........
-
-2008-01-11 17:17 +0000 [r98218] Russell Bryant <russell@digium.com>
-
- * codecs/codec_g722.c: At one point during working on this module,
- I had the lin/lin16 versions of the framein callbacks different.
- However, they are now the same again, so remove the duplicate
- code and use the same functions for the lin/lin16 versions.
-
-2008-01-11 16:08 +0000 [r98152-98193] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 98164 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r98164 | tilghman | 2008-01-11 09:52:31 -0600 (Fri, 11 Jan 2008)
- | 2 lines Back out changes from revision 97077, since it wasn't
- perfect ........
-
- * doc/manager_1_1.txt: Documentation updates
-
-2008-01-11 12:51 +0000 [r98124] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_sip.c: Ascom phones send Flash events as SIP INFO
- using '!' as the 'digit'
-
-2008-01-11 03:40 +0000 [r98081-98083] Russell Bryant <russell@digium.com>
-
- * codecs/codec_g722.c, main/frame.c: - Fix the last set of places
- where incorrect assumptions were made about the sample length
- with g722. It is _2_ samples per byte, not 1. This was all over
- the place, and I believed it, and it is what caused me to take so
- long to figure out what was broken. - Update copyright
- information on codec_g722.
-
-2008-01-11 00:54 +0000 [r98047] Mark Michelson <mmichelson@digium.com>
-
- * main/translate.c: Fix "core show translation" to not output
- information for "unknown" codecs. This fix was made in favor of
- the proposed patch since it doesn't involve changing a core codec
- define. (closes issue #11722, reported and initially patched by
- caio1982, final patch by me)
-
-2008-01-11 00:38 +0000 [r98024-98027] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add a new
- global and per-peer option to chan_sip, qualifyfreq, which allows
- you to set the qualify frequency. (closes issue #11597) Reported
- by: wilder Patches: qualifyfreq5.patch uploaded by wilder
- (license 362) -- with some mods by me
-
- * main/translate.c: Simplify this code with a suggestion from Luigi
- on the asterisk-dev list. Instead of using is16kHz(), implement a
- format_rate() function.
-
-2008-01-10 23:40 +0000 [r97978] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c, main/translate.c: Merged revisions 97973
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97973 | tilghman | 2008-01-10 17:08:36 -0600 (Thu, 10 Jan 2008)
- | 6 lines 1) When we get a translated frame out, clone it,
- because if the translator pvt is freed before we use the frame,
- bad things happen. 2) Getting a failure from ast_sched_delete
- means that the schedule ID is currently running. Don't just
- ignore it. (Closes issue #11698) ........
-
-2008-01-10 23:33 +0000 [r97974-97977] Russell Bryant <russell@digium.com>
-
- * /, main/translate.c: Merged revisions 97976 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97976 | russell | 2008-01-10 17:30:40 -0600 (Thu, 10 Jan 2008) |
- 3 lines Fix various timing calculations that made assumptions
- that the audio being processed was at a sample rate of 8 kHz.
- ........
-
- * codecs/codec_g722.c: Fix various issues in codec_g722. - The most
- common fix being made here is to fix all of the places where the
- number of output samples and output bytes gets updated in the
- translator state structure. - Fix a number of other places where
- the number of samples provided as an initialization value to a
- struct was incorrect.
-
- * codecs/codec_resample.c: Fix the buffer_samples value. For signed
- linear, the number of samples needed to fill the buffer is half
- the buffer size.
-
-2008-01-10 21:58 +0000 [r97933] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 97925 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97925 | mmichelson | 2008-01-10 15:57:06 -0600 (Thu, 10 Jan
- 2008) | 6 lines Let us leave a voicemail for ourself if we have
- logged into VoiceMailMain and chosen to leave a message. (closes
- issue #11735, reported and patched by jamessan) ........
-
-2008-01-10 21:46 +0000 [r97850-97890] Steve Murphy <murf@digium.com>
-
- * /, res/ael/ael_lex.c, res/Makefile, res/ael/ael.flex: Merged
- revisions 97889 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97889 | murf | 2008-01-10 14:37:10 -0700 (Thu, 10 Jan 2008) | 1
- line Applied the same fixes for ael.flex as was done in 97849 for
- ast_expr2.fl; overrode the normally generate yyfree func with our
- own version that checks the pointer for non-null before passing
- to free(). Also takes care of a little problem with 2.5.33 and
- the use of the __STDC_VERSION__ macro. ........
-
- * /, main/Makefile, main/ast_expr2f.c, main/ast_expr2.fl: Merged
- revisions 97849 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97849 | murf | 2008-01-10 13:21:27 -0700 (Thu, 10 Jan 2008) | 1
- line This is a fix for 2 things: a problem Terry was having in
- OSX with null pointers, which was my fault, as I probably forgot
- to run the sed script last time I made mods. So, I moved the fix
- into the flex input itself. Then, I found when I used flex
- 2.5.33, that it was using __STDC_VERSION__, and that's not real
- good; so I added back in a DIFFERENT sed script to fix that
- little mess. Tested everything, a couple different ways. Hope I
- did no harm, at the least. ........
-
-2008-01-10 20:13 +0000 [r97848] Jason Parker <jparker@digium.com>
-
- * /, include/asterisk/frame.h: Merged revisions 97847 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r97847 | qwell | 2008-01-10 14:12:37 -0600 (Thu, 10 Jan
- 2008) | 1 line Fix a comment that is no longer true. ........
-
-2008-01-10 20:05 +0000 [r97846] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Use the appropriate line ending for the
- X-Asterisk-VM-Message-Type header. (closes issue #11734, reported
- and patched by jaroth)
-
-2008-01-10 19:07 +0000 [r97825-97826] Terry Wilson <twilson@digium.com>
-
- * main/ast_expr2f.c: heh, remove patch to generated file.
-
- * main/ast_expr2f.c, main/cli.c: Check pointers before freeing (was
- getting WARNINGS under MALLOC_DEBUG)
-
-2008-01-10 17:38 +0000 [r97805] Tilghman Lesher <tlesher@digium.com>
-
- * cdr/cdr_odbc.c: Fix problem with timestr going out of scope
- (Closes issue #11726, closes issue #11731)
-
-2008-01-10 17:30 +0000 [r97745-97804] Russell Bryant <russell@digium.com>
-
- * formats/format_sln16.c: minor formatting changes
-
- * main/translate.c: spaces to tabs
-
- * configure, configure.ac: Use AST_EXT_TOOL_CHECK() for the GTK
- check again. I changed this to an inline implementation to fix a
- small bug, but after a discussion with rizzo, I went to change it
- back. Also, it turns out that the implementation of the macro
- already supported what was needed to fix the problem.
-
- * pbx/pbx_kdeconsole.h (removed), /, configs/modules.conf.sample,
- pbx/kdeconsole_main.cc (removed): Merged revisions 97753 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97753 | russell | 2008-01-10 10:19:47 -0600 (Thu, 10 Jan 2008) |
- 2 lines Remove other remnants of pbx_kdeconsole ........
-
- * /, build_tools/menuselect-deps.in, configure,
- include/asterisk/autoconfig.h.in, configure.ac, makeopts.in,
- pbx/pbx_kdeconsole.cc (removed): Merged revisions 97734 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97734 | russell | 2008-01-10 10:10:09 -0600 (Thu, 10 Jan 2008) |
- 4 lines Remove pbx_kdeconsole from the tree. It hasn't worked in
- ages, and nobody has complained. (closes issue #11706, reported
- by caio1982) ........
-
-2008-01-10 15:12 +0000 [r97698] Joshua Colp <jcolp@digium.com>
-
- * funcs/func_groupcount.c, /: Merged revisions 97697 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r97697 | file | 2008-01-10 11:07:12 -0400 (Thu, 10 Jan
- 2008) | 6 lines Don't try to copy the category from the group if
- no category exists. (closes issue #11724) Reported by: IgorG
- Patches: group_count.v1.patch uploaded by IgorG (license 20)
- ........
-
-2008-01-10 00:54 +0000 [r97657] Russell Bryant <russell@digium.com>
-
- * include/asterisk.h: These prototypes are not supposed to be in
- asterisk.h. They are already in version.h.
-
-2008-01-10 00:50 +0000 [r97656] Steve Murphy <murf@digium.com>
-
- * include/asterisk.h, channels/console_video.c, utils/astman.c,
- channels/console_board.c, channels/vgrabbers.c: The fixes in this
- commit are mainly to allow compiling of trunk with
- --enable-dev-mode, mutex profiling, lock debugging, etc. Mainly,
- the version.c needs to be in the OBJS line; asterisk.h was chosen
- to have the prototypes for ast_get_version, ast_get_version_num;
- and the ASTERISK_FILE_VERSION macro needs to be used after
- including asterisk.h in a few files. I hope I did the right
- thing. If not, let me know.
-
-2008-01-10 00:39 +0000 [r97655] Tilghman Lesher <tlesher@digium.com>
-
- * main/manager.c: oops, missed the case of a 0 permission (which
- should mean everybody is allowed, not nobody)
-
-2008-01-10 00:22 +0000 [r97653] Terry Wilson <twilson@digium.com>
-
- * res/res_phoneprov.c: Attempt at making lookup_iface work under
- FreeBSD. Not yet tested, but it compiles under OS X. And still
- works under linux.
-
-2008-01-10 00:17 +0000 [r97652] Russell Bryant <russell@digium.com>
-
- * codecs/Makefile: Fix this so it doesn't force codec_g722 to get
- relinked every time
-
-2008-01-10 00:12 +0000 [r97651] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c, main/manager.c, channels/chan_sip.c,
- res/res_features.c, pbx/pbx_realtime.c,
- configs/manager.conf.sample, CHANGES, channels/chan_iax2.c,
- include/asterisk/manager.h, apps/app_stack.c, main/db.c,
- apps/app_voicemail.c: Several manager changes: 1) Add the
- Dialplan class, for NewExten and VarSet events, which should cut
- down on the volume of traffic in the Call class. 2) Permit some
- commands to be run from multiple classes, such as allowing DBGet
- to be run from either the System or the Reporting class. 3)
- Heavily document each class in the sample config, as there were
- several that made no sense to be in the write= line, and two that
- made no sense to be in the read= line (since they controlled no
- permissions there). (Closes issue #10386)
-
-2008-01-10 00:11 +0000 [r97641-97650] Russell Bryant <russell@digium.com>
-
- * codecs/Makefile: Ensure that libg722.a gets rebuilt if one of the
- files changes
-
- * /, pbx/pbx_gtkconsole.c: Merged revisions 97645 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97645 | russell | 2008-01-09 17:01:48 -0600 (Wed, 09 Jan 2008) |
- 2 lines Strip terminal sequences from the verbose messages
- ........
-
- * configure: re-gen configure
-
- * configure.ac: re-add check for gtk1, which is used for
- pbx_gtkconsole (related to issue #11706)
-
- * /, pbx/pbx_gtkconsole.c: Merged revisions 97640 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97640 | russell | 2008-01-09 16:26:33 -0600 (Wed, 09 Jan 2008) |
- 3 lines Make pbx_gtkconsole build ... but doesn't actually load
- on my system still (related to issue #11706) ........
-
-2008-01-09 21:37 +0000 [r97634] Terry Wilson <twilson@digium.com>
-
- * phoneprov/000000000000.cfg, phoneprov/000000000000-directory.xml,
- phoneprov/polycom.xml, res/res_phoneprov.c (added),
- funcs/func_strings.c, phoneprov/000000000000-phone.cfg,
- configs/modules.conf.sample, main/acl.c,
- include/asterisk/localtime.h, CHANGES,
- configs/phoneprov.conf.sample (added), Makefile, phoneprov
- (added), doc/tex/phoneprov.tex (added), main/stdtime/localtime.c,
- doc/tex/asterisk.tex: Added a new module, res_phoneprov, which
- allows auto-provisioning of phones based on configuration
- templates that use Asterisk dialplan function and variable
- substitution. It should be possible to create phone profiles and
- templates that work for the majority of phones provisioned over
- http. It is currently only intended to provision a single user
- account per phone. An example profile and set of templates for
- Polycom phones is provided. NOTE: Polycom firmware is not
- included, but should be placed in AST_DATA_DIR/phoneprov/configs
- to match up with the included templates.
-
-2008-01-09 20:30 +0000 [r97620-97623] Jason Parker <jparker@digium.com>
-
- * /, main/cli.c: Merged revisions 97622 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #11718) ........ r97622 | qwell | 2008-01-09 14:28:43 -0600
- (Wed, 09 Jan 2008) | 5 lines Correctly display a message if a
- command could not be found. Also fix a comment which may have led
- to this happening. Issue 11718, reported by kshumard. ........
-
- * /, main/cli.c: Merged revisions 97618 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97618 | qwell | 2008-01-09 14:05:45 -0600 (Wed, 09 Jan 2008) | 1
- line Fix some locking and return value funkiness. We really
- shouldn't be unlocking this lock inside of a function, unless we
- locked it there too. ........
-
-2008-01-09 18:53 +0000 [r97577] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 97575 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97575 | mmichelson | 2008-01-09 12:48:15 -0600 (Wed, 09 Jan
- 2008) | 3 lines Part 2 of app_queue doxygen improvements. Some
- smaller functions this time ........
-
-2008-01-09 18:12 +0000 [r97532-97533] Luigi Rizzo <rizzo@icir.org>
-
- * channels/console_gui.c: remove a wrong 'const'
-
- * images/kpad2.jpg: add annotations for the two message windows we
- use.
-
-2008-01-09 18:04 +0000 [r97531] Russell Bryant <russell@digium.com>
-
- * /, res/res_features.c: Merged revisions 97529 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97529 | russell | 2008-01-09 12:02:08 -0600 (Wed, 09 Jan 2008) |
- 2 lines Fix saying the parking space number to the caller doing
- the parking ... ........
-
-2008-01-09 18:03 +0000 [r97530] Luigi Rizzo <rizzo@icir.org>
-
- * channels/console_gui.c, channels/console_board.c,
- channels/console_video.h: Two changes: - support scrolling of
- message window; - simplify the code for creating a message
- window, and try it using a second one in the top of the keypad
- (where we echo the dialed number). The 'skin' that supports these
- two windows will be committed separately.
-
-2008-01-09 17:30 +0000 [r97495] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, codecs/codec_zap.c: Merged revisions 97491 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97491 | kpfleming | 2008-01-09 11:21:14 -0600 (Wed, 09 Jan 2008)
- | 2 lines report the same message whether Zaptel does not have
- transcoder support loaded or no transcoders were found ........
-
-2008-01-09 16:59 +0000 [r97490] Philippe Sultan <philippe.sultan@gmail.com>
-
- * /, channels/chan_gtalk.c: Merged revisions 97489 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r97489 | phsultan | 2008-01-09 17:44:24 +0100 (Wed, 09
- Jan 2008) | 7 lines Set the caller id within the gtalk_alloc
- function. As underlined in issue #10437 by Josh, we need to
- prevent a possible memory leak. We only set the name part of the
- caller id, the number part is not relevant when dealing with
- JIDs. Closes issue #11549. ........
-
-2008-01-09 16:44 +0000 [r97488] Luigi Rizzo <rizzo@icir.org>
-
- * channels/console_gui.c, channels/console_video.c,
- channels/console_board.c, channels/console_video.h: Implement
- keyboard handling, and use it to enter a number to dial in the
- 'message' area under the keypad. Now you can make calls using the
- keypad as a regular phone (or the keyboard for chars not present
- on the keypad)
-
-2008-01-09 16:13 +0000 [r97451] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 97450 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97450 | file | 2008-01-09 12:11:17 -0400 (Wed, 09 Jan 2008) | 6
- lines Don't do conferencing totally in Zaptel if Monitor is
- running on the channel. (closes issue #11709) Reported by:
- BigJimmy Patches: patch-meetmerec uploaded by BigJimmy (license
- 371) ........
-
-2008-01-09 15:45 +0000 [r97421-97449] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 97448 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97448 | kpfleming | 2008-01-09 09:43:19 -0600 (Wed, 09 Jan 2008)
- | 2 lines pass the right variable to get an error string... oops
- ........
-
- * channels/chan_zap.c, /: Merged revisions 97410 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97410 | kpfleming | 2008-01-09 09:26:23 -0600 (Wed, 09 Jan 2008)
- | 2 lines add error number output to ioctl failure messages to
- help with debugging ........
-
-2008-01-09 12:23 +0000 [r97389-97390] Luigi Rizzo <rizzo@icir.org>
-
- * channels/console_video.c, channels/console_video.h: implement the
- "console startgui" and "console stopgui" commands so you can
- start and stop the gui even outside of a call. This is convenient
- for testing, and also for using the keypad to pick up a call, and
- to dial a number (the latter not yet implemented, but should be
- close).
-
- * channels/chan_oss.c: make get_video_desc() return the active
- console if passed a null argument (channel).
-
-2008-01-09 00:58 +0000 [r97364-97365] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c: New option in trunk, needs strdupa to be safe,
- too
-
- * /, main/editline/readline.c, main/cli.c: Merged revisions 97350
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97350 | tilghman | 2008-01-08 18:44:14 -0600 (Tue, 08 Jan 2008)
- | 5 lines Allow filename completion on zero-length modules,
- remove a memory leak, remove a file descriptor leak, and make
- filename completion thread-safe. Patched and tested by tilghman.
- (Closes issue #11681) ........
-
-2008-01-09 00:18 +0000 [r97307-97309] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 97308 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97308 | mmichelson | 2008-01-08 18:17:40 -0600 (Tue, 08 Jan
- 2008) | 3 lines use the \retval doxygen command properly ........
-
- * /, apps/app_queue.c: Merged revisions 97304 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97304 | mmichelson | 2008-01-08 17:49:11 -0600 (Tue, 08 Jan
- 2008) | 5 lines Part 1 of N of adding doxygen comments to
- app_queue. I picked some of the most common functions used (which
- also happen to be some the biggest/ugliest functions too) to
- document first. I'm pretty new to doxygen so criticism is
- welcome. ........
-
-2008-01-08 23:51 +0000 [r97305] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: Add a new flag 'd' (with optional context)
- permitting any extension within that context to be entered as a
- new extension during the playback of a voicemail greeting. Patch
- inspired by bluecrow76, by tilghman. (Closes issue #7063)
-
-2008-01-08 23:35 +0000 [r97280-97303] Luigi Rizzo <rizzo@icir.org>
-
- * channels/console_board.c: add copyright (most of this code was
- written by Marta Carbone), remove some unused code, add/clarify
- some comments.
-
- * images/kpad2.jpg: Add the annotation for the textarea used for
- messages, and also change the background from white to something
- different to show that we can make use of fonts with transparent
- background.
-
- * images/font.png (added): add a font suitable for use with the
- console GUI. The background of this particular image is
- transparent so we can preserve the original background when we
- draw strings.
-
- * channels/console_gui.c, channels/console_video.c,
- channels/console_board.c (added), channels/Makefile: add support
- for textareas, used for various dialog windows on the gui. The
- main code to implement the textarea is in console_board.c, and
- uses a simple png image with the font, blitting characters on the
- designated areas of the main screen. Additionally we provide some
- annotations in the image used as a skin to indicate which areas
- are used for text messages. (images will be committed
- separately). At the moment the dialog area is only used to
- display a running counter, just as a proof of concept.
-
-2008-01-08 21:56 +0000 [r97248] Terry Wilson <twilson@digium.com>
-
- * apps/app_queue.c: Initialize new variable to NULL
-
-2008-01-08 21:28 +0000 [r97203-97208] Mark Michelson <mmichelson@digium.com>
-
- * configs/queues.conf.sample, CHANGES, apps/app_queue.c: Adding the
- option of specifying a second interface in a member definition
- for a queue. app_queue will monitor this second device's state
- for the member, even though it actually calls the first
- interface. This ability has been added for statically defined
- queue members, realtime queue members, and dynamic queue members
- added through the CLI, dialplan, or manager. (closes issue
- #11603, reported by acidv)
-
-2008-01-08 21:01 +0000 [r97199-97200] Olle Johansson <oej@edvina.net>
-
- * channels/chan_console.c: Change reference to external library so
- it appears on the extref listing
- http://www.asterisk.org/doxygen/trunk/extref.html
-
- * res/res_jabber.c: Iksemel is alive in a new home. Release 1.3 is
- out with bug fixes.
-
-2008-01-08 20:56 +0000 [r97198] Tilghman Lesher <tlesher@digium.com>
-
- * main/autoservice.c, /, main/utils.c: Merged revisions 97194 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97194 | tilghman | 2008-01-08 14:47:07 -0600 (Tue, 08 Jan 2008)
- | 3 lines Increase constants to where we're less likely to hit
- them while debugging. (Closes issue #11694) ........
-
-2008-01-08 20:52 +0000 [r97196-97197] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: One line documentation ftw!
-
- * /, channels/chan_mgcp.c: Merged revisions 97195 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97195 | file | 2008-01-08 16:48:20 -0400 (Tue, 08 Jan 2008) | 6
- lines Fix various DTMF issues in chan_mgcp. (closes issue #11443)
- Reported by: eferro Patches:
- dtmf_control_hybrid-inband-mode.patch uploaded by eferro (license
- 337) ........
-
-2008-01-08 20:45 +0000 [r97193] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 97192 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97192 | mmichelson | 2008-01-08 14:42:07 -0600 (Tue, 08 Jan
- 2008) | 9 lines Making some changes designed to not allow for a
- corrupted mailstream for a vm_state. 1. Add locking to the
- vm_state retrieval functions so that no linked list corruption
- occurs. 2. Make sure to always grab the persistent vm_state when
- mailstream access is necessary. 3. Correct an incorrect return
- value in the init_mailstream function. (closes issue #11304,
- reported by dwhite) ........
-
-2008-01-08 20:06 +0000 [r97153-97154] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Move common code for setting T38
- capabilities and fix a bug with fax detection in the SIP RTP read
- callback. It's still sort of silly... but more on that later.
- (closes issue #11239) Reported by: dimas Patches:
- sipt38prop.patch uploaded by dimas (license 88)
-
- * funcs/func_groupcount.c, /: Merged revisions 97152 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r97152 | file | 2008-01-08 15:53:52 -0400 (Tue, 08 Jan
- 2008) | 4 lines If no group has been provided to the GROUP_COUNT
- dialplan function then use the first one specific to the channel.
- (closes issue #11077) Reported by: m4him ........
-
-2008-01-08 19:06 +0000 [r97125] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c, main/asterisk.c: Merged revisions 97077
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97077 | tilghman | 2008-01-08 12:02:13 -0600 (Tue, 08 Jan 2008)
- | 3 lines Apply multiple crash fixes, found in issue #11386, but
- not completely closing that issue. ........
-
-2008-01-08 18:42 +0000 [r97041-97103] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 97093 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97093 | file | 2008-01-08 14:36:40 -0400 (Tue, 08 Jan 2008) | 4
- lines Make app_queue calls work with directed pickup. (closes
- issue #11700) Reported by: jbauer ........
-
- * utils/extconf.c: Make ast_atomic_fetchadd_int_slow magically
- appear in extconf. (closes issue #11703) Reported by: dmartin
-
-2008-01-07 23:03 +0000 [r96988] Luigi Rizzo <rizzo@icir.org>
-
- * channels/console_gui.c: add support for cropping the keypad image
- while displaying it. This way it can contain additional elements
- (e.g. fonts, buttons, widgets) without having to use a zillion
- files to store them.
-
-2008-01-07 22:31 +0000 [r96987] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Explicitly make literal constants long
- where they are expected to be.
-
-2008-01-07 21:12 +0000 [r96936] Jason Parker <jparker@digium.com>
-
- * main/config.c: Display a message if no config mappings are found
- with "core show config mappings". Closes issue #11704, patch by
- kshumard.
-
-2008-01-07 21:10 +0000 [r96934-96935] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Document some weird casting magic that's
- necessary to interface with the c-client
-
- * doc/tex/imapstorage.tex, apps/app_voicemail.c: Adding
- user-configurable TCP timeout settings to IMAP voicemail. This
- could go a long way towards preventing unexplainable hangs
- experienced by people. In the case of MWI hangs, this also will
- mean that the SIP port isn't blocked anymore. (closes issue
- #11665, reported by yehavi)
-
-2008-01-07 20:48 +0000 [r96885-96933] Russell Bryant <russell@digium.com>
-
- * /, configs/extensions.conf.sample: Merged revisions 96932 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r96932 | russell | 2008-01-07 14:47:52 -0600
- (Mon, 07 Jan 2008) | 10 lines Merged revisions 96931 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r96931 | russell | 2008-01-07 14:46:22 -0600 (Mon, 07
- Jan 2008) | 2 lines Change misery.digium.com to pbx.digium.com
- ........ ................
-
- * configs/http.conf.sample: Add a note about viewing the default
- set of documentation using the built-in http server
-
- * Makefile: If the HTML documentation exists, install it in the
- static-http/docs directory so that it can be viewed through the
- Asterisk http server if it is turned on.
-
- * build_tools/prep_tarball: Build the HTML version of the doc files
- for tarballs, as well
-
- * res/res_smdi.c, /: Merged revisions 96884 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r96884 | russell | 2008-01-07 10:39:23 -0600 (Mon, 07 Jan 2008) |
- 3 lines Don't crash if something happens when setting up an SMDI
- interface and it gets destroyed before the SMDI port handling
- thread gets created. ........
-
-2008-01-07 16:17 +0000 [r96862] Kevin P. Fleming <kpfleming@digium.com>
-
- * formats/format_sln16.c (added): add a file-format driver for
- 16KHz signed linear... which may or may not work
-
-2008-01-07 15:52 +0000 [r96858] Joshua Colp <jcolp@digium.com>
-
- * main/manager.c, main/loader.c: Move ModuleLoad and ModuleCheck
- manager commands from loader.c to manager.c. Previously they
- would get registered twice because of the way manager.c operates.
- (closes issue #11699) Reported by: caio1982 Patches:
- manager_module_commands1.diff uploaded by caio1982 (license 22)
-
-2008-01-07 15:06 +0000 [r96776-96836] Luigi Rizzo <rizzo@icir.org>
-
- * channels/console_gui.c: update comments to reflect reality (or at
- least planned behaviour). minor code cleanups
-
- * channels/console_gui.c: resolve a load-time problem avoiding a
- call to console_do_answer. On passing, fix dialling from the
- keypad.
-
-2008-01-05 23:05 +0000 [r96645-96743] Russell Bryant <russell@digium.com>
-
- * res/snmp/agent.c: Convert this file over the new method of
- getting the Asterisk version. (I don't have this building on this
- machine, so caio1982 on IRC is going to test it for me. :) )
-
- * Makefile, funcs/func_version.c, main/manager.c,
- channels/chan_sip.c, main/Makefile, build_tools/make_version_c
- (added), include/asterisk/version.h (added), res/res_agi.c, main,
- main/http.c, build_tools/make_version_h (removed),
- include/asterisk, main/asterisk.c: Now that the version.h file
- was getting properly regenerated every time the svn revision
- changed, every module that used the version was getting rebuilt
- after every svn update. This severly annoyed me pretty quickly,
- so I have improved the situation. Now, instead of generating
- version.h, main/version.c is generated. version.c includes the
- version information, as well as a couple of API calls for modules
- to retrieve the version. So now, only version.c will get rebuilt,
- and the main asterisk binary relinked, which is must faster than
- rebuilding http.c, manager.c, asterisk.c, relinking the asterisk
- binary, chan_sip.c, func_version.c, res_agi ... The only minor
- change in behavior here is that the version information reported
- by chan_sip, for example, is the version of the Asterisk core,
- and not necessarily the Asterisk version that the chan_sip module
- came from.
-
- * main/pbx.c: Print out the name of a function being registered in
- color, just like the name of applications when they get
- registered.
-
- * UPGRADE.txt: Add a note about changing modules.conf since another
- console channel driver is now present that can not be used at the
- same time as chan_alsa or chan_oss.
-
- * channels/chan_console.c: Add the URL to the home page for
- portaudio. Also add the location of the svn repository to check
- out portaudio v19.
-
- * /, main/devicestate.c: Merged revisions 96644 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r96644 | russell | 2008-01-04 20:09:19 -0600 (Fri, 04 Jan 2008) |
- 2 lines Don't pass an empty string as the device name. ........
-
-2008-01-05 01:05 +0000 [r96621] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_usbradio.c: improve chan_usbradio to use
- indications just like chan_alsa/chan_oss do now
-
-2008-01-04 23:12 +0000 [r96576] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/devicestate.c: Merged revisions 96575 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r96575 | tilghman | 2008-01-04 17:03:40 -0600 (Fri, 04 Jan 2008)
- | 7 lines Fix the problem of notification of a device state
- change to a device with a '-' in the name. Could probably do with
- a better fix in trunk, but this bug has been open way too long
- without a better solution. Reported by: stevedavies Patch by:
- tilghman (Closes issue #9668) ........
-
-2008-01-04 22:57 +0000 [r96574] Jason Parker <jparker@digium.com>
-
- * /, res/res_features.c: Merged revisions 96573 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes
- issue #11237) ........ r96573 | qwell | 2008-01-04 16:55:56 -0600
- (Fri, 04 Jan 2008) | 4 lines Properly continue in the dialplan if
- using PARKINGEXTEN and the slot is full. Issue 11237, patch by
- me. ........
-
-2008-01-04 19:35 +0000 [r96547] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 96525 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r96525 | tilghman | 2008-01-04 13:27:25 -0600 (Fri, 04 Jan 2008)
- | 4 lines If you change the bindaddr in sip.conf to a non-bound
- address and reload, sip goes kablooie. Reported and patched by:
- one47 (Closes issue #11535) ........
-
-2008-01-04 17:21 +0000 [r96500] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c, configure, include/asterisk/autoconfig.h.in,
- configure.ac, acinclude.m4: [commit message] (closes issue
- #10393) Reported by: tzafrir Patches: chan_alarm_asterisk.diff
- uploaded by tzafrir (license 46) (modified by me and added
- configure script support)
-
-2008-01-04 17:19 +0000 [r96499] Philippe Sultan <philippe.sultan@gmail.com>
-
- * res/res_jabber.c: Use SASL DIGEST-MD5 authentication over
- unsecured network connections only. This authentication mechanism
- is implemented under the iksemel API, which makes use of GnuTLS,
- whereas we use OpenSSL. Note : there's ongoing dicsussion at the
- SASL IETF WG in order to deprecate SASL DIGEST-MD5, see
- http://ietfreport.isoc.org/ids-wg-sasl.html.
-
-2008-01-04 16:21 +0000 [r96450] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 96449 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r96449 | russell | 2008-01-04 10:19:22 -0600 (Fri, 04 Jan 2008) |
- 7 lines Make use of the temporary channel pointer while the pvt
- is unlocked. (closes issue #11675) Reported by: flefoll Patches:
- chan_zap.c.patch-store-owner-before-unlock uploaded by flefoll
- (license 244) ........
-
-2008-01-03 23:14 +0000 [r96397-96398] Kevin P. Fleming <kpfleming@digium.com>
-
- * Makefile: we have to *always* use a completely silent 'make'
- invocation for generating the module embedding rules
-
- * Makefile: there was no reason to add this define for non-Solaris
- platforms
-
-2008-01-03 22:46 +0000 [r96395] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 96394 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r96394 | russell | 2008-01-03 16:44:22 -0600 (Thu, 03 Jan 2008) |
- 3 lines Don't crash if the iax2 pvt structure has been destroyed
- before we get to this point (closes issue #11672, reported by
- snuffy, patched by me) ........
-
-2008-01-03 21:58 +0000 [r96301-96368] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/channel.h: Document recent API addition
-
- * res/res_config_pgsql.c, /: Merged revisions 96318 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r96318 | tilghman | 2008-01-03 15:37:02 -0600 (Thu, 03
- Jan 2008) | 4 lines Missed initialization caused crash. Reported
- and fixed by: tiziano (Closes issue #11671) ........
-
- * main/channel.c: Allow the uniqueid to be used for searching for a
- channel in the list. Reported and initially patched by:
- michael-fig (Closes issue #11340)
-
-2008-01-03 20:04 +0000 [r96245-96272] Kevin P. Fleming <kpfleming@digium.com>
-
- * Makefile, tests/Makefile (added), tests/test_skel.c (added),
- tests (added): add some simple infrastructure for modules to be
- used for testing parts of Asterisk
-
- * channels/answer.h (removed), channels/ring10.h (removed),
- channels/busy.h (removed), channels/ringtone.h (removed),
- channels/Makefile, channels/chan_oss.c, channels/gentone.c
- (removed), channels: eliminiate sound_thread() and other stuff
- from chan_oss since Asterisk indications can handle it remove
- gentone and all the headers containing tones that are no longer
- needed
-
- * channels/chan_alsa.c: coding guidelines cleanup remove background
- thread and all sound generation mechanisms, as the built-in
- indications can handle everything that is needed
-
-2008-01-03 14:47 +0000 [r96221] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /: Merged revisions 96198 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r96198 | crichter | 2008-01-03 13:08:40 +0100 (Do, 03
- Jan 2008) | 1 line when overlapdial was used and no number was
- dialed, the call was dropped, now we just jump into the s
- extension, which makes a lot more sense. ........
-
-2008-01-03 06:16 +0000 [r96147-96174] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_agi.c: Add coordination between AMI and AGI applications,
- with an asyncagi method Feature proposed and patched by: moy
- (Closes issue #11282)
-
- * apps/app_mp3.c, apps/app_ices.c, main/asterisk.c: Compatibility
- fix for OpenBSD Report and fix by: mvanbaak (Closes issue #11669)
-
-2008-01-02 23:48 +0000 [r96103] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 96102 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r96102 | mmichelson | 2008-01-02 17:46:02 -0600 (Wed, 02 Jan
- 2008) | 4 lines We need to reset the membername to NULL on each
- iteration of this loop, otherwise the result is that multiple
- members can have the same name, since the variable was not reset
- on each iteration of the loop. ........
-
-2008-01-02 23:22 +0000 [r96076-96079] Russell Bryant <russell@digium.com>
-
- * channels/chan_console.c: Add support for generating a ringing
- sound on an incoming call. This is a bit of a hack. It just asks
- the core to generate the same tone that it would when you hear
- ringback when making an outbound call. But hey, it works, and you
- get the localized ring tone for the appropriate language set on
- the channel.
-
- * channels/chan_console.c: Note that this module doesn't actually
- play a ringing sound for an incoming call ... oops
-
- * channels/chan_console.c: Show the correct CLI command to answer
- the call
-
-2008-01-02 22:41 +0000 [r96073] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c: actually parse and store echocan parameters
- from zapata.conf... this *should* work <G>
-
-2008-01-02 22:40 +0000 [r96071] Joshua Colp <jcolp@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac: Don't
- use AST_C_DEFINE_CHECK for the two pthread things that may not
- actually be definitions, they could be enums for example.
-
-2008-01-02 22:29 +0000 [r96028] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_zap.c: Add curly braces around a compound if
- statement so that trunk will build properly
-
-2008-01-02 21:51 +0000 [r96019] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c, configs/zapata.conf.sample: another
- checkpoint... chan_zap can now use the new ZT_ECHOCAN_PARAMS
- ioctl if it is present, but doesn't parse any supplied parameters
- yet (this implementation is not very memory efficient as the
- parameters and their values will be duplicated for each channel
- that has the same settings, but we can worry about that later
- once it is working)
-
-2008-01-02 21:49 +0000 [r96018] Russell Bryant <russell@digium.com>
-
- * main/libresample/include/libresample.h: Add doxygen documentation
- to libresample.h while it's still fresh on my mind
-
-2008-01-02 21:08 +0000 [r95994] Mark Michelson <mmichelson@digium.com>
-
- * funcs/func_odbc.c, channels/chan_agent.c, funcs/func_strings.c,
- apps/app_rpt.c: Change instances of AST_NONSTANDARD_APP_ARGS(foo,
- bar, ',') to AST_STANDARD_APP_ARGS(foo, bar) (closes issue
- #11668, reported and patched by mvanbaak)
-
-2008-01-02 20:26 +0000 [r95947] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 95946 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r95946 | file | 2008-01-02 16:24:09 -0400 (Wed, 02 Jan 2008) | 4
- lines Allocate a SIP refer structure when performing a transfer
- using BYE with Also so that the transfer information is properly
- stored. (AST-2008-001) (closes issue #11637) Reported by:
- greyvoip ........
-
-2008-01-02 20:23 +0000 [r95944-95945] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Since ',' is the standard argument separator in
- trunk, change app_queue to use AST_STANDARD_APP_ARGS instead of
- AST_NONSTANDARD_APP_ARGS for determining member data.
-
- * include/asterisk/app.h: Fix a typo in a comment.
- AST_STANDARD_APP_ARGS uses ',' as the separator, not '|'.
-
-2008-01-02 19:47 +0000 [r95893-95939] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c: clean up hwgain CLI command and improve docs
- for swgain CLI command
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac,
- acinclude.m4: improve AC_C_DEFINE_CHECK to not try to evaluate
- the macro being checked for, but just check for its existence
- finish implementation of check for Zaptel HWGAIN support add
- check for Zaptel ECHOCANCEL_PARAMS support
-
- * codecs/Makefile, include/asterisk/libresample.h (added),
- codecs/codec_resample.c: and now just to keep the libresample
- party going... if the functions from libresample are going to be
- in the main Asterisk binary, it makes sense for the header that
- defines them to be available without any special CFLAGS and to
- out-of-tree modules building against /usr/include/asterisk
-
- * channels/chan_zap.c: umm... this did not compile on x86-64, and
- could not possibly have worked on any platform as it was passing
- string pointers to a function expecting ints
-
-2008-01-02 18:05 +0000 [r95891] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 95890 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r95890 | mmichelson | 2008-01-02 11:51:22 -0600 (Wed, 02 Jan
- 2008) | 9 lines A change to improve the accuracy of queue logging
- in the case where a member does not answer during the specified
- timeout period. Prior to this change, there was a small chance
- that the member name recorded in this case would be blank. Also
- prior to this change, if using the ringall strategy, if no one
- answered the call during the specified timeout, the member name
- listed in the queue log would randomly be one of the members that
- was rung. (closes issue #11498, reported and tested by hloubser,
- patched by me) ........
-
-2008-01-02 17:38 +0000 [r95888] Jason Parker <jparker@digium.com>
-
- * apps/app_osplookup.c: Update osplookup documentation to use
- commas instead of pipes. Closes issue #11666, patch by Laureano.
-
-2008-01-02 16:20 +0000 [r95864] Russell Bryant <russell@digium.com>
-
- * main/Makefile, main/translate.c: For some odd reason, the last
- set of libresample build changes from Kevin did not work for
- everyone, but it did for some. This set of changes makes trunk
- start again for those having problems. Instead of building
- libresample as a static library, it just links the object files
- in directly with the asterisk binary.
-
-2008-01-02 14:53 +0000 [r95816-95841] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/Makefile: fix some long-time breakage that kept
- chan_misdn from being embedded
-
- * channels/Makefile: use the proper technique for including
- submodules so that embedding will work
-
- * CHANGES: note that chan_console requires portaudio v19
-
- * configure, configure.ac: actually check for a function present in
- libiconv (don't know how this test could have worked before) and
- don't do the check on Linux/GNU systems because libiconv is not
- present there and attempting to link with '-liconv' always fails
- (it's not necessary as the iconv functionality is always
- available)
-
- * main/libresample/src/filterkit.h,
- main/libresample/src/resample.c,
- main/libresample/win/libresample.dsp, main/libresample/configure,
- main/libresample/Makefile.in, res/Makefile,
- main/libresample/configure.in, main/libresample/src,
- main/libresample/tests/testresample.c,
- main/libresample/win/libresample.vcproj,
- main/libresample/tests/compareresample.c, main/libresample/tests,
- codecs/codec_resample.c, res/res_resample.c (removed),
- main/libresample/README.txt, main/libresample/src/resamplesubs.c,
- main/libresample/tests/resample-sndfile.c,
- main/libresample/src/configtemplate.h,
- main/libresample/install-sh, main/Makefile, main/translate.c,
- main/libresample/include, main/libresample/src/resample_defs.h,
- codecs/Makefile, main/libresample/config.guess,
- main/libresample/config.sub, main/libresample/win,
- main/libresample/LICENSE.txt, main/libresample (added),
- main/libresample/Makefile.asterisk, build_tools/strip_nonapi,
- res/libresample (removed), main/libresample/src/filterkit.c,
- main/libresample/include/libresample.h: go back to including
- libresample in the main Asterisk binary, but this time including
- a small hack to ensure that it does get linked in (and also
- modify the strip_nonapi script to leave the resample_<foo>
- symbols alone)
-
-2008-01-02 11:34 +0000 [r95794] Philippe Sultan <philippe.sultan@gmail.com>
-
- * res/res_jabber.c: Set stream flags to zero upon initialization.
- When the XMPP over TLS/SSL connection resets for some reason, it
- is wrongly believed as being secured, which makes the
- re-connection process endlessly fail. This was reported by
- mvanbaak in issue #11644.
-
-2008-01-02 09:16 +0000 [r95771-95772] Luigi Rizzo <rizzo@icir.org>
-
- * main/loader.c: some cleanup of this code while I am trying to
- debug a problem with gdb dying while debugging asterisk. The
- problem seems to be related with a race in the handling of
- module_list, which in turn is triggeded by calling dlopen() on a
- system which uses initializers to create locks.
-
- * include/asterisk/module.h: There are three instances of the
- module definition macros, which make maintaining this file very
- error prone. This commit merges the embedded and !embedded
- versions, and fixes the C++ version. Eventually we should move to
- a single version of the macro. Too bad C++ doesn't like the
- C-style struct initializers .foo = some_value
-
-2008-01-02 04:33 +0000 [r95697-95746] Russell Bryant <russell@digium.com>
-
- * res/libresample/src/resample_defs.h,
- res/libresample/src/resample.c: Don't make libresample print out
- debugging output
-
- * main/translate.c: Make the translation table show slin16
-
- * apps/app_meetme.c: fix a spacing issue introduced in revision
- 95443.
-
- * main/Makefile, res/libresample/README.txt, res/Makefile,
- res/libresample/install-sh, res/libresample/configure,
- res/libresample/Makefile.in, res/libresample/include,
- codecs/Makefile, res/libresample/configure.in,
- res/libresample/src, res/libresample/config.guess,
- main/libresample (removed), res/libresample/config.sub,
- res/libresample/win, codecs/codec_resample.c,
- res/libresample/LICENSE.txt, res/libresample (added),
- res/libresample/Makefile.asterisk, res/libresample/tests,
- res/res_resample.c (added): Instead of linking libresample into
- the main Asterisk binary, build it as res_resample, and mark
- codec_resample as dependent upon res_resample. This prevents the
- linker from optimizing away libresample, and also makes it so the
- libresample code isn't linked in to multiple places. (I have
- another module in a branch that needs it, too.)
-
-2008-01-01 23:55 +0000 [r95671-95673] Luigi Rizzo <rizzo@icir.org>
-
- * channels/console_gui.c: call directly the cli command to
- implement hangup.
-
- * channels/vcodecs.c: prevent a panic when destroying a channel
- with no incoming video.
-
- * channels/console_video.c: remove a leftover sleep(1) used for
- debugging
-
-2008-01-01 23:09 +0000 [r95648] Joshua Colp <jcolp@digium.com>
-
- * codecs/Makefile: Fix building of codec_resample on platforms
- other then Cygwin. On everything else it actually gets built
- after codec_resample, so you can't exactly link it in since it
- doesn't exist.
-
-2008-01-01 22:21 +0000 [r95624-95625] Luigi Rizzo <rizzo@icir.org>
-
- * codecs/Makefile, codecs/codec_resample.c: make codec_resample
- build on __CYGWIN__, and make it load on FreeBSD (and probably
- other systems as well). Both need libresample.a to be specified
- in the linking phase, and cygwin needs <float.h> as other BSD.
- The checks for OS-specific headers should really be moved to some
- common header though.
-
- * build_tools/menuselect-deps.in, configure,
- include/asterisk/autoconfig.h.in, configure.ac,
- funcs/func_iconv.c, makeopts.in: implement "configure" checks for
- libiconv, and add the iconv dependency for func_iconv. This fixes
- some build issues on CYGWIN and FreeBSD and probably other
- platforms where libiconv is not there by default
-
-2007-12-31 23:44 +0000 [r95578] Mark Michelson <mmichelson@digium.com>
-
- * main/pbx.c, /: Merged revisions 95577 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r95577 | mmichelson | 2007-12-31 17:43:13 -0600 (Mon, 31 Dec
- 2007) | 9 lines Avoiding a potentially bad locking situation.
- ast_merge_contexts_and_delete writelocks the conlock, then calls
- ast_hint_extension, which attempts to readlock the same lock.
- Recursion with read-write locks is dangerous, so the inner lock
- needs to be removed. I did this by copying the "guts" of
- ast_hint_extension into ast_merge_contexts_and_delete (sans the
- extra lock). (this change is inspired by the locking problems
- seen in issue #11080, but I have no idea if this is the
- problematic area experienced by the reporters of that issue)
- ........
-
-2007-12-31 22:41 +0000 [r95501-95550] Russell Bryant <russell@digium.com>
-
- * codecs/codec_resample.c: Use float.h to fix the build on FreeBSD.
- Also, add some other platforms as they are likely the same.
-
- * channels/chan_console.c: Update chan_console to natively use a 16
- kHz sample rate. If it is talking to an 8 kHz endpoint, then
- codec_resample will automatically be used to properly resample
- the audio before sending it to/from chan_console.
-
- * main/libresample/src/filterkit.h, main/libresample/README.txt,
- main/libresample/tests/resample-sndfile.c,
- main/libresample/src/resamplesubs.c, main/Makefile,
- main/libresample/install-sh,
- main/libresample/src/configtemplate.h,
- main/libresample/src/resample.c,
- main/libresample/win/libresample.dsp, main/libresample/configure,
- main/libresample/Makefile.in, main/libresample/include, CHANGES,
- main/libresample/src/resample_defs.h,
- main/libresample/configure.in, main/libresample/src,
- main/libresample/config.guess, codecs/Makefile,
- main/libresample/tests/testresample.c, codecs/slin_resample_ex.h
- (added), main/libresample/config.sub, main/libresample/win,
- main/libresample/win/libresample.vcproj,
- main/libresample/LICENSE.txt, main/libresample (added),
- main/libresample/Makefile.asterisk, main/libresample/tests,
- main/libresample/tests/compareresample.c, codecs/codec_resample.c
- (added), main/libresample/src/filterkit.c,
- main/libresample/include/libresample.h: Merge changes from
- team/russell/codec_resample This commit imports libresample for
- use in Asterisk. It also adds a new codec module, codec_resample.
- This module uses libresample to re-sample signed linear audio
- between 8 kHz and 16 kHz. It also provides an alternative for
- converting between 16 kHz G.722 and 8 kHz signed linear when
- using G.722, which will likely be useful as some people have
- complained about volume issues when the current codec_g722
- converts to 8 kHz signed linear. But, to test this, you will have
- to disable the g722-to-slin and g722-to-slin16 translators in
- codec_g722.c.
-
-2007-12-31 20:33 +0000 [r95490] Tilghman Lesher <tlesher@digium.com>
-
- * /, funcs/func_env.c: Merged revisions 95470 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r95470 | tilghman | 2007-12-31 14:27:26 -0600 (Mon, 31 Dec 2007)
- | 3 lines Allow the default "0" to be returned if the STAT fails
- (Closes issue #11659) ........
-
-2007-12-31 18:46 +0000 [r95443] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_meetme.c: Fix a compiler warning (closes issue #11658,
- reported and patched by eliel)
-
-2007-12-31 16:13 +0000 [r95383-95412] Russell Bryant <russell@digium.com>
-
- * configs/console.conf.sample (added), configs/modules.conf.sample,
- channels/chan_console.c (added), CHANGES: Merge the main set of
- changes from team/russell/chan_console. Add a new console channel
- driver, chan_console, which is a console channel driver that uses
- portaudio as a cross platform audio interface. It was written to
- provide a console channel driver that works with Mac CoreAudio,
- but it supports a number of other audio interfaces, as well,
- including OSS and ALSA. It could one day be the single console
- channel driver, but does not yet have as many features as
- chan_oss.
-
- * include/asterisk/channel.h: fix a spelling error in a comment
-
- * include/asterisk/config.h: Add CV_STRINGFIELD() macro. This lets
- you set a config variable to a string field. (from
- team/russell/chan_console)
-
- * configure, include/asterisk/autoconfig.h.in: Regenerate configure
- script to include check for portaudio.
-
- * build_tools/menuselect-deps.in, configure.ac, makeopts.in: Add
- configure script checking for portaudio.
-
-2007-12-29 02:02 +0000 [r95262-95313] Luigi Rizzo <rizzo@icir.org>
-
- * channels/vcodecs.c, channels/console_video.c, channels/Makefile,
- channels/console_video.h, channels/vgrabbers.c (added): Move
- grabbers definitions to a separate file, vgrabbers.c, so it is
- easier to add more entries. This required moving struct grab_desc
- to the common header, and adding an entry in the Makefile. On
- passing, cleanup some comments and file headers (some are still
- missing).
-
- * channels/console_gui.c, channels/console_video.c: virtualize the
- interface for video grabbers, which should make it easier to add
- support for more grabbers (V4L2, firewire, and so on).
-
- * channels/console_video.c: Add a few entries up to 1408x1152 in
- the table of known video resolutions. This makes it very
- convenient to enlarge images using the right-click on the video
- window.
-
- * channels/vcodecs.c, channels/console_video.c: change the
- interface of video encapsulation routines, they only need the
- buffer and mtu as input.
-
- * channels/console_gui.c, channels/vcodecs.c,
- channels/console_video.c, channels/console_video.h: various
- rearrangements and renaming of console_video stuff
-
-2007-12-28 18:39 +0000 [r95233] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: The diff for this change looks really bad, but
- all I did here was decrease the indentation of most of the
- queue_exec function by reversing the logic of an if statement.
- This change makes the function comply better with the coding
- guidelines. Since this change is purely a cosmetic change to the
- code, I am only committing the change to trunk.
-
-2007-12-28 18:26 +0000 [r95192] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 95191 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r95191 | russell | 2007-12-28 12:24:59 -0600 (Fri, 28 Dec 2007) |
- 6 lines Remove duplicate increment of the header count in the
- add_header() function. (closes issue #11648) Reported by: makoto
- Patch provided by sergee, committed patch by me, inspired by
- comments from putnopvut ........
-
-2007-12-28 16:12 +0000 [r95167] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_amd.c, CHANGES: Some changes to app_amd. The channel
- name is printed in verbose messages maximumWordLength option
- added. Duration of words that do not meet the minimum word
- duration will be logged The duration of pre-greeting silence will
- be logged Only consider us in the greeting if we actually
- detected a valid word duration. (closes issue #11650, reported
- and patched by davevg)
-
-2007-12-28 08:57 +0000 [r95139] Luigi Rizzo <rizzo@icir.org>
-
- * channels/console_video.c: fix a small bug in printing out
- geometries - wrong input.
-
-2007-12-28 00:17 +0000 [r95096] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 95095 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r95095 | mmichelson | 2007-12-27 18:16:15 -0600 (Thu, 27 Dec
- 2007) | 8 lines I found a bug while browsing the queue code and
- managed to reproduce it in a small setup. If a queue uses the
- ringall strategy, it was possible through unfortunate coincidence
- for a single member at a given penalty level to make app_queue
- think that all members at that penalty level were unavailable and
- cause the members at the next penalty level to be rung. With this
- patch, we will only move to the next penalty level if ALL the
- members at a given penalty level are unreachable. ........
-
-2007-12-27 23:32 +0000 [r95073] Luigi Rizzo <rizzo@icir.org>
-
- * apps/app_dictate.c, apps/app_mp3.c, apps/app_voicemail.c: remove
- more unnecessary casts for NULL. main/say.c is a big offender in
- this respect.
-
-2007-12-27 23:28 +0000 [r95070] Jason Parker <jparker@digium.com>
-
- * doc/asterisk.8, main/asterisk.c: Fix -s socket option, and
- document it as well. Closes issue #11645, patch by Laureano.
-
-2007-12-27 23:13 +0000 [r95068-95069] Luigi Rizzo <rizzo@icir.org>
-
- * apps/app_ices.c, apps/app_queue.c, apps/app_voicemail.c: NULL
- does not need to be cast to (char *)
-
- * channels/chan_oss.c: remove useless casts
-
-2007-12-27 21:41 +0000 [r95025] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /: Merged revisions 95024 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r95024 | russell | 2007-12-27 15:40:02 -0600 (Thu, 27 Dec 2007) |
- 9 lines Don't report a syntax error when an empty string is
- passed to ast_get_group. Just return 0. (closes issue #11540)
- Reported by: tzafrir Patches: group_empty.diff uploaded by
- tzafrir (license 46) -- slightly changed by me ........
-
-2007-12-27 20:11 +0000 [r94978] Mark Michelson <mmichelson@digium.com>
-
- * /, main/io.c: Merged revisions 94977 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94977 | mmichelson | 2007-12-27 14:09:06 -0600 (Thu, 27 Dec
- 2007) | 3 lines Fixing a typo in a comment. ........
-
-2007-12-27 17:34 +0000 [r94908-94934] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_h323.c: Merged revisions 94924 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94924 | file | 2007-12-27 13:32:15 -0400 (Thu, 27 Dec 2007) | 6
- lines Include types.h in chan_h323 as without it it can not be
- compiled on some operating systems like FreeBSD to name one.
- (closes issue #11585) Reported by: sobomax Patches:
- chan_h323.c.diff uploaded by sobomax (license 359) ........
-
- * /, channels/chan_sip.c: Merged revisions 94905 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94905 | file | 2007-12-27 13:27:11 -0400 (Thu, 27 Dec 2007) | 4
- lines Use ast_strlen_zero to see if our_contact is set or not on
- the dialog. It is possible for it to be a pointer to NULL.
- (closes issue #11557) Reported by: FuriousGeorge ........
-
-2007-12-27 17:26 +0000 [r94904] Luigi Rizzo <rizzo@icir.org>
-
- * channels/console_gui.c, channels/console_video.c: more
- localization of gui stuff
-
-2007-12-27 17:18 +0000 [r94903] Mark Michelson <mmichelson@digium.com>
-
- * doc/manager_1_1.txt: Adding documentation for new manager actions
- and events in app_queue
-
-2007-12-27 16:51 +0000 [r94902] Luigi Rizzo <rizzo@icir.org>
-
- * CHANGES: clarify the type of video support in chan_oss
-
-2007-12-27 16:11 +0000 [r94830-94877] Russell Bryant <russell@digium.com>
-
- * codecs/codec_g722.c: I went looking for where we downloaded the
- g722 implementation and came across these two links. So, I'm
- adding them so they are available for reference later.
-
- * /, main/translate.c, include/asterisk/translate.h: Merged
- revisions 94828-94829 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94828 | russell | 2007-12-27 08:33:21 -0600 (Thu, 27 Dec 2007) |
- 9 lines Change ast_translator_best_choice() to only pay attention
- to audio formats. This fixes a problem where Asterisk claims that
- a translation path can not be found for channels involving video.
- (closes issue #11638) Reported by: cwhuang Tested by: cwhuang
- Patch suggested by cwhuang, with some additional changes by me.
- ........ r94829 | russell | 2007-12-27 08:44:29 -0600 (Thu, 27
- Dec 2007) | 2 lines Use the constant that I really meant to use
- here ... ........
-
-2007-12-27 09:13 +0000 [r94826-94827] Olle Johansson <oej@edvina.net>
-
- * funcs/func_dialplan.c: This function checks more than just
- contexts...
-
- * apps/app_pickupchan.c: - Add Copyright - Doxygen fixes Note: -
- This application needs better documentation and a RESULT code in
- the dialplan.
-
-2007-12-27 01:03 +0000 [r94825] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/manager.c, /: Merged revisions 94824 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94824 | kpfleming | 2007-12-26 18:01:47 -0700 (Wed, 26 Dec 2007)
- | 2 lines make this comment explain the situation in an even more
- explicit fashion ........
-
-2007-12-27 00:48 +0000 [r94819-94823] Luigi Rizzo <rizzo@icir.org>
-
- * channels/console_gui.c: more steps to decouple the gui from the
- rest of the code.
-
- * channels/console_gui.c, channels/console_video.c,
- channels/console_video.h: Enable building the code even if SDL is
- not present (similarly, SDL is also detected at runtime). Now we
- should be able to stream video even without a rendering device
- (useful for remote monitoring).
-
- * channels/console_gui.c, channels/console_video.c: more
- localizations around sdl_setup
-
- * channels/console_gui.c: use fread instead of mmap to read in the
- comment area from the keypad. fread is simpler and more portable,
- and there is no performance gain in using mmap.
-
- * images/kpad2.jpg: update the region description with an empty
- line at the beginning.
-
-2007-12-26 22:38 +0000 [r94818] Tilghman Lesher <tlesher@digium.com>
-
- * build_tools/cflags.xml, channels/chan_zap.c: Allow more spans
- than 32. Also, rearrange compiler flags so the most often used
- flags appear closer to the top. Reported by: tzafrir Patch by:
- tzafrir,tilghman (Closes issue #11528)
-
-2007-12-26 22:29 +0000 [r94817] Luigi Rizzo <rizzo@icir.org>
-
- * channels/console_gui.c, channels/console_video.c: another bunch
- of gui localizations
-
-2007-12-26 22:14 +0000 [r94814] Jason Parker <jparker@digium.com>
-
- * apps/app_exec.c: Make 'else' argument to ExecIf optional. Clean
- up the description and usage text a bit. Closes issue #11564,
- patch by pnlarsson (with some extra cleanup by me).
-
-2007-12-26 22:10 +0000 [r94810-94813] Luigi Rizzo <rizzo@icir.org>
-
- * channels/console_gui.c, channels/console_video.c: more
- localization of sdl stuff
-
- * channels/console_gui.c, channels/console_video.c,
- channels/console_video.h: move more gui stuff into console_gui.c
-
-2007-12-26 20:49 +0000 [r94809] Tilghman Lesher <tlesher@digium.com>
-
- * main/manager.c, /: Merged revisions 94808 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94808 | tilghman | 2007-12-26 14:43:38 -0600 (Wed, 26 Dec 2007)
- | 6 lines Workaround for what is probably a glibc bug (but we'll
- see this crop up again and again, if we don't add the
- workaround). Reported by: rolek Patch by: tilghman (Closes issue
- #11601, closes issue #11426) ........
-
-2007-12-26 20:02 +0000 [r94806] Jason Parker <jparker@digium.com>
-
- * pbx/pbx_loopback.c, apps/app_zapbarge.c, pbx/pbx_spool.c,
- apps/app_authenticate.c, apps/app_zapscan.c, apps/app_zapras.c,
- apps/app_alarmreceiver.c, apps/app_amd.c, pbx/pbx_realtime.c,
- pbx/pbx_dundi.c, apps/app_zapateller.c, pbx/pbx_config.c,
- pbx/pbx_gtkconsole.c, apps/app_adsiprog.c, apps/app_cdr.c: Use
- defined return values in load_module in more places. (closes
- issue #11096) Patches: pbx_config.c.patch uploaded by moy
- (license 222) pbx_dundi.c.patch uploaded by moy (license 222)
- pbx_gtkconsole.c.patch uploaded by moy (license 222)
- pbx_loopback.c.patch uploaded by moy (license 222)
- pbx_realtime.c.patch uploaded by moy (license 222)
- pbx_spool.c.patch uploaded by moy (license 222)
- app_adsiprog.c.patch uploaded by moy (license 222)
- app_alarmreceiver.c.patch uploaded by moy (license 222)
- app_amd.c.patch uploaded by moy (license 222)
- app_authenticate.c.patch uploaded by moy (license 222)
- app_cdr.c.patch uploaded by moy (license 222)
- app_zapateller.c.patch uploaded by moy (license 222)
- app_zapbarge.c.patch uploaded by moy (license 222)
- app_zapras.c.patch uploaded by moy (license 222)
- app_zapscan.c.patch uploaded by moy (license 222)
-
-2007-12-26 20:01 +0000 [r94805] Luigi Rizzo <rizzo@icir.org>
-
- * channels/console_gui.c, channels/vcodecs.c,
- channels/console_video.c, channels/console_video.h: more
- preparation for untangling of the various console_video stuff
-
-2007-12-26 19:09 +0000 [r94796-94802] Russell Bryant <russell@digium.com>
-
- * main/autoservice.c, /: Merged revisions 94801 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94801 | russell | 2007-12-26 13:04:31 -0600 (Wed, 26 Dec 2007) |
- 4 lines Just in case the AST_FLAG_END_DTMF_ONLY flag was already
- set before starting autoservice, remember it and ensure that the
- channel has the same setting when autoservice gets stopped.
- (pointed out by d1mas, patched up by me) ........
-
- * funcs/func_dialplan.c (added), CHANGES: Add a new dialplan
- function, DIALPLAN_EXISTS(), which allows you to check for the
- existence of a dialplan target. (closes issue #11579) Reported
- by: irroot Patches: func_dialplan2.c uploaded by irroot (license
- 52) -- Additional changes by me.
-
- * main/autoservice.c, /: Merged revisions 94797 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94797 | russell | 2007-12-26 12:46:39 -0600 (Wed, 26 Dec 2007) |
- 4 lines When a channel is in autoservice, mark a flag on the
- channel that says that we only care about the END of a digit.
- That way, no magic digit emulation stuff will happen when all
- we're doing is queueing up END frames. ........
-
- * main/channel.c: Leave a note for a minor bug that was pointed out
- by d1mas
-
-2007-12-26 18:05 +0000 [r94795] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_zap.c: Convert raw bits for callprogress bitfield
- to use constants, for greater code clarity Reported by: dimas
- Patch by: dimas (Closes issue #11280)
-
-2007-12-26 17:26 +0000 [r94787-94794] Russell Bryant <russell@digium.com>
-
- * /, res/res_features.c: Merged revisions 94793 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94793 | russell | 2007-12-26 11:24:17 -0600 (Wed, 26 Dec 2007) |
- 3 lines Don't try to send a parked call back to itself. (closes
- issue #11622, reported by djrodman, patched by me) ........
-
- * Makefile, /: Merged revisions 94789 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94789 | russell | 2007-12-26 11:00:03 -0600 (Wed, 26 Dec 2007) |
- 5 lines List include/asterisk/version.h as a .PHONY target
- because we want the commands listed for this target to be
- executed regardless of whether the file exists or not. This fixes
- having the version not up to date when running from svn. (closes
- issue #11619, reported by plack, fixed by me) ........
-
- * main/autoservice.c, /: Merged revisions 94790 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94790 | russell | 2007-12-26 11:06:26 -0600 (Wed, 26 Dec 2007) |
- 5 lines Don't store DTMF BEGIN frames while a channel is in
- autoservice. It's just going to make ast_read() do a lot of extra
- work when the channel comes back out of autoservice. (closes
- issue #11628, patched by me) ........
-
- * channels/chan_iax2.c: Fix a bug in peer handling that caused
- multiple instances of a peer to end up in the peers container
- after a reload. Somehow, this bug doesn't exist in 1.4 ...
- (closes issue #11626) (reported by pnlarsson, additional info
- from mvanbaak, fixed by me)
-
- * utils: update svn:ignore for astcanary
-
-2007-12-26 15:58 +0000 [r94782] Mark Michelson <mmichelson@digium.com>
-
- * configs/extconfig.conf.sample, main/logger.c, CHANGES: Adding
- support for storing the queue log entries in a realtime backend.
- (closes issue #11625, reported and patched by sergee) Thank you
- very much to sergee for adding this new feature!
-
-2007-12-26 10:14 +0000 [r94774] Luigi Rizzo <rizzo@icir.org>
-
- * channels/console_gui.c (added), channels/vcodecs.c (added),
- channels/console_video.c: Split console_video.c so that video
- codecs and gui functions are in separate files (still #include'd
- because of tangling in the data structures, but this is going to
- be cleaned up). The video grabbing functions still need to be
- moved to a separate file.
-
-2007-12-25 04:10 +0000 [r94771-94773] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_pickupchan.c (added): Add pickup by channel (Closes
- issue #11161)
-
- * channels/chan_zap.c, configs/zapata.conf.sample: Change the
- abbreviated TON from 'A' to 'V', since 'A' is a legitimate DTMF
- character. Also, fix the documentation to match the code.
-
- * res/res_agi.c: Add channel thread ID to the information passed to
- AGI. Reported by: dror99 Patch by: tilghman (Closes issue #11162)
-
-2007-12-24 19:43 +0000 [r94764-94768] Tilghman Lesher <tlesher@digium.com>
-
- * main/channel.c, /: Merged revisions 94767 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94767 | tilghman | 2007-12-24 13:36:59 -0600 (Mon, 24 Dec 2007)
- | 5 lines Race: we need to wait to queue a NewChannel event until
- after the channel is inserted into the channel list. The reason
- is because some manager users immediately queue requests from the
- channel when they see that event and are confused when Asterisk
- reports no such channel. (Closes issue #11632) ........
-
- * /: Merged revisions 94763 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94763 | tilghman | 2007-12-24 09:39:56 -0600 (Mon, 24 Dec 2007)
- | 5 lines Another bit of bad logic in realtime_peer Reported by:
- dimas Patch by: dimas (Closes issue #11631) ........
-
-2007-12-23 14:51 +0000 [r94713-94741] Luigi Rizzo <rizzo@icir.org>
-
- * channels/console_video.c, channels/console_video.h: support
- sdl_videodriver to send output to x11/aalib/console
-
- * channels/console_video.c: move reading info from the keypad to a
- separate function. Remove an unused keypad field and some
- debugging messages. Adjust formatting on config file parsing
-
- * channels/console_video.c: make sure the minimum surface depth is
- 16bpp so we can create YUVoverlays. With this change we can do
- setenv SDL_VIDEODRIVER aalib and output to an ascii window (which
- is still in an X11 window). If you also do unsetenv DISPLAY then
- the output goes into the main asterisk window, unfortunately it
- interferes with the normal output so you don't see much. In any
- case, i don't think we are very far away from having a working
- xterm videophone!
-
- * channels/Makefile: avoid rebuilding dependent files if the
- generated busy.h and ringtone.h do not change. Ths masks (but
- does not solve) a but that i am seeing in doing a 'gmake install'
- without donig a 'gmake all' first.
-
-2007-12-23 01:38 +0000 [r94662] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 94660 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94660 | tilghman | 2007-12-22 19:21:03 -0600 (Sat, 22 Dec 2007)
- | 2 lines Argh... I suppose third time's the charm. ........
-
-2007-12-22 22:44 +0000 [r94615-94638] Luigi Rizzo <rizzo@icir.org>
-
- * configs/oss.conf.sample, channels/console_video.c: Change the
- name of config file entries for keypad regions from
- 'keypad_entry' to 'region'. Fix the example file accordingly.
- Also make some fixes in the code do reset entries on reload of
- the keypad. The recently committed kpad2.jpg has the correct
- names.
-
- * images/kpad2.jpg (added): add a sample keypad (with annotations)
- for console video
-
- * channels/console_video.c, channels/Makefile, channels/chan_oss.c,
- channels/console_video.h (added): Build console_video support by
- linking in, as opposed to including, console_video.c This will
- ease the task of splitting console_video.c into its components
- (V4L and X11 grabbers, various video codecs and packetizers,
- SDL), as well as ease future extensions (e.g. additional video
- sources, codecs and rendering engines). For the time being
- nothing changes for users: video support is off by default, and
- requires -DHAVE_VIDEO_CONSOLE on the command line to be included
- (if SDL and FFMPEG are available).
-
-2007-12-21 21:19 +0000 [r94593] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Something I've been itching to do for a
- while now. A minor optimization in app_voicemail. Since the
- dtable in base_encode always gets populated with the same values
- every time and never changes, make it static and const and only
- initialize it once. Also, there's no reason to define
- BASEMAXINLINE twice, so remove the redundant #define.
-
-2007-12-21 20:50 +0000 [r94549-94551] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: We should only clear this value if we have
- to
-
- * channels/chan_zap.c: Commit non TCP transport part of #11506.
- Includes numerous additional parameters, as well as RLT support
- for DMS type switches
-
-2007-12-21 20:38 +0000 [r94542-94548] Mark Michelson <mmichelson@digium.com>
-
- * res/res_config_sqlite.c: Store dates using local time instead of
- UTC (closes issue #11610, reported and patched by
- rbraun_performatique)
-
- * apps/app_queue.c: Fix a memory leak when reloading queue rules.
-
- * CHANGES: The one documentation source I forgot to update after
- the merge of the queue-penalty branch was the CHANGES file. No
- longer!
-
- * apps/app_voicemail.c: Lots of coding guidelines cleanup.
-
- * /, apps/app_voicemail.c: Merged revisions 94540 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94540 | mmichelson | 2007-12-21 14:11:34 -0600 (Fri, 21 Dec
- 2007) | 8 lines Better quota support for using IMAP storage
- voicemail (closes issue #11415, reported by jaroth) (closes issue
- #11152, reported by selsky) Patch provided by jaroth ........
-
-2007-12-21 20:12 +0000 [r94541] Jason Parker <jparker@digium.com>
-
- * codecs/codec_lpc10.c, codecs/codec_g722.c, codecs/codec_adpcm.c,
- codecs/codec_alaw.c, codecs/codec_speex.c, codecs/codec_g726.c,
- codecs/codec_gsm.c, codecs/codec_ulaw.c, codecs/codec_zap.c:
- codecs.conf really shouldn't be mandatory.. it never had been
- before, so let's go back to being optional. A big "thank you" to
- pnlarsson on IRC for allowing me access to his system to debug
- this. Closes issue #11584.
-
-2007-12-21 20:01 +0000 [r94477-94539] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 94538 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94538 | mmichelson | 2007-12-21 13:59:45 -0600 (Fri, 21 Dec
- 2007) | 5 lines The mail_copy c-client function does not expect a
- full imap mailbox string, just the name of the mailbox. (closes
- issue #11419, reported and patched by jaroth, with additional
- patchwork from me) ........
-
- * main/dial.c: AST_LIST_REMOVE_CURRENT only takes one argument in
- trunk
-
- * main/dial.c, /: Merged revisions 94468 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94468 | mmichelson | 2007-12-21 10:49:35 -0600 (Fri, 21 Dec
- 2007) | 6 lines Since we are freeing list elements within a list
- traversal, we need to use the safe traversal and remove the item
- from the list before freeing it. (closes issue 11612, reported by
- dtyoo) ........
-
-2007-12-21 16:12 +0000 [r94463-94465] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 94464 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94464 | mmichelson | 2007-12-21 10:11:44 -0600 (Fri, 21 Dec
- 2007) | 3 lines Removing a debug message I accidentally just
- committed ........
-
- * /, main/say.c, apps/app_queue.c: Merged revisions 94420 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94420 | mmichelson | 2007-12-21 09:45:14 -0600 (Fri, 21 Dec
- 2007) | 5 lines Fixing Portuguese syntax for saying dates and
- times. Also some coding guidelines cleanup. (closes issue #11599,
- reported and patched by caio1982, coding guidelines cleanup by
- me) ........
-
-2007-12-21 15:14 +0000 [r94419] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/asterisk.c: Merged revisions 94418 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94418 | tilghman | 2007-12-21 09:07:42 -0600 (Fri, 21 Dec 2007)
- | 2 lines Fix for restart-as-user problem reported via the -dev
- list ........
-
-2007-12-21 01:14 +0000 [r94345-94396] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Moved the update of the queue_ent's rule list
- to just before we try to call queue members. This allows for the
- change in penalty levels to be executed at the most logical time
- frame.
-
- * configs/queues.conf.sample, doc/tex/channelvariables.tex,
- apps/app_queue.c, configs/queuerules.conf.sample (added): Merging
- the queue-penalty branch. In short, this allows one to
- dynamically adjust the QUEUE_MAX_PENALTY and the newly introduced
- QUEUE_MIN_PENALTY during a call depending on the amount of time
- passed. The purpose is to allow the call to open up to more (or
- maybe just different) members without the caller's losing his
- place in the queue. See configs/queuerules.conf.sample for an
- example of how to set up queue rules and
- configs/queues.conf.sample for how to associate a rule with a
- queue. Along with the functional changes, new CLI and manager
- commands exist to show the rules defined and there is an
- additional CLI command to reload the queue rules. Future
- enhancements that may be made: support for realtime queue rules
- and support for dynamically adding a rule through the manager or
- CLI. Also a manager command to reload the queue rules (I'll
- probably write this myself very soon).
-
- * apps/app_voicemail.c: The changes to header inclusion in trunk
- broke compilation of app_voicemail when using IMAP storage. The
- reason is that c-client has its own definitions for LOG_WARNING
- and LOG_DEBUG, so we need to be sure to include asterisk's
- definitions last so that we use the proper values in
- app_voicemail. (closes issue #11437, reported by blitzrage, patch
- suggested by blitzrage)
-
-2007-12-20 22:39 +0000 [r94320] Russell Bryant <russell@digium.com>
-
- * configs/zapata.conf.sample: Add a bit more to the description of
- the "mwimonitor" option.
-
-2007-12-20 22:28 +0000 [r94319] Steve Murphy <murf@digium.com>
-
- * build_tools/make_buildopts_h: closes issue #11287; thanks to
- snuffy for this fix, which will surely make all solaris owners
- shout praises to his name.
-
-2007-12-20 20:25 +0000 [r94252-94257] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 94256 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r94256 | russell | 2007-12-20 14:22:22 -0600
- (Thu, 20 Dec 2007) | 13 lines Merged revisions 94255 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r94255 | russell | 2007-12-20 14:21:41 -0600 (Thu, 20
- Dec 2007) | 5 lines Fix another potential seg fault ... (closes
- issue #11606) Reported by: dimas ........ ................
-
- * channels/chan_zap.c, /: Merged revisions 94251 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94251 | russell | 2007-12-20 14:08:42 -0600 (Thu, 20 Dec 2007) |
- 10 lines Fix a deadlock in d-channel handling in chan_zap. This
- deadlock was introduced by the fix to ensure that channels are
- properly locked when handling channel variables. There were
- sections of this code where the channel pvt was locked before the
- channel lock, when in fact it _must_ be the other way around.
- (closes issue #11582) Reported by: bugi ........
-
-2007-12-20 12:56 +0000 [r94168-94191] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_usbradio.c, include/asterisk/config.h,
- channels/console_video.c, channels/chan_oss.c: add some macros to
- simplify parsing the config file, see description in config.h .
- They are a variant of the set of macros i used in chan_oss.c,
- structured in a way to be more robust to the presence of spurious
- ';' - basically, they define wrappers for 'do {' and '} while
- (0)', plus some helper functions to deal with simple cases such
- as ast_copy_string, ast_malloc, strtoul, ast_true ... The prefix
- (CV_ as 'Config Variable') tries to be easy to remember and has
- been chosen to not conflict with other existing macros in the
- tree. For the time being, I have only updated the three source
- files in the tree that used the old M_* macros. Hopefully, more
- files will be converted. NOTE: I understand that inventing my own
- dialect of C is generally wrong; however, the lack of adequate
- support in the language encourages lazy programming practices
- (such as ignoring errors, bounds, etc.) and this increases the
- chance of vulnerability in the code, especially because we are
- parsing user input here. Hopefully, these macros and the use of
- ast_parse_arg (in config.h) should encourage the programmer to
- write more robust code.
-
- * include/asterisk/paths.h, res/snmp/agent.c, utils/ael_main.c,
- utils/extconf.c, main/asterisk.c, utils/conf2ael.c: modify
- http://svn.digium.com/view/asterisk?view=rev&rev=93603 so that
- paths and filename are writable by asterisk.c without causing
- segfaults. This involves defining the variables as const char *,
- and having them point to as static, writable buffer defined in
- asterisk.c On passing, fix some errors in using these variables
- in some files in utils/ , and in res/snmp/agent.c which was
- redefining a variable without using paths.h (not applicable to
- 1.4)
-
-2007-12-19 23:17 +0000 [r94123-94124] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: 1. Unify the check for a penalty < 0 into the
- set_member_penalty code. 2. Fix an error when checking the CLI
- command for setting a member's penalty. 3. Fix a logging error if
- the incorrect parameter was the queue name or interface. (closes
- issue #11544, reported and patched by Laureano)
-
- * /, res/res_monitor.c: Merged revisions 94122 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94122 | mmichelson | 2007-12-19 17:02:22 -0600 (Wed, 19 Dec
- 2007) | 6 lines Sox versions 13.0.0 and newer do not have
- "soxmix" and instead use sox -m. res_monitor needs to use this if
- the user does not have soxmix. (closes issue #11589, reported by
- amessina, patch inspired by amessina but with a flourish from me)
- ........
-
-2007-12-19 22:51 +0000 [r94085] Russell Bryant <russell@digium.com>
-
- * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
- Merged revisions 94077 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94077 | russell | 2007-12-19 16:48:48 -0600 (Wed, 19 Dec 2007) |
- 4 lines Check for the existence of the soxmix application on the
- target platform and have the result available in autoconfig.h.
- (part of issue #11589) ........
-
-2007-12-19 20:20 +0000 [r94052-94053] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: Add 'voicemail reload' command. Reported
- by: eliel Patch by: eliel (Closes issue #11365)
-
- * apps/app_waituntil.c (added): Add contributed WaitUntil app.
- Original code by pprindeville, updated for trunk by tilghman.
- (Closes issue #11487)
-
-2007-12-19 19:29 +0000 [r94029] Russell Bryant <russell@digium.com>
-
- * include/asterisk/time.h: Add a couple of new time API calls -
- ast_tvdiff_sec and ast_tvdiff_usec (closes issue #11270) Reported
- by: dimas Patches: tvdiff_us-4.patch uploaded by dimas (license
- 88)
-
-2007-12-19 17:58 +0000 [r94002] Luigi Rizzo <rizzo@icir.org>
-
- * channels/console_video.c: Add instructions on how to generate
- your own font.
-
-2007-12-19 17:31 +0000 [r93956] Joshua Colp <jcolp@digium.com>
-
- * /: Merged revisions 93955 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r93955 | file | 2007-12-19 13:29:20 -0400 (Wed, 19 Dec 2007) | 2
- lines Make the 1.4 builders happy, ensure var is NULL. ........
-
-2007-12-19 17:13 +0000 [r93952] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 93949 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r93949 | tilghman | 2007-12-19 11:04:13 -0600 (Wed, 19 Dec 2007)
- | 3 lines Avoid segfault in chan_iax when peer isn't defined
- (Closes issue #11602) ........
-
-2007-12-19 17:09 +0000 [r93925-93950] Luigi Rizzo <rizzo@icir.org>
-
- * main/utils.c, include/asterisk/strings.h: Add a new API function,
- written at least twice in app_voicemail.c and likely in other
- places too. This is quite useful when placing mail/html stuff in
- config files. /*! \brief Convert some C escape sequences
- (\b\f\n\r\t) into the equivalent characters. \brief s The string
- to be converted (will be modified). \return The converted string.
- */ char *ast_unescape_c(char *s);
-
- * include/asterisk/config.h, main/config.c: add support for
- PARSE_DOUBLE, and remove identifiers for types not supported
- (INT16 and UINT16)
-
-2007-12-19 09:20 +0000 [r93899] Olle Johansson <oej@edvina.net>
-
- * CHANGES: Reorganize CHANGES a bit. The "misc" section grew too
- large...
-
-2007-12-19 08:57 +0000 [r93898] Luigi Rizzo <rizzo@icir.org>
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac,
- acinclude.m4, makeopts.in: Properly document AST_EXT_TOOL_CHECK()
- and use it to check for NETSMP and GTK (GTK is not used thoug).
- AST_EXT_TOOL_CHECK() could be used for checking curl status as
- well, perhaps with a small addition because we currently seem to
- require a curl version greater than X.Y.Z Add a NETSMP_INCLUDE
- entry in makeopts.in We don't have yet any macros for using
- pkg-config to check for a specific package (right now there is
- only gtk2+ in the category).
-
-2007-12-19 08:57 +0000 [r93897] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Adding the
- ability to specify the To: header in an outbound INVITE by adding
- an exclamation mark to the dial string. This patch also exists
- for 1.4 in the fixtoheader-1.4 branch and has been in production
- for quite some time.
-
-2007-12-19 08:12 +0000 [r93875] Luigi Rizzo <rizzo@icir.org>
-
- * res/snmp/agent.c: make netsmp build under AST_DEVMODE.
- Description, included in the source, is below. I should note that
- the PACKAGE_* macros that asterisk defines in autoconfig.h are
- not used anywhere in the tree so they should just be removed. /*
- * There is some collision collision between netsmp and asterisk
- names, * causing build under AST_DEVMODE to fail. * * The
- following PACKAGE_* macros are one place. * Also netsnmp has an
- improper check for HAVE_DMALLOC_H, using * #if HAVE_DMALLOC_H
- instead of #ifdef HAVE_DMALLOC_H * As a countermeasure we define
- it to 0, however this will fail * when the proper check is
- implemented. */ No
-
-2007-12-19 07:01 +0000 [r93854] Olle Johansson <oej@edvina.net>
-
- * CHANGES, main/asterisk.c, doc/asterisk.sgml: Add option for
- starting remote Asterisk by naming the actual runtime socket
- instead of pointing to configuration file with -C Reported by:
- sobomax Patches: asterisk.c.diff.trunk uploaded by sobomax
- (license 359) doc changes by committer (closes issue #11598)
-
-2007-12-19 00:09 +0000 [r93827] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * apps/app_osplookup.c: add missing header file
-
-2007-12-18 23:38 +0000 [r93804-93805] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c: Making the canary error message a little more
- obvious.
-
- * utils/Makefile, utils/astcanary.c (added), main/asterisk.c: Add a
- canary process, for high priority mode (asterisk -p) to ensure
- that if Asterisk goes into a busy loop, the machine will be
- recoverable. We'd still need to do a restart to put Asterisk back
- into high priority mode, but at least a reboot won't be required.
- (Closes issue #11559)
-
-2007-12-18 21:13 +0000 [r93741] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Move some warnings away to debug since some
- devices send a packet with a silly string as a NAT keepalive
- packet.
-
-2007-12-18 18:39 +0000 [r93672] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions
- 93668 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r93668 | tilghman | 2007-12-18 12:29:39 -0600
- (Tue, 18 Dec 2007) | 10 lines Merged revisions 93667 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r93667 | tilghman | 2007-12-18 12:23:06 -0600 (Tue, 18
- Dec 2007) | 2 lines Fixing AST-2007-027 (Closes issue #11119)
- ........ ................
-
-2007-12-18 18:20 +0000 [r93666] Luigi Rizzo <rizzo@icir.org>
-
- * include/asterisk/paths.h: remove a leftover line with only a '#'
- (wonder why the compiler does not complain!) and variables that
- are only used in asterisk.c
-
-2007-12-18 17:05 +0000 [r93626] Mark Michelson <mmichelson@digium.com>
-
- * main/channel.c, /: Merged revisions 93625 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r93625 | mmichelson | 2007-12-18 11:02:48 -0600 (Tue, 18 Dec
- 2007) | 6 lines Rework deadlock avoidance used in ast_write,
- since it meant that agent channels which were being monitored had
- one audio file recorded and one empty audio file saved. (closes
- issue #11529, reported by atis patched by me) ........
-
-2007-12-18 10:24 +0000 [r93558-93603] Luigi Rizzo <rizzo@icir.org>
-
- * include/asterisk/paths.h, channels/chan_sip.c, res/res_crypto.c,
- utils/ael_main.c, utils/extconf.c, main/asterisk.c,
- res/res_monitor.c, utils/conf2ael.c: make configuration variable
- const so they are not accidentally modified. This requires
- casting the strings in asterisk.c when writing to them, so we do
- it through a macro to do it consistently.
-
- * channels/chan_unistim.c, res/res_crypto.c, main/astmm.c,
- apps/app_ices.c, utils/extconf.c, channels/chan_iax2.c,
- main/asterisk.c, main/config.c, main/db.c, apps/app_adsiprog.c,
- cdr/cdr_csv.c: remove unnecessary (char *) casts for
- ast_config_AST_* variables. There are some left in the .flex
- files, left to the maintainer...
-
- * build_tools/make_defaults_h, main/asterisk.c: Rename the macros
- in defaults.h - they are not meant to be globally visible.
- Document the fact that DEFAULT_TMP_DIR cannot be overridden from
- the default configuration (this needs to be fixed, as you could
- have a totally different spooldir configured at runtime, and yet
- DEFAULT_TMP_DIR keeps the compile-time default). Remove two
- unused entries for sounds and images.
-
- * Makefile.moddir_rules: make the code match documentation - now
- you can specify multiple words in MODULE_PREFIX.
-
- * CREDITS: Name the people responsible for some recent
- contributions to the tree.
-
- * Makefile: Two small changes: + document the difference between
- "A=foo make ..." and "make A=foo ..." and suggest using
- COPTS/LDOPTS if you need to use the second form to pass compiler
- and loader flags; + define only in one place the environment used
- to build stuff in menuselect/
-
-2007-12-18 07:56 +0000 [r93557] Olle Johansson <oej@edvina.net>
-
- * doc/CODING-GUIDELINES: A minor update, caused by a recent bug
- report ;-)
-
-2007-12-18 07:22 +0000 [r93536] Luigi Rizzo <rizzo@icir.org>
-
- * doc/CODING-GUIDELINES: small documentation update (nothing
- important).
-
-2007-12-18 02:57 +0000 [r93514] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_unistim.c: You... will... build! I say so and
- therefore you will.
-
-2007-12-18 02:42 +0000 [r93493] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_unistim.c, include/asterisk/threadstorage.h: minor
- cleanups
-
-2007-12-17 23:10 +0000 [r93464] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_unistim.c: fix building under cygwin. At this point
- WINARCH should go away.
-
-2007-12-17 22:54 +0000 [r93405] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_unistim.c: remove some unnecessary includes
-
-2007-12-17 22:50 +0000 [r93390] Jason Parker <jparker@digium.com>
-
- * /, main/translate.c: Merged revisions 93381 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r93381 | qwell | 2007-12-17 16:45:57 -0600 (Mon, 17 Dec 2007) | 4
- lines What was I thinking when I wrote this masterpiece? -1 + 1 =
- 0.. who woulda thunk it?. ........
-
-2007-12-17 22:38 +0000 [r93380] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_oss.c: surprising as it may be, chan_oss compiles
- correctly under cygwin as well, provided you look for soundcard.h
- in the right place...
-
-2007-12-17 22:29 +0000 [r93378] Joshua Colp <jcolp@digium.com>
-
- * /, main/utils.c: Merged revisions 93377 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r93377 | file | 2007-12-17 18:28:09 -0400 (Mon, 17 Dec 2007) | 7
- lines Do not try to access information about a lock when printing
- out a trylock attempt. It is possible for the lock that it
- references to no longer be valid. This would have caused
- segfaults or deadlocks. (issue #BE-263) (closes issue #11080)
- Reported by: callguy (closes issue #11100) Reported by: callguy
- ........
-
-2007-12-17 21:14 +0000 [r93337] Tilghman Lesher <tlesher@digium.com>
-
- * /, include/asterisk/time.h: Merged revisions 93336 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r93336 | tilghman | 2007-12-17 15:12:42 -0600 (Mon, 17
- Dec 2007) | 6 lines Today is tomorrow's yesterday, and
- yesterday's tomorrow is today, and tomorrow's tomorrow is the day
- after tomorrow, so who cares if you recycle anyway? If this
- confuses you, that's nothing compared to what this fixes. ;-)
- ........
-
-2007-12-17 21:12 +0000 [r93335] Olle Johansson <oej@edvina.net>
-
- * channels/chan_zap.c, /, channels/chan_sip.c, apps/app_queue.c,
- channels/chan_iax2.c, channels/chan_mgcp.c: Merged revisions
- 93182 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r93182 | oej | 2007-12-17 08:15:13 +0100 (MÃ¥n, 17 Dec 2007) | 8
- lines Issue 11574: Add dependencies on res_monitor and
- res_features. I wonder if Asterisk can run at all without
- res_features. My guess is that there's propably a lot of more
- modules and the core that depends on it. Reported by: caio1982
- (closes issue #11574) ........
-
-2007-12-17 20:42 +0000 [r93293-93297] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Removing some leftover debug messages from a
- while back.
-
- * /, apps/app_voicemail.c: Merged revisions 93291 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r93291 | mmichelson | 2007-12-17 13:53:48 -0600 (Mon, 17 Dec
- 2007) | 6 lines We need to create the directory for a voicemail
- user even if they are using IMAP storage since greetings are
- stored in the filesystem. (closes issue #11388, reported by
- spditner, patch by me inspired by a patch by spditner) ........
-
-2007-12-17 18:07 +0000 [r93252] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 93250 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r93250 | file | 2007-12-17 14:05:55 -0400 (Mon, 17 Dec 2007) | 6
- lines If a call is received with a called number IE containing
- nothing go to the 's' extension. (closes issue #9099) Reported
- by: kb1_kanobe2 Patches: 20070906__9099.diff.txt uploaded by
- Corydon76 (license 14) ........
-
-2007-12-17 17:16 +0000 [r93191-93224] Kevin P. Fleming <kpfleming@digium.com>
-
- * utils: all created files need to be listed in the ignore property
-
- * channels/chan_unistim.c, build_tools/menuselect-deps.in,
- configure, configure.ac, channels/Makefile, channels/chan_oss.c:
- make the configure script detect that it is running on a Windows
- platform, and report that information so that menuselect can use
- it (all information that is used to decide whether to build
- modules or not must be fed to menuselect so the user knows what
- will be built and why... don't make module build decisions in the
- makefiles, please)
-
- * Makefile: make using PRINT_DIR a little easier
-
-2007-12-17 15:18 +0000 [r93187-93190] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Fix usage of rtptimeout. It can be used
- without rtpkeepalive, and the value can not be accessed directly
- in the SIP pvt structure. All RTP related timeouts have to be
- retrieved using the ast_rtp_* function calls. (closes issue
- #11562) Reported by: ibc
-
- * channels/chan_unistim.c: If no timezone is available use the
- default message. (closes issue #11576) Reported by: junky
-
- * channels/chan_unistim.c: Make chan_unistim actually be able to
- unload. When creating a thread that you want to pthread_join you
- have to explicitly create it as joinable, and also if using
- pthread_cancel you have to have a pthread_testcancel to see if it
- has been called.
-
-2007-12-17 07:27 +0000 [r93184-93185] Kevin P. Fleming <kpfleming@digium.com>
-
- * codecs, /, build_tools/make_version,
- include/asterisk/autoconfig.h.in, configure.ac, apps,
- Makefile.moddir_rules, res/Makefile, pbx/Makefile,
- build_tools/prep_moduledeps (removed), channels/Makefile, cdr,
- formats, Makefile, codecs/Makefile, funcs, apps/Makefile,
- configure, build_tools/embed_modules.xml, cdr/Makefile,
- build_tools/prep_tarball, makeopts.in, formats/Makefile, res,
- pbx, channels, funcs/Makefile: Merged revisions 93180 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r93180 | kpfleming | 2007-12-16 22:44:51 -0800 (Sun, 16 Dec 2007)
- | 23 lines In
- http://lists.digium.com/pipermail/asterisk-dev/2007-December/031145.html,
- rizzo brought up some issues related to the way that the metadata
- required for menuselect and the rest of the build system is
- extracted from the source files. Since I had a few hours to kill
- on an airplane today, I decided to improve this situation... so
- now the system caches the extracted metadata and uses it to build
- the menuselect 'tree' as much as it can. The result of this is
- that when a single source file is changed, only the metadata for
- that file needs to be extracted again, and the rest is used from
- the cache files. I also reduced the number of forked processes
- required to do the metadata extraction; it was actually possible
- to do most of what we needed in the Makefiles themselves without
- using any shell scripts at all! On my laptop, these changes
- resulted in an 80% decrease in the time required for the
- 'menuselect.makeopts' automatic check to occur after editing a
- single source file. While doing this work I also cleaned up a few
- minor things in the Makefiles, adding a check for 'awk' to the
- configure script and changed all remaining places we use 'grep'
- or 'awk' to use the ones found by the configure script, and
- changed the 'prep_tarball' script to build the menuselect
- metadata so that tarballs of Asterisk will include it and won't
- require the user to wait while it is extracted after unpacking.
- ........
-
-2007-12-16 19:06 +0000 [r93173] Luigi Rizzo <rizzo@icir.org>
-
- * Makefile: menuselect.makeopts is not a .PHONY target
-
-2007-12-16 13:38 +0000 [r93163-93167] Olle Johansson <oej@edvina.net>
-
- * pbx/pbx_dundi.c: Convert from LOG_DEBUG etc to ast_debug. Thanks,
- dimas! (closes issue #11572) Reported by: dimas Patches:
- dundilog-trunk.patch uploaded by dimas (license 88)
-
- * main/manager.c, CHANGES: Adding a new CLI command for "manager
- reload", which is important now that you need to reload after
- changes. Thanks YS. Reported by: ys Patches:
- trunk93163_manager_reload.c.diff uploaded by ys (license 281)
- (related to issue #11414)
-
- * main/manager.c, CHANGES: Change manager so that registered
- accounts are stored in memory. This opens for a manager realtime
- implementation. If you change accounts in manager.conf, you now
- need to reload to activate the changes (deletions, additions).
- This was not the case with 1.4. Reported by: ys Patches:
- trunk93163_manager_reload.c.diff uploaded by ys (license 281)
- (closes issue #11414)
-
- * CHANGES: Adding console_video to CHANGES. It's important that we
- keep this file up to date, even with experimental stuff.
-
- * channels/chan_unistim.c, main/udptl.c, configs/dundi.conf.sample,
- channels/chan_sip.c, include/asterisk/rtp.h,
- include/asterisk/netsock.h, channels/iax2-provision.c,
- UPGRADE.txt, doc/tex/qos.tex, configs/skinny.conf.sample,
- CHANGES, channels/chan_iax2.c, main/rtp.c, main/netsock.c,
- configs/h323.conf.sample, configs/iax.conf.sample,
- channels/chan_skinny.c, configs/mgcp.conf.sample,
- configs/unistim.conf.sample, channels/chan_h323.c,
- configs/iaxprov.conf.sample, pbx/pbx_dundi.c,
- configs/sip.conf.sample, channels/chan_mgcp.c: 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)
-
-2007-12-16 10:34 +0000 [r93162] Luigi Rizzo <rizzo@icir.org>
-
- * Makefile: use a simpler idiom for 'cmp -s ...'
-
-2007-12-16 09:37 +0000 [r93152-93161] Olle Johansson <oej@edvina.net>
-
- * main/asterisk.c: Don't drop the first character randomly in long
- listings in the CLI. Reported by: slavon Patches:
- asterisk-consolerefresh2.diff.txt uploaded by jamesgolovich
- (license 176) Tested by: eliel (closes issue #9325)
-
- * configs/sip.conf.sample, CHANGES: Update documentation
-
- * channels/chan_sip.c, configs/sip.conf.sample: Make more timers
- settable in SIP so that we can force timeout earlier on
- non-responsive SIP servers. Thanks, jcmoore, for the patch!
- Reported by: jcmoore Patches: peer_t1_timerb_trunk_v3.patch.txt
- uploaded by jcmoore (license 9) (closes issue #9771)
-
- * include/asterisk/file.h: Typo fixed earlier, that wasn't a typo
- after all. Didn't a clever guy once say "Compile before you
- commit" ? :-)
-
-2007-12-15 08:10 +0000 [r93151] Russell Bryant <russell@digium.com>
-
- * include/asterisk/file.h: fix a typo from revision 93138
-
-2007-12-15 00:44 +0000 [r93138-93145] Luigi Rizzo <rizzo@icir.org>
-
- * configs/oss.conf.sample: configuration options related to video
- support.
-
- * channels/console_video.c (added): Bring in video console support
- for chan_oss (and later chan_alsa too). This is disabled in the
- default build, you need to explicitly enable it compiling with
- make COPTS=-DHAVE_VIDEO_CONSOLE In return, you will be able to do
- a video call with chan_oss, using the webcam (or X11 grabbing) as
- local source, and rendering the incoming stream on your screen.
- Currently supported formats are h261, h263, h263+, h264, mpeg4
- (all through the avcodec lib, part of ffmpeg). Incoming video is
- on the left, outgoing video is on the right, while the center
- displays a keypad (if configured so). Right clicking on the video
- windows increases the size, center clicking reduces the size.
- Dragging the mouse (with the left key) on the right window while
- the X11 grabber is active moves the grab area. This is the result
- of work by Sergio Fadda, Marta Carbone and myself, all properly
- disclaimed to digium. Note, there is a lot of work left to do in
- this module, including adding support for Video4LinuxV2 (I have
- patches from Matteo Brancaleoni which should be integrated), and
- making the GUI a lot more friendly than it is now (e.g.
- supporting merging or switching among multiple sources, a text
- window, and more).
-
- * channels/chan_oss.c: remove some redundant headers
-
- * include/asterisk/file.h: include mmap header if detected by
- configure
-
-2007-12-14 22:02 +0000 [r93094-93115] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Resolve a compiler warning
-
- * apps/app_voicemail.c: Change places where the name "INBOX" was
- hardcoded to use the imapfolder setting from voicemail.conf
- instead. This commit will help to get issue #11415 moving towards
- commitment.
-
-2007-12-14 21:09 +0000 [r93090] Tilghman Lesher <tlesher@digium.com>
-
- * Makefile, channels/chan_unistim.c, codecs/ilbc/iLBC_define.h:
- Solaris compat fixes Reported by: snuffy Patch by:
- snuffy,tilghman (Closes issue #11315)
-
-2007-12-14 19:31 +0000 [r93067] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_dundi.c: make something static
-
-2007-12-14 19:27 +0000 [r93066] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_privacy.c, UPGRADE.txt, CHANGES,
- configs/privacy.conf.sample (removed): Remove use of privacy.conf
- by the Privacy app. Reported by: eliel Patch by: eliel (Closes
- issue #11344)
-
-2007-12-14 19:19 +0000 [r93042-93065] Mark Michelson <mmichelson@digium.com>
-
- * main/pbx.c, main/manager.c, funcs/func_timeout.c: I needed to
- increment the numbers used on the VERBOSITY_ATLEAST calls by 1.
- Thanks to kpfleming for pointing this out.
-
- * include/asterisk/logger.h, main/pbx.c, main/manager.c,
- funcs/func_timeout.c: Changed VERBOSITY_LEVEL to
- VERBOSITY_ATLEAST to be more accurate.
-
- * include/asterisk/logger.h, main/pbx.c, main/manager.c,
- funcs/func_timeout.c, main/logger.c: After reading Russell's
- e-mail to the dev list stating that checking option_verbose is
- not equivalent to the check done by ast_verb, I wrote a macro,
- VERBOSITY_LEVEL, which does this check. I did a quick look in the
- source and used this macro in some places where option_verbose
- was used. I also converted some verbose messages in logger.c to
- use ast_verb instead of ast_verbose.
-
-2007-12-14 18:24 +0000 [r93041] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_meetme.c: gcc 4.1.3 wants a union used here.
-
-2007-12-14 17:49 +0000 [r93001-93004] Russell Bryant <russell@digium.com>
-
- * main/config.c: Print an error message if a #included file does
- not exist
-
-2007-12-14 17:29 +0000 [r92999] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_agi.c: Publish the AGI events to manager. Reported by:
- moy Patch by: moy,tilghman (Closes issue #11337)
-
-2007-12-14 15:59 +0000 [r92976] Mark Michelson <mmichelson@digium.com>
-
- * funcs/func_timeout.c: Reintroduce an optimization that was lost
- when converting trunk to use ast_verb.
-
-2007-12-14 15:49 +0000 [r92939] Tilghman Lesher <tlesher@digium.com>
-
- * main/editline/sys.h: If malloc.h is included in a Solaris build,
- the compilation breaks. Reported by: snuffy Patch by: snuffy
- (Closes issue #11313)
-
-2007-12-14 15:18 +0000 [r92938] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 92937 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r92937 | file | 2007-12-14 11:16:15 -0400 (Fri, 14 Dec 2007) | 4
- lines Up the length of the format on the SIP channel since it can
- now be rather long. (closes issue #11552) Reported by:
- francesco_r ........
-
-2007-12-14 15:14 +0000 [r92936] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_agi.c: Merged revisions 92933 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r92933 | tilghman | 2007-12-14 09:01:10 -0600 (Fri, 14 Dec 2007)
- | 5 lines Change help documentation to match actual behavior
- (FAILURE vs FAILED). Reported by: angeloxx-sir Patch by: tilghman
- (Closes issue #11548) ........
-
-2007-12-14 15:08 +0000 [r92935] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /: Merged revisions 92934 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r92934 | crichter | 2007-12-14 16:05:28 +0100 (Fr, 14
- Dez 2007) | 1 line fixed the sequencing of WAITING_4DIGS state
- setting and overlap_task thread starting. ........
-
-2007-12-14 14:48 +0000 [r92913] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_dial.c, main/pbx.c, main/srv.c, channels/chan_skinny.c,
- res/res_features.c, apps/app_minivm.c, apps/app_amd.c,
- res/snmp/agent.c, apps/app_chanspy.c, apps/app_mixmonitor.c,
- main/asterisk.c, main/netsock.c, apps/app_voicemail.c: Convert
- ast_verbose to ast_verb. Reported by: snuffy Patch by: snuffy
- (Closes issue #11547)
-
-2007-12-14 01:25 +0000 [r92876] Mark Michelson <mmichelson@digium.com>
-
- * /, include/asterisk/lock.h: Merged revisions 92875 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r92875 | mmichelson | 2007-12-13 19:24:06 -0600 (Thu, 13
- Dec 2007) | 7 lines When compiling with DETECT_DEADLOCKS, don't
- spam the CLI with messages about possible deadlocks. Instead just
- print the intended single message every five seconds. (closes
- issue 11537, reported and patched by dimas) ........
-
-2007-12-13 23:10 +0000 [r92816-92855] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_meetme.c: When working with dates, use numeric form
- whenever possible, as it's faster. Also, a bunch of coding
- guidelines fixes.
-
- * channels/chan_zap.c, /: Merged revisions 92815 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r92815 | tilghman | 2007-12-13 15:28:39 -0600 (Thu, 13 Dec 2007)
- | 5 lines Properly initialize polarity statuses, so that they are
- detected properly. Reported by: julianjm Patch by: julianjm
- (Closes issue #10238) ........
-
-2007-12-13 20:23 +0000 [r92811] Joshua Colp <jcolp@digium.com>
-
- * include/asterisk/app.h, include/asterisk/module.h, res/res_agi.c,
- apps/app_rpt.c: Move usage of the old LOCAL_USER_* macros to the
- new ast_module_user_* functions in a few documentation places.
- (closes issue #11533) Reported by: IgorG Patches:
- oldmacroclean.v1.diff uploaded by IgorG (license 20)
-
-2007-12-13 20:14 +0000 [r92810] Jason Parker <jparker@digium.com>
-
- * main/pbx.c, /: Merged revisions 92809 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r92809 | qwell | 2007-12-13 14:13:48 -0600 (Thu, 13 Dec 2007) | 1
- line Make application help text a little more clear about the use
- of extensions in a filename. ........
-
-2007-12-13 20:12 +0000 [r92806-92808] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 92807 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r92807 | mmichelson | 2007-12-13 14:03:20 -0600 (Thu, 13 Dec
- 2007) | 3 lines Prevent another potential fd leak ........
-
- * /, apps/app_voicemail.c: Merged revisions 92803 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r92803 | mmichelson | 2007-12-13 13:49:55 -0600 (Thu, 13 Dec
- 2007) | 3 lines Prevent a possible fd leak. ........
-
-2007-12-13 17:46 +0000 [r92779] Tilghman Lesher <tlesher@digium.com>
-
- * cdr/cdr_adaptive_odbc.c: Don't use backslash as an escape
- character, unless it really is an escape character.
-
-2007-12-13 16:23 +0000 [r92758] Jason Parker <jparker@digium.com>
-
- * channels/chan_sip.c: Remove remnants of a poorly merged commit.
- (92697)
-
-2007-12-13 15:40 +0000 [r92737] Doug Bailey <dbailey@digium.com>
-
- * apps/app_voicemail.c: Tag voicemails with UTC time as opposed to
- local time zone
-
-2007-12-13 00:18 +0000 [r92697] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_sip.c, channels/chan_h323.c, main/config.c:
- Merged revisions 92696 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #10690) ........ r92696 | qwell | 2007-12-12 18:11:09 -0600
- (Wed, 12 Dec 2007) | 7 lines If a typo is found in a config file,
- we previous continued on with what was already loaded. We do not
- want to do this (see bug below for details). This makes it so
- that if a [ is found without a ], the entire config will fail,
- and nothing in it will be loaded. Issue 10690. ........
-
-2007-12-12 23:44 +0000 [r92676] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Revert an "optimization" that I added in
- revision 89887, as the user who reported issue #11449 has
- demonstrated that it actually was a performance hit on his
- machine. I think that it is possible that it could still be a
- benefit on systems under higher load, especially SMP systems, but
- I don't have enough time or interest to find out at the moment.
- (closes issue #11449)
-
-2007-12-12 21:22 +0000 [r92618] Jason Parker <jparker@digium.com>
-
- * /, apps/app_meetme.c, channels/ringtone.h: Merged revisions 92617
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #11048) ........ r92617 | qwell | 2007-12-12 15:15:45 -0600
- (Wed, 12 Dec 2007) | 4 lines Don't increment user count until
- after name has been recorded (if enabled). Issue 11048, tested by
- pep. ........
-
-2007-12-12 20:05 +0000 [r92594] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_dial.c, main/logger.c, main/utils.c,
- apps/app_mixmonitor.c: Conversions of free to ast_free, where
- applicable, and several other formatting fixes. Reported by:
- eliel Patch by: eliel,tilghman (Closes issue #11209)
-
-2007-12-12 19:50 +0000 [r92562] Russell Bryant <russell@digium.com>
-
- * res/res_features.c: Merged revisions 92556 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r92556 | russell | 2007-12-12 13:40:02 -0600 (Wed, 12 Dec 2007) |
- 1 line resolve compiler warning ........
-
-2007-12-12 17:51 +0000 [r92511-92526] Mark Michelson <mmichelson@digium.com>
-
- * res/res_features.c: Same change to trunk as revision 92510. I'm
- not sure why I merged this way, but I did.
-
-2007-12-12 17:15 +0000 [r92476-92507] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c: Correctly handle possible memory allocation
- failure Reported by: eliel Patch by: eliel (Closes issue #11512)
-
- * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
- Merged revisions 92463 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r92463 | tilghman | 2007-12-12 10:52:56 -0600 (Wed, 12 Dec 2007)
- | 4 lines Test directly for the API that fixed AST-2007-026, to
- ensure that older versions of PostgreSQL are no longer
- acceptable. (Closes issue #11526) ........
-
-2007-12-12 16:11 +0000 [r92444] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 92443 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r92443 | mmichelson | 2007-12-12 10:08:55 -0600 (Wed, 12 Dec
- 2007) | 3 lines Removing an unused variable. ........
-
-2007-12-11 22:20 +0000 [r92423] Olle Johansson <oej@edvina.net>
-
- * include/asterisk/term.h, channels/misdn/isdn_msg_parser.c,
- channels/ringtone.h, include/asterisk/ulaw.h,
- include/jitterbuf.h, include/asterisk/manager.h,
- include/asterisk/transcap.h, channels/misdn/isdn_lib.c,
- channels/gentone.c, include/asterisk/zapata.h,
- channels/misdn/isdn_lib.h, include/asterisk/doxyref.h,
- channels/DialTone.h, channels/misdn/ie.c,
- channels/misdn/chan_misdn_config.h, channels/iax2.h,
- channels/misdn/portinfo.c, include/asterisk/udptl.h,
- main/cygload.c, include/asterisk/translate.h: Doxygen updates,
- formatting. misdn stuff needs a lot of doxygenification (Hello,
- Qwell :-) )
-
-2007-12-11 22:10 +0000 [r92422] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_zap.c, configure, include/asterisk/autoconfig.h.in,
- configure.ac: Trunk build would fail due to the nonexistence of
- zaptel hwgain structures missing. Patched configure to check for
- this stuff and put a #ifdef around the offending code in
- chan_zap. Thanks to file for overseeing this.
-
-2007-12-11 21:58 +0000 [r92421] Jason Parker <jparker@digium.com>
-
- * channels/chan_sip.c: We need to set the address we want to match
- against before we actually do the match.. Closes issue #11518.
-
-2007-12-11 21:46 +0000 [r92402] Mark Michelson <mmichelson@digium.com>
-
- * res/res_musiconhold.c: Removing a pointless memset. The memory
- was just calloc'd, so the memory is already zeroed out
-
-2007-12-11 21:17 +0000 [r92401] Jason Parker <jparker@digium.com>
-
- * apps/app_controlplayback.c: Add variable to show which key was
- pressed to stop playback. Issue #11377, initial patch by johan.
-
-2007-12-11 20:06 +0000 [r92364-92365] Joshua Colp <jcolp@digium.com>
-
- * res/res_monitor.c: Only look to see if options are set if some
- have been provided. (closes issue #11505) Reported by: Mike
- Anikienko
-
- * main/global_datastores.c, /: Merged revisions 92363 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r92363 | file | 2007-12-11 15:51:40 -0400 (Tue, 11 Dec
- 2007) | 6 lines Fix potential memory leak with the dialed
- interfaces list if another memory allocation fails. (closes issue
- #11507) Reported by: eliel Patches: global_datastores.c.patch
- uploaded by eliel (license 64) ........
-
-2007-12-11 17:44 +0000 [r92324] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 92323 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r92323 | mmichelson | 2007-12-11 11:42:25 -0600 (Tue, 11 Dec
- 2007) | 10 lines Fixing autofill to be more accurate.
- Specifically, if calls ahead of the current caller were ringing
- members (but not yet bridged) there could be available members
- and waiting callers who would not get matched up. The member
- availability checker was correctly determining the number of
- available members in this scenario, but the queue itself did not
- parallelly reflect this status on the pending calls. This commit
- corrects the issue. (closes issue #11459, reported by
- equissoftware, patched by me) ........
-
-2007-12-11 16:29 +0000 [r92305] Russell Bryant <russell@digium.com>
-
- * include/asterisk/unaligned.h, main/event.c: * In unaligned.h,
- remove some unnecessary casts and mark the arg of the
- get_unaligned functions as const * In event.c, use
- get_unaligned_uint32() in a couple of places to fix issues on
- architectures that don't allow unaligned access
-
-2007-12-11 14:17 +0000 [r92267-92285] Olle Johansson <oej@edvina.net>
-
- * include/asterisk/devicestate.h, include/asterisk/agi.h,
- include/asterisk/astobj2.h, include/asterisk/extconf.h,
- include/asterisk/io.h, include/asterisk/cdr.h,
- include/asterisk/aes.h, include/asterisk/_private.h,
- include/asterisk/localtime.h, include/asterisk/hashtab.h,
- include/asterisk/callerid.h, include/asterisk/logger.h,
- include/asterisk/doxyref.h, include/asterisk/app.h,
- include/asterisk/adsi.h, include/asterisk/event.h,
- include/asterisk/causes.h, include/asterisk/alaw.h,
- include/asterisk/ast_expr.h, include/asterisk/dsp.h,
- include/asterisk/mod_format.h, include/asterisk/ael_structs.h,
- include/asterisk/astdb.h: A lot of doxygen updates
-
- * include/asterisk/frame.h: Doxygen updates
-
-2007-12-10 20:18 +0000 [r92243] Doug Bailey <dbailey@digium.com>
-
- * channels/chan_zap.c: Add CLI commands to dynamically set hw and
- sw gains
-
-2007-12-10 16:48 +0000 [r92205-92206] Joshua Colp <jcolp@digium.com>
-
- * utils/check_expr.c: Add ast_atomic_fetchadd_int_slow to
- check_expr for platforms that need it. (closes issue #11484)
- Reported by: snuffy
-
- * /, main/rtp.c: Merged revisions 92204 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r92204 | file | 2007-12-10 12:36:15 -0400 (Mon, 10 Dec 2007) | 6
- lines Add G729A as another possible payload name for G729. Some
- devices use this instead of G729, which is perfectly normal since
- the payload number itself is defined and can't be used by
- anything else so the name doesn't matter that much. (closes issue
- #11483) Reported by: revolution Patches: rtp.diff uploaded by
- revolution (license 346) ........
-
-2007-12-10 16:30 +0000 [r92203] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 92202 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r92202 | mmichelson | 2007-12-10 10:29:44 -0600 (Mon, 10 Dec
- 2007) | 7 lines If there are no members in a queue, then the loop
- where the datastore for detecting duplicate dialed numbers will
- be skipped, meaning the datastore isn't created. This means that
- when we try to free it, there's a crash. This stops that crash
- from occurring. (closes issue #11499, reported by slavon, patched
- by eliel) ........
-
-2007-12-10 16:15 +0000 [r92199-92201] Joshua Colp <jcolp@digium.com>
-
- * res/res_agi.c: Only send a SIGHUP if the pid is greater than -1,
- otherwise all PIDs greater than -1 will get the SIGHUP... and
- that is bad. (closes issue #11453) Reported by: alanmcmillan
-
-2007-12-10 14:18 +0000 [r92140-92160] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Removing some LOG_DEBUG items
-
- * /, channels/chan_sip.c: Merged revisions 92158 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r92158 | oej | 2007-12-10 15:04:44 +0100 (MÃ¥n, 10 Dec 2007) | 16
- lines Avoid reinvite race situations with two Asterisks trying to
- reinvite each other in 1.4 and trunk. This patch implements
- support for the 491 error code that Asterisk 1.4 generates on
- situations where we get an incoming INVITE and already has one in
- progress. Thanks to mavetju for reporting and to Raj Jain for an
- excellent explanation of the problem. Patch by myself. Tested
- with 8 Asterisk servers connected to each other in a training
- network. Closes issue #10481 ........
-
- * doc/manager_1_1.txt, apps/app_voicemail.c: Add a few extra
- headers in the voicemail users listing in manager 1.1. Update
- documentation too. (closes issue #11495) Reported by: caio1982
- Patches: extra_vm_manager_info1.diff uploaded by caio1982
- (license 22)
-
-2007-12-10 09:00 +0000 [r91929-92122] Luigi Rizzo <rizzo@icir.org>
-
- * build_tools/make_version, build_tools/make_version_h:
- simplify/cleanup the scripts
-
- * utils/Makefile: remove relative paths and use ASTTOPDIR instead.
- Give a default value to ASTTOPDIR if unset so we can at least do
- a 'make clean' without too much trouble. The proper fix, however,
- is to partition the top level Makefile in a 'setup' and a 'main'
- part, in a way that the 'setup' part can be included from
- subdirs' Makefiles and allow targets to be built without going
- through the top level Makefile.
-
- * utils/clicompat.c: simplify this file
-
- * doc/CODING-GUIDELINES: add a bit of info on the build
- infrastructure
-
- * Makefile: Fix the detection of modules installed from this build.
- You can now add the path of local module subdirs from the command
- line with make LOCAL_MOD_SUBDIRS= ....
-
- * codecs/Makefile, apps/Makefile, Makefile.moddir_rules,
- cdr/Makefile, pbx/Makefile, res/Makefile, channels/Makefile,
- formats/Makefile, funcs/Makefile: Put into Makefile.moddir_rules
- the common instructions used to generate loadable and embedded
- module lists. Individual Makefiles now are a lot simpler,
- possibly as simple as this: -include
- $(ASTTOPDIR)/menuselect.makeopts $(ASTTOPDIR)/menuselect.makedeps
- MODULE_PREFIX=cdr_ all: _all include
- $(ASTTOPDIR)/Makefile.moddir_rules and also more flexible because
- in a single directory we can combine various types of modules
- (app_, cdr_, func_, ... ) by simply listing them in the
- MODULE_PREFIX variable. The individual Makefiles can also create
- list of modules to be excluded by listing them in the variablel
- MODULE_EXCLUDE (see an example in channels/Makefile). With this
- change it becomes trivial to integrate a directory with locally
- created/modified sources into the main build.
-
- * Makefile, Makefile.moddir_rules: make the install target a bit
- less noisy
-
- * Makefile: document usage of several exported variables
-
- * utils/Makefile: add hashtab.c to the list of files deleted
-
- * Makefile.moddir_rules: another place where ../ should have been
- ASTTOPDIR
-
- * codecs/Makefile, utils/Makefile, apps/Makefile, cdr/Makefile,
- pbx/Makefile, res/Makefile, channels/Makefile, formats/Makefile,
- funcs/Makefile: normalize subdirs' Makefile by using ASTTOPDIR
- and not .. to reference the top level directory.
-
- * Makefile: Implement the outcome of a discussion on the -dev list
- re. the use of DESTDIR and INSTALL_PATH - many thanks to Tzafrir
- Cohen and Simon Perreault for extremely useful feedback: 1.
- comment out the [directories] section the created asterisk.conf ;
- you can set the correct defaults at build time using
- INSTALL_PATH, so the repetition here is redundant and often
- wrong. (The next step now is move asterisk.conf outside the
- Makefile to asterisk.conf.sample, because there is little if
- anything here that needs to be constructed at build/install
- time). 2. use DESTDIR?=$(INSTALL_PATH) so you only need to
- specify a path once if the two coincide. This should have no ill
- side effects, because if you don't specify DESTDIR, you really
- need INSTALL_PATH="" to set the correct defaults, and if you
- specify DESTDIR the value is not overridden. The second part
- required moving the 'export DESTDIR' right after the assignment
- to prevent DESTDIR getting set by the export (this is documented
- in the Makefile).o hopefully avoid the mistake)$ With this change
- you can now do something like this from your source tree: make
- INSTALL_PATH=/some/place install samples and then main/asterisk
- -vdc which will pick up the correct config files and libraries
- from /some/place - i.e. great for developers!
-
- * main/config.c: remove unused code, and simplify the logic for
- #include/#exec (still a lot of cleanup needed here).
-
- * main/config.c: Implement comment_buffer and lline_buffer in terms
- of the ast_str_*() API. I don't know if comment_buffers etc are
- actually used at all...
-
- * main/config.c: unify some common code
-
- * main/config.c: normalize formatting
-
- * main/config.c: document a nice technique to exit from a block in
- case of errors.
-
- * main/config.c: a little bit of documentation on how lines are
- parsed.
-
- * utils/ael_main.c: normalize header order, and add a comment on
- the need to clean up this file.
-
- * include/asterisk/network.h: some platforms (e.g. FreeBSD4) need
- netinet/in.h to be included before arpa/inet.h
-
-2007-12-07 23:32 +0000 [r91832-91891] Jason Parker <jparker@digium.com>
-
- * /, main/dsp.c: Merged revisions 91890 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #11273) ........ r91890 | qwell | 2007-12-07 17:29:01 -0600
- (Fri, 07 Dec 2007) | 4 lines We need to make sure we free the
- input frame if we return a different frame in ast_dsp_process.
- Issue 11273, pointed out by dimas, with a patch by eliel.
- ........
-
- * pbx/pbx_lua.c, configs/extensions.lua.sample: Update
- documentation for pbx_lua. Closes issue #11492, patch by
- mnicholson.
-
-2007-12-07 21:25 +0000 [r91784-91831] Russell Bryant <russell@digium.com>
-
- * /, main/utils.c: Merged revisions 91830 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r91830 | russell | 2007-12-07 15:24:33 -0600 (Fri, 07 Dec 2007) |
- 5 lines Make the lock protecting each thread's list of locks it
- currently holds recursive. I think that this will fix the
- situation where some people have said that "core show locks"
- locks up the CLI. (related to issue #11080) ........
-
- * /, include/asterisk/lock.h: Merged revisions 91828 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r91828 | russell | 2007-12-07 15:17:24 -0600 (Fri, 07
- Dec 2007) | 6 lines Fix another bug in the DEBUG_THREADS code.
- The ast_mutex_init() function had the mutex attribute object
- marked as static. This means that multiple threads initializing
- locks at the same time could step on each other and end up with
- improperly initialized locks. (found when tracking down locking
- issues related to issue #11080) ........
-
- * /, include/asterisk/lock.h: Merged revisions 91826 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r91826 | russell | 2007-12-07 15:11:08 -0600 (Fri, 07
- Dec 2007) | 6 lines I love fixing lock related errors in the lock
- debugging code. That's about as ironic as it gets in Asterisk
- programming land. Anyway, I spotted this bug while trying to
- track down why systems are locking up and acting weird in issue
- #11080. The mutex attribute object was marked as static in this
- function when it should not have been. ........
-
- * apps/app_dial.c, /: Merged revisions 91783 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r91783 | russell | 2007-12-07 10:38:48 -0600 (Fri, 07 Dec 2007) |
- 6 lines * Add channel locking around datastore operations that
- expect the channel to be locked. * Document why we don't record
- Local channels in the dialed interfaces list. * Remove the dialed
- variable as it isn't needed. * Restructure some code for clarity
- and coding guidelines stuff ........
-
-2007-12-07 16:37 +0000 [r91782] Jason Parker <jparker@digium.com>
-
- * channels/chan_sip.c: Fix a small typo in a comment. Closes issue
- #11490
-
-2007-12-07 16:28 +0000 [r91781] Russell Bryant <russell@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 91780 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r91780 | russell | 2007-12-07 10:25:25 -0600 (Fri, 07 Dec 2007) |
- 7 lines * Add channel locking around datastore operations that
- expect the channel to be locked. * Document why we don't record
- Local channels in the dialed interfaces list. * Handle memory
- allocation failure. * Remove the dialed variable, as it wasn't
- actually needed. * Tweak some formatting to conform to coding
- guidelines. ........
-
-2007-12-07 16:11 +0000 [r91779] Jason Parker <jparker@digium.com>
-
- * doc/asterisk-mib.txt, main/pbx.c, res/snmp/agent.c,
- include/asterisk/pbx.h, main/cli.c: Add count of total number of
- calls processed by asterisk during it's lifetime. Add number of
- total calls and current calls to SNMP. Closes issue #10057, patch
- by jcmoore.
-
-2007-12-07 16:11 +0000 [r91778] Russell Bryant <russell@digium.com>
-
- * main/autoservice.c, /: Merged revisions 91777 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r91777 | russell | 2007-12-07 10:08:35 -0600 (Fri, 07 Dec 2007) |
- 6 lines * Add a bit more of a verbose comment as to why a hangup
- frame needs to be queued up if autoservice gets a NULL return
- from ast_read(). * Make the process of queueing the hangup frame
- more efficient by putting the frame where it is going to end up
- and avoiding some locking and extra memory allocations and
- freeing. ........
-
-2007-12-07 15:40 +0000 [r91738] Mark Michelson <mmichelson@digium.com>
-
- * main/autoservice.c, /: Merged revisions 91737 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r91737 | mmichelson | 2007-12-07 09:39:58 -0600 (Fri, 07 Dec
- 2007) | 7 lines Hangups that happen during autoservice were not
- processed appropriately. This is because a hangup actually causes
- a NULL frame to be received, not a hangup frame. Queueing a
- hangup if we receive a NULL frame during autoservice corrects
- this problem (closes issue #11467, reported by jmls, patched by
- me) ........
-
-2007-12-07 02:52 +0000 [r91676-91700] Russell Bryant <russell@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 91693 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r91693 | russell | 2007-12-06 20:51:22 -0600 (Thu, 06 Dec 2007) |
- 2 lines Don't unlock the dialed_interfaces list until we're done
- messing with the iterator. ........
-
- * apps/app_dial.c, /, apps/app_queue.c: Merged revisions 91677 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r91677 | russell | 2007-12-06 20:38:40 -0600 (Thu, 06 Dec 2007) |
- 4 lines Allow dialing local channels from Queue() and Dial()
- again. There was a slight flaw in the code to prevent call
- forwards from looping that caused this problem. (related to issue
- #11486) ........
-
- * /, apps/app_queue.c: Merged revisions 91675 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r91675 | russell | 2007-12-06 20:19:45 -0600 (Thu, 06 Dec 2007) |
- 7 lines Fix in an issue in the call forwarding handling code that
- was causing crashes on every call into a queue. I'm not entirely
- sure about the logic in this part of the code, so I want to look
- at it some more tomorrow. However, this makes it safe and keeps
- it from crashing. (closes issue #11486, reported by adamg,
- patched by me) ........
-
-2007-12-07 00:58 +0000 [r91617-91638] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/rtp.c: Merged revisions 91637 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r91637 | tilghman | 2007-12-06 18:52:17 -0600 (Thu, 06 Dec 2007)
- | 5 lines At the end of a call, when we're reporting, RTCP may
- already be partially torn down, so check for NULL dereference
- Reported by: blitzrage Patch by: tilghman (Closes issue #11450)
- ........
-
- * channels/chan_zap.c: Add a manager event for PRI events: this
- will help manager users detect when a D-channel goes down
-
- * main/cdr.c: If duration or billsec are not yet calculated,
- calculate them on demand.
-
-2007-12-06 21:57 +0000 [r91598] Jason Parker <jparker@digium.com>
-
- * cdr/cdr_sqlite3_custom.c: Fix a problem with quoting in sqlite3
- cdr module.. Closes issue #11070, patch by seanbright.
-
-2007-12-06 21:03 +0000 [r91579] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Handle allocation failure of the heard and
- deleted arrays of the vm_state. (closes issue #11408, reported
- and patched by jaroth)
-
-2007-12-06 20:52 +0000 [r91561] Tilghman Lesher <tlesher@digium.com>
-
- * /, cdr/cdr_pgsql.c: Merged revisions 90166,90736,90753 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90166 | tilghman | 2007-11-29 13:48:10 -0600 (Thu, 29 Nov 2007)
- | 3 lines Properly escape cdr->src and cdr->dst and ensure we use
- thread-safe escaping (Fixes AST-2007-026) ........ r90736 |
- tilghman | 2007-12-03 17:23:55 -0600 (Mon, 03 Dec 2007) | 5 lines
- If both dbhost and dbsock were not set, a NULL deref could result
- Reported by: xrg Patch by: tilghman (Closes issue #11387)
- ........ r90753 | tilghman | 2007-12-03 17:50:51 -0600 (Mon, 03
- Dec 2007) | 5 lines Solaris requires the inclusion of
- sys/loadavg.h for getloadavg(). Reported by: snuffy Patch by:
- snuffy,tilghman (Closes issue #11430) ........
-
-2007-12-06 16:54 +0000 [r91472] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Make sure we clear these flags when libpri
- is not installed
-
-2007-12-06 16:51 +0000 [r91440-91458] Joshua Colp <jcolp@digium.com>
-
- * main/udptl.c, /: Merged revisions 91450 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r91450 | file | 2007-12-06 12:49:42 -0400 (Thu, 06 Dec 2007) | 6
- lines Fix various in the udptl implementation. It could return
- empty modem frames, have an incorrect sequence number on packets,
- and display the wrong sequence number in the debug messages.
- (closes issue #11228) Reported by: Cache Patches: udptl-4.patch
- uploaded by dimas (license 88) ........
-
- * /, channels/chan_sip.c: Merged revisions 91439 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r91439 | file | 2007-12-06 12:14:26 -0400 (Thu, 06 Dec 2007) | 4
- lines Add support for accepting and sending T.38 in the initial
- INVITE. (closes issue #9402) Reported by: thdei ........
-
-2007-12-06 15:56 +0000 [r91347-91438] Olle Johansson <oej@edvina.net>
-
- * doc/manager_1_1.txt (added), UPGRADE.txt: Adding documentation
- for the massive manager changes to manager version 1.1 -
- hopefully a more consistent manager interface.
-
- * main/manager.c: - The Ping Action - Now use Response: success -
- New header "Ping: pong" :-) - The Events action - Now use
- Response: Success - The new status is reported as "Events: On" or
- "Events: Off" - Report if manager is enabled in the reload event
- Small cleanups... From moremanager
-
- * main/channel.c: Changes to manager events in channel.c - Newstate
- event - Now has "CalleridNum" for numeric caller id, like
- Newchannel - The event does not send "<unknown>" for unknown
- caller IDs just an empty field - Newstate and Newchannel events -
- these have changed headers "State" -> ChannelStateDesc Text based
- channel state -> ChannelState Numeric channel state - The events
- does not send "<unknown>" for unknown caller IDs just an empty
- field - Newstate event - Now has "CalleridNum" for numeric caller
- id, like Newchannel - The event does not send "<unknown>" for
- unknown caller IDs just an empty field - Link and Unlink events -
- The "Link" and "Unlink" bridge events in channel.c are now
- renamed to "Bridge" - The link state is in the bridgestate:
- header as "Link" or "Unlink" - For channel.c bridges,
- "Bridgetype: core" is added. This opens up for bridge events in
- rtp.c and channel drivers - The "Rename" manager event has a
- renamed header, to use the same terminology for the current
- channel as other events - Oldname -> Channel (Moremanager)
-
- * main/cdr.c: New manager event when a channel changes account
- code. Maybe belongs in the new cdr category? ---moremanager---
- Event: NewAccountCode Modules: cdr.c Purpose: To report a change
- in account code for a live channel Example: Event: NewAccountCode
- Privilege: call,all Channel: SIP/olle-01844600 Uniqueid:
- 1177530895.2 AccountCode: Stinas account 1234848484
- OldAccountCode: Olles Account 12345
-
- * apps/app_dial.c: - Dial event - Event Dial has new headers, to
- comply with other events - Source -> Channel Channel name
- (caller) - SrcUniqueID -> UniqueID Uniqueid (new) -> Dialstring
- Dialstring in app data (moremanager)
-
- * apps/app_meetme.c: Adding small missing but important comma...
-
- * apps/app_meetme.c: A big oops...
-
- * apps/app_meetme.c: The MeetmeJoin now has caller ID name and
- Caller ID number fields (like MeetMeLeave) (Moremanager)
-
- * channels/chan_zap.c: Update ZapShowChannels so that you can
- specify one channel. Action ZapShowChannels Header changes -
- Channel: -> ZapChannel For active channels, the Channel: and
- Uniqueid: headers are added You can now add a "ZapChannel: "
- argument to zapshowchannels actions to only get information about
- one channel. From the moremanager branch
-
- * main/logger.c: Doxygen updates
-
- * include/asterisk/logger.h, /, main/logger.c, main/loader.c:
- Merged revisions 91366 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r91366 | oej | 2007-12-06 13:54:11 +0100 (Tor, 06 Dec 2007) | 4
- lines Make sure logger is reloaded at general reload in the cli.
- (Discovered during Asterisk training in Portugal) ........
-
- * main/manager.c: Change description of new manager command
-
- * main/manager.c, CHANGES: Add manager command for showing all
- current channels. Thanks, eliel, for writing the original patch.
- Modified by me to follow other manager events and the new
- "moremanager" style. (closes issue #11478) Reported by: eliel
- Patches: manager.c.patch uploaded by eliel (license 64)
-
-2007-12-06 04:37 +0000 [r91328] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: Instead of iterating through the entire epoll
- events array just look at the ones that will actually contain
- data. (props to eliel on IRC for this)
-
-2007-12-05 22:57 +0000 [r91291-91293] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 91292 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r91292 | mmichelson | 2007-12-05 16:57:13 -0600 (Wed, 05 Dec
- 2007) | 3 lines Reverting extra stuff I didn't mean to commit
- ........
-
- * apps/app_dial.c, /, apps/app_voicemail.c: Merged revisions 91273
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r91273 | mmichelson | 2007-12-05 16:35:52 -0600 (Wed, 05 Dec
- 2007) | 11 lines The 'G' option for Dial() did not properly
- handle the case where only a label was provided. This was due to
- the fact that the answering channel did not have an extension
- set, so ast_parseable_goto would fail. This fix eliminates the
- call to ast_parseable_goto on the answering channel since it is a
- wasteful call. The answering channel and the calling channel are
- both directed to the same extension and context, just different
- priorities, so we can just copy the values from the calling
- channel to the answering channel and increment the answering
- channel's priority. (closes issue #11382, reported by jon, patch
- by me with correction by jon) ........
-
-2007-12-05 21:46 +0000 [r91238] Tilghman Lesher <tlesher@digium.com>
-
- * /, sounds/Makefile: Merged revisions 91237 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r91237 | tilghman | 2007-12-05 15:38:13 -0600 (Wed, 05 Dec 2007)
- | 2 lines Upgrade to the latest version of extra sounds ........
-
-2007-12-05 17:49 +0000 [r91193-91197] Russell Bryant <russell@digium.com>
-
- * /, main/threadstorage.c: Merged revisions 91192 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r91192 | russell | 2007-12-05 11:31:42 -0600 (Wed, 05 Dec 2007) |
- 10 lines Make the lock in the threadstorage debugging code
- untracked to avoid a deadlock on thread destruction. (closes
- issue #11207) Reported by: ys Patches: threadstorage.c.diff
- uploaded by ys (license 281) Also fixes an open bug report:
- (closes issue #11446) ........
-
- * apps/app_directory.c: Resolve compiler warnings.
-
-2007-12-05 16:46 +0000 [r91172-91173] Tilghman Lesher <tlesher@digium.com>
-
- * main/manager.c, UPGRADE.txt, configs/manager.conf.sample,
- CHANGES, include/asterisk/manager.h, cdr/cdr_manager.c: Change
- cdr_manager to use a "CDR" level, rather than the (overcrowded)
- "call" level. (Closes issue #11015)
-
- * CHANGES, apps/app_directory.c: Added multiple name listing.
- (Closes issue #10413)
-
-2007-12-05 16:14 +0000 [r91171] Joshua Colp <jcolp@digium.com>
-
- * configs/http.conf.sample: Remove second prefix line. Only need it
- documented once in the same file. (closes issue #11472) Reported
- by: eserra Patches: http.conf.sample.diff uploaded by eserra
- (license 45)
-
-2007-12-05 13:09 +0000 [r91151-91152] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c, UPGRADE.txt, configs/sip.conf.sample: Rename
- "username" to "defaultuser" to match with "defaultip". "Username"
- still works, but is deprecated.
-
- * channels/chan_sip.c: Remove the cseqs from "sip show channel" and
- make more place for the call ID.
-
-2007-12-05 03:48 +0000 [r91133] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c: revert part of my changes from earlier today
- since this code is no longer dependent on libpri.h
-
-2007-12-05 03:34 +0000 [r91029-91131] Russell Bryant <russell@digium.com>
-
- * res/res_odbc.c: Use ast_free() instead of free(). (closes issue
- #11309) Reported by: Laureano Patches: res_odbc.c.patch uploaded
- by Laureano (license 265)
-
- * /, include/asterisk/lock.h: Merged revisions 91070 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r91070 | russell | 2007-12-04 18:35:31 -0600 (Tue, 04
- Dec 2007) | 11 lines Fix some crashes in chan_iax2 that were
- reported as happening on Mac systems. It turns out that the
- problem was the Mac version of the ast_atomic_fetchadd_int()
- function. The Mac atomic add function returns the _new_ value,
- while this function is supposed to return the old value. So, the
- crashes happened on unreferencing objects. If the reference count
- was decreased to 1, ao2_ref() thought that it had been decreased
- to zero, and called the destructor. However, there was still an
- outstanding reference around. (closes issue #11176) (closes issue
- #11289) ........
-
- * /, main/utils.c: Merged revisions 91074 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r91074 | russell | 2007-12-04 18:48:47 -0600 (Tue, 04 Dec 2007) |
- 4 lines When DEBUG_THREADS is enabled, we only have the details
- about who is holding a lock that we are waiting on for a mutex,
- not rwlocks. This should fix the problem where people have
- reported "core show locks" crashing sometimes. ........
-
- * channels/chan_zap.c: Fix mwimonitornotify on reload ... again.
- This option was only read at startup so a reload would erase it
- and not reset it. (pointed out by tzafrir)
-
- * utils/astman.c: Fix the build of astman. Any file that includes
- any asterisk sub-headers needs to first include asterisk.h.
- (closes issue #11394)
-
-2007-12-04 22:44 +0000 [r91012] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Don't error when we don't have libpri
- installed with libss7 support. Also, print the debug message
- anyway if we can't find the right PRI
-
-2007-12-04 22:07 +0000 [r91010-91011] Russell Bryant <russell@digium.com>
-
- * main/pbx.c, /: Merged revisions 90967 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90967 | russell | 2007-12-04 13:57:39 -0600 (Tue, 04 Dec 2007) |
- 7 lines Make some changes to some additions I made recently for
- doing channel autoservice when looking up extensions. This code
- was added to handle the case where a dialplan switch was in use
- that could block for a long time. However, the way that I added
- it, it did this for all extension lookups. However, lookups in
- the in-memory tree of extensions should _not_ take long enough to
- matter. So, move the autoservice stuff to be only around
- executing a switch. ........
-
- * channels/chan_zap.c: Fix resetting mwimonitornotify on reload. I
- guess I only added this line in my head. (thanks to tzafrir for
- pointing it out)
-
-2007-12-04 21:46 +0000 [r90993] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_usbradio.c: Coding guidelines fixups (Closes issue
- #11412)
-
-2007-12-04 21:23 +0000 [r90991] Jason Parker <jparker@digium.com>
-
- * channels/chan_sip.c, CHANGES: Add manager action
- 'sipshowregistry'. Closes issue #11464, patch by eliel.
-
-2007-12-04 19:08 +0000 [r90949] Russell Bryant <russell@digium.com>
-
- * include/asterisk/callerid.h, channels/chan_zap.c,
- main/callerid.c, CHANGES, configs/zapata.conf.sample: Add support
- for monitoring MWI on FXO lines. This introduces two new options
- for zapata.conf: mwimonitor and mwimonitornotify. The mwimonitor
- option enables MWI monitoring. When the MWI state on a line
- changes, then the script specified by mwimonitornotify will be
- executed for custom handling of the state change, similar to the
- externnotify option of voicemail.conf. Also, when the MWI state
- on an FXO line changes, an internal Asterisk event is generated
- to indicate the new state of the associated mailbox. That may,
- any module that cares about MWI information will get notified and
- can handle it just as if app_voicemail had sent this
- notification. (BE-253, original patch from markster, with some
- minor modifications by me to add comments, documentation, and
- internal event support)
-
-2007-12-04 18:29 +0000 [r90930] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Kevin suggested doing the reverse of my
- last commit, since imap_retrieve_file does not modify the
- contents of the "mailbox" string. In other words, I'm changing
- the imap_retrieve_file function to take a const char* as the
- third argument so that I don't need to cast const char*'s as
- char*'s to suppress compiler warnings.
-
-2007-12-04 18:15 +0000 [r90929] Jason Parker <jparker@digium.com>
-
- * Makefile: Add Makefile alias target 'pdf' which does the same
- thing as asterisk.pdf. Issue 11452, reported by blitzrage.
-
-2007-12-04 18:14 +0000 [r90928] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Suppress a compiler warning due to
- discarding a "const" qualifier
-
-2007-12-04 18:09 +0000 [r90927] Jason Parker <jparker@digium.com>
-
- * main/global_datastores.c: Fix build, that some people aren't
- seeing for some reason.
-
-2007-12-04 17:51 +0000 [r90899] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Wrong locking style got merged from 1.4 to
- trunk. My mistake.
-
-2007-12-04 17:40 +0000 [r90880] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c: fix build of this module when libpri and/or
- libss7 are or are not present
-
-2007-12-04 17:38 +0000 [r90879] Jason Parker <jparker@digium.com>
-
- * main/channel.c, /: Merged revisions 90876 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #11454) ........ r90876 | qwell | 2007-12-04 11:28:08 -0600
- (Tue, 04 Dec 2007) | 4 lines If we fail to create a channel after
- allocating a timing fd, we need to make sure to close it. Issue
- 11454, patch by eliel. ........
-
-2007-12-04 17:36 +0000 [r90878] Russell Bryant <russell@digium.com>
-
- * main/Makefile: Fix a silly little typo :)
-
-2007-12-04 17:35 +0000 [r90877] Jason Parker <jparker@digium.com>
-
- * apps/app_dial.c: Fix build in trunk. This was fixed in 1.4, but
- blocked in trunk since this hadn't been merged yet.
-
-2007-12-04 17:08 +0000 [r90873] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_dial.c, main/global_datastores.c (added),
- channels/chan_local.c, /, main/Makefile,
- include/asterisk/channel.h, include/asterisk/global_datastores.h
- (added), apps/app_queue.c: Merged revisions 90735 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03
- Dec 2007) | 22 lines A big one... This is the merge of the
- forward-loop branch. The main change here is that call-forwards
- can no longer loop. This is accomplished by creating a datastore
- on the calling channel which has a linked list of all devices
- dialed. If a forward happens, then the local channel which is
- created inherits the datastore. If, through this progression of
- forwards and datastore inheritance, a device is attempted to be
- dialed a second time, it will simply be skipped and a warning
- message will be printed to the CLI. After the dialing has been
- completed, the datastore is detached from the channel and
- destroyed. This change also introduces some side effects to the
- code which I shall enumerate here: 1. Datastore inheritance has
- been backported from trunk into 1.4 2. A large chunk of code has
- been removed from app_dial. This chunk is the section of code
- which handles the call forward case after the channel has been
- requested but before it has been called. This was removed because
- call-forwarding still works fine without it, it makes the code
- less error-prone should it need changing, and it made this set of
- changes much less painful to just have the forwarding handled in
- one place in each module. 3. Two new files, global_datastores.h
- and .c have been added. These are necessary since the datastore
- which is attached to the channel may be created and attached in
- either app_dial or app_queue, so they need a common place to find
- the datastore info. This approach was taken in case similar
- datastores are needed in the future, there will be a common place
- to add them. ........
-
-2007-12-04 15:16 +0000 [r90852-90854] Olle Johansson <oej@edvina.net>
-
- * apps/app_queue.c: (closes issue #11431) Reported by: Laureano
- Patches: app_queue.c.patch uploaded by Laureano (license 265)
-
- * main/pbx.c, CHANGES: (closes issue #11422) Reported by: eliel
- Patches: core.show.hint.patch uploaded by eliel (license 64)
-
- * CHANGES: (closes issue #11462) Reported by: eliel Patches:
- CHANGES.patch uploaded by eliel (license 64)
-
-2007-12-04 15:01 +0000 [r90851] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_agi.c: Pass the Asterisk version to AGI scripts as part
- of the initial dump of info Reported by: acunningham Patch by:
- acunningham (Closes issue #11398)
-
-2007-12-04 11:50 +0000 [r90834] Luigi Rizzo <rizzo@icir.org>
-
- * res/Makefile: fix build on cygwin
-
-2007-12-03 23:52 +0000 [r90760] Tilghman Lesher <tlesher@digium.com>
-
- * /, include/asterisk/compat.h: Merged revisions 90753 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r90753 | tilghman | 2007-12-03 17:50:51 -0600 (Mon, 03
- Dec 2007) | 5 lines Solaris requires the inclusion of
- sys/loadavg.h for getloadavg(). Reported by: snuffy Patch by:
- snuffy,tilghman (Closes issue #11430) ........
-
-2007-12-03 23:49 +0000 [r90746] Steve Murphy <murf@digium.com>
-
- * main/hashtab.c: A small fix from snuffy
-
-2007-12-03 23:48 +0000 [r90738] Jason Parker <jparker@digium.com>
-
- * res/res_monitor.c: Add manager events for when a monitor is
- started or stopped. Closes issue #10191, patch by dgradecak.
-
-2007-12-03 23:29 +0000 [r90737] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_config_pgsql.c, /: Merged revisions 90736 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r90736 | tilghman | 2007-12-03 17:23:55 -0600 (Mon, 03
- Dec 2007) | 5 lines If both dbhost and dbsock were not set, a
- NULL deref could result Reported by: xrg Patch by: tilghman
- (Closes issue #11387) ........
-
-2007-12-03 22:07 +0000 [r90697] Jason Parker <jparker@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 90696 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes
- issue #11383) ........ r90696 | qwell | 2007-12-03 16:06:36 -0600
- (Mon, 03 Dec 2007) | 4 lines Make sure we always close the
- conference fd if we have an open one. Issue 11383, reported by
- markmhy, patch by eliel. ........
-
-2007-12-03 21:24 +0000 [r90670] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Replacing some calls to free() with
- ast_free(). (closes issue #11448, reported and patched by jaroth)
-
-2007-12-03 21:03 +0000 [r90656] Joshua Colp <jcolp@digium.com>
-
- * include/asterisk/agi.h, res/res_agi.c, CHANGES: Add AGI commands
- for speech recognition. These mirror the dialplan applications
- mostly but present the information in a nicer fashion. The SPEECH
- RECOGNIZE command for example will return the results instead of
- having to query the dialplan functions.
-
-2007-12-03 21:00 +0000 [r90644] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_mgcp.c: Merged revisions 90639 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90639 | mmichelson | 2007-12-03 14:59:51 -0600 (Mon, 03 Dec
- 2007) | 5 lines Changing some bad logic when calculating the
- interdigit timeout. (closes issue #11402, reported and patched by
- eferro) ........
-
-2007-12-03 20:58 +0000 [r90631] Jason Parker <jparker@digium.com>
-
- * /, res/res_features.c: Merged revisions 90607 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes
- issue #11436) ........ r90607 | qwell | 2007-12-03 14:51:17 -0600
- (Mon, 03 Dec 2007) | 4 lines Fix crash in ParkAndAnnounce
- application. Issue #11436, reported by lytledd, patch by eliel.
- ........
-
-2007-12-03 20:30 +0000 [r90591] Tilghman Lesher <tlesher@digium.com>
-
- * main/channel.c: Avoid an additional function call. Reported by:
- eliel Patch by: eliel (Closes issue #11438)
-
-2007-12-03 20:07 +0000 [r90550-90589] Joshua Colp <jcolp@digium.com>
-
- * /, main/rtp.c: Merged revisions 90588 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90588 | file | 2007-12-03 16:05:42 -0400 (Mon, 03 Dec 2007) | 2
- lines Do not create a smoother for G723.1 frames, they need to be
- left alone to their native 20/24 byte size. ........
-
- * main/channel.c, /, include/asterisk/channel.h, .cleancount:
- Merged revisions 90548 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90548 | file | 2007-12-03 14:40:56 -0400 (Mon, 03 Dec 2007) | 2
- lines Preserve the indication currently playing on a channel when
- a masquerade operation happens. (issue #BE-88) ........
-
-2007-12-03 16:46 +0000 [r90528] Mark Michelson <mmichelson@digium.com>
-
- * configs/queues.conf.sample: Updating sample queues.conf file to
- show how multiple periodic announcements may be specified since
- this was not documented previously (closes issue #11432, reported
- and patched by Laureano)
-
-2007-12-03 14:14 +0000 [r90508] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c: Remove the file descriptors from the main poll
- channel when the channel is hung up during the dialing attempt,
- and make sure a channel exists before trying to remove it at the
- end. (closes issue #11441) Reported by: blitzrage
-
-2007-12-02 18:20 +0000 [r90471] Russell Bryant <russell@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 90470 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90470 | russell | 2007-12-02 12:18:52 -0600 (Sun, 02 Dec 2007) |
- 6 lines The other day when I went through making changes as a
- result of the ao2_link() change, I added some code to set
- pointers to NULL after they were unreferenced. This pointed out
- that in this place, the object was unreferenced before the code
- was done using it. So, move the unref down a little bit. (crash
- reported by jmls on IRC) ........
-
-2007-12-02 09:42 +0000 [r90433] Tilghman Lesher <tlesher@digium.com>
-
- * main/autoservice.c, /: Merged revisions 90432 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90432 | tilghman | 2007-12-02 03:34:23 -0600 (Sun, 02 Dec 2007)
- | 7 lines Clarify the return value on autoservice. Specifically,
- if you started autoservice and autoservice was already on, it
- would erroneously return an error. Reported by: adiemus Patch by:
- dimas (Closes issue #11433) ........
-
-2007-12-01 01:37 +0000 [r90410] Jason Parker <jparker@digium.com>
-
- * res/res_adsi.c: Only reload if the config file has changed.
- Closes issue #11281, patch by eliel.
-
-2007-11-30 21:19 +0000 [r90388] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_dial.c, include/asterisk/app.h,
- include/asterisk/audiohook.h, res/res_features.c,
- include/asterisk/channel.h, main/audiohook.c, apps/app_queue.c,
- configs/features.conf.sample: Adding support for the
- "automixmonitor" dial and queue options. This works in much the
- same way as the automonitor, except that instead of using the
- monitor app, it uses the mixmonitor app. By providing an 'x' or
- 'X' as a dial or queue option, a DTMF sequence may be entered (as
- defined in features.conf) to start the one-touch mixmonitor. This
- patch also introduces some new API calls to the audiohooks code
- for searching for an audiohook by type and for searching for a
- running audiohook by type. Big thanks to joetester for writing
- the initial patch, testing it and patiently waiting for it to be
- committed. (closes issue #10185, reported and patched by
- xmarksthespot)
-
-2007-11-30 19:34 +0000 [r90311-90351] Russell Bryant <russell@digium.com>
-
- * main/manager.c, /, include/asterisk/astobj2.h, apps/app_queue.c,
- channels/chan_iax2.c, main/astobj2.c, main/config.c: Merged
- revisions 90348 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90348 | russell | 2007-11-30 13:26:04 -0600 (Fri, 30 Nov 2007) |
- 8 lines Change the behavior of ao2_link(). Previously, in
- inherited a reference. Now, it automatically increases the
- reference count to reflect the reference that is now held by the
- container. This was done to be more consistent with ao2_unlink(),
- which automatically releases the reference held by the container.
- It also makes it so it is no longer possible for a pointer to be
- invalid after ao2_link() returns. ........
-
- * /, include/asterisk/astobj2.h: Merged revisions 90310 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90310 | russell | 2007-11-30 12:46:46 -0600 (Fri, 30 Nov 2007) |
- 2 lines Add some notes on the behavior of ao2_unlink() after a
- discussion with Tilghman ........
-
-2007-11-30 14:45 +0000 [r90270] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 90269 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90269 | file | 2007-11-30 10:43:15 -0400 (Fri, 30 Nov 2007) | 6
- lines Fix locking issues under one legged replaces scenarios.
- (closes issue #11420) Reported by: irroot Patches:
- chan_sip_oneleg.patch uploaded by irroot (license 52) ........
-
-2007-11-30 00:16 +0000 [r90164-90232] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_mgcp.c: Merged revisions 90231 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90231 | mmichelson | 2007-11-29 18:16:04 -0600 (Thu, 29 Nov
- 2007) | 5 lines Clear the DTMF buffer if the call times out.
- (closes issue #11418, reported and patched by eferro) ........
-
- * /, apps/app_queue.c: Merged revisions 90163 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90163 | mmichelson | 2007-11-29 13:38:39 -0600 (Thu, 29 Nov
- 2007) | 6 lines This patch handles the case where a queue member
- with a negative penalty is added via the manager. If a negative
- value is submitted for a member penalty, we set it to 0. (closes
- issue #11411, reported and patched by Laureano) ........
-
-2007-11-29 19:35 +0000 [r90156-90162] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_config_pgsql.c, /: Merged revisions 90160 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r90160 | tilghman | 2007-11-29 13:24:11 -0600 (Thu, 29
- Nov 2007) | 2 lines Properly escape input buffers (Fixes
- AST-2007-025) ........
-
- * /, formats/format_wav.c, formats/format_pcm.c,
- formats/format_ogg_vorbis.c, main/file.c,
- include/asterisk/mod_format.h, formats/format_h263.c,
- formats/format_h264.c, formats/format_wav_gsm.c,
- formats/format_g726.c: Merged revisions 90155 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90155 | tilghman | 2007-11-29 11:29:59 -0600 (Thu, 29 Nov 2007)
- | 5 lines Use of "private" as a field name in a header file
- messes with C++ projects Reported by: chewbacca Patch by: casper
- (Closes issue #11401) ........
-
- * include/asterisk/lock.h: Fix build of trunk
-
- * /, sounds/Makefile: Merged revisions 90154 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90154 | tilghman | 2007-11-29 11:18:09 -0600 (Thu, 29 Nov 2007)
- | 2 lines Upgrade the core sounds release version ........
-
-2007-11-29 13:38 +0000 [r90149-90150] Kevin P. Fleming <kpfleming@digium.com>
-
- * utils/Makefile, utils/hashtest.c: let's try this again... *all*
- compilation and linking in Asterisk should be done using the
- standard compilation rules, not manually created ones. changing
- hashtest.c to use these rules caused the compiler to notice a
- large number of coding guidelines violations, so those are fixed
- too.
-
- * main/manager.c: restore behavior from the 1.4 branch... manager
- users created via users.conf should default to *all* permissions,
- not none
-
-2007-11-29 00:37 +0000 [r90139-90148] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /, include/asterisk/channel.h,
- funcs/func_callerid.c: Merged revisions 90145 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90145 | russell | 2007-11-28 18:20:34 -0600 (Wed, 28 Nov 2007) |
- 5 lines This set of changes is to make some callerID handling
- thread-safe. The ast_set_callerid() function needed to lock the
- channel. Also, the handlers for the CALLERID() dialplan function
- needed to lock the channel when reading or writing callerid
- values directly on the channel structure. ........
-
- * include/asterisk/file.h, /, main/file.c: Merged revisions 90142
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90142 | russell | 2007-11-28 18:06:08 -0600 (Wed, 28 Nov 2007) |
- 4 lines Merge a change from team/russell/chan_refcount ... This
- makes ast_stopstream() thread-safe. ........
-
- * include/asterisk/audiohook.h: Merge another small doxygen change
- from team/russell/chan_refcount to indicate that a channel
- doesn't need to be locked before calling a certain function.
-
- * include/asterisk/channel.h: Merge some channel.h doxygen updates
- from team/russell/chan_refcount This was mostly to note whether a
- channel needed to be locked or not before calling these
- functions. However, I added some other things, too.
-
-2007-11-28 23:03 +0000 [r90102] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_musiconhold.c, apps/app_queue.c: Merged revisions
- 90101 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90101 | file | 2007-11-28 18:59:28 -0400 (Wed, 28 Nov 2007) | 6
- lines Fix a few memory leaks. (closes issue #11405) Reported by:
- eliel Patches: load_realtime.patch uploaded by eliel (license 64)
- ........
-
-2007-11-28 22:44 +0000 [r90100] Kevin P. Fleming <kpfleming@digium.com>
-
- * configs/users.conf.sample, main/manager.c, /: Merged revisions
- 90098 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90098 | kpfleming | 2007-11-28 16:30:46 -0600 (Wed, 28 Nov 2007)
- | 2 lines it is impossible to set permissions for manager
- accounts created by users.conf (reported internally, patched by
- me) ........
-
-2007-11-28 22:32 +0000 [r90099] Joshua Colp <jcolp@digium.com>
-
- * main/cli.c: file says... compile before you commit!
-
-2007-11-28 22:17 +0000 [r90060-90061] Mark Michelson <mmichelson@digium.com>
-
- * main/pbx.c: Removing a pointless check of option_debug
-
- * main/pbx.c, /: Merged revisions 90059 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90059 | mmichelson | 2007-11-28 16:08:50 -0600 (Wed, 28 Nov
- 2007) | 13 lines Removing some seemingly pointless code. This
- sets a channel variable for every priority executed in the
- dialplan if you have debug set to anything non-zero. This seems
- pointless due to the fact that these channel variables are not
- referenced anywhere else in the code and their names are esoteric
- enough that they would not be practical to reference in the
- dialplan. Plus the fact that this behavior isn't documented
- anywhere means that the change is not likely to cause any
- disruption. If anything, this may actually cause a slight
- performance increase if running with debug on. The motivating
- influence for this code change is the eventwhencalled option for
- queues. If set to vars, all channel variables will be output to
- the manager. These unnecessary channel variables make the output
- a lot more difficult to deal with. ........
-
-2007-11-28 20:33 +0000 [r90039] Steve Murphy <murf@digium.com>
-
- * main/ast_expr2f.c, main/ast_expr2.fl: Made expr2 parser 8-bit
- transparent
-
-2007-11-28 20:27 +0000 [r90038] Jason Parker <jparker@digium.com>
-
- * main/pbx.c, res/res_crypto.c, include/asterisk/cli.h, main/cli.c:
- Remove "old"-style CLI handler, since nothing uses it anymore.
- Closes issue #11403, patch by eliel. This also completes the
- janitor project.
-
-2007-11-28 15:48 +0000 [r89981-89982] Joshua Colp <jcolp@digium.com>
-
- * main/cli.c: Hide CLI commands starting with _ from tab completion
- as was done previously. (closes issue #11395) Reported by: eliel
- Patches: cli.c.patch uploaded by eliel (license 64)
-
- * main/abstract_jb.c, res/res_agi.c: Fix a few log messages.
- (closes issue #11396) Reported by: IgorG Patches: spell.v1.diff
- uploaded by IgorG (license 20)
-
-2007-11-28 00:49 +0000 [r89947] Russell Bryant <russell@digium.com>
-
- * apps/app_voicemail.c: Merge some little changes from
- team/russell/chan_refcount to help reduce the diff to trunk. This
- just removes some checks on the return value of alloca(), as
- behavior is undefined if it runs out of stack space, and we don't
- check it anywhere else.
-
-2007-11-28 00:47 +0000 [r89946] Mark Michelson <mmichelson@digium.com>
-
- * configs/musiconhold.conf.sample, configs/extconfig.conf.sample,
- res/res_musiconhold.c, CHANGES: Adding support for realtime music
- on hold. The following are the main points: 1. When moh is
- started, we search first in memory to find the class. If we do
- not find it in memory, we search realtime instead. 2. When moh is
- restarted (as in, it had been started on this particular channel,
- stopped, and now we're starting it again), if using the "files"
- mode, then realtime will always be rechecked. If you are using
- other modes, however, we will simply reattach to the external
- running process which was playing moh earlier in the call. This
- is a necessary compromise so that we don't end up with too many
- background processes. 3. musiconhold.conf has a general section
- now. It has one option: cachertclasses. If set to yes, then moh
- classes found in realtime will be added to the in-memory list.
- This has the advantage of not requiring database lookups each
- time moh is started, but it has the disadvantage of not truly
- being realtime. I have tested this for functionality, and it
- passes. I also tested this under valgrind and there are no memory
- problems reported under typical use. Special thanks to Sergee for
- implementing this feature and enduring my complaints on the
- bugtracker! (closes issue #11196, reported and patched by sergee)
-
-2007-11-28 00:24 +0000 [r89840-89915] Russell Bryant <russell@digium.com>
-
- * main/pbx.c, /, include/asterisk/pbx.h: Merged revisions 89893 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89893 | russell | 2007-11-27 18:20:13 -0600 (Tue, 27 Nov 2007) |
- 4 lines - update documentation for some of the goto functions to
- note that they handle locking the channel as needed - update
- ast_explicit_goto() to lock the channel as needed ........
-
- * include/asterisk/channel.h: Document that the channel is not
- locked when the send_digit_begin and end callbacks get called.
-
- * main/autoservice.c, /: Merged revisions 89886 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89886 | russell | 2007-11-27 17:47:28 -0600 (Tue, 27 Nov 2007) |
- 2 lines Don't do frame processing if ast_read() returned NULL.
- ........
-
- * channels/chan_iax2.c: Merge changes from
- team/russell/iax2_frame_queue This patch is an optimization for
- chan_iax2. This module is now heavily multi-threaded. However,
- there is still a good number of globally shared resources that
- prevent things from happen asynchronously. One of those things
- was the global IAX frame queue. This queue was used to hold
- frames that have been deferred for transmitting by another
- thread, and frames that may need to get retransmitted. I changed
- the frame queue to be per-call, since almost all of the frame
- queue handling only cares about frames specific to a call number.
-
- * /, apps/app_queue.c: Merged revisions 89844 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89844 | russell | 2007-11-27 17:21:13 -0600 (Tue, 27 Nov 2007) |
- 3 lines Instead of depending on the return value of ast_true(),
- explicitly set the eventwhencalled variable to 1. ........
-
- * main/pbx.c, /: Merged revisions 89839 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89839 | russell | 2007-11-27 17:16:00 -0600 (Tue, 27 Nov 2007) |
- 2 lines Don't start/stop autoservice in pbx_extension_helper()
- unless a channel exists ........
-
-2007-11-27 23:11 +0000 [r89838] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 89837 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89837 | mmichelson | 2007-11-27 17:10:05 -0600 (Tue, 27 Nov
- 2007) | 12 lines Two changes with regards to the
- 'eventwhencalled' option of queues.conf 1) Due to some signed vs.
- unsigned silliness, setting 'eventwhencalled' to 'vars' or 'yes'
- did exactly the same thing. Thus the sign change of the ast_true
- call. 2) The vars2manager function overwrote a \n for every
- channel variable it parsed, resulting in bizarre output for the
- channel variables. This patch remedies this. (related to issue
- #11385, however I'm not sure if this will actually be enough to
- close it) ........
-
-2007-11-27 22:42 +0000 [r89835] Russell Bryant <russell@digium.com>
-
- * channels/chan_misdn.c: Bring in a small change from
- team/russell/chan_refcount This replaces tab completion code with
- the use of a public function that does the same thing
-
-2007-11-27 22:14 +0000 [r89792] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, pbx/pbx_config.c: closes issue #11294; missed the
- conditional unlock of the contexts when the hash table is used
- instead; also, used the ast_free_ptr as advised.
-
-2007-11-27 22:05 +0000 [r89791] Russell Bryant <russell@digium.com>
-
- * main/autoservice.c, main/pbx.c, /: Merged revisions 89790 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89790 | russell | 2007-11-27 15:45:51 -0600 (Tue, 27 Nov 2007) |
- 41 lines Merge changes from team/russell/autoservice_1.4 This set
- of changes fixes an issue that was reported to me on IRC
- yesterday. The user, d1mas, was using chan_zap for incoming calls
- and was having DTMF recognition issues in some situations.
- Specifically, he noticed that the problem occurred when using
- DISA or WaitExten. He also noticed that when using Read, the
- problem did not occur. His system also used DUNDi for dialplan
- lookups. So, he theorized that if the DUNDi lookups blocked for
- some period of time, that audio from the zap channel could get
- lost. If the audio got lost, then it wouldn't be run through the
- DTMF detector, and digits could get lost. He was correct, and the
- following set of changes fixes the problem. However, the changes
- go a little bit further than what was necessary to fix this exact
- problem. 1) I updated pbx_extension_helper() to autoservice the
- associated channel to handle cases where extension lookups may
- take a long time. This would normally be a dialplan switch that
- does some lookup over the network, such as the DUNDi or IAX2
- switches. This ensures that even while a DUNDi lookup is
- blocking, the channel will be continuously serviced. 2) I made a
- change to the autoservice code. This is actually something that
- has bothered me for a long time. When a channel is in
- autoservice, _all_ frames get thrown away. However, some frames
- really shouldn't be thrown away. The most notable examples are
- signalling (CONTROL) frames, and DTMF. So, this patch queues up
- important frames while a channel is in autoservice. When
- autoservice is stopped on the channel, the queued up frames get
- stuck back on the channel so that they can get processed instead
- of thrown away. 3) I made another change to the autoservice code
- to handle the case where autoservice is started on channels
- recursively. Previously, you could call ast_autoservice_start()
- multiple times on a channel, and it would stop the first time
- ast_autoservice_stop() gets called. Now, it will ensure that
- autoservice doesn't actually stop until the final call to
- ast_autoservice_stop(). ........
-
-2007-11-27 21:10 +0000 [r89769-89772] Olle Johansson <oej@edvina.net>
-
- * main/dnsmgr.c, res/res_jabber.c, main/enum.c, main/asterisk.c: A
- few more "moremanager" fixes
-
- * include/asterisk.h, main/asterisk.c, main/loader.c: More
- "moremanager" fixes. Manager commands to check module status.
-
- * include/asterisk/manager.h: More "moremanager" changes - doxygen
- docs and changing manager version (finally) before making more
- dramatic changes.
-
- * channels/chan_iax2.c: More additions from the "moremanager"
- branch, this time for IAX2.
-
-2007-11-27 20:21 +0000 [r89721] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, main/app.c: Merged revisions 89709 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89709 | kpfleming | 2007-11-27 14:16:56 -0600 (Tue, 27 Nov 2007)
- | 2 lines on second thought... revert all the other changes i've
- made in app options parsing leaving only one: if an empty
- argument is supplied for an option, set that argument pointer to
- point to an empty string rather than NULL, so that the
- application can do normal checks on it without worrying about it
- being NULL ........
-
-2007-11-27 20:17 +0000 [r89710] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: remove a duplicate manager event
-
-2007-11-27 19:50 +0000 [r89706] Olle Johansson <oej@edvina.net>
-
- * channels/chan_gtalk.c: Manager events from the "moremanager"
- branch
-
-2007-11-27 19:47 +0000 [r89704] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, main/app.c: Merged revisions 89701 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89701 | kpfleming | 2007-11-27 13:36:55 -0600 (Tue, 27 Nov 2007)
- | 2 lines generate a warning when an application option that
- requires an argument is ignored due to lack of an argument
- ........
-
-2007-11-27 19:45 +0000 [r89698-89702] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Starting to merge changes from the
- "moremanager" branch. Documentation will follow.
-
- * /, channels/chan_sip.c, include/asterisk/rtp.h, main/rtp.c: The
- following patch with updates for trunk. Works much better in
- trunk. Also by accident fixed a bad typo by a previous committer,
- which actually made video calls not work fully... Merged
- revisions 89630 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89630 | oej | 2007-11-27 16:23:17 +0100 (Tis, 27 Nov 2007) | 12
- lines If we get a codec offer using a well-known payload type,
- but using it for another codec that we don't know, Asterisk did
- not remove that codec from the list. With this patch, we remove
- the codec from audio and video rtp objects and deny it ever
- existed. Thanks to lasse for testing. (closes issue #11376)
- Reported by: lasse Patches: bug11376.txt uploaded by oej (license
- 306) Tested by: lasse ........
-
-2007-11-27 19:12 +0000 [r89683] Jason Parker <jparker@digium.com>
-
- * include/asterisk/strings.h: Add an S_COR macro, which is similar
- to the existing S_OR macro, except with an additional boolean
- arg. A hack such as: foo ? S_OR(bar, "baz") : "baz" becomes:
- S_COR(foo, bar, "baz")
-
-2007-11-27 18:50 +0000 [r89682] Steve Murphy <murf@digium.com>
-
- * res/ael/ael.y, pbx/ael/ael-test/ref.ael-test11,
- pbx/ael/ael-test/ref.ael-test20, pbx/ael/ael-test/ref.ael-test14,
- pbx/ael/ael-test/ref.ael-test15, pbx/ael/ael-test/ref.ael-ntest9,
- pbx/ael/ael-test/ref.ael-test16, pbx/ael/ael-test/ref.ael-test18,
- pbx/ael/ael-test/ref.ael-test19,
- pbx/ael/ael-test/ref.ael-ntest10, res/ael/ael.tab.c,
- pbx/ael/ael-test/ref.ael-test1, pbx/ael/ael-test/ref.ael-ntest12,
- pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-ntest22,
- res/ael/ael_lex.c, pbx/ael/ael-test/ref.ael-test3,
- pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test5,
- pbx/ael/ael-test/ref.ael-test6, res/ael/ael.flex,
- pbx/ael/ael-test/ref.ael-test7, pbx/ael/ael-test/ref.ael-test8:
- made AEL 8-bit transparent; mainly the lexer was tossing chars
- with the hi-order bit set. Not nice. Also, allow @ in extension
- names, and a backslash, also.
-
-2007-11-27 17:01 +0000 [r89637] Joshua Colp <jcolp@digium.com>
-
- * main/utils.c: Ensure the value returned from ast_random is
- between 0 and RAND_MAX on 64-bit platforms. (closes issue #11348)
- Reported by: sperreault
-
-2007-11-27 16:13 +0000 [r89635] Russell Bryant <russell@digium.com>
-
- * /, configs/voicemail.conf.sample: Merged revisions 89634 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89634 | russell | 2007-11-27 10:12:33 -0600 (Tue, 27 Nov 2007) |
- 3 lines Add a note to the sample voicemail config noting that
- when using IMAP storage, only the first format specified will be
- attached to the message. ........
-
-2007-11-27 15:41 +0000 [r89632] Tilghman Lesher <tlesher@digium.com>
-
- * /, funcs/func_env.c: Merged revisions 89631 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89631 | tilghman | 2007-11-27 09:38:03 -0600 (Tue, 27 Nov 2007)
- | 3 lines Default result of STAT should be "0" not "". Reported
- via the -users mailing list, fixed by me. ........
-
-2007-11-27 07:36 +0000 [r89625] Olle Johansson <oej@edvina.net>
-
- * /, configs/sip.conf.sample: Merged revisions 89624 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r89624 | oej | 2007-11-27 08:34:19 +0100 (Tis, 27 Nov
- 2007) | 6 lines Clarify limitonpeers=yes (closes issue #11304)
- Reported by: pj ........
-
-2007-11-27 06:47 +0000 [r89623] Steve Murphy <murf@digium.com>
-
- * apps/app_dial.c, main/cdr.c, /, configs/cdr.conf.sample,
- include/asterisk/cdr.h: Merged revisions 89622 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89622 | murf | 2007-11-26 23:24:02 -0700 (Mon, 26 Nov 2007) | 1
- line closes issue #11379; OK, this is an attempt to make both
- sides happy. To the cdr.conf file, I added the option
- 'unanswered', which defaults to 'no'. In this mode, you will see
- a cdr for a call, whether it was answered or not. The disposition
- will be NO ANSWER or ANSWERED, as appropriate. The src is as
- you'd expect, the destination channel will be one of the channels
- from the Dial() call, usually the last in the list if more than
- one chan was specified. With unanswered set to 'yes', you will
- still see this cdr entry in both cases. But in the case where the
- dial timed out, you will also see a cdr for each line attempted,
- marked NO ANSWER, with no destination channel name. The new
- option defaults to 'no', so you don't see the pesky extra cdr's
- by default, and you will not see the irritating 'not posted'
- messages. ........
-
-2007-11-26 23:15 +0000 [r89617-89621] Mark Michelson <mmichelson@digium.com>
-
- * pbx/ael/ael-test/ael-test19/extensions.ael,
- pbx/ael/ael-test/ael-vtest13/extensions.ael, doc/osp.txt,
- pbx/ael/ael-test/ael-test3/extensions.ael,
- pbx/ael/ael-test/ref.ael-vtest13,
- pbx/ael/ael-test/ael-test7/extensions.ael: Change all instances
- of "CALLERID(number)" to "CALLERID(num)" for consistency's sake
- (closes issue #11381, reported and patched by jon)
-
- * /, apps/app_playback.c: Merged revisions 89618 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89618 | mmichelson | 2007-11-26 17:10:49 -0600 (Mon, 26 Nov
- 2007) | 7 lines After issuing a "say load new", if a caller hangs
- up during the middle of playback of a number, app_playback will
- continue to try to play the remaining files. With this change, no
- more files will be played back upon hangup. (closes issue #11345,
- reported and patched by IgorG) ........
-
-2007-11-26 22:52 +0000 [r89615] Russell Bryant <russell@digium.com>
-
- * configure, configure.ac: Update the configure script check for
- libpri to check for the newest function that was just added.
- Cresl1n, please keep this in mind when making these changes to
- libpri or libss7.
-
-2007-11-26 21:23 +0000 [r89613] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c, configs/sip.conf.sample: Rename
- "limitonpeer" to "counteronpeer" since the call-limit is
- deprecated. Both still works in this version.
-
-2007-11-26 21:14 +0000 [r89612] Joshua Colp <jcolp@digium.com>
-
- * main/dial.c, /: Merged revisions 89610 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89610 | file | 2007-11-26 17:10:29 -0400 (Mon, 26 Nov 2007) | 2
- lines Fix issues with async dialing with an application
- executing. The application has to be terminated and control
- returned to the thread before hanging things up. (issue #BE-252)
- ........
-
-2007-11-26 21:12 +0000 [r89606-89611] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Formatting, doxygenification
-
- * channels/chan_sip.c: Formatting changes, cleaning up some code
-
- * include/asterisk/doxyref.h, channels/chan_sip.c: Start using
- Doxygen groupings to group variables and defines.
-
- * apps/app_meetme.c, UPGRADE.txt, CHANGES, main/cli.c: - Mark
- "concise" as deprecated - Restructure other changes to
- UPGRADE.txt and CHANGES We're still looking for scripts that
- replace asterisk -rx "show shannels concise" by using the manager
- interface, but still produces the same output. Anyone?
-
-2007-11-26 18:11 +0000 [r89600-89602] Joshua Colp <jcolp@digium.com>
-
- * res/res_features.c, apps/app_queue.c: Perform some module use
- counting audits. This is now done outside the scope of the
- application/dialplan function so they do not need to worry about
- it.
-
- * /, res/res_features.c: Merged revisions 89599 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89599 | file | 2007-11-26 14:02:56 -0400 (Mon, 26 Nov 2007) | 6
- lines Add module counting removal for error conditions. (closes
- issue #11333) Reported by: Laureano Patches:
- res_features_v2.c.patch uploaded by Laureano (license 265)
- ........
-
-2007-11-26 17:49 +0000 [r89596] Russell Bryant <russell@digium.com>
-
- * main/pbx.c, /: Merged revisions 89594 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89594 | russell | 2007-11-26 11:41:04 -0600 (Mon, 26 Nov 2007) |
- 3 lines Add channel locking to a function that needed to be doing
- it. This is just a little something I noticed while working on a
- completely unrelated issue. ........
-
-2007-11-26 17:46 +0000 [r89595] Steve Murphy <murf@digium.com>
-
- * utils/ael_main.c, utils/conf2ael.c, utils/check_expr.c: closes
- issue #11341; made changes to make utils again right with the
- MTX_PROFILE world.
-
-2007-11-26 17:38 +0000 [r89593] Joshua Colp <jcolp@digium.com>
-
- * /, pbx/pbx_config.c: Merged revisions 89592 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89592 | file | 2007-11-26 13:36:45 -0400 (Mon, 26 Nov 2007) | 6
- lines Use ast_free to free memory, or else we shall implode if
- MALLOC_DEBUG is enabled. (closes issue #11347) Reported by: ys
- Patches: pbx.pbx_config.c.diff uploaded by ys (license 281)
- ........
-
-2007-11-26 17:26 +0000 [r89591] Steve Murphy <murf@digium.com>
-
- * main/hashtab.c: closes issue #11356; Many thanks to snuffy for
- his code review and changes to cut down duplication. I tested
- this against hashtest, and it passes. I reviewed the changes, and
- they look reasonable. I had to remove a few const decls to make
- things compile on my workstation,
-
-2007-11-26 17:25 +0000 [r89590] Russell Bryant <russell@digium.com>
-
- * Makefile: make sure we check to see if the configure script has
- been executed on a new checkout or after a distclean
-
-2007-11-26 17:23 +0000 [r89589] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_mixmonitor.c: Merged revisions 89587 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r89587 | file | 2007-11-26 13:20:58 -0400 (Mon, 26 Nov
- 2007) | 6 lines Close the audio file before sending it to the
- post processing application. (closes issue #11357) Reported by:
- reformed Patches: mixmonitor.patch uploaded by reformed (license
- 330) ........
-
-2007-11-26 17:21 +0000 [r89588] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, main/app.c, apps/app_controlplayback.c: Merged revisions 89586
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89586 | kpfleming | 2007-11-26 11:20:36 -0600 (Mon, 26 Nov 2007)
- | 2 lines when parsing application options that take arguments,
- don't indicate that the option was supplied unless a
- non-zero-length argument was found for it ........
-
-2007-11-26 16:24 +0000 [r89583] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, CHANGES, configs/extensions.conf.sample: Thanks to
- pnlarsson for noting the spelling error in the cli commands.
- Also, added some verbage about the new algorithm to CHANGES.
-
-2007-11-26 16:20 +0000 [r89582] Joshua Colp <jcolp@digium.com>
-
- * main/utils.c: Revert change for 11348 until it can be looked at
- even more.
-
-2007-11-26 15:50 +0000 [r89581] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 89580 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89580 | mmichelson | 2007-11-26 09:48:06 -0600 (Mon, 26 Nov
- 2007) | 6 lines Revert vmu->email back to an empty string if it
- was empty when imap_store_file was called. This prevents sending
- a duplicate e-mail. (closes issue #11204, reported by spditner,
- patched by me) ........
-
-2007-11-26 15:36 +0000 [r89570-89578] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /: Merged revisions 89577 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89577 | file | 2007-11-26 11:34:38 -0400 (Mon, 26 Nov 2007) | 6
- lines If channel allocation fails because the alert pipe could
- not be created also free the scheduler context. (closes issue
- #11355) Reported by: eliel Patches: main.channel.c.patch uploaded
- by eliel (license 64) ........
-
- * main/utils.c: Make the behavior of using /dev/urandom for random
- numbers the same as random(). (closes issue #11348) Reported by:
- sperreault Patches: ast_random2.diff uploaded by sperreault
- (license 252)
-
- * channels/chan_sip.c: Instead of printing out one codec in sip
- show channels print out all of the native ones (this is for
- video). (closes issue #11366) Reported by: ovi
-
- * /, apps/app_meetme.c: Merged revisions 89571 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89571 | file | 2007-11-26 10:41:03 -0400 (Mon, 26 Nov 2007) | 4
- lines When unloading app_meetme destroy any auto created contexts
- created by SLA. (closes issue #11367) Reported by: eliel ........
-
- * apps/app_controlplayback.c: Don't crash if the 'o' option of
- ControlPlayback is used without any value. (closes issue #11375)
- Reported by: johan
-
-2007-11-25 21:12 +0000 [r89564-89566] Olle Johansson <oej@edvina.net>
-
- * channels/chan_usbradio.c: Formatting changes
-
- * main/channel.c, include/asterisk/channel.h: Try to get channel.h
- and channel.c aligned in regards to ast_set_callerid as well as
- change name of variables to follow the rest of the naming.
-
-2007-11-25 17:50 +0000 [r89560-89561] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/res_odbc.h, res/res_config_odbc.c, /,
- res/res_odbc.c, configs/res_odbc.conf.sample: Merged revisions
- 89559 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89559 | tilghman | 2007-11-25 11:17:10 -0600 (Sun, 25 Nov 2007)
- | 14 lines We previously attempted to use the ESCAPE clause to
- set the escape delimiter to a backslash. Unfortunately, this does
- not universally work on all databases, since on databases which
- natively use the backslash as a delimiter, the backslash itself
- needs to be delimited, but on other databases that have no
- delimiter, backslashing the backslash causes an error. So the
- only solution that I can come up with is to create an option in
- res_odbc that explicitly specifies whether or not backslash is a
- native delimiter. If it is, we use it natively; if not, we use
- the ESCAPE clause to make it one. Reported by: elguero Patch by:
- tilghman (Closes issue #11364) ........
-
- * channels/chan_sip.c: Typo (someone needs to test compile before
- committing his changes)
-
-2007-11-25 12:18 +0000 [r89551-89557] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: More doxygen changes
-
- * channels/chan_sip.c: Housekeeping
-
- * channels/chan_sip.c: Formatting, doxygen updates
-
- * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: -
- Deprecate "call-limit" in chan_sip. No other channel driver
- enforces call-limits and we now have the groupcount system to
- implement call-limits in the dialplan. You can use the "setvar"
- option in realtime/sip.conf to set limits per device. - Implement
- "callcounter" as a new option to enable the call counting we need
- to report device status to queue, manager and SIP subscriptions.
- The call counter setting is now enabled in the code by setting
- the device call-limit to 999. When we remove the call limit, we
- can simply enable this with a boolean setting.
-
- * channels/chan_sip.c, include/asterisk/channel.h: Housekeeping...
- - Fix typo in chan_sip - Remove changes to caller ID structure,
- moving it to branch (russellb)
-
-2007-11-24 21:00 +0000 [r89547] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, include/asterisk/pbx.h, pbx/pbx_config.c,
- configs/extensions.conf.sample: closes issue #11363; where the
- pattern _20x. buried in an included context, didn't match 2012;
- There were a small set of problems to fix: 1. I needed NOT to
- score patterns unless you are at the end of the data string. 2.
- Capital N,X,Z and small n,x,z are OK in patterns. I canonicalize
- the patterns in the trie to caps. 3. When a pattern ends with dot
- or exclamation, CANMATCH/MATCHMORE should always report this
- pattern, no matter the length. With this commit, I also supplied
- the wish of Luigi, where the user can select which pattern
- matching algorithm to use, the old (legacy) pattern matcher, or
- the new, trie based matcher. The OLD matcher is the default. A
- new [general] section variable, extenpatternmatchnew, is added to
- the extensions.conf, and the example config has it set to false.
- If true, the new matcher is used. In all other respects, the
- context/exten structs are the same; the tries and hashtabs are
- formed, but in the new mode the tries are not used. A new CLI
- command 'dialplan set extenpatternmatch true/false' is provided
- to allow switching at run time. I beg users that are forced to
- return to the old matcher to please report the reason in the bug
- tracker. Measured the speed benefit of the new matcher against an
- impossibly large context with 10,000 extensions: the new matcher
- is 374 times faster.
-
-2007-11-24 17:07 +0000 [r89546] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_adsi.c: Merged revisions 89545 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89545 | tilghman | 2007-11-24 10:59:59 -0600 (Sat, 24 Nov 2007)
- | 5 lines Free some frames that would otherwise leak on error.
- Reported by: Laureano Patch by: Laureano,tilghman (Closes issue
- #11351) ........
-
-2007-11-24 16:53 +0000 [r89544] Steve Murphy <murf@digium.com>
-
- * main/app.c: Added <sys/file.h> include to allow trunk to compile.
- Hope this doesn't louse thing up.
-
-2007-11-24 13:57 +0000 [r89542-89543] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_h323.c: remove a DEBUG_THREADS message that
- accesses private lock fields. If needed, the code to extract this
- information should be implemented in some generic header or
- library and the function called here. (closed bug #11362)
-
- * main/acl.c, main/http.c, main/app.c: remove some unnecessary
- includes
-
-2007-11-24 06:24 +0000 [r89535-89541] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/app.c, apps/app_voicemail.c: Merged revisions 89540 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89540 | tilghman | 2007-11-24 00:19:23 -0600 (Sat, 24 Nov 2007)
- | 9 lines Currently, zero-length voicemail messages cause a
- hangup in VoicemailMain. This change fixes the problem, with a
- multi-faceted approach. First, we do our best to avoid these
- messages from being created in the first place, and second, if
- that fails, we detect when the voicemail message is zero-length
- and avoid exiting at that point. Reported by: dtyoo Patch by:
- gkloepfer,tilghman (Closes issue #11083) ........
-
- * main/manager.c, /: Merged revisions 89536 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89536 | tilghman | 2007-11-23 11:18:26 -0600 (Fri, 23 Nov 2007)
- | 10 lines Up until this point, the XML output of the manager has
- been technically invalid, due to the repetition of certain
- parameters in a single event. This caused various issues for XML
- parsers, some of which refused to parse at all, given the
- invalidity of the rendered XML. So this commit fixes the XML
- output, ensuring that each entity parameter has a unique name,
- thus ensuring valid XML. Reported by: msetim Patch by: tilghman
- (Closes issue #10220) ........
-
- * res/res_config_odbc.c, /: Merged revisions 89534 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r89534 | tilghman | 2007-11-23 11:05:10 -0600 (Fri, 23
- Nov 2007) | 5 lines Use ESCAPE clause for the first parameter,
- not just 2nd-Nth parameters. Reported by: apsaras Patch by:
- tilghman (Closes issue #11353) ........
-
-2007-11-23 15:54 +0000 [r89532-89533] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_oss.c: put in the necessary hooks for video support
- in the console. This is a NOP as far as the current code is
- concerned, but there is already support in ./configure and the
- Makefiles for the various libraries used by console_video.c (not
- yet in the tree) so addition is trivial.
-
- * channels/chan_sip.c: set rtpmap video info according to what is
- read from SDP; make the format explicit in a debug message; print
- the audio instead of aggregated peer capability in a debugging
- msg.
-
-2007-11-23 09:40 +0000 [r89531] Olle Johansson <oej@edvina.net>
-
- * include/asterisk/channel.h: Let's start with implementing the
- base architecture for UTF8 caller ID's so we can handle multiple
- formats properly. This is not carved in stone, but a proposal to
- start with. We need to add support for transliterations as well
- as UTF8 handling, propably with libiconv. Murf is looking into
- that for the dialplan.
-
-2007-11-23 09:03 +0000 [r89530] Luigi Rizzo <rizzo@icir.org>
-
- * include/asterisk/image.h, formats/format_jpeg.c: formatting
- cleanup on the header, normalization of the assignment of
- descriptor fields.
-
-2007-11-23 02:37 +0000 [r89529] Russell Bryant <russell@digium.com>
-
- * configs/agents.conf.sample, /: Merged revisions 89527 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89527 | russell | 2007-11-22 12:29:41 -0500 (Thu, 22 Nov 2007) |
- 3 lines mvanbaak pointed out a spelling error in this sample
- configuration file. While I was at it, I went ahead and tweaked
- it a little bit more. ........
-
-2007-11-22 07:10 +0000 [r89514-89526] Luigi Rizzo <rizzo@icir.org>
-
- * doc/CODING-GUIDELINES: new info on the management of headers
-
- * apps/app_echo.c, apps/app_sendtext.c, apps/app_verbose.c,
- apps/app_milliwatt.c: more header removal
-
- * include/asterisk/channel.h: formatting cleanup
-
- * include/asterisk.h, apps/app_read.c, apps/app_record.c,
- apps/app_echo.c, apps/app_readexten.c,
- include/asterisk/channel.h, apps/app_system.c,
- apps/app_transfer.c, res/ael/pval.c, include/asterisk/app.h,
- apps/app_dumpchan.c, include/asterisk/module.h, apps/app_url.c,
- include/asterisk/pbx.h, apps/app_senddtmf.c, pbx/pbx_config.c,
- apps/app_mixmonitor.c, apps/app_stack.c, apps/app_verbose.c,
- apps/app_milliwatt.c, apps/app_cdr.c, apps/app_while.c: shuffle a
- little bit the content of header files to reduce dependencies. In
- this commit: - move the ast_register/unregister_app functions to
- module.h to avoid the need to include pbx.h for the simpler apps;
- - move the ast_group structure to channel.h to remove the
- dependency of app.h on linkedlists.h Note, this is a long process
- that I am doing in small steps. The main difficulty is that now
- for each subsystem we have a single header (e.g. channel.h)
- included by the subsystem provider (usually one file, e.g.
- channel.c) and by its clients (dozens of them, e.g. we have some
- 70+ apps and 30+ functions). This requires the clients to include
- all the extra headers required by the provider (eg. lock.h,
- linkedlists.h, definitions of substructures...) even though many
- of the clients would be just happy with opaque struct
- declarations and function prototypes. The long term plan is to
- eventually rectify this structure so that the compilation can
- become faster, and also APIs are more stable.
-
- * funcs/func_md5.c, funcs/func_module.c, funcs/func_blacklist.c,
- apps/app_url.c, funcs/func_sha1.c, funcs/func_global.c: remove
- some useless includes
-
- * include/asterisk/audiohook.h, apps/app_dictate.c,
- apps/app_readexten.c, apps/app_directory.c, apps/app_senddtmf.c,
- apps/app_mixmonitor.c, apps/app_stack.c,
- apps/app_controlplayback.c: more removal of redundant headers
-
- * apps/app_read.c, apps/app_echo.c, apps/app_record.c,
- apps/app_userevent.c, apps/app_image.c, apps/app_system.c,
- apps/app_verbose.c, apps/app_milliwatt.c, apps/app_playback.c,
- apps/app_while.c: remove redundant headers
-
- * main/file.c, main/netsock.c: more removal of fcntl.h and other
- system headers
-
- * codecs/codec_lpc10.c, codecs/codec_g722.c, codecs/codec_a_mu.c,
- codecs/codec_speex.c, codecs/codec_alaw.c, codecs/codec_adpcm.c,
- res/res_crypto.c, codecs/codec_g726.c, apps/app_test.c,
- formats/format_ogg_vorbis.c, codecs/codec_gsm.c, res/res_agi.c,
- apps/app_mp3.c, main/app.c, codecs/codec_ulaw.c,
- codecs/codec_ilbc.c: remove a number of #include <fcntl.h> which
- are either useless or done elsewhere
-
- * formats/format_sln.c, formats/format_wav.c,
- formats/format_ogg_vorbis.c, include/asterisk/_private.h,
- formats/format_wav_gsm.c, formats/format_ilbc.c,
- include/asterisk/file.h, formats/format_vox.c,
- formats/format_pcm.c, main/file.c, formats/format_h263.c,
- formats/format_g723.c, formats/format_h264.c,
- include/asterisk/frame.h, formats/format_jpeg.c,
- formats/format_g726.c, formats/format_gsm.c,
- formats/format_g729.c: implement the split of file.h and
- mod_format.h
-
- * include/asterisk/mod_format.h (added): Add a specific header for
- providers of file and format handling routines, moving here
- structs and function declarations formerly in file.h
-
-2007-11-21 23:54 +0000 [r89513] Steve Murphy <murf@digium.com>
-
- * apps/app_dial.c, channels/chan_sip.c, channels/chan_skinny.c,
- res/res_features.c, apps/app_queue.c, channels/chan_iax2.c:
- 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.
-
-2007-11-21 23:24 +0000 [r89511-89512] Luigi Rizzo <rizzo@icir.org>
-
- * funcs/func_rand.c, cdr/cdr_sqlite3_custom.c, apps/app_readfile.c,
- channels/chan_local.c, apps/app_record.c, funcs/func_strings.c,
- apps/app_sayunixtime.c, apps/app_test.c,
- apps/app_alarmreceiver.c, cdr/cdr_adaptive_odbc.c,
- apps/app_image.c, apps/app_chanisavail.c, apps/app_ices.c,
- channels/chan_iax2.c, apps/app_exec.c, pbx/pbx_loopback.c,
- pbx/pbx_spool.c, channels/chan_skinny.c, apps/app_dumpchan.c,
- apps/app_zapscan.c, apps/app_zapras.c, pbx/pbx_realtime.c,
- channels/chan_alsa.c, apps/app_amd.c, apps/app_url.c,
- apps/app_externalivr.c, cdr/cdr_odbc.c, apps/app_dial.c,
- funcs/func_timeout.c, apps/app_page.c, apps/app_privacy.c,
- channels/chan_agent.c, apps/app_disa.c, apps/app_morsecode.c,
- channels/iax2-provision.c, funcs/func_cut.c,
- apps/app_talkdetect.c, apps/app_transfer.c, apps/app_db.c,
- apps/app_playback.c, funcs/func_curl.c, channels/chan_misdn.c,
- apps/app_zapbarge.c, apps/app_waitforring.c, apps/app_sendtext.c,
- channels/chan_features.c, apps/app_macro.c, funcs/func_iconv.c,
- pbx/pbx_config.c, apps/app_mixmonitor.c, apps/app_chanspy.c,
- apps/app_voicemail.c, channels/chan_unistim.c,
- channels/chan_vpb.cc, apps/app_meetme.c, apps/app_authenticate.c,
- apps/app_readexten.c, funcs/func_vmcount.c,
- channels/chan_gtalk.c, cdr/cdr_pgsql.c, apps/app_followme.c,
- cdr/cdr_radius.c, apps/app_controlplayback.c, cdr/cdr_csv.c,
- channels/chan_phone.c, funcs/func_enum.c, apps/app_osplookup.c,
- funcs/func_odbc.c, apps/app_mp3.c, apps/app_minivm.c,
- apps/app_rpt.c, channels/chan_mgcp.c, apps/app_parkandannounce.c,
- apps/app_while.c, apps/app_adsiprog.c, apps/app_nbscat.c,
- funcs/func_version.c, funcs/func_db.c, channels/chan_zap.c,
- apps/app_read.c, channels/chan_sip.c, apps/app_festival.c,
- apps/app_waitforsilence.c, funcs/func_lock.c, pbx/pbx_lua.c,
- apps/app_system.c, apps/app_getcpeid.c, apps/app_queue.c,
- channels/chan_oss.c, cdr/cdr_tds.c, funcs/func_realtime.c,
- channels/chan_jingle.c, channels/chan_usbradio.c,
- apps/app_channelredirect.c, apps/app_flash.c,
- apps/app_directed_pickup.c, funcs/func_blacklist.c,
- channels/chan_h323.c, pbx/pbx_dundi.c, apps/app_sms.c,
- channels/chan_nbs.c, apps/app_senddtmf.c, funcs/func_callerid.c,
- apps/app_verbose.c, apps/app_stack.c, pbx/pbx_gtkconsole.c:
- remove another set of redundant #include "asterisk/options.h"
-
- * main/udptl.c, main/autoservice.c, main/frame.c, res/res_snmp.c,
- main/say.c, res/res_features.c, main/devicestate.c, main/utils.c,
- res/res_musiconhold.c, res/res_jabber.c, main/indications.c,
- main/enum.c, res/res_config_sqlite.c, main/config.c,
- main/loader.c, main/term.c, main/cli.c, main/io.c,
- main/channel.c, main/cdr.c, main/dial.c, res/res_smdi.c,
- res/res_config_odbc.c, main/manager.c, res/res_agi.c,
- main/http.c, main/logger.c, res/res_realtime.c, main/app.c,
- main/image.c, main/dns.c, main/db.c, res/res_speech.c,
- main/sched.c, main/pbx.c, res/res_config_pgsql.c, main/dnsmgr.c,
- main/translate.c, res/res_crypto.c, res/res_adsi.c,
- main/jitterbuf.c, main/acl.c, formats/format_ogg_vorbis.c,
- res/res_ael_share.c, res/res_monitor.c, main/rtp.c,
- main/netsock.c, main/srv.c, main/hashtab.c, main/privacy.c,
- main/adsistub.c, main/abstract_jb.c, main/file.c,
- main/callerid.c, main/astmm.c, main/audiohook.c,
- formats/format_g726.c, main/asterisk.c, res/res_odbc.c,
- main/dsp.c: remove a bunch of useless #include "options.h"
-
-2007-11-21 22:37 +0000 [r89509-89510] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Remove unneccessary explicit case for BRI
-
- * channels/chan_zap.c: Take some debug code out :-)
-
-2007-11-21 22:20 +0000 [r89508] Luigi Rizzo <rizzo@icir.org>
-
- * main/cygload.c: add a missing return
-
-2007-11-21 22:07 +0000 [r89507] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Add BRI support to chan_zap
-
-2007-11-21 21:30 +0000 [r89506] Luigi Rizzo <rizzo@icir.org>
-
- * utils/Makefile, configure, configure.ac: enable support for stack
- backtrace for stuff built in utils/ (this was present in the main
- tree but forgotten here).
-
-2007-11-21 20:38 +0000 [r89505] Steve Murphy <murf@digium.com>
-
- * main/pbx.c: closes issue #11290; the proposed patch was a good
- guess, and would solve the bug to some extent, but was really
- masking the real issue, that there were bad entries in the table.
- This fix removes the condition that the hashtab updates be done
- on exten removal only when the pattern_tree was present, which is
- silly. The operations that apply to the pattern tree are instead
- made conditional. Also, threw back in routines that kpfleming
- deleted because of probs in the 64-bit world. Tested on both 32
- and 64-bit machines (compile). Tested the reload problem with
- over 20 reloads, and no problems. If you find more problems,
- please reopen 11290.
-
-2007-11-21 20:22 +0000 [r89504] Terry Wilson <twilson@digium.com>
-
- * res/res_features.c: Simplify comparison in parking fix
-
-2007-11-21 19:28 +0000 [r89494-89496] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 89495 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89495 | mmichelson | 2007-11-21 13:27:51 -0600 (Wed, 21 Nov
- 2007) | 3 lines Fix a small error I made in my previous commit
- ........
-
- * /, apps/app_queue.c: Merged revisions 89493 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89493 | mmichelson | 2007-11-21 13:24:22 -0600 (Wed, 21 Nov
- 2007) | 5 lines Changing an inaccurate debug message to be less
- inaccurate. Under the circumstances, this message would always
- report that there were 0 members available, even though that may
- not be true. ........
-
-2007-11-21 19:20 +0000 [r89492] Terry Wilson <twilson@digium.com>
-
- * /, res/res_features.c: Merged revisions 89491 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89491 | twilson | 2007-11-21 12:59:27 -0600 (Wed, 21 Nov 2007) |
- 4 lines If a channel gets masqueraded in the middle of a park,
- don't play the announcement to the masqueraded channel, and dial
- back to the original channel on timeout. ........
-
-2007-11-21 18:52 +0000 [r89490] Russell Bryant <russell@digium.com>
-
- * main/dsp.c: Remove obsolete OLD_DSP_ROUTINES code. Also, remove
- the FAX_DETECT define and only do the calculations if fax
- detection is enabled on the dsp. (closes issue #11331) Reported
- by: dimas Patches: dsp.patch uploaded by dimas (license 88)
-
-2007-11-21 18:38 +0000 [r89489] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_read.c, UPGRADE.txt, CHANGES: Change Read to set
- READSTATUS as an indication of the result Also, some cleanup to
- CHANGES. Reported by: michael-fig Patch by: michael-fig,tilghman
- (Closes issue #11004)
-
-2007-11-21 18:24 +0000 [r89488] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: fix a small gramatical error in a comment
-
-2007-11-21 18:19 +0000 [r89487] Mark Michelson <mmichelson@digium.com>
-
- * main/utils.c: There existed about a 1 in 4 billion chance that
- reading from /dev/urandom would return LONG_MIN (1 in 9
- quintillion if using 64-bit longs). Since there is no positive
- equivalent of LONG_MIN, the result of labs() in this case is
- unpredictable. This fixes that situation. (closes issue #11336,
- reported and patched by sperreault)
-
-2007-11-21 16:24 +0000 [r89484] Russell Bryant <russell@digium.com>
-
- * channels/chan_unistim.c: Fix some code that was supposed to
- ensure that a buffer was terminated, but was writing to the wrong
- byte. Also, remove some non-thread safe test code. (closes issue
- #11317) Reported by: IgorG Patches: unistim-2.patch uploaded by
- IgorG (license 20) - additional changes by me
-
-2007-11-21 16:08 +0000 [r89483] Mark Michelson <mmichelson@digium.com>
-
- * main/pbx.c: I introduced a deadlock avoidance into 1.4, which I
- attempted to port to trunk as well. Unfortunately, since trunk
- uses read/write locks for the context lock, it means that I have
- actually *introduced* a deadlock condition since they are not
- recursive. Removing this change for now and will look into
- introducing a different one.
-
-2007-11-21 16:07 +0000 [r89480-89482] Kevin P. Fleming <kpfleming@digium.com>
-
- * include/asterisk.h, include/asterisk/compat.h, utils/ael_main.c,
- utils/conf2ael.c: move these forward declarations back to
- asterisk.h where they belong... even though asterisk.h includes
- compat.h, these declarations have nothing to do with the being
- platform-compatible and are directly related to being part of
- Asterisk
-
- * channels/chan_usbradio.c: get this to actually compile...
-
- * main/pbx.c: remove some debugging code that doesn't compile on
- 64-bit platforms
-
-2007-11-21 15:17 +0000 [r89478-89479] Steve Murphy <murf@digium.com>
-
- * res/res_features.c: OOOps! All the debug stuff I inserted was
- accidentally committed. I hereby revert it.
-
- * main/hashtab.c, res/res_features.c: closes issue #11265; Thanks
- to snuffy for his work on neatening up the code and removing
- duplicated code.
-
-2007-11-21 08:28 +0000 [r89475-89477] Luigi Rizzo <rizzo@icir.org>
-
- * channels/gentone-ulaw.c (removed): remove this file, it is not
- used anywhere.
-
- * main/astmm.c: add missing paths.h
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac: add
- check for video4linux
-
-2007-11-21 01:09 +0000 [r89474] Steve Murphy <murf@digium.com>
-
- * main/pbx.c: A free in add_pri was ultimately the source of the
- grief we were having with parking. This set of changes fixes that
- problem, and introduces some more error messages, and puts debugs
- into ifdefs for what could be short-term usage. Txs to Terry W.
- for his help, guidance, and especially patience.
-
-2007-11-21 00:23 +0000 [r89472-89473] Luigi Rizzo <rizzo@icir.org>
-
- * main/sha1.c, agi/eagi-test.c, utils/smsq.c, utils/hashtest2.c,
- main/minimime/mm.h, utils/check_expr.c: more header
- removal/normalization
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac: X11
- checks (at least some - for other platforms with unusual X11
- locations you might need to add more directories)
-
-2007-11-21 00:21 +0000 [r89470] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c, CHANGES: Merge changes from
- team/russell/sla_trunk_moh ... * Added the ability to specify the
- music on hold class used to play into the conference when there
- is only one member and the M option is used. * Added the ability
- to specify a music on hold class to play instead of ringing for
- the SLATrunk application. (patched by me, and tested internally)
-
-2007-11-21 00:20 +0000 [r89469] Luigi Rizzo <rizzo@icir.org>
-
- * makeopts.in: complete support for X11
-
-2007-11-20 23:29 +0000 [r89467-89468] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_meetme.c, cdr/cdr_sqlite.c, pbx/pbx_lua.c: Make trunk
- build again
-
- * main/say.c: Add support for new recorded character sounds Closes
- issue #5208
-
-2007-11-20 23:16 +0000 [r89465-89466] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_unistim.c, cdr/cdr_sqlite3_custom.c,
- apps/app_dictate.c, apps/app_test.c, apps/app_ices.c,
- apps/app_followme.c, channels/chan_iax2.c, main/config.c,
- main/loader.c, main/cli.c, cdr/cdr_csv.c, main/channel.c,
- main/manager.c, pbx/pbx_spool.c, include/asterisk/compat.h,
- res/res_agi.c, apps/app_minivm.c, main/logger.c, main/http.c,
- main/app.c, main/image.c, apps/app_directory.c, main/db.c,
- cdr/cdr_custom.c, apps/app_adsiprog.c, apps/app_dial.c,
- include/asterisk/utils.h, include/asterisk.h, main/pbx.c,
- channels/chan_sip.c, res/res_crypto.c,
- include/asterisk/channel.h, res/res_monitor.c,
- include/asterisk/paths.h, main/file.c, apps/app_sms.c,
- include/asterisk/ael_structs.h, pbx/pbx_config.c,
- apps/app_mixmonitor.c, apps/app_chanspy.c, apps/app_voicemail.c:
- move asterisk/paths.h outside asterisk.h and into those files who
- really need it.
-
- * main/pbx.c, include/asterisk.h, main/frame.c, main/dnsmgr.c,
- main/threadstorage.c, main/devicestate.c,
- include/asterisk/_private.h (added), main/astobj2.c,
- main/loader.c, main/term.c, main/cli.c, main/channel.c,
- main/manager.c, main/logger.c, build_tools/strip_nonapi,
- main/event.c, main/asterisk.c, main/db.c: move internal function
- declarations to include/asterisk/_private.h
-
-2007-11-20 19:29 +0000 [r89464] Russell Bryant <russell@digium.com>
-
- * configure, configure.ac: i got a little carried away with commas
- ...
-
-2007-11-20 19:28 +0000 [r89463] Kevin P. Fleming <kpfleming@digium.com>
-
- * include/asterisk/module.h, build_tools/make_buildopts_h,
- main/loader.c: switch compile-time option checking to string
- storage mode in this branch too
-
-2007-11-20 19:11 +0000 [r89460] Russell Bryant <russell@digium.com>
-
- * configure, configure.ac: fix the zaptel configure script check
-
-2007-11-20 18:20 +0000 [r89459] Luigi Rizzo <rizzo@icir.org>
-
- * acinclude.m4: the 'version' is now $7 not $6 (wait a bit before
- regenerating configure, i have more changes)
-
-2007-11-20 17:59 +0000 [r89458] Mark Michelson <mmichelson@digium.com>
-
- * main/pbx.c, /: Merged revisions 89457 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89457 | mmichelson | 2007-11-20 11:50:31 -0600 (Tue, 20 Nov
- 2007) | 9 lines According to comments in main/pbx.c, it is
- essential that if we are going to lock the conlock as well as the
- hints lock, it must be locked in that respective order. In order
- to prevent a potential deadlock, we need to lock the conlock
- prior to locking the hints lock in ast_hint_state_changed (see
- the call stack example on issue #11323 for how this can happen).
- (closes issue #11323, reported by eelcob, suggestion for patch by
- eelcob, patch by me) ........
-
-2007-11-20 17:11 +0000 [r89454-89455] Luigi Rizzo <rizzo@icir.org>
-
- * makeopts.in: prepare to support console_video
-
- * apps/Makefile, Makefile.moddir_rules, pbx/Makefile, res/Makefile,
- channels/Makefile: Fix building of modules under cygwin. After
- this commit we can actually load modules under windows, and we
- can start debugging more interesting problems related to the load
- order and functionality of modules.
-
-2007-11-20 16:11 +0000 [r89453] Mark Michelson <mmichelson@digium.com>
-
- * configs/sip.conf.sample: Changed occurrences of "busy-level" to
- "busylevel" in sip.conf.sample in light of commit 89441. Thanks
- to pj for pointing out the need for this (closes issue #11307,
- reported by pj)
-
-2007-11-20 15:39 +0000 [r89452] Luigi Rizzo <rizzo@icir.org>
-
- * configure, configure.ac, acinclude.m4: add an argument for extra
- headers to AC_EXT_LIB_CHECK, and on passing simplify the code.
- Too bad that every time we need to regenerate configure...
-
-2007-11-20 15:30 +0000 [r89451] Steve Murphy <murf@digium.com>
-
- * /, doc/tex/queues-with-callback-members.tex: Merged revisions
- 89450 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89450 | murf | 2007-11-20 08:22:08 -0700 (Tue, 20 Nov 2007) | 1
- line closes issue #11324; break statements missing in switch
- cases. ........
-
-2007-11-20 15:00 +0000 [r89449] Joshua Colp <jcolp@digium.com>
-
- * main/translate.c: Minor documentation tweak and if an incorrect
- parameter is given to core show translation return the usage
- information. (closes issue #11316) Reported by: eliel Patches:
- translate.c.patch uploaded by eliel (license 64)
-
-2007-11-20 14:54 +0000 [r89448] Luigi Rizzo <rizzo@icir.org>
-
- * configure, acinclude.m4: comment a bit the code in acinclude.m4
- There is still a lot of code to clean up there, but hopefully
- this should clarify what goes on in there.
-
-2007-11-20 14:49 +0000 [r89447] Joshua Colp <jcolp@digium.com>
-
- * channels/h323/ast_h323.cxx: Include the compatibility header file
- in ast_h323.cxx for compatibility reasons. (closes issue #11311)
- Reported by: falves11
-
-2007-11-20 14:44 +0000 [r89444-89446] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Fix sip show history. Closes issue #11312
-
- * channels/chan_sip.c: Change terminology a bit for CLI commands
- handling SIP channels/calls/dialogs/whatever. Closes issue #11312
-
-2007-11-20 07:42 +0000 [r89443] Luigi Rizzo <rizzo@icir.org>
-
- * Makefile, main/Makefile, Makefile.moddir_rules: initial makefile
- changes to build loadable modules under cygwin (not complete yet
- - still need to sort out dependecies on res_*)
-
-2007-11-20 00:17 +0000 [r89442] Steve Murphy <murf@digium.com>
-
- * main/pbx.c: Get rid of some debug messages in pbx.c
-
-2007-11-19 23:24 +0000 [r89441] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c, CHANGES: Changed the "busy-level" option in
- sip.conf to "busylevel" to be more parallel with the SIPPEER()
- argument of the same name. The deprecation procedure is not being
- used here since this is a trunk-only option. (closes issue
- #11307, reported by pj, patched by me)
-
-2007-11-19 23:03 +0000 [r89439-89440] Russell Bryant <russell@digium.com>
-
- * include/asterisk/module.h: Be a bit more pedantic about the type
- for holding the md5 sum for the build options. Also, doxygenify
- the comment.
-
- * funcs/func_sysinfo.c: Make the SYSINFO documentation reflect
- which options were compiled in
-
-2007-11-19 22:55 +0000 [r89438] Steve Murphy <murf@digium.com>
-
- * main/pbx.c: These changes were made in response to
- niklas@tese.se's letter of 11-17-2007, where he had 20 and 201 in
- two different contexts, included in the same context. In that
- particular case, we were behaving the same as 1.4, but after
- experimenting, I quickly found that if 20 and 201 were in the
- same extension, 1.4 would return 201, and this code returns 20.
- These changes now enable the current code to replicate the
- behavior of 1.4 in respect to MATCHMORE in cases like this.
-
-2007-11-19 21:18 +0000 [r89430-89433] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_vpb.cc, channels/misdn_config.c, main/dsp.c:
- another few errno.h removals
-
- * pbx/pbx_loopback.c, apps/app_zapbarge.c, pbx/pbx_spool.c,
- apps/app_meetme.c, pbx/pbx_ael.c, pbx/pbx_lua.c,
- pbx/pbx_realtime.c, pbx/pbx_dundi.c, apps/app_externalivr.c,
- apps/app_directory.c, apps/app_system.c, pbx/pbx_config.c,
- apps/app_milliwatt.c: more errno.h removal
-
- * funcs/func_sysinfo.c: remove unnecessary headers
-
- * funcs/func_base64.c, funcs/func_volume.c: remove some unnecessary
- includes.
-
-2007-11-19 20:13 +0000 [r89429] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_sip.c: Change delimiter of SIPPEER to be comma
- (instead of pipe) and further deprecate the old ':' delimiter
- Reported by: pj Patch by: tilghman Closes issue #11305
-
-2007-11-19 19:51 +0000 [r89424-89428] Luigi Rizzo <rizzo@icir.org>
-
- * codecs/codec_lpc10.c, codecs/codec_a_mu.c, codecs/codec_g722.c,
- codecs/codec_adpcm.c, codecs/codec_alaw.c, codecs/codec_speex.c,
- codecs/codec_g726.c, codecs/codec_gsm.c, codecs/codec_ulaw.c,
- codecs/codec_ilbc.c, codecs/codec_zap.c: remove some useless
- includes from codecs
-
- * formats/format_ilbc.c, formats/format_sln.c,
- formats/format_vox.c, formats/format_wav.c, formats/format_pcm.c,
- formats/format_ogg_vorbis.c, formats/format_g723.c,
- formats/format_h263.c, formats/format_h264.c,
- formats/format_wav_gsm.c, formats/format_g726.c,
- formats/format_jpeg.c, formats/format_gsm.c,
- formats/format_g729.c: format handlers don't need network, lock,
- channel and scheduler headers
-
- * include/asterisk.h, include/asterisk/compat.h,
- include/asterisk/lock.h, utils/extconf.c,
- include/asterisk/abstract_jb.h: move the declaration of struct
- ast_channel ast_frame and ast_module to compat.h so it is always
- available - hopefully this will let us reduce the number of
- inclusions of channel.h and frame.h
-
- * main/udptl.c, main/autoservice.c, funcs/func_rand.c,
- cdr/cdr_sqlite3_custom.c, main/frame.c, funcs/func_module.c,
- main/threadstorage.c, main/say.c, funcs/func_env.c,
- funcs/func_strings.c, main/devicestate.c,
- cdr/cdr_adaptive_odbc.c, main/indications.c, main/config.c,
- main/loader.c, main/term.c, main/cli.c, funcs/func_shell.c,
- main/http.c, cdr/cdr_odbc.c, main/db.c, cdr/cdr_manager.c,
- main/sched.c, main/pbx.c, funcs/func_timeout.c,
- funcs/func_math.c, funcs/func_cut.c, main/chanvars.c,
- main/netsock.c, funcs/func_curl.c, main/srv.c, main/privacy.c,
- funcs/func_cdr.c, funcs/func_channel.c, main/audiohook.c,
- funcs/func_iconv.c, main/alaw.c, main/asterisk.c,
- funcs/func_base64.c, funcs/func_md5.c, funcs/func_sysinfo.c,
- main/utils.c, funcs/func_sha1.c, cdr/cdr_pgsql.c,
- funcs/func_logic.c, cdr/cdr_radius.c, main/enum.c,
- funcs/func_uri.c, main/io.c, cdr/cdr_csv.c, main/ulaw.c,
- main/channel.c, main/cdr.c, funcs/func_enum.c, main/dial.c,
- funcs/func_groupcount.c, main/manager.c, main/tdd.c,
- funcs/func_odbc.c, cdr/cdr_sqlite.c, main/logger.c, main/app.c,
- main/image.c, main/dns.c, cdr/cdr_custom.c, funcs/func_version.c,
- funcs/func_db.c, main/dnsmgr.c, main/translate.c,
- main/slinfactory.c, funcs/func_lock.c, main/acl.c, main/rtp.c,
- cdr/cdr_tds.c, funcs/func_realtime.c, main/hashtab.c,
- funcs/func_blacklist.c, main/abstract_jb.c, main/cryptostub.c,
- main/adsistub.c, main/file.c, main/callerid.c, main/astmm.c,
- funcs/func_callerid.c, main/dsp.c: another bunch of include
- removals (errno.h and asterisk/logger.h)
-
- * channels/chan_local.c, apps/app_record.c,
- apps/app_alarmreceiver.c, apps/app_chanisavail.c,
- apps/app_ices.c, apps/app_exec.c, channels/chan_iax2.c,
- channels/chan_skinny.c, formats/format_pcm.c,
- apps/app_dumpchan.c, apps/app_zapras.c, formats/format_h263.c,
- codecs/codec_g722.c, formats/format_wav.c, apps/app_softhangup.c,
- codecs/codec_g726.c, formats/format_ogg_vorbis.c,
- apps/app_morsecode.c, apps/app_talkdetect.c, apps/app_db.c,
- apps/app_speech_utils.c, apps/app_sendtext.c,
- formats/format_g726.c, apps/app_mixmonitor.c, res/res_odbc.c,
- apps/app_voicemail.c, channels/chan_vpb.cc, formats/format_sln.c,
- res/res_snmp.c, apps/app_dictate.c, apps/app_authenticate.c,
- apps/app_readexten.c, codecs/codec_gsm.c, apps/app_userevent.c,
- channels/chan_gtalk.c, res/res_jabber.c, apps/app_setcallerid.c,
- res/res_config_odbc.c, apps/app_osplookup.c, apps/app_mp3.c,
- apps/app_minivm.c, res/res_realtime.c, formats/format_h264.c,
- apps/app_directory.c, apps/app_rpt.c, channels/chan_mgcp.c,
- apps/app_adsiprog.c, codecs/codec_lpc10.c,
- res/res_config_pgsql.c, apps/app_read.c, channels/chan_sip.c,
- codecs/codec_alaw.c, res/res_adsi.c, res/res_crypto.c,
- channels/chan_jingle.c, apps/app_channelredirect.c,
- apps/app_forkcdr.c, formats/format_vox.c, apps/app_sms.c,
- formats/format_g723.c, apps/app_verbose.c, apps/app_stack.c,
- apps/app_readfile.c, res/res_features.c, codecs/codec_adpcm.c,
- apps/app_sayunixtime.c, apps/app_test.c, apps/app_image.c,
- formats/format_wav_gsm.c, res/res_smdi.c,
- include/asterisk/compat.h, apps/app_skel.c, apps/app_zapscan.c,
- channels/chan_alsa.c, apps/app_url.c, apps/app_externalivr.c,
- formats/format_jpeg.c, formats/format_gsm.c,
- apps/app_milliwatt.c, apps/app_dial.c, apps/app_page.c,
- apps/app_privacy.c, codecs/codec_speex.c, apps/app_echo.c,
- channels/chan_agent.c, apps/app_disa.c,
- channels/iax2-provision.c, res/res_ael_share.c,
- apps/app_transfer.c, res/res_monitor.c, apps/app_playback.c,
- channels/chan_misdn.c, apps/app_waitforring.c,
- apps/app_zapbarge.c, channels/chan_features.c, apps/app_macro.c,
- apps/app_zapateller.c, res/res_indications.c,
- codecs/codec_ilbc.c, apps/app_chanspy.c, channels/chan_unistim.c,
- apps/app_meetme.c, res/res_musiconhold.c, apps/app_followme.c,
- codecs/codec_zap.c, res/res_config_sqlite.c,
- channels/misdn_config.c, apps/app_controlplayback.c,
- formats/format_ilbc.c, channels/chan_phone.c, res/res_agi.c,
- main/logger.c, apps/app_ivrdemo.c, apps/app_parkandannounce.c,
- res/res_clioriginate.c, apps/app_while.c, include/asterisk.h,
- apps/app_nbscat.c, channels/chan_zap.c, codecs/codec_a_mu.c,
- res/res_limit.c, apps/app_festival.c, apps/app_waitforsilence.c,
- res/res_convert.c, apps/app_getcpeid.c, apps/app_system.c,
- apps/app_queue.c, channels/chan_oss.c, channels/chan_usbradio.c,
- apps/app_flash.c, apps/app_directed_pickup.c,
- channels/chan_h323.c, codecs/codec_ulaw.c, channels/chan_nbs.c,
- apps/app_senddtmf.c, formats/format_g729.c: include "logger.h"
- and errno.h from asterisk.h - usage shows that they were included
- almost everywhere. Remove some of the instances.
-
-2007-11-19 17:18 +0000 [r89422] Steve Murphy <murf@digium.com>
-
- * main/pbx.c: a correction to code involved in an extension removal
-
-2007-11-19 16:29 +0000 [r89421] Mark Michelson <mmichelson@digium.com>
-
- * funcs/func_sysinfo.c (added), CHANGES: Adding SYSINFO() dialplan
- function for retrieval of system information
-
-2007-11-19 15:55 +0000 [r89417-89420] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_features.c: Merged revisions 89419 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89419 | file | 2007-11-19 11:53:32 -0400 (Mon, 19 Nov 2007) | 6
- lines Print out the correct filename (features.conf) in the log
- message when parkpos options are incorrect. (closes issue #11295)
- Reported by: Laureano Patches: res_features.c.patch uploaded by
- Laureano (license 265) ........
-
- * /, doc/tex/localchannel.tex: Merged revisions 89416 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r89416 | file | 2007-11-19 11:24:12 -0400 (Mon, 19 Nov
- 2007) | 4 lines Clarify documentation a bit, include that a frame
- has to pass through the core in order for the Local channel
- optimization to happen. (closes issue #11246) Reported by: jon
- ........
-
-2007-11-19 14:36 +0000 [r89412] Luigi Rizzo <rizzo@icir.org>
-
- * include/asterisk/logger.h: revert inclusion of options.h
-
-2007-11-19 14:03 +0000 [r89410] Joshua Colp <jcolp@digium.com>
-
- * apps/app_playback.c: Change warning messages (which are really
- debug messages) into debug messages. (closes issue #11288)
- Reported by: IgorG Patches: saydebug-89394-1-trunk.patch uploaded
- by IgorG (license 20)
-
-2007-11-19 09:16 +0000 [r89404-89407] Olle Johansson <oej@edvina.net>
-
- * CHANGES: Update CHANGES
-
- * channels/chan_sip.c: Adding busy-level to the SIP_PEER() dialplan
- function. With this, you can control the peer in the dialplan, so
- you avoid placing outbound calls when the device has reached
- busy-level. Reported by pj. Closes bug #11180
-
- * main/acl.c: Add some debugging to the routines that finds our
- local IP address. Related to bug #9225
-
- * channels/chan_sip.c: Make some notes about a problem I found with
- the OPTIONs handler while working with the bug tracker. Since we
- don't authenticate devices (peers/users) on OPTIONS we don't have
- the proper context set for the user/peer. However, we might not
- want to process an authentication for every OPTIONS, so we could
- have a config option for this, "optionsforceok" to always answer
- 200 OK on the request and not check device or destination, nor
- add a SDP. If Asterisk sends the OPTIONs request, it doesn't care
- about the reply. Some devices use OPTIONs to discover
- capabilities, since we should answer like an INVITE from the
- device and we need to support that properly too, which we don't
- today. So much to do :-)
-
-2007-11-18 21:50 +0000 [r89394-89399] Joshua Colp <jcolp@digium.com>
-
- * build_tools/make_buildopts_h: Add OSX into the logic that uses
- md5 instead of md5sum.
-
- * include/asterisk/compat.h: Use the easy way that rizzo mentioned,
- only include malloc.h on the Windows platform.
-
- * include/asterisk/compat.h: Revert last commit, apparently
- buildbot lied to me.
-
- * include/asterisk/compat.h: Change how we handle alloca to conform
- with how it is suggested in the autoconf manual for
- AC_FUNC_ALLOCA. FreeBSD 6 now builds again and no other platforms
- should be broken by this.
-
- * configure, configure.ac: Change autoconf logic a bit so it says
- what it is looking for in two instances where it didn't.
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac,
- include/asterisk/lock.h, include/asterisk/network.h: Use autoconf
- logic to determine the presence of
- PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP and
- PTHREAD_MUTEX_RECURSIVE_NP. Enclose error message from network.h
- in "
-
-2007-11-17 21:47 +0000 [r89393] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Add SS7 Generic address support (#11156)
-
-2007-11-17 19:29 +0000 [r89389-89392] Luigi Rizzo <rizzo@icir.org>
-
- * include/asterisk/compat.h: if alloca.h is not present, try
- malloc.h
-
- * agi/Makefile: temporarily disable this target in mingw
-
- * Makefile: will i ever get precedences for windows right ? in the
- meantime, use a variable to ease enabling/disabling print
- subdirectories.
-
- * Makefile: reformulate dependencies in a more correct way
-
-2007-11-17 17:46 +0000 [r89388] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, pbx/pbx_dundi.c: a quick fix to pbx_dundi.c to make
- it so it will compile. Hope I did the right thing. And some
- additions to removal of extens to take care of hashtab pointers
- in all cases.
-
-2007-11-17 17:27 +0000 [r89363-89387] Luigi Rizzo <rizzo@icir.org>
-
- * Makefile.moddir_rules, Makefile.rules: as discussed some time ago
- on the -dev list, create embedde object with a .eo suffix even if
- they are coming from .cc sources. This simplifies the handling in
- the build scripts.
-
- * include/asterisk/network.h: prefer socket.h over other variants
- (winsock etc.)
-
- * channels/chan_local.c, main/translate.c,
- channels/chan_features.c, main/http.c, main/config.c: trim more
- redundant headers
-
- * main/acl.c: remove unnecessary includes
-
- * main/udptl.c, main/dnsmgr.c, channels/chan_sip.c, main/acl.c,
- main/dns.c, main/rtp.c, main/netsock.c: fix breakage induced by
- previous mistake
-
- * Makefile: wrong variable, wrong order -> broken build.
-
- * include/asterisk/acl.h, include/asterisk/utils.h,
- include/asterisk/autoconfig.h.in, include/asterisk/rtp.h,
- configure.ac, main/acl.c, include/asterisk/netsock.h,
- main/utils.c, include/asterisk/manager.h, main/netsock.c,
- main/manager.c, res/res_agi.c, pbx/pbx_dundi.c,
- include/asterisk/udptl.h, include/asterisk/dnsmgr.h,
- main/asterisk.c: start using asterisk/network.h for network
- related headers. Also remove some unnecessary includes.
-
- * include/asterisk/network.h (added): wrapper for all generic
- network headers that have different names and locations on the
- various systems.
-
- * main/cygload.c: main is called main not amain!
-
- * main/Makefile: conditional targets for building the windows
- version
-
- * Makefile: support cygwin targets
-
- * Makefile.moddir_rules: and this is the last one to have asterisk
- compile (not run yet) natively under cygwin.
-
- * apps/app_sms.c: another cygwin compatibility fix. This one must
- be handled in a better way in configure, also for other
- architectures
-
- * utils/Makefile, main/Makefile, utils/extconf.c: more
- cygwin/mingw32 compatibility fixes
-
- * include/asterisk/channel.h: use autoconf results to conditionally
- compile timersub
-
- * include/asterisk/lock.h: compatibility fixes for cygwin
-
- * include/asterisk/compat.h: some version of flex produce code that
- wants __STDC_VERSION__ defined, but the compiler does not always
- define it.
-
- * Makefile: these linker flags apply to both cygwin and mingw32
-
- * utils/hashtest2.c: add a return NULL to a function that is
- expected to return a value so compilers that don't understand
- that this code is NOTREACHED will not complain (the fault is not
- much on the compiler but on the declaration of pthread_exit on
- certain platforms) s/certain platform/cygwin/ if you are really
- curious
-
- * main/loader.c: define RTLD_LOCAL for platforms that don't have
- it. This is only to complete the build, clearly the linker
- behaviour will be completely different and likely to cause
- trouble in those cases.
-
- * channels/Makefile: filter out modules that do not compile under
- windows (this should be handled with the dependencies generated
- by configure and menuselect, but will be fixed later)
-
- * main/utils.c: netdb.h is used for gethostbyname, and it was not
- included in some platforms.
-
- * main/cygload.c (added): Loader for cygwin where asterisk is
- really a big dll (something like this is already in 1.2)
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac:
- timersub is a macro not a function, so write the check in a way
- that detects both formats.
-
-2007-11-17 06:34 +0000 [r89359-89362] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_lua.c: fix the build of pbx_lua
-
- * configure, include/asterisk/autoconfig.h.in,
- include/asterisk/compat.h, configure.ac, include/asterisk/io.h,
- include/asterisk/channel.h: Update the configure script check for
- sys/poll.h to also provide the result in
- include/asterisk/autoconfig.h. Also, move the conditional include
- of sys/poll.h or asterisk/poll-compat.h into asterisk/config.h
- instead of the two headers it existed in before.
-
- * build_tools/make_buildopts_h: actually let this compile, oops :(
-
- * build_tools/make_buildopts_h: Use the fix suggested by Tilghman
- on the -dev to make cutting up the BUILDSUM friendly to non-bash
- shells. I think this should work for BSD/mingw as well, but did
- not yet remove the switch statement.
-
-2007-11-17 04:19 +0000 [r89348-89358] Luigi Rizzo <rizzo@icir.org>
-
- * Makefile: linker flags for mingw32
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac: add
- detection for timersub() and winsock.h/winsock2.h
-
- * include/asterisk/endian.h: provide definitions for
- __LITTLE_ENDIAN and __BIG_ENDIAN if not present.
-
- * main/Makefile, include/asterisk/io.h, include/asterisk/channel.h:
- use poll as detected by configure
-
- * configure, configure.ac, makeopts.in: use autoconf to check for
- the existence of sys/poll.h
-
- * build_tools/make_buildopts_h: this script is run on the build
- system, not on the host.
-
- * Makefile.moddir_rules: compatibility fix for mingw32
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac,
- acinclude.m4, makeopts.in: acinclude.m4: add a function to help
- checking sdl-config, gtk-config and the like (this could be used
- for gtk and gtk2 as well) Other files: add tests for sdl,
- sdl_image and avcodec and regenerate configure and
- autoconfig.h.in
-
- * include/asterisk/autoconfig.h.in, configure.ac: add check for the
- presence of glob
-
- * channels/chan_jingle.c, channels/chan_unistim.c,
- funcs/func_enum.c, channels/chan_local.c, channels/chan_misdn.c,
- channels/chan_skinny.c, funcs/func_odbc.c, channels/chan_h323.c,
- utils/ael_main.c, cdr/cdr_pgsql.c, channels/chan_gtalk.c,
- apps/app_db.c, channels/chan_mgcp.c: more removal of duplicate
- #include lines
-
- * main/udptl.c, funcs/func_module.c, res/res_features.c,
- funcs/func_lock.c, res/res_adsi.c, funcs/func_strings.c,
- channels/chan_agent.c, pbx/dundi-parser.c, main/rtp.c,
- pbx/pbx_loopback.c, funcs/func_blacklist.c,
- channels/chan_features.c, apps/app_dumpchan.c, res/res_agi.c,
- main/logger.c, pbx/pbx_realtime.c, pbx/pbx_dundi.c,
- apps/app_rpt.c, main/asterisk.c, apps/app_parkandannounce.c:
- remove a bunch of duplicate includes Reproduce with grep -r
- #include . | grep -v .svn | grep -v Binary | sort | uniq -c |
- sort -nr
-
-2007-11-16 23:44 +0000 [r89347] Terry Wilson <twilson@digium.com>
-
- * res/res_features.c: Fix broken parking dial-back
-
-2007-11-16 23:33 +0000 [r89346] Steve Murphy <murf@digium.com>
-
- * main/pbx.c: My goodness, haven't handled an extension deletion.
- Add code to ast_context_remove_extension2() to remove an
- extension from the trie. Done by marking it deleted. The
- scoreboard won't update for it any more. Also, a couple of calls
- to insert hashtab had a spurious ->exten, which was removed.
-
-2007-11-16 23:28 +0000 [r89341-89345] Luigi Rizzo <rizzo@icir.org>
-
- * include/asterisk/paths.h, include/asterisk.h: paths are already
- in include/asterisk/paths.h so don't duplicate them in
- include/asterisk.h
-
- * include/asterisk/utils.h, include/asterisk/lock.h: whitespace
- only change - adjust indentation and add some comments on the
- content of these two files. utils.h (which is included in over
- 150 files) contains a lot of unrelated functions which require
- the inclusion of a large number of other headers. At some point
- we should partition its content in a better way.
-
-2007-11-16 21:23 +0000 [r89333-89338] Luigi Rizzo <rizzo@icir.org>
-
- * include/asterisk/logger.h: logger.h does not need options.h
-
- * include/asterisk/utils.h, channels/chan_sip.c,
- include/asterisk/astobj.h, include/asterisk/compat.h,
- include/asterisk/channel.h, include/asterisk/strings.h,
- utils/extconf.c, include/asterisk/frame.h,
- include/asterisk/stringfields.h, include/asterisk/endian.h:
- remove redundant #include "asterisk/compat.h", but make sure that
- asterisk/compiler.h is included everywhere
-
- * main/acl.c, main/asterisk.c: remove duplicate headers. Properly
- check for netdb.h (there is actually tens of places to fix)
-
- * Makefile.rules: put back default optimization to -O6 (previously
- changed by mistake)
-
- * main/frame.c, main/threadstorage.c, apps/app_alarmreceiver.c,
- apps/app_ices.c, channels/chan_iax2.c, apps/app_exec.c,
- channels/chan_skinny.c, main/strcompat.c, pbx/pbx_ael.c,
- apps/app_zapras.c, formats/format_h263.c, cdr/cdr_odbc.c,
- include/asterisk/sha1.h, main/db.c, cdr/cdr_manager.c,
- main/pbx.c, funcs/func_timeout.c, formats/format_wav.c,
- apps/app_softhangup.c, codecs/codec_g726.c, funcs/func_cut.c,
- apps/app_talkdetect.c, apps/app_db.c, funcs/func_channel.c,
- main/privacy.c, funcs/func_iconv.c, pbx/pbx_config.c,
- main/asterisk.c, res/res_odbc.c, include/asterisk/stringfields.h,
- apps/app_voicemail.c, formats/format_sln.c,
- apps/app_authenticate.c, apps/app_readexten.c,
- apps/app_userevent.c, codecs/codec_gsm.c, Makefile.rules,
- apps/app_setcallerid.c, include/asterisk/astmm.h,
- res/res_config_odbc.c, apps/app_osplookup.c, funcs/func_odbc.c,
- apps/app_mp3.c, formats/format_h264.c, apps/app_directory.c,
- main/md5.c, res/res_config_pgsql.c, main/dnsmgr.c,
- funcs/func_version.c, channels/chan_sip.c, funcs/func_lock.c,
- res/res_crypto.c, include/asterisk/cli.h, channels/chan_jingle.c,
- apps/app_forkcdr.c, funcs/func_blacklist.c, main/abstract_jb.c,
- main/file.c, apps/app_sms.c, formats/format_g723.c, main/astmm.c,
- apps/app_stack.c, apps/app_verbose.c, main/dsp.c, main/udptl.c,
- main/autoservice.c, funcs/func_module.c, codecs/codec_adpcm.c,
- cdr/cdr_adaptive_odbc.c, main/devicestate.c, apps/app_image.c,
- formats/format_wav_gsm.c, main/indications.c, pbx/pbx_loopback.c,
- funcs/func_shell.c, include/asterisk/compat.h, apps/app_skel.c,
- main/plc.c, channels/chan_alsa.c, apps/app_externalivr.c,
- formats/format_gsm.c, apps/app_milliwatt.c, res/res_speech.c,
- main/sched.c, apps/app_dial.c, apps/app_page.c, apps/app_disa.c,
- channels/iax2-provision.c, res/res_monitor.c, main/netsock.c,
- apps/app_waitforring.c, main/fixedjitterbuf.c,
- include/asterisk/lock.h, apps/app_chanspy.c, apps/app_cdr.c,
- channels/chan_unistim.c, funcs/func_base64.c, funcs/func_md5.c,
- apps/app_meetme.c, main/sha1.c, funcs/func_vmcount.c,
- res/res_musiconhold.c, cdr/cdr_radius.c, apps/app_followme.c,
- res/res_config_sqlite.c, main/fskmodem.c,
- channels/misdn_config.c, apps/app_controlplayback.c,
- cdr/cdr_csv.c, formats/format_ilbc.c, main/cdr.c,
- channels/chan_phone.c, funcs/func_enum.c, main/dial.c,
- main/manager.c, funcs/func_groupcount.c, cdr/cdr_sqlite.c,
- main/logger.c, main/image.c, apps/app_ivrdemo.c,
- res/res_clioriginate.c, apps/app_nbscat.c, codecs/codec_a_mu.c,
- channels/chan_zap.c, main/slinfactory.c, res/res_convert.c,
- pbx/pbx_lua.c, apps/app_queue.c, apps/app_system.c,
- channels/chan_oss.c, cdr/cdr_tds.c, funcs/func_realtime.c,
- channels/chan_usbradio.c, main/hashtab.c, apps/app_flash.c,
- include/asterisk/strings.h, apps/app_senddtmf.c,
- funcs/func_callerid.c, include/asterisk/time.h,
- channels/chan_local.c, funcs/func_dialgroup.c, funcs/func_env.c,
- apps/app_record.c, funcs/func_strings.c, apps/app_chanisavail.c,
- pbx/pbx_spool.c, apps/app_dumpchan.c, formats/format_pcm.c,
- main/http.c, main/stdtime/localtime.c, codecs/codec_g722.c,
- apps/app_morsecode.c, formats/format_ogg_vorbis.c,
- channels/iax2-parser.c, apps/app_speech_utils.c,
- include/asterisk/logger.h, main/srv.c, apps/app_sendtext.c,
- funcs/func_cdr.c, include/asterisk/md5.h, utils/hashtest2.c,
- utils/ael_main.c, main/audiohook.c, apps/app_mixmonitor.c,
- formats/format_g726.c, channels/chan_vpb.cc, apps/app_dictate.c,
- channels/chan_gtalk.c, funcs/func_logic.c, cdr/cdr_pgsql.c,
- res/res_jabber.c, funcs/func_uri.c, main/io.c,
- include/asterisk/abstract_jb.h, main/channel.c,
- apps/app_minivm.c, res/res_realtime.c, main/dns.c,
- apps/app_rpt.c, channels/chan_mgcp.c, apps/app_adsiprog.c,
- codecs/codec_lpc10.c, apps/app_read.c, codecs/codec_alaw.c,
- res/res_adsi.c, include/asterisk/plc.h,
- apps/app_channelredirect.c, formats/format_vox.c,
- main/cryptostub.c, main/callerid.c, pbx/pbx_dundi.c,
- funcs/func_devstate.c, funcs/func_rand.c, apps/app_readfile.c,
- cdr/cdr_sqlite3_custom.c, main/say.c, res/res_features.c,
- apps/app_sayunixtime.c, apps/app_test.c, main/config.c,
- main/loader.c, main/term.c, main/cli.c, res/res_smdi.c,
- include/asterisk/astobj.h, apps/app_zapscan.c, apps/app_amd.c,
- pbx/pbx_realtime.c, apps/app_url.c, formats/format_jpeg.c,
- include/asterisk/utils.h, apps/app_privacy.c,
- codecs/codec_speex.c, apps/app_echo.c, channels/chan_agent.c,
- funcs/func_math.c, res/res_ael_share.c, pbx/dundi-parser.c,
- apps/app_transfer.c, include/asterisk/manager.h,
- apps/app_playback.c, main/chanvars.c, apps/app_zapbarge.c,
- channels/chan_misdn.c, funcs/func_curl.c,
- channels/chan_features.c, apps/app_macro.c, codecs/codec_ilbc.c,
- res/res_indications.c, apps/app_zapateller.c, main/dlfcn.c,
- include/asterisk/slinfactory.h, utils/hashtest.c, main/utils.c,
- funcs/func_sha1.c, codecs/codec_zap.c, main/enum.c,
- include/asterisk/file.h, main/tdd.c, funcs/func_volume.c,
- res/res_agi.c, main/app.c, apps/app_parkandannounce.c,
- cdr/cdr_custom.c, apps/app_while.c, funcs/func_db.c,
- res/res_limit.c, apps/app_festival.c, apps/app_waitforsilence.c,
- main/translate.c, include/asterisk/config.h, main/jitterbuf.c,
- main/acl.c, apps/app_getcpeid.c, funcs/func_global.c, main/rtp.c,
- funcs/func_extstate.c, apps/app_directed_pickup.c,
- main/adsistub.c, channels/chan_h323.c, codecs/codec_ulaw.c,
- main/event.c, channels/chan_nbs.c, pbx/pbx_gtkconsole.c,
- formats/format_g729.c: 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.).
-
-2007-11-16 19:51 +0000 [r89331-89332] Mark Michelson <mmichelson@digium.com>
-
- * main/manager.c: Fixing a problem pointed out by Qwell
-
- * main/manager.c: Added some locks that should have been around
- astman_send_error, at least according to the comments. (closes
- issue #11258, reported and patched by eliel)
-
-2007-11-16 19:26 +0000 [r89329-89330] Steve Murphy <murf@digium.com>
-
- * main/pbx.c: This corrects a hashtab removal, given a bad argument
-
- * main/pbx.c, res/res_features.c: This fixes a problem with pattern
- ranges; and corrects a situation in res_features, where an
- extension would be created with the name Zap/51, as an example.
- THe / is bad because it would tend to mean that the 51 is to be
- cid matched.
-
-2007-11-16 18:48 +0000 [r89328] Luigi Rizzo <rizzo@icir.org>
-
- * build_tools/make_buildopts_h: both md5sum and variable
- substitutions such as ${BUILDSUM:0:8} are not available in
- FreeBSD. For the time being, put in a workaround so we can build
- the system, and wait for the result of the discussion on whether
- we can store the md5 as a string rather than 4 ints (if so, we
- won't need more complex tricks with awk or sed for splitting the
- md5). 1.4 will be fixed when we decide the issue.
-
-2007-11-16 17:11 +0000 [r89327] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Adding confirmation playback when
- forwarding voicemail messages. This will attempt to play the
- name(s) of the person(s) to whom you are forwarding the message
- prior to prompting for prepending. If no name is found, the
- extension is read back verbatim. (closes issue #9046, reported
- and patched by jaroth)
-
-2007-11-16 16:56 +0000 [r89326] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, include/asterisk/module.h, build_tools/make_buildopts_h,
- main/loader.c: Merged revisions 89325 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89325 | kpfleming | 2007-11-16 10:47:46 -0600 (Fri, 16 Nov 2007)
- | 4 lines To help combat problems where people build external
- modules (asterisk-addons or others) and then change the build
- options of the Asterisk build in a way that makes the
- incompatible without warning, this commit introduces an MD5
- signature of the important build-time options and includes that
- signature into modules when they are built. When the loader loads
- one of these modules and notices the problem, it will emit a
- warning to console and refuse to initialize the module, as doing
- so could cause the system to be unstable or even crash. If you
- upgrade to this version of Asterisk, you must rebuild *all* of
- your modules that came from other sources before trying to run
- this version. If you are using Digium's G.729 binary codec
- module, you will need v33 or newer. ........
-
-2007-11-16 15:44 +0000 [r89324] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 89323 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89323 | mmichelson | 2007-11-16 09:28:22 -0600 (Fri, 16 Nov
- 2007) | 5 lines Make realtime queues accessible from the
- QUEUE_MEMBER_COUNT function. (closes issue #11271, reported and
- patched by atis, with small modifications from me) ........
-
-2007-11-16 10:07 +0000 [r89322] Luigi Rizzo <rizzo@icir.org>
-
- * include/asterisk/config.h, main/config.c: add a small new
- function to retrieve variables from a config once we have a
- pointer to the category.
-
-2007-11-16 10:06 +0000 [r89321] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c: fixed #10631, about one way audio. thanks
- IgorG again.
-
-2007-11-16 09:51 +0000 [r89320] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_oss.c: move the inner part of config file parsing
- to a separate function, so it can be reused in the implementation
- of cli commands when they have a similar syntax.
-
-2007-11-16 08:54 +0000 [r89319] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c: fixed compilation of chan_misdn, #11269,
- thanks IgorG.
-
-2007-11-15 23:50 +0000 [r89299-89312] Tilghman Lesher <tlesher@digium.com>
-
- * main/utils.c, include/asterisk/stringfields.h: If we're going to
- be passing a negative value for the size of a stringfield, in
- order to indicate something, then using an UNSIGNED parameter is
- bad, mmmmmkay?
-
- * Makefile, /: Merged revisions 89302 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89302 | tilghman | 2007-11-15 12:37:38 -0600 (Thu, 15 Nov 2007)
- | 2 lines Start Asterisk in Debian at a more reasonable time
- (since zaptel is at level 20) ........
-
- * /, channels/misdn/isdn_lib.c: Merged revisions 89301 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r89301 | tilghman | 2007-11-15 12:23:14 -0600 (Thu, 15
- Nov 2007) | 2 lines Fix an uninitialized memory read found by
- valgrind ........
-
- * apps/app_zapscan.c: Fix trunk breakage due to chan->lock being
- renamed.
-
- * /, channels/chan_iax2.c: Merged revisions 89298 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89298 | tilghman | 2007-11-15 12:05:56 -0600 (Thu, 15 Nov 2007)
- | 5 lines Yet another memory corruption issue. Reported by: atis
- Patch by: tilghman Fixes issue #10923 ........
-
-2007-11-15 17:27 +0000 [r89297] Russell Bryant <russell@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 89296 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89296 | russell | 2007-11-15 11:19:28 -0600 (Thu, 15 Nov 2007) |
- 8 lines Update the SLAStation application to account for the case
- where the SLA thread has a call out to the station, but the user
- has pressed a line button to answer the call instead of picking
- up the handset. If they do, the phone sends out a new INVITE. So,
- the SLAStation app must check to see if it is picking up a
- ringing trunk, and ensure that the other stations stop ringing.
- (reported internally, patched by me, tested by mogorman) ........
-
-2007-11-15 16:50 +0000 [r89294-89295] Steve Murphy <murf@digium.com>
-
- * main/pbx.c: Get rid of a previously missed ast_log call for
- debug, no longer nec.
-
- * main/pbx.c: Perhaps I went overboard on initializing things. I
- can remove unnecc. stuff later. A few bug fixes. Killing small
- bugs on the way to killing bigger ones. Removed locking on
- hashtabs; there's plenty of locks already being taken. A small
- bug in the root_tree hashtab compare func.
-
-2007-11-15 16:20 +0000 [r89293] Luigi Rizzo <rizzo@icir.org>
-
- * main/channel.c, apps/app_channelredirect.c, main/manager.c,
- res/res_features.c, apps/app_softhangup.c,
- include/asterisk/channel.h, include/asterisk/lock.h,
- apps/app_senddtmf.c: access channel locks through
- ast_channel_lock/unlock/trylock and not through ast_mutex
- primitives. To detect all occurrences, I have renamed the lock
- field in struct ast_channel so it is clear that it shouldn't be
- used directly. There are some uses in res/res_features.c (see
- details of the diff) that are error prone as they try and lock
- two channels without caring about the order (or without
- explaining why it is safe).
-
-2007-11-15 15:39 +0000 [r89290-89291] Joshua Colp <jcolp@digium.com>
-
- * UPGRADE.txt: Fix typo in UPGRADE.txt. 'increase' should have been
- used, not 'increasing'.
-
- * channels/chan_sip.c, channels/chan_h323.c,
- channels/misdn_config.c: And file said... let trunk build again!
- Accomplished by some more constification, and marking a function
- in chan_sip as purposely unused until it is fixed up.
-
-2007-11-15 14:58 +0000 [r89287-89289] Mark Michelson <mmichelson@digium.com>
-
- * main/manager.c, /: Merged revisions 89288 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89288 | mmichelson | 2007-11-15 08:57:28 -0600 (Thu, 15 Nov
- 2007) | 3 lines Undoing previous commit since I realize it was
- wrong ........
-
- * main/manager.c, /: Merged revisions 89286 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89286 | mmichelson | 2007-11-15 08:54:10 -0600 (Thu, 15 Nov
- 2007) | 4 lines Adding a missing mutex unlock. (closes issue
- 11256, reported and patched by ys) ........
-
-2007-11-15 12:21 +0000 [r89278-89285] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Always relying on the responses when
- crossing NAT's are not a good solution, it breaks communication.
- Rizzo - you need to implement a configuration option for this
- code. It's good, but maybe should be off by default.
-
- * /, channels/chan_sip.c: Merged revisions 89281 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89281 | oej | 2007-11-15 12:26:22 +0100 (Tor, 15 Nov 2007) | 6
- lines Don't send re-invites during pending INVITE transactions.
- Patch by one47 - thanks! Closes issue #9305 ........
-
- * /, channels/chan_sip.c: Merged revisions 89280 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89280 | oej | 2007-11-15 12:15:09 +0100 (Tor, 15 Nov 2007) | 5
- lines Improve support for multipart messages. Code by gasparz,
- changes by me (mostly formatting). Thanks, gasparz! Closes issue
- #10947 ........
-
- * channels/chan_sip.c: Exit early instead of deciding to exit after
- processing the message.
-
- * channels/chan_sip.c, configs/sip.conf.sample: Add support for
- application/dtmf SIP INFO dtmf handling. Yep, another way of
- handling DTMF in SIP. Totally undocumented, but implemented in
- enough devices so we have to support it. Code by sergee, small
- changes by oej. Closes issue #11049
-
-2007-11-15 01:42 +0000 [r89277] Steve Murphy <murf@digium.com>
-
- * main/pbx.c: Had trouble playing with parking; spent a long time
- trying to reason out MATCHMORE mode. made these updates and xfers
- on zaptel lines seem to work ok now
-
-2007-11-15 00:01 +0000 [r89273-89276] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/app.c: Merged revisions 89275 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89275 | tilghman | 2007-11-14 17:23:58 -0600 (Wed, 14 Nov 2007)
- | 5 lines When a recording ends with '#', we are improperly
- trimming an extra 200ms from the recording. Reported by: sim
- Patch by: tilghman Closes issue #11247 ........
-
- * main/channel.c: Typo
-
- * main/channel.c: Add callerid to the Hangup manager event.
- Reported by: outtolunc Patch by: outtolunc Closes issue #11248
-
-2007-11-14 18:05 +0000 [r89271-89272] Steve Murphy <murf@digium.com>
-
- * main/pbx.c: Rescaled the weights of the patterns to give
- something more independent of pattern length; and make . less
- likely to win. Question: which should win for 14102241145--
- _1xxxxxxx. or _XXXXXXXXXXX -- right now, the pure X pattern will
- win.
-
- * main/pbx.c: A further problem highlighted by 11233 has been
- resolved; a certain combination of patterns in a certain order,
- led to a malformed trie, due to a ptr not being initialized in
- the loop. Also, some tree printing prettifications.
-
-2007-11-14 15:13 +0000 [r89269-89270] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_phone.c, channels/chan_zap.c, res/res_jabber.c,
- res/res_config_sqlite.c, main/config.c, res/res_odbc.c: One more
- typo in config.c; and missed conversions due to the constifying
- of ast_variable_new parameters
-
- * main/config.c: Typo
-
-2007-11-14 13:18 +0000 [r89268] Luigi Rizzo <rizzo@icir.org>
-
- * include/asterisk/acl.h, channels/chan_sip.c,
- include/asterisk/config.h, channels/chan_agent.c, res/res_adsi.c,
- main/acl.c, pbx/dundi-parser.c, apps/app_queue.c,
- channels/chan_iax2.c, main/enum.c, channels/chan_oss.c,
- apps/app_playback.c, main/config.c, pbx/dundi-parser.h,
- include/asterisk/abstract_jb.h, main/manager.c,
- channels/chan_skinny.c, apps/app_minivm.c, main/abstract_jb.c,
- main/logger.c, pbx/pbx_dundi.c, apps/app_directory.c,
- apps/app_voicemail.c: 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.
-
-2007-11-14 03:22 +0000 [r89263-89266] Russell Bryant <russell@digium.com>
-
- * main/hashtab.c, include/asterisk/hashtab.h: Fix up various coding
- guidelines issues ... - handle memory allocation failures - add
- an ast_ prefix to a publicly exported function - put curly braces
- in the right places - add a bunch of spaces where they should be
- be used
-
- * res/res_clioriginate.c: - Use the ARRAY_LEN macro in a couple
- places - return errors from load_module / unload_module
-
- * apps/app_dial.c: Use BEGIN_OPTIONS / END_OPTIONS to make the
- syntax highlighting in my editor happy
-
- * apps/app_queue.c: Instead of reserving 800 bytes for periodic
- announcements, use an array of ast_str pointers and only alloate
- space for the strings as needed.
-
-2007-11-14 01:16 +0000 [r89262] Joshua Colp <jcolp@digium.com>
-
- * main/srv.c, /: Merged revisions 89260 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89260 | file | 2007-11-13 21:15:12 -0400 (Tue, 13 Nov 2007) | 4
- lines Return the proper value when the srv_callback function
- executes properly. (closes issue #11240) Reported by: jtodd
- ........
-
-2007-11-14 01:15 +0000 [r89261] Russell Bryant <russell@digium.com>
-
- * apps/app_queue.c: Convert most of the strings in the call_queue
- struct to use stringfields.
-
-2007-11-14 00:54 +0000 [r89259] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/channel.c, main/pbx.c: use simpler technique for removing
- known entries from lists
-
-2007-11-14 00:33 +0000 [r89258] Russell Bryant <russell@digium.com>
-
- * main/image.c: - Simplify removing an item from a list - move a
- verbose message to after the item is added to the list - make use
- of the ARRAY_LEN macro in one spot
-
-2007-11-13 23:43 +0000 [r89256-89257] Steve Murphy <murf@digium.com>
-
- * main/pbx.c: This hopefully will fix the re-opened 11233. Hadn't
- covered the case of a context with no patterns. (blush)
-
- * main/pbx.c: closes issue #11233 -- where some fine points in the
- algorithm to build the tree needed to be corrected. Many thanks
- for the test case, jtodd
-
-2007-11-13 21:01 +0000 [r89250-89253] Russell Bryant <russell@digium.com>
-
- * include/asterisk/lock.h: This fixes a build error on my mac. It
- also works on my linux box. Let me know if it breaks any other
- platform ...
-
- * res/res_features.c: Fix a typo pointed out by outtolunc, thanks
- :)
-
- * channels/chan_sip.c: - Convert initialization of a struct to C99
- style instead of GNU style - Fix a minor spelling error in a
- comment
-
- * res/res_features.c, CHANGES: Update the ParkedCall application to
- grab the first available parked call if no parked extension is
- provided as an argument. (closes issue #10803) Reported by:
- outtolunc Patches: res_features-parkedcall-any.diff4 uploaded by
- outtolunc (license 237) - modified by me to work a bit
- differently ...
-
-2007-11-13 19:48 +0000 [r89249] Jason Parker <jparker@digium.com>
-
- * /, res/res_features.c: Merged revisions 89248 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #11237) ........ r89248 | qwell | 2007-11-13 13:47:45 -0600
- (Tue, 13 Nov 2007) | 7 lines Revert change from revision 67064.
- It is documented behavior that if a parking extension already
- exists while using PARKINGEXTEN, dialplan execution will
- continue. If blind transferring to a Park with PARKINGEXTEN, you
- must keep this in mind, and handle the failure yourself. Issue
- 11237, reported by jon. ........
-
-2007-11-13 17:41 +0000 [r89247] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 89246 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89246 | tilghman | 2007-11-13 11:34:11 -0600 (Tue, 13 Nov 2007)
- | 2 lines If we set a value for qualify, we should actually pay
- attention to it, instead of overriding the value ........
-
-2007-11-13 16:03 +0000 [r89242] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_mixmonitor.c: Merged revisions 89241 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r89241 | mmichelson | 2007-11-13 10:02:02 -0600 (Tue, 13
- Nov 2007) | 5 lines Reverting commit made in revision 89205 since
- it is unnecessary. Thanks to Kevin for pointing this out ........
-
-2007-11-13 14:03 +0000 [r89240] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/utils.c: Merged revisions 89239 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89239 | tilghman | 2007-11-13 07:51:53 -0600 (Tue, 13 Nov 2007)
- | 4 lines Debugging is running into the 16-lock limit. Increase
- to avoid. (This define is only effective when debugging is turned
- on, so there's no effect for most installations.) ........
-
-2007-11-13 01:19 +0000 [r89206-89207] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_mixmonitor.c: There is the potential to copy
- uninitialized memory into the mixmonitor->post_process string.
- This fix prevents that.
-
- * /, apps/app_mixmonitor.c: Merged revisions 89205 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r89205 | mmichelson | 2007-11-12 18:56:46 -0600 (Mon, 12
- Nov 2007) | 5 lines Some sanity checking for MixMonitor. If only
- 1 argument is given, then the args.options and args.post_process
- strings are uninitialized and could contain garbage. This change
- handles this situation properly by only using arguments that we
- have parsed. ........
-
-2007-11-13 00:19 +0000 [r89202-89203] Jason Parker <jparker@digium.com>
-
- * Makefile: oops, somebody left out the directory here...
-
- * channels/chan_unistim.c, res/res_features.c, main/ast_expr2f.c,
- include/asterisk/config.h, res/res_convert.c, res/res_crypto.c,
- pbx/pbx_lua.c, include/asterisk/cli.h, include/asterisk/pbx.h,
- res/res_config_sqlite.c, res/res_monitor.c,
- include/asterisk/stringfields.h, res/res_clioriginate.c: Doxygen
- fixes. Also fix a common typo I kept seeing (arguement) in
- various files. Closes issue #11222, patch by snuffy (with
- arguement > argument by me).
-
-2007-11-12 23:33 +0000 [r89196-89201] Steve Murphy <murf@digium.com>
-
- * utils/hashtest.c: Don't forget the ASTERISK_VERSION for the sake
- of the mtx_prof stuff.
-
- * include/asterisk/hashtab.h: Thanks to snuffy for this doxygen
- update to hashtab.h; closes issue #11223
-
- * main/hashtab.c, include/asterisk/hashtab.h: Thanks to snuff-work,
- who brought up that these fixes might need to be made.
-
-2007-11-12 20:48 +0000 [r89195] Jason Parker <jparker@digium.com>
-
- * main/pbx.c, /: Merged revisions 89194 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89194 | qwell | 2007-11-12 14:46:52 -0600 (Mon, 12 Nov 2007) | 1
- line Fix a typo pointed out by De_Mon on #asterisk-dev ........
-
-2007-11-12 20:16 +0000 [r89190] Kevin P. Fleming <kpfleming@digium.com>
-
- * utils/Makefile, utils/hashtest.c: (closes issue #11221) Reported
- by: eliel Patches: utils.Makefile.patch uploaded by eliel
- (modified by me) (license 64)
-
-2007-11-12 18:44 +0000 [r89186] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, pbx/pbx_realtime.c, pbx/pbx_dundi.c,
- funcs/func_logic.c, apps/app_exec.c, apps/app_queue.c,
- apps/app_mixmonitor.c, cdr/cdr_manager.c: Based on a note in
- asterisk-dev by Brian Capouch, I determined I too agressive in
- not initializing arrays passed to pbx_substitute_variables_xxxx;
- I reviewed the code (again) and hopefully found every possible
- spot where substitute_variables is called conditionally, and made
- sure the char array involved was set to a null string.
-
-2007-11-12 17:44 +0000 [r89185] Tilghman Lesher <tlesher@digium.com>
-
- * main/channel.c, /, channels/chan_sip.c: Merged revisions 89184
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89184 | tilghman | 2007-11-12 11:29:17 -0600 (Mon, 12 Nov 2007)
- | 5 lines Fix two cases of memory corruption caused by background
- threads. Reported by: atis Patch by: tilghman Fixes issue #10923
- ........
-
-2007-11-12 13:36 +0000 [r89178-89179] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /, configs/misdn.conf.sample: Merged
- revisions 89173 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89173 | crichter | 2007-11-12 12:26:48 +0100 (Mo, 12 Nov 2007) |
- 1 line if we're NT and no number was dialed and overlapdial is
- set, we wait for the ISDN timeout instead of starting our own
- timer. added a comment for the misdn.conf.sample for the
- overlapdial config option. ........
-
- * channels/misdn/isdn_lib_intern.h, channels/chan_misdn.c, /,
- channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c:
- Merged revisions 89172 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89172 | crichter | 2007-11-12 12:23:57 +0100 (Mo, 12 Nov 2007) |
- 1 line added restart all interfaces Restart_Indicator, to
- automatically send a RESTART after the L2 of a PTP Port comes up.
- Also fixed some places where we have send a RELEASE without need
- for it. ........
-
-2007-11-12 13:26 +0000 [r89177] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_unistim.c, utils/hashtest.c: Fix building on
- FreeBSD by including/not including some headers. (closes issue
- #11218) Reported by: ys Patches: trunk89169.diff uploaded by ys
- (license 281)
-
-2007-11-12 13:22 +0000 [r89174-89176] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged
- revisions 89171 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89171 | crichter | 2007-11-12 12:13:13 +0100 (Mo, 12 Nov 2007) |
- 1 line fixed a state/event issue with overlapdial=yes when no
- extension matched. removed the general sending of a
- RELEASE_COMPLETE when we receive a RELEASE, this is done by
- mISDNuser/mISDN. This makes it possible to use asterisk-1.4 with
- mISDN trunk, but requires users of mISDN/mISDNuser-1.1.X to
- upgrade to at least mISDNuser-1.1.6 (when using the NT mode at
- all) ........
-
- * /, channels/misdn/isdn_lib.c: Merged revisions 89170 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r89170 | crichter | 2007-11-12 10:57:23 +0100 (Mo, 12
- Nov 2007) | 1 line fixed the support for CW and therefore for the
- reject_cause option. ........
-
- * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
- channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample,
- channels/misdn/isdn_lib.c, channels/misdn_config.c: Merged
- revisions 89169 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89169 | crichter | 2007-11-12 10:45:36 +0100 (Mo, 12 Nov 2007) |
- 1 line aded ntkeepcalls option, to avoid droÃpping calls when the
- L2 goes down on a PTP link. There are some pbx which do turn off
- the L1 for a very short while and restart it immediately.
- normally T310 should be started and after 10 seconds or so the
- calls should be dropped, this is a simple fix wihtout this timer.
- ........
-
-2007-11-09 18:57 +0000 [r89130-89132] Jason Parker <jparker@digium.com>
-
- * configs/usbradio.conf.sample (added): Add usbradio.conf.sample
- from branches/1.4/configs - r84162. It was mistakenly deleted in
- 1.4 without ever being merged to trunk. Reported by eliel on
- #asterisk-dev.
-
- * cdr/cdr_sqlite3_custom.c, configs/cdr_sqlite3_custom.conf
- (removed), configs/cdr_sqlite3_custom.conf.sample (added): Fix a
- few potential deadlocks in cdr_sqlite3_custom. (also rename
- sample config to .sample) Closes issue #11208, patch by Laureano.
-
-2007-11-09 16:00 +0000 [r89129] Steve Murphy <murf@digium.com>
-
- * res/ael/pval.c, utils/Makefile, main/pbx.c, main/hashtab.c
- (added), main/Makefile, utils/hashtest.c (added), pbx/pbx_ael.c,
- include/asterisk/hashtab.h (added), main/config.c: This is the
- perhaps the biggest, boldest, most daring change I've ever
- committed to trunk. Forgive me in advance any disruption this may
- cause, and please, report any problems via the bugtracker. The
- upside is that this can speed up large dialplans by 20 times (or
- more). Context, extension, and priority matching are all fairly
- constant-time searches. I introduce here my hashtables
- (hashtabs), and a regression for them. I would have used the
- ast_obj2 tables, but mine are resizeable, and don't need the
- object destruction capability. The hashtab stuff is well tested
- and stable. I introduce a data structure, a trie, for extension
- pattern matching, in which knowledge of all patterns is
- accumulated, and all matches can be found via a single traversal
- of the tree. This is per-context. The trie is formed on the first
- lookup attempt, and stored in the context for future lookups.
- Destruction routines are in place for hashtabs and the pattern
- match trie. You can see the contents of the pattern match trie by
- using the 'dialplan show' cli command when 'core set debug' has
- been done to put it in debug mode. The pattern tree traversal
- only traverses those parts of the tree that are interesting. It
- uses a scoreboard sort of approach to find the best match. The
- speed of the traversal is more a function of the length of the
- pattern than the number of patterns in the tree. The tree also
- contains the CID matching patterns. See the source code comments
- for details on how everything works. I believe the approach
- general enough that any issues that might come up involving fine
- points in the pattern matching algorithm, can be solved by just
- tweaking things. We shall see. The current pattern matcher is
- fairly involved, and replicating every nuance of it is difficult.
- If you find and report problems, I will try to resolve than as
- quickly as I can. The trie and hashtabs are added to the existing
- context and exten structs, and none of the old machinery has been
- removed for the sake of the multitude of functions that use them.
- In the future, we can (maybe) weed out the linked lists and save
- some space.
-
-2007-11-08 23:53 +0000 [r89124-89126] Jason Parker <jparker@digium.com>
-
- * /, main/say.c: Merged revisions 89125 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #11203) ........ r89125 | qwell | 2007-11-08 17:52:35 -0600
- (Thu, 08 Nov 2007) | 4 lines Properly say the seconds here..
- Issue 11203, fix described by vma. ........
-
- * pbx/pbx_lua.c: Add check_hangup() method to pbx_lua, which can be
- used to check whether it is time to hangup a channel. Closes
- issue #11202, patch by mnicholson
-
-2007-11-08 22:33 +0000 [r89122-89123] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: app_voicemail failed to build when
- compiling with IMAP_STORAGE Now it does not.
-
- * main/threadstorage.c: AST_LIST_REMOVE_CURRENT takes only one
- argument. Thanks to snuffy for pointing this out on IRC
-
-2007-11-08 21:27 +0000 [r89121] Joshua Colp <jcolp@digium.com>
-
- * funcs/func_env.c: Make func_env build again.
-
-2007-11-08 21:01 +0000 [r89120] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 89119 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89119 | mmichelson | 2007-11-08 15:00:08 -0600 (Thu, 08 Nov
- 2007) | 7 lines Rework of the commit I made yesterday to use the
- already built-in ast_uri_decode function as opposed to my
- home-rolled one. Also added comments. Thanks to oej for pointing
- me in the right direction ........
-
-2007-11-08 20:39 +0000 [r89118] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_features.c: convert this code to a more efficient
- idiom
-
-2007-11-08 18:49 +0000 [r89116-89117] Jason Parker <jparker@digium.com>
-
- * res/res_smdi.c: Change a warning to a notice. Issue #11195, patch
- by eliel
-
- * /, configs/cdr_adaptive_odbc.conf.sample,
- configs/res_odbc.conf.sample: Merged revisions 89115 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #11195) ........ r89115 | qwell | 2007-11-08 12:45:15 -0600
- (Thu, 08 Nov 2007) | 4 lines Avoid warnings on load when using
- sample configuration files. Issue 11195, patch by eliel. ........
-
-2007-11-08 17:32 +0000 [r89113-89114] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_readfile.c, funcs/func_env.c: Add the FILE() dialplan
- function and deprecate ReadFile.
-
- * channels/chan_features.c: Fix missed conversion to linkedlists
- macro change
-
-2007-11-08 16:51 +0000 [r89112] Mark Michelson <mmichelson@digium.com>
-
- * /: Blocking changes from previous 1.4 commit
-
-2007-11-08 09:21 +0000 [r89108-89110] Luigi Rizzo <rizzo@icir.org>
-
- * apps/app_voicemail.c: use %f instead of %lf (the 'l' is ignored
- anyways).
-
- * main/audiohook.c: use %d and cast to int instead of %zd for
- size_t object, this helps portability.
-
- * channels/chan_unistim.c: initialize a variable to silence
- compiler. The type of warnings emitted depends on the
- optimization level, at the lower levels the compiler doesn't
- always understand what the programmer has in mind. In this case I
- could not understand it either.
-
-2007-11-08 05:36 +0000 [r89106-89107] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/srv.c, /: Merged revisions 89105 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89105 | kpfleming | 2007-11-08 00:26:47 -0500 (Thu, 08 Nov 2007)
- | 2 lines fix a glaring bug in the new SRV record handling that
- would cause incorrect weight sorting ........
-
- * main/autoservice.c, main/frame.c, apps/app_meetme.c,
- res/res_features.c, funcs/func_strings.c, main/devicestate.c,
- res/res_musiconhold.c, channels/chan_iax2.c, apps/app_followme.c,
- codecs/codec_zap.c, res/res_jabber.c, main/indications.c,
- main/astobj2.c, main/config.c, main/loader.c, main/cli.c,
- main/cdr.c, main/channel.c, main/manager.c, res/res_agi.c,
- main/logger.c, main/app.c, main/image.c, res/res_speech.c,
- main/sched.c, main/pbx.c, main/translate.c, res/res_crypto.c,
- channels/chan_agent.c, utils/astman.c, apps/app_queue.c,
- channels/iax2-parser.c, main/srv.c,
- include/asterisk/linkedlists.h, main/file.c, pbx/pbx_dundi.c,
- main/event.c, main/audiohook.c, res/res_odbc.c, main/asterisk.c,
- apps/app_voicemail.c: improve linked-list macros in two ways: -
- the *_CURRENT macros no longer need the list head pointer
- argument - add AST_LIST_MOVE_CURRENT to encapsulate the
- remove/add operation when moving entries between lists
-
-2007-11-08 05:00 +0000 [r89104] Tilghman Lesher <tlesher@digium.com>
-
- * /, doc/valgrind.txt: Merged revisions 89103 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89103 | tilghman | 2007-11-07 22:55:19 -0600 (Wed, 07 Nov 2007)
- | 2 lines Typo ........
-
-2007-11-08 02:28 +0000 [r89096-89102] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 89101 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89101 | file | 2007-11-07 22:26:48 -0400 (Wed, 07 Nov 2007) | 4
- lines Do not add a sip: to the beginning of the To URI unless
- needed. (closes issue #10756) Reported by: goestelecom ........
-
- * /, channels/chan_sip.c: Merged revisions 89099 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89099 | file | 2007-11-07 21:28:56 -0400 (Wed, 07 Nov 2007) | 6
- lines Improve the devicestate logic for multiple devices. If any
- are available then the extension is considered available. (closes
- issue #10164) Reported by: nic_bellamy Patches:
- sip-hinting-svn-branch-1.4.patch uploaded by nic (license 299)
- ........
-
- * /, channels/chan_sip.c: Merged revisions 89097 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89097 | file | 2007-11-07 21:11:25 -0400 (Wed, 07 Nov 2007) | 8
- lines Add support for allowing one outgoing transaction. This
- means if a response comes back out of order chan_sip will still
- handle it. I dream of a chan_sip with real transaction support.
- (closes issue #10946) Reported by: flefoll (closes issue #10915)
- Reported by: ramonpeek (closes issue #9567) Reported by:
- atca_pres ........
-
- * /, channels/chan_sip.c: Merged revisions 89095 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89095 | file | 2007-11-07 19:53:25 -0400 (Wed, 07 Nov 2007) | 4
- lines If callerid is configured in sip.conf use that for checking
- the presence of an extension in the dialplan. (closes issue
- #11185) Reported by: spditner ........
-
-2007-11-07 23:47 +0000 [r89094] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 89093 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89093 | tilghman | 2007-11-07 17:39:37 -0600 (Wed, 07 Nov 2007)
- | 7 lines The member refcount must be incremented, to avoid using
- it after deallocation. A huge thanks go to lvl- for patiently
- providing the necessary valgrind output that was necessary to
- finding this problem of memory corruption. Reported by: lvl-
- Patch by: tilghman Closes issue #11174 ........
-
-2007-11-07 23:18 +0000 [r89091-89092] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: If imapfolder has been specified in
- voicemail.conf, we should not connect to INBOX... ever. It may
- not exist. (closes issue #11151, reported by selsky, patched by
- me)
-
- * /, channels/chan_sip.c: Merged revisions 89090 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89090 | mmichelson | 2007-11-07 16:40:35 -0600 (Wed, 07 Nov
- 2007) | 6 lines This patch makes it possible for SIP phones to
- dial extensions defined with '#' characters in extensions.conf
- AND maintain their escaped characters when forming URI's (closes
- issue #10681, reported by cahen, patched by me, code review by
- file) ........
-
-2007-11-07 22:09 +0000 [r89089] Steve Murphy <murf@digium.com>
-
- * /, res/res_jabber.c, cdr/cdr_tds.c: Merged revisions 89088 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89088 | murf | 2007-11-07 14:40:28 -0700 (Wed, 07 Nov 2007) | 1
- line In response to 10578, I just ran 1.4 thru valgrind; some of
- the config leakage I've already fixed, but it doesn't hurt to
- double check. I found and fixed leaks in res_jabber, cdr_tds,
- pbx_ael. Nothing major, tho. ........
-
-2007-11-07 17:45 +0000 [r89086] Joshua Colp <jcolp@digium.com>
-
- * channels/h323/ast_h323.cxx: Minor change so chan_h323 builds
- again.
-
-2007-11-07 13:12 +0000 [r89082-89084] Luigi Rizzo <rizzo@icir.org>
-
- * Makefile: remove enter/exit comments when handling subdirectory.
- If we really want them we can remove the --no-print-directory
-
- * main/loader.c: remove a debugging message which i forgot in.
-
- * Makefile: match changes in menuselect's Makefile
-
-2007-11-07 04:21 +0000 [r89077-89081] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_playback.c: Suppress erroneous warnings on load.
- Reported by: eliel Patch by: eliel Closes issue #11177
-
- * /, configs/extensions.ael.sample: Merged revisions 89079 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89079 | tilghman | 2007-11-06 22:07:49 -0600 (Tue, 06 Nov 2007)
- | 5 lines Suppress AEL warnings on load. Reported by: eliel Patch
- by: eliel Closes issue #11178 ........
-
- * channels/chan_zap.c, configs/zapata.conf.sample: Provide the
- ability to directly manipulate the TON/NPI bits in the
- dialstring. Reported by: thetatag Patch by:
- thetatag/stevens/tilghman Closes issue #5331
-
- * contrib/utils/eagi_proxy.c (added): Add contributed EAGI proxy,
- which provides FastAGI functionality for EAGI, while also
- buffering the audio stream. Reported by: devil_slayer Patch by:
- devil_slayer Closes issue #8921
-
-2007-11-07 00:16 +0000 [r89076] Russell Bryant <russell@digium.com>
-
- * main/astmm.c: Fix another CLI command so it doesn't run the real
- code when called for initialization.
-
-2007-11-07 00:04 +0000 [r89075] Mark Michelson <mmichelson@digium.com>
-
- * doc/tex/imapstorage.tex: Adding documentation regarding
- imapfolder, imapgreetings, and greetingsfolder options in
- voicemail.conf (closes issue #11133, reported by selsky, patched
- by blitzrage)
-
-2007-11-07 00:00 +0000 [r89073-89074] Russell Bryant <russell@digium.com>
-
- * include/asterisk/agi.h, res/res_agi.c, CHANGES: Print out the
- channel name as a prefix to the "agi debug" output. This makes
- AGI debugging on busy systems much easier. (closes issue #10730)
- Reported by: junky Patches: agi_debug_chan.diff uploaded by junky
- (license 177) 20070923_10730.diff uploaded by mvanbaak (license
- 7)
-
- * apps/app_meetme.c, CHANGES: Added the ability to do "meetme
- concise" with the "meetme" CLI command. This extends the concise
- capabilities of this CLI command to include listing all
- conferences, instead of an addition to the other sub commands for
- the "meetme" command. (closes issue #11078) Reported by: jthomas
- Patches: meetme-concise.patch uploaded by jthomas (license 293)
-
-2007-11-06 23:08 +0000 [r89072] Joshua Colp <jcolp@digium.com>
-
- * main/pbx.c: Fix up some PBX logic that became broken. The code
- would exit prematurely when it should have been collecting more
- digits. (closes issue #11175) Reported by: pj
-
-2007-11-06 22:51 +0000 [r89071] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_jingle.c, channels/chan_phone.c,
- codecs/codec_g722.c, main/frame.c, channels/chan_sip.c,
- channels/chan_skinny.c, main/translate.c, channels/chan_h323.c,
- main/file.c, channels/chan_gtalk.c, include/asterisk/frame.h,
- main/rtp.c, channels/chan_mgcp.c, include/asterisk/translate.h:
- 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.)
-
-2007-11-06 22:36 +0000 [r89070] Mark Michelson <mmichelson@digium.com>
-
- * configs/queues.conf.sample, CHANGES, apps/app_queue.c: Adding the
- queue strategy wrandom (closes issue #10942, reported and patched
- by julianjm, documentation changes by me)
-
-2007-11-06 22:15 +0000 [r89069] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c, doc/tex/channelvariables.tex, CHANGES: Added
- the S() and L() options to the MeetMe application. These are
- pretty much identical to the S() and L() options to Dial(). They
- let you set timeouts for the conference, as well as have warning
- sounds played to let the caller know how much time is left, and
- when it is running out. (closes issue #8030) Reported by: areski
- Patches: meetme_timeout_timelimit_v2.patch uploaded by areski
- (license 29)
-
-2007-11-06 22:05 +0000 [r89068] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Added CLI and manager commands for changing a
- queue member's penalty (closes issue #9374, reported and
- initially patched by wuwu, intermediate patch by eliel, and final
- patch by me)
-
-2007-11-06 22:01 +0000 [r89067] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Add some more locking as well as API update
- for libss7 for new transport types
-
-2007-11-06 21:08 +0000 [r89062] Steve Murphy <murf@digium.com>
-
- * /, main/config.c: Merged revisions 89036 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89036 | murf | 2007-11-06 10:52:50 -0700 (Tue, 06 Nov 2007) | 1
- line closes issue #8786 - where the [catname](!) and
- [catname](othercat1,othercat2,...) notation gets dropped across a
- ConfigUpdate (or any other thing that would cause a config file
- to be written). While I was at it, I also cleaned up some of the
- destroy routines to free up comments, which was not being done.
- Made sure the new struct I introduced is also cleaned up properly
- at destruction time. My code handles multiple template
- inclusions. Many thanks to ssokol for his patch, which, while not
- literally used in the final merge, served as a foundation for the
- fix. ........
-
-2007-11-06 20:55 +0000 [r89057] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: Remove native bridging check for DTMF based
- transfers. Thanks to the last batch of RTP changes it is no
- longer required for the media stream to go through Asterisk if
- DTMF is going over signalling. It will simply reinvite back as
- needed. (closes issue #11172) Reported by: ibc
-
-2007-11-06 20:32 +0000 [r89055] Mark Michelson <mmichelson@digium.com>
-
- * res/res_features.c: Instead of trying to callback a local channel
- on a failed attended transfer, call the device that made the
- transfer instead. This makes for much smoother calling back when
- queues are involved. (closes issue #11155, reported by IPetrov)
- Tremendous thanks to Russell for pulling me out of my block I was
- having on this one
-
-2007-11-06 20:22 +0000 [r89052-89054] Russell Bryant <russell@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 89053 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r89053 | russell | 2007-11-06 14:18:49 -0600 (Tue, 06
- Nov 2007) | 3 lines Fix init_classes() so that classes that
- actually do have files loaded aren't treated as empty, and
- immediately destroyed ... ........
-
- * main/astmm.c: Fix the memory show allocations CLI command so that
- it doesn't spew out all of the current memory allocations when
- you start Asterisk, when the command's handler gets called for
- initialization.
-
-2007-11-06 19:40 +0000 [r89051] Steve Murphy <murf@digium.com>
-
- * main/ast_expr2f.c, main/ast_expr2.fl: Hoping to avoid a crash in
- OSX for a problem blitzrage found
-
-2007-11-06 19:23 +0000 [r89050] Olle Johansson <oej@edvina.net>
-
- * main/fskmodem.c: Formatting. Illegaly using some spare spaces
- from Russell's space-bucket.
-
-2007-11-06 19:16 +0000 [r89049] Tilghman Lesher <tlesher@digium.com>
-
- * /, include/asterisk/lock.h: Merged revisions 89045 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r89045 | tilghman | 2007-11-06 13:09:06 -0600 (Tue, 06
- Nov 2007) | 2 lines We went to the trouble of creating a method
- of tracking failed trylocks, then never turned it on (oops).
- ........
-
-2007-11-06 19:10 +0000 [r89048] Olle Johansson <oej@edvina.net>
-
- * main/tdd.c, include/asterisk/tdd.h: Additional TDD changes
- (preparing for SIP changes - adding TDD support to SIP)
-
-2007-11-06 19:10 +0000 [r89047] Jason Parker <jparker@digium.com>
-
- * /, codecs/codec_zap.c: Merged revisions 89046 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89046 | qwell | 2007-11-06 13:09:30 -0600 (Tue, 06 Nov 2007) | 4
- lines Correctly set the total number of channels from a zaptel
- transcoder board. SPD-49, patch by Matthew Nicholson. ........
-
-2007-11-06 19:04 +0000 [r89044] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_readfile.c, res/res_features.c, apps/app_sayunixtime.c,
- apps/app_test.c, apps/app_chanisavail.c, res/res_musiconhold.c,
- apps/app_exec.c, apps/app_followme.c, apps/app_minivm.c,
- apps/app_mp3.c, apps/app_amd.c, apps/app_while.c, main/pbx.c,
- apps/app_nbscat.c, channels/chan_sip.c, apps/app_festival.c,
- apps/app_softhangup.c, apps/app_waitforsilence.c,
- channels/chan_agent.c, apps/app_morsecode.c, apps/app_getcpeid.c,
- apps/app_playback.c, res/res_monitor.c, apps/app_speech_utils.c,
- apps/app_forkcdr.c, apps/app_waitforring.c,
- apps/app_directed_pickup.c, apps/app_macro.c, apps/app_sms.c,
- res/res_indications.c, apps/app_chanspy.c, apps/app_mixmonitor.c,
- apps/app_stack.c: "show application <foo>" changes for clarity.
- (closes issue #11171, reported and patched by blitzrage) Many
- thanks!
-
-2007-11-06 19:04 +0000 [r89043] Olle Johansson <oej@edvina.net>
-
- * /, main/tdd.c: Merged revisions 89042 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89042 | oej | 2007-11-06 19:53:37 +0100 (Tis, 06 Nov 2007) | 2
- lines Bug fixes to tdd support in zaptel. ........ (Small changes
- for trunk)
-
-2007-11-06 18:44 +0000 [r89041] Jason Parker <jparker@digium.com>
-
- * channels/chan_jingle.c, include/asterisk/jabber.h,
- channels/chan_gtalk.c, res/res_jabber.c: Allow gtalk and jingle
- to use TLS connections again. Closes issue #9972
-
-2007-11-06 18:23 +0000 [r89038] Russell Bryant <russell@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 89037 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r89037 | russell | 2007-11-06 12:20:07 -0600 (Tue, 06
- Nov 2007) | 11 lines If someone were to delete the files used by
- an existing MOH class, and then issue a reload, further use of
- that class could result in a crash due to dividing by zero. This
- set of changes fixes up some places to prevent this from
- happening. (closes issue #10948) Reported by: jcomellas Patches:
- res_musiconhold_division_by_zero.patch uploaded by jcomellas
- (license 282) Additional changes added by me. ........
-
-2007-11-06 17:10 +0000 [r89034] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 89032 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89032 | file | 2007-11-06 13:08:05 -0400 (Tue, 06 Nov 2007) | 4
- lines Make it so that if a peer is determined to be unreachable
- using qualify their devicestate will report back unavailable.
- (closes issue #11006) Reported by: pj ........
-
-2007-11-06 17:05 +0000 [r89031] Luigi Rizzo <rizzo@icir.org>
-
- * main/loader.c: Fix embedding of modules on FreeBSD: the
- constructor for the list of modules was run after the
- constructors for the embedded modules (which appended entries to
- the list). As a result, the list appeared empty when it was time
- to use it. On linux the order of execution of constructor was
- evidently different (it may depend on the ordering of modules in
- the ELF file). This is only a workaround - there may be other
- situations where the execution of constructors causes problems,
- so if we manage to find a more general solution this workaround
- can go away.
-
-2007-11-06 16:29 +0000 [r88974-88995] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_zap.c, /, configs/zapata.conf.sample: Merged
- revisions 88994 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r88994 | file | 2007-11-06 12:24:56 -0400 (Tue, 06 Nov 2007) | 6
- lines Fix improbable but possible memory leaks in chan_zap.
- (closes issue #11166) Reported by: eliel Patches:
- chan_zap.c.patch uploaded by eliel (license 64) ........
-
- * channels/chan_agent.c: Update chan_agent documentation. Change a
- | to , as that is now the required way. (closes issue #11167)
- Reported by: eliel Patches: chan_agent.c.patch uploaded by eliel
- (license 64)
-
-2007-11-06 15:01 +0000 [r88973] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_unistim.c, configure,
- include/asterisk/autoconfig.h.in, configure.ac: Set up detection
- of IP_PKTINFO in autoconf for chan_unistim
-
-2007-11-06 14:17 +0000 [r88932-88937] Russell Bryant <russell@digium.com>
-
- * channels/chan_unistim.c: convert uses of LOG_DEBUG to use
- ast_debug()
-
- * channels/chan_unistim.c, configs/unistim.conf.sample: Add
- jitterbuffer support to chan_unistim. (closes issue #11168)
- Reported by: IgorG Patches: unistimjb-88863-1.patch uploaded by
- IgorG (license 20)
-
- * main/pbx.c, /, channels/busy.h, channels/ringtone.h,
- include/asterisk/pbx.h: Merged revisions 88805 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r88805 | russell | 2007-11-05 16:07:54 -0600 (Mon, 05 Nov 2007) |
- 12 lines After seeing crashes related to channel variables, I
- went looking around at the ways that channel variables are
- handled. In general, they were not handled in a thread-safe way.
- The channel _must_ be locked when reading or writing from/to the
- channel variable list. What I have done to improve this situation
- is to make pbx_builtin_setvar_helper() and friends lock the
- channel when doing their thing. Asterisk API calls almost all
- lock the channel for you as necessary, but this family of
- functions did not. (closes issue #10923, reported by atis)
- (closes issue #11159, reported by 850t) ........
-
- * /, include/asterisk/lock.h: Merged revisions 88931 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r88931 | russell | 2007-11-06 07:50:15 -0600 (Tue, 06
- Nov 2007) | 8 lines Remove some checks to see if locks are
- initialized from the non-DEBUG_THREADS versions of the lock
- routines. These are incorrect for a number of reasons: - It
- breaks the build on mac. - If there is a problem with locks not
- getting initialized, then the proper fix is to find that place
- and fix the code so that it does get initialized. - If additional
- debug code is needed to help find the problem areas, then this
- type of things should _only_ be put in the DEBUG_THREADS
- wrappers. ........
-
-2007-11-06 08:17 +0000 [r88898-88913] Luigi Rizzo <rizzo@icir.org>
-
- * channels/Makefile: explain that the host environment must be used
- to build gentone; Remove unset variables, they would be
- misleading.
-
- * Makefile: don't export variables that can be retrieved from
- makeopts in child subdirs
-
-2007-11-06 02:53 +0000 [r88863] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, include/asterisk/srv.h: Merged revisions 88862 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r88862 | kpfleming | 2007-11-05 20:52:05 -0600 (Mon, 05
- Nov 2007) | 2 lines update comment to match the state of the code
- ........
-
-2007-11-05 23:31 +0000 [r88827] Mark Michelson <mmichelson@digium.com>
-
- * main/channel.c, /: Merged revisions 88826 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r88826 | mmichelson | 2007-11-05 17:29:29 -0600 (Mon, 05 Nov
- 2007) | 6 lines Reworked deadlock avoidance in __ast_read.
- Restored audio to callback agents. (closes issue #11071, reported
- by callguy, patched by me, tested by callguy and Ted Brown)
- ........
-
-2007-11-05 21:36 +0000 [r88770] Luigi Rizzo <rizzo@icir.org>
-
- * Makefile, utils/Makefile: Move AUDIO_LIBS outside the top level
- Makefile. This too is used only in one place.
-
-2007-11-05 21:35 +0000 [r88769] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 88768 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r88768 | russell | 2007-11-05 15:33:56 -0600 (Mon, 05 Nov 2007) |
- 8 lines When traversing the list of channel variables here in
- transmit_invite(), the asterisk channel must be locked, as this
- data may change at any time. (I have seen numerous reports of
- crashes related to the handling of channel variables. There are a
- couple of issues on the bug tracker related to it, but it has
- also been noted on IRC and mailing lists. So, I am finding and
- fixing some places where channel variables are handled
- improperly.) ........
-
-2007-11-05 21:27 +0000 [r88767] Luigi Rizzo <rizzo@icir.org>
-
- * Makefile, main/Makefile: Move the last instance of AST_LIBS to
- the only place it is used, namely main/Makefile . I am unclear
- where decisions on the build environment (CFLAGS, LDFLAGS, LIBS
- and so on) should be made - right now they are split here and
- there. As a first step in cleaning up this situation, i am trying
- to at least collect all instances of each variable in one place.
-
-2007-11-05 21:23 +0000 [r88766] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 88765 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r88765 | russell | 2007-11-05 15:21:39 -0600 (Mon, 05 Nov 2007) |
- 2 lines Fix up some indentation. ........
-
-2007-11-05 20:50 +0000 [r88764] Luigi Rizzo <rizzo@icir.org>
-
- * Makefile.moddir_rules: comment out an unused variable. Remove it
- in a few days if no problems arise.
-
-2007-11-05 20:44 +0000 [r88710-88740] Russell Bryant <russell@digium.com>
-
- * main/srv.c, /, include/asterisk/srv.h: Merged revisions 88719 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r88719 | russell | 2007-11-05 14:40:01 -0600 (Mon, 05 Nov 2007) |
- 7 lines Merge changes from
- asterisk/team/kpfleming/SRV-priority-handling Previously, the SRV
- record support in Asterisk was broken. There was no guarantee on
- what record Asterisk would choose to actually use. This set of
- changes improves the situation by ensuring that Asterisk will
- choose the highest priority record. ........
-
- * main/channel.c, /: Merged revisions 88709 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r88709 | russell | 2007-11-05 14:11:04 -0600 (Mon, 05 Nov 2007) |
- 20 lines Merge the last bit of changes from
- asterisk/team/russell/readq-1.4 The issue here is that the
- channel frame readq handling got broken when the code was
- converted to use the linked list macros. It caused corruption of
- the list head and tail pointers. So, I fixed up the usage of the
- linked list macros and in passing, simplified the code. I also
- documented what the code is doing, as it was a bit difficult to
- figure out at first. This bug showed itself with crashes showing
- messed up head/tail pointers for the readq. However, there are a
- couple of crashes that aren't quite as obvious, but I think may
- be related. So, if your bug gets closed by this commit, but you
- still have a problem, please reopen or create a new bug report.
- (closes issue #10936) (closes issue #10595) (closes issue #10368)
- (closes issue #11084) (closes issue #10040) (closes issue #10840)
- ........
-
-2007-11-05 19:22 +0000 [r88675] Luigi Rizzo <rizzo@icir.org>
-
- * Makefile: Cleanup the installation of samples, avoiding
- repetitions. I am preserving the behaviour on *.adsi files, i.e.
- overwrite anything there without making a backup. However I am
- not sure that this is the intended behaviour.
-
-2007-11-05 18:52 +0000 [r88673] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 88671 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r88671 | file | 2007-11-05 14:47:13 -0400 (Mon, 05 Nov 2007) | 7
- lines If a SIP channel is put on hold multiple times do not keep
- incrementing the onHold value. (closes issue #11085) Reported by:
- francesco_r Tested by: blitzrage (closes issue #10474) Reported
- by: acennami ........
-
-2007-11-05 18:22 +0000 [r88653] Tilghman Lesher <tlesher@digium.com>
-
- * CHANGES: Change wording to that suggested by MasterYoda
-
-2007-11-05 18:00 +0000 [r88652] Luigi Rizzo <rizzo@icir.org>
-
- * Makefile: simplify (hopefully) the printing of $(MAKE) in aligned
- output.
-
-2007-11-05 17:52 +0000 [r88651] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /: Merged revisions 88624 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r88624 | russell | 2007-11-05 11:46:02 -0600 (Mon, 05 Nov 2007) |
- 5 lines Fix up datastore handling in ast_do_masquerade(). The
- code is intended to move any channel datastores from the old
- channel to the new one. However, it did not use the linked list
- macros properly to accomplish the task. The existing code would
- only work if there was only a single datastore on the old
- channel. ........
-
-2007-11-05 17:44 +0000 [r88587-88615] Luigi Rizzo <rizzo@icir.org>
-
- * Makefile: print messages when entering/leaving a directory so we
- know where we are (sometimes it is obvious, sometimes it is not).
-
- * Makefile.moddir_rules: merge two rules with the same right hand;
- document a bit what is done here.
-
-2007-11-05 17:21 +0000 [r88586] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 88585 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #11163) ........ r88585 | qwell | 2007-11-05 11:19:41 -0600
- (Mon, 05 Nov 2007) | 4 lines Make sure we destroy the config
- structure on configuration failure. Issue 11163, patch by eliel.
- ........
-
-2007-11-05 17:00 +0000 [r88584] Kevin P. Fleming <kpfleming@digium.com>
-
- * Makefile.rules: use a variable name that actually indicates what
- it is for
-
-2007-11-05 16:41 +0000 [r88553] Luigi Rizzo <rizzo@icir.org>
-
- * Makefile.rules: Put extra compiler flags into a variable so they
- are not repeated too many times. On passing, add some comments
- and fix indentation a bit. On passing, i suspect that the
- following pattern is wrong %.eoo: %.o but in case it will be
- fixed in a later commit.
-
-2007-11-05 16:30 +0000 [r88540] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_odbc.c: Merged revisions 88539 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r88539 | tilghman | 2007-11-05 10:20:13 -0600 (Mon, 05 Nov 2007)
- | 4 lines Don't check used pooled connections for connection
- status, as it will cause issues for prepared queries. Reported
- by: Nick Gorham (via -dev list) Patch by: tilghman ........
-
-2007-11-05 15:15 +0000 [r88525] Luigi Rizzo <rizzo@icir.org>
-
- * main/db.c: remove a cygwin-specific function remap that does not
- work.
-
-2007-11-05 13:11 +0000 [r88510] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_unistim.c: Fix memory leaks and deadlocks in
- chan_unistim. (closes issue #11158) Reported by: eliel Patches:
- chan_unistim.c.patch uploaded by eliel (license 64)
-
-2007-11-04 22:42 +0000 [r88454-88490] Luigi Rizzo <rizzo@icir.org>
-
- * /: block merging of not-applicable patch
-
- * main/channel.c, main/pbx.c, apps/app_meetme.c,
- channels/chan_sip.c, res/res_features.c, main/utils.c,
- channels/chan_iax2.c, include/asterisk/stringfields.h: Simplify
- the implementation and the API for stringfields; details and
- examples are in include/asterisk/stringfields.h. Not applicable
- to older branches except for 1.4 which will receive a fix for the
- routines that free memory pools.
-
-2007-11-03 14:19 +0000 [r88437] Tilghman Lesher <tlesher@digium.com>
-
- * main/term.c: Revert commit #86119. Some users intentionally do
- not want colorized terminals, so this was a misfeature.
-
-2007-11-03 04:55 +0000 [r88422] James Golovich <james@gnuinter.net>
-
- * main/db.c: Set CLI command to the correct name. Rev 85460
- introduced two 'database show' commands when this one should have
- been 'database showkey'
-
-2007-11-02 22:36 +0000 [r88368-88409] Russell Bryant <russell@digium.com>
-
- * channels/chan_unistim.c: fix some issues with crashing on unload,
- when it didn't completely load cleanly
-
- * channels/chan_unistim.c: Convert the CLI commands to the new
- format
-
- * pbx/pbx_lua.c: propagate the DECLINE return value back to the
- loader
-
- * pbx/pbx_lua.c: Don't kill asterisk if extensions.lua is not
- present.
-
- * main/cli.c: Show the channel unique ID in the "show channel
- concise" output (closes issue #11148, requested by falves11,
- patched by me)
-
- * channels/chan_unistim.c (added), CREDITS,
- configs/unistim.conf.sample (added), CHANGES, doc/unistim.txt
- (added): Merge the code from asterisk/team/group/chan_unistim:
- This introduces a new channel driver, chan_unistim, that supports
- the Unistim VoIP protocol for Nortel phones. The following models
- have been confirmed to work: i2002, i2004 and i2050. (closes
- issue #8864) Reported by: c_hans Patches: chan_unistim.patch
- uploaded by c (license 304) ustm_no_conf.diff uploaded by junky
- (license 177) Tested by: c_hans, dbowerman, math, junky, loloski
-
-2007-11-02 20:51 +0000 [r88329-88367] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 88366 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r88366 | file | 2007-11-02 17:49:45 -0300 (Fri, 02 Nov 2007) | 4
- lines Make subscribecontext behave as advertised. It will now
- look for the presence of a hint in the given context (be it
- subscribecontext or context). (closes issue #10702) Reported by:
- slavon ........
-
- * /, channels/chan_sip.c: Merged revisions 88328 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r88328 | file | 2007-11-02 17:20:21 -0300 (Fri, 02 Nov 2007) | 6
- lines If an INFO request within a dialog is received with a
- content length of 0 simply send back a 200 OK. It is valid to do
- this and the remote side is probably using it to make sure the
- signalling is still alive. (closes issue #5747) Reported by:
- chandi Patches: infofix-81430-1.patch uploaded by IgorG (license
- 20) ........
-
-2007-11-02 20:13 +0000 [r88327] Russell Bryant <russell@digium.com>
-
- * doc/tex/Makefile: Fix replacing the version number when it has a
- '/' in it, like SVN-group-chan_unistim-r88326M-/trunk
-
-2007-11-02 17:34 +0000 [r88287] Tilghman Lesher <tlesher@digium.com>
-
- * pbx/pbx_lua.c: Oops, some dev-mode changes for ISO C90
-
-2007-11-02 16:54 +0000 [r88284] Jason Parker <jparker@digium.com>
-
- * /, main/say.c: Merged revisions 88283 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #11147) ........ r88283 | qwell | 2007-11-02 11:51:08 -0500
- (Fri, 02 Nov 2007) | 4 lines We need to make sure to specify a
- language to ast_fileexists, otherwise it may fail for anything
- besides en Issue 11147, fix discovered by both citats and myself
- (independently), with input from Corydon76 ........
-
-2007-11-02 16:26 +0000 [r88209-88267] Tilghman Lesher <tlesher@digium.com>
-
- * CHANGES: Add a few bytes on LUA
-
- * main/pbx.c, utils/build-extensions-conf.lua (added),
- build_tools/menuselect-deps.in, configure,
- include/asterisk/autoconfig.h.in, configure.ac, pbx/pbx_lua.c
- (added), configs/extensions.lua.sample (added),
- include/asterisk/pbx.h, makeopts.in: Add pbx_lua as a method of
- doing extensions Reported by: mnicholson Patch by: mnicholson
- Closes issue #11140
-
- * main/config.c: Don't re-cache the filename, but check to see if
- it already exists Reported by: jamesgolovich Patch by:
- jamesgolovich Closes issue #11144
-
- * /, include/asterisk/lock.h: Merged revisions 88210 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r88210 | tilghman | 2007-11-02 08:03:03 -0500 (Fri, 02
- Nov 2007) | 5 lines Fix build on Solaris Reported by: snuffy
- Patch by: ys Closes issue #11143 ........
-
- * main/pbx.c: 'h' extension doesn't execute past first priority
- Reported by: dimas Patch by: dimas Closes bug #11146
-
-2007-11-02 03:09 +0000 [r88197] Joshua Colp <jcolp@digium.com>
-
- * cdr/cdr_odbc.c: Restore building under 64-bit platforms.
-
-2007-11-01 23:26 +0000 [r88184] Jason Parker <jparker@digium.com>
-
- * channels/chan_jingle.c, configure,
- include/asterisk/autoconfig.h.in, configure.ac,
- include/asterisk/jabber.h, channels/chan_gtalk.c, makeopts.in:
- Remove traces of gnutls, since we no longer use/need it.
-
-2007-11-01 23:26 +0000 [r88182-88183] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c: Modify WaitExten to include an optional dialtone
- Closes issue #10783
-
- * UPGRADE.txt, cdr/cdr_odbc.c: Convert cdr_odbc to use res_odbc
- managed connections Closes issue #10614
-
-2007-11-01 22:26 +0000 [r88166] Steve Murphy <murf@digium.com>
-
- * apps/app_dial.c, main/pbx.c, cdr/cdr_sqlite3_custom.c,
- funcs/func_strings.c, funcs/func_cut.c, funcs/func_logic.c,
- apps/app_exec.c, apps/app_queue.c, apps/app_playback.c,
- res/ael/pval.c, pbx/pbx_loopback.c, funcs/func_odbc.c,
- apps/app_minivm.c, res/res_agi.c, main/logger.c,
- pbx/pbx_realtime.c, apps/app_macro.c, pbx/pbx_dundi.c,
- utils/extconf.c, include/asterisk/pbx.h, pbx/pbx_config.c,
- apps/app_mixmonitor.c, apps/app_rpt.c, cdr/cdr_custom.c,
- cdr/cdr_manager.c: This commits the performance mods that give
- the priority processing engine in the pbx, a 25-30% speed boost.
- The two updates used, are, first, to merge the
- ast_exists_extension() and the ast_spawn_extension() where they
- are called sequentially in a loop in the code, into a slightly
- upgraded version of ast_spawn_extension(), with a few extra args;
- and, second, I modified the substitute_variables_helper_full, so
- it zeroes out the byte after the evaluated string instead of
- demanding you pre-zero the buffer; I also went thru the code and
- removed the code that zeroed this buffer before every call to the
- substitute_variables_helper_full. The first fix provides about a
- 9% speedup, and the second the rest. These figures come from the
- 'PIPS' benchmark I describe in blogs, conf. reports, etc.
-
-2007-11-01 22:19 +0000 [r88164-88165] Jason Parker <jparker@digium.com>
-
- * /: Crap, accidentally copied the props. Thanks for pointing this
- out mvanbaak. The odds are quite high that this will break
- automerge on every team branch.
-
- * /, include/asterisk/jabber.h, res/res_jabber.c: Switch res_jabber
- to use openssl rather than gnutls. Closes issue #9972, patch by
- phsultan. Copied from branch at
- http://svn.digium.com/svn/asterisk/team/phsultan/res_jabber-openssl/
-
-2007-11-01 17:25 +0000 [r88117] Tilghman Lesher <tlesher@digium.com>
-
- * /, doc/valgrind.txt (added): Merged revisions 88116 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r88116 | tilghman | 2007-11-01 12:17:56 -0500 (Thu, 01
- Nov 2007) | 2 lines Add some notes on using valgrind ........
-
-2007-11-01 16:22 +0000 [r88079] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 88078 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r88078 | qwell | 2007-11-01 11:21:22 -0500 (Thu, 01 Nov 2007) | 4
- lines Make sure we set the poll fds to NULL after free()ing it.
- Part of issue 11017, patch by tzafrir. ........
-
-2007-11-01 15:56 +0000 [r88062-88077] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c, pbx/pbx_dundi.c: Change some uses of free()
- to ast_free(). (No functional differences.) (closes issue #11138)
- Reported by: eliel Patches: pbx_dundi.c.patch uploaded by eliel
- (license 64) chan_sip.c.patch uploaded by eliel (license 64)
-
- * utils/Makefile: Remove another copied source file on "make
- clean". (closes issue #11137) Reported by: IgorG Patches:
- addonclean-87971-1.patch uploaded by IgorG (license 20)
-
-2007-11-01 13:30 +0000 [r88027] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 88026 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r88026 | file | 2007-11-01 10:27:37 -0300 (Thu, 01 Nov 2007) | 2
- lines Fix up commit for my Zap channel with spies in Meetme fix.
- (thanks Tony Mountifield!) ........
-
-2007-11-01 06:12 +0000 [r88007-88010] Tilghman Lesher <tlesher@digium.com>
-
- * main/utils.c: Conditionally free lock_info->thread_name to avoid
- a useless warning Reported by: snuffy Patch by: snuffy Closes
- issue #11125
-
- * apps/app_meetme.c, channels/chan_iax2.c: Janitor: use ast_free to
- pair calls of ast_malloc and ast_calloc Reported by: eliel Patch
- by: eliel Closes issue #11135
-
- * cdr/cdr_adaptive_odbc.c: Fix memory leak Reported by: eliel Fixed
- by: tilghman Closes issue #11136
-
-2007-11-01 01:55 +0000 [r87953-87971] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 87970 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r87970 | file | 2007-10-31 22:53:55 -0300 (Wed, 31 Oct 2007) | 4
- lines If a Zap channel contains a spy or a spy is added take it
- out of the conference in kernel space and make it go through
- Asterisk so the spy gets audio from both sides. (closes issue
- #10060) Reported by: mparker ........
-
- * main/pbx.c: Drop any more references to type in the Exception
- dialplan function. (closes issue #11134) Reported by: blitzrage
- Patches: exception_patch.txt uploaded by blitzrage (license 10)
-
-2007-10-31 21:23 +0000 [r87889-87909] Jason Parker <jparker@digium.com>
-
- * /, res/res_jabber.c: Merged revisions 87908 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #11131) ........ r87908 | qwell | 2007-10-31 16:23:11 -0500
- (Wed, 31 Oct 2007) | 4 lines Make sure we free some allocated
- memory before returning. Issue 11131, patch by eliel. ........
-
- * channels/chan_jingle.c, /, channels/chan_gtalk.c: Merged
- revisions 87906 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #11130) (closes issue #11132) ........ r87906 | qwell |
- 2007-10-31 16:16:20 -0500 (Wed, 31 Oct 2007) | 4 lines Don't try
- to allocate memory that we're just going to re-allocate later
- anyways. Issues 11130 and 11132, patch by eliel. ........
-
- * formats/format_sln.c, codecs/codec_adpcm.c, codecs/codec_gsm.c,
- formats/format_wav_gsm.c, res/res_musiconhold.c,
- codecs/codec_zap.c, formats/format_ilbc.c, res/res_smdi.c,
- formats/format_pcm.c, formats/format_h263.c,
- formats/format_h264.c, formats/format_jpeg.c,
- formats/format_gsm.c, res/res_speech.c, res/res_clioriginate.c,
- codecs/codec_lpc10.c, codecs/codec_g722.c, codecs/codec_a_mu.c,
- formats/format_wav.c, codecs/codec_speex.c, codecs/codec_alaw.c,
- res/res_adsi.c, res/res_convert.c, codecs/codec_g726.c,
- formats/format_ogg_vorbis.c, res/res_ael_share.c,
- formats/format_vox.c, codecs/codec_ulaw.c, formats/format_g723.c,
- res/res_indications.c, codecs/codec_ilbc.c,
- formats/format_g726.c, formats/format_g729.c: More changes to
- change return values from load_module functions. (issue #11096)
- Patches: codec_adpcm.c.patch uploaded by moy (license 222)
- codec_alaw.c.patch uploaded by moy (license 222)
- codec_a_mu.c.patch uploaded by moy (license 222)
- codec_g722.c.patch uploaded by moy (license 222)
- codec_g726.c.diff uploaded by moy (license 222) codec_gsm.c.patch
- uploaded by moy (license 222) codec_ilbc.c.patch uploaded by moy
- (license 222) codec_lpc10.c.patch uploaded by moy (license 222)
- codec_speex.c.patch uploaded by moy (license 222)
- codec_ulaw.c.patch uploaded by moy (license 222)
- codec_zap.c.patch uploaded by moy (license 222)
- format_g723.c.patch uploaded by moy (license 222)
- format_g726.c.patch uploaded by moy (license 222)
- format_g729.c.patch uploaded by moy (license 222)
- format_gsm.c.patch uploaded by moy (license 222)
- format_h263.c.patch uploaded by moy (license 222)
- format_h264.c.patch uploaded by moy (license 222)
- format_ilbc.c.patch uploaded by moy (license 222)
- format_jpeg.c.patch uploaded by moy (license 222)
- format_ogg_vorbis.c.patch uploaded by moy (license 222)
- format_pcm.c.patch uploaded by moy (license 222)
- format_sln.c.patch uploaded by moy (license 222)
- format_vox.c.patch uploaded by moy (license 222)
- format_wav.c.patch uploaded by moy (license 222)
- format_wav_gsm.c.patch uploaded by moy (license 222)
- res_adsi.c.patch uploaded by eliel (license 64)
- res_ael_share.c.patch uploaded by eliel (license 64)
- res_clioriginate.c.patch uploaded by eliel (license 64)
- res_convert.c.patch uploaded by eliel (license 64)
- res_indications.c.patch uploaded by eliel (license 64)
- res_musiconhold.c.patch uploaded by eliel (license 64)
- res_smdi.c.patch uploaded by eliel (license 64)
- res_speech.c.patch uploaded by eliel (license 64)
-
-2007-10-31 18:53 +0000 [r87888] Steve Murphy <murf@digium.com>
-
- * /: Merged revisions 87849 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r87849 | murf | 2007-10-31 11:49:39 -0600 (Wed, 31 Oct 2007) | 1
- line closes issue #11108 -- where the 'dialplan save' cli command
- saves a file where the semicolon is not escaped. Fixed this; User
- also wanted comments to be preserved across dialplan save, but
- this is impossible at this point in time, because comments are
- not stored in the dialplan. They are 'compiled' out of
- extensions.conf. The only way to preserve those comments is to
- use the config file reader/writer that the GUI uses to allow
- online user edits. extensions.conf is first and foremost, a
- config file, and is read in by the normal config-file reading
- routines. Then, it is processed into a dialplan (context/exten
- structs). (in the case of trunk, tho, no mods needed to be made
- -- works OK there -- just make sure you use ',' to sep app args!)
- ........
-
-2007-10-31 18:09 +0000 [r87854] Tilghman Lesher <tlesher@digium.com>
-
- * Makefile, /: Merged revisions 87852 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r87852 | tilghman | 2007-10-31 13:03:53 -0500 (Wed, 31 Oct 2007)
- | 2 lines Create samples for ALL of the available options in
- asterisk.conf ........
-
-2007-10-31 18:03 +0000 [r87833-87851] Joshua Colp <jcolp@digium.com>
-
- * apps/app_mixmonitor.c: Add volume adjustment in.
-
- * apps/app_mixmonitor.c: Restore operation of the option that only
- writes when the channel is bridged.
-
- * apps/app_chanspy.c: Add volume adjustment to spy audiohook in
- app_chanspy.
-
-2007-10-31 16:13 +0000 [r87817] Tilghman Lesher <tlesher@digium.com>
-
- * CREDITS: Formatting cleanups, remove obsolete contributions
- (modules no longer in Asterisk), and obfuscate email addresses
- enough to stop most spam harvesters.
-
-2007-10-31 16:07 +0000 [r87815] Joshua Colp <jcolp@digium.com>
-
- * include/asterisk/channel.h: Remove old whisper remnants from
- channel.h
-
-2007-10-31 15:46 +0000 [r87811] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c: Optimize pbx_substitute_variables
-
-2007-10-31 04:20 +0000 [r87776] Steve Murphy <murf@digium.com>
-
- * res/ael/pval.c, /: Merged revisions 87775 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r87775 | murf | 2007-10-30 21:51:52 -0600 (Tue, 30 Oct 2007) | 1
- line Included some verbage in the check_includes func, to inform
- the user that included contexts that have no match in the AEL,
- might be OK, as AEL cannot check in the extensions.conf or the
- in-memory contexts, as they may not be there at the time of the
- check. ........
-
-2007-10-30 23:08 +0000 [r87724-87740] Tilghman Lesher <tlesher@digium.com>
-
- * /, include/asterisk/lock.h: Merged revisions 87739 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r87739 | tilghman | 2007-10-30 18:02:22 -0500 (Tue, 30
- Oct 2007) | 5 lines Fix for uninitialized mutexes on *BSD
- Reported by: ys Fixed by: ys Closes issue #11116 ........
-
- * apps/app_exec.c: If no '?' is found in the arguments, don't
- attempt to continue. Reported by: blitzrage Fixed by: tilghman
- Closes issue #11111
-
-2007-10-30 21:22 +0000 [r87687] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 87686 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r87686 | russell | 2007-10-30 16:19:09 -0500 (Tue, 30 Oct 2007) |
- 11 lines Merge the changes from team/russell/iax2_poke_fix and
- iax2-poke-fix-trunk There was a race condition related to the
- handling of POKEing peers. Essentially, a reference to a peer is
- held by the scheduler when there are pending callbacks, but the
- reference count didn't reflect it. So, it was possible for a peer
- to hit a reference count of zero and have its destructor begin to
- be called at the same time that the scheduler thread ran a POKE
- related callback. If that happened, a crash would likely occur.
- (closes issue #11082, closes issue #11094) ........
-
-2007-10-30 20:30 +0000 [r87626-87651] Jason Parker <jparker@digium.com>
-
- * /, channels/Makefile: Merged revisions 87650 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r87650 | qwell | 2007-10-30 15:29:41 -0500 (Tue, 30 Oct 2007) | 1
- line Only try to clean out h323/ if the h323/Makefile exists.
- ........
-
- * main/pbx.c: Update documentation to give an example of how to use
- the return status of RaiseException Closes issue #11117, patch by
- blitzrage (yay blitzrage)
-
-2007-10-30 17:07 +0000 [r87573-87608] Mark Michelson <mmichelson@digium.com>
-
- * main/pbx.c: The priority gets incremented after raising an
- exception, so the priority should be set to 0
-
- * main/pbx.c: Jumped the gun a bit in the RaiseException app. It
- would always return -1 since it checked for the existence of
- something that will never exist.
-
-2007-10-30 16:15 +0000 [r87572] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_features.c: Merged revisions 87571 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r87571 | file | 2007-10-30 13:13:39 -0300 (Tue, 30 Oct 2007) | 4
- lines Add two more checks before printing out a warning message
- about bridging. If either channel has hungup of course the bridge
- will have failed. (closes issue #10009) Reported by: dimas
- ........
-
-2007-10-30 15:47 +0000 [r87568] Jason Parker <jparker@digium.com>
-
- * /, main/editline/np/vis.c: Merged revisions 87567 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #11113) ........ r87567 | qwell | 2007-10-30 10:45:35 -0500
- (Tue, 30 Oct 2007) | 4 lines Fix build of editline on Solaris.
- Issue 11113, patch by snuffy. ........
-
-2007-10-29 22:44 +0000 [r87462-87498] Kevin P. Fleming <kpfleming@digium.com>
-
- * utils/Makefile, utils, utils/hashtest2.c: UGH... while trying to
- fix #10995, I found all kinds of cruft in this Makefile. It
- should all be gone now, and as a side effect hashtest2 now builds
- with --enable-dev-mode enabled without a host of errors
-
- * agi/Makefile, utils/Makefile, codecs/g722/Makefile,
- main/editline/Makefile.in, Makefile.moddir_rules,
- codecs/ilbc/Makefile, codecs/lpc10/Makefile,
- main/db1-ast/Makefile: clean up assembler and preprocessor files
- if they are here too
-
- * utils, agi, codecs, apps, cdr, codecs/ilbc, formats, funcs,
- codecs/lpc10, main/db1-ast, codecs/g722, main/editline, main,
- codecs/gsm, main/minimime, pbx, res, channels: ignore
- preprocessor and assembler files if they are present
-
- * Makefile, /: Merged revisions 87460 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r87460 | kpfleming | 2007-10-29 17:04:29 -0500 (Mon, 29 Oct 2007)
- | 2 lines don't put '-pipe' into ASTCFLAGS if '-save-temps' is
- already there (used when debugging preprocessor issues) because
- the compiler will whine about each compile command ........
-
-2007-10-29 21:34 +0000 [r87397-87428] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c: If a caller is listen-only, then don't bother
- with doing talker detection. (closes issue #10911, reported by
- junky, patched by me)
-
- * /, main/utils.c, include/asterisk/lock.h: Merged revisions 87396
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r87396 | russell | 2007-10-29 15:22:07 -0500 (Mon, 29 Oct 2007) |
- 5 lines Add some more details to the output of "core show locks".
- When a thread is waiting for a lock, this will now show the
- details about who currently has it locked. (inspired by issue
- #11100) ........
-
-2007-10-29 20:13 +0000 [r87395] Mark Michelson <mmichelson@digium.com>
-
- * UPGRADE.txt, apps/app_queue.c: Adding the more flexible
- QUEUE_MEMBER function to replace the QUEUE_MEMBER_COUNT function.
- A deprecation notice will be issued the first time
- QUEUE_MEMBER_COUNT is used.
-
-2007-10-29 20:02 +0000 [r87394] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Drop the RTCP Read too short message to debug. There
- are some phones out there that send a sort of keep alive packet
- in the RTCP that trigger this every 5 seconds.
-
-2007-10-29 19:56 +0000 [r87393] Jason Parker <jparker@digium.com>
-
- * apps/app_record.c: Make sure we set flags to a 0 value before
- trying to use it. Pointed out by seanbright while I was debugging
- issue 11109.
-
-2007-10-29 19:47 +0000 [r87392] Russell Bryant <russell@digium.com>
-
- * /, main/astmm.c: Merged revisions 87373 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r87373 | russell | 2007-10-29 14:21:06 -0500 (Mon, 29 Oct 2007) |
- 5 lines Remove a lock that doesn't make any sense. The regions
- lock needs to be held when traversing the list of allocated
- chunks so that they can be printed out to the CLI. (Thanks to
- eliel on #asterisk-dev for pointing this out!) ........
-
-2007-10-29 17:22 +0000 [r87343] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 87342 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r87342 | file | 2007-10-29 14:20:28 -0300 (Mon, 29 Oct 2007) | 6
- lines Fix issue where if both sides of the dialog cancelled the
- dialog at the same time chan_sip could kepe retransmitting a
- response for no reason. (closes issue #9566) Reported by:
- atca_pres Patches: bug9566.patch uploaded by oej ........
-
-2007-10-29 16:38 +0000 [r87295-87327] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c: Remove duplicate stdlib.h include. (closes
- issue #11105) Reported by: eliel Patches: app_voicemail.c.patch
- uploaded by eliel (license 64)
-
- * channels/chan_misdn.c, configure,
- include/asterisk/autoconfig.h.in, configure.ac: Add autoconf
- checks for extra suppserv definitions that are not present in
- releases yet. chan_misdn should now build against the latest
- release. (closes issue #11103) Reported by: IgorG
-
- * /, main/utils.c: Merged revisions 87294 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r87294 | file | 2007-10-29 11:23:49 -0300 (Mon, 29 Oct 2007) | 6
- lines Fix issue with ast_unescape_semicolon going into an endless
- loop. (closes issue #10550) Reported by: ramonpeek Patches:
- unescape-85177-1.patch uploaded by IgorG (license 20) ........
-
-2007-10-28 14:16 +0000 [r87263-87264] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_dialgroup.c (added): Add a simple dialgroup function.
- By taking one of the simpler uses of Queue away from Queue, we
- simplify the lives of people who do not need all the bells and
- whistles. Also, this is part of the functions that people need to
- reimplement Queue in the dialplan, as a set of logic, rather than
- as a single app with hundreds of options.
-
- * /, funcs/func_odbc.c, funcs/func_strings.c, funcs/func_cut.c,
- funcs/func_realtime.c: Merged revisions 87262 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r87262 | tilghman | 2007-10-28 08:46:55 -0500 (Sun, 28 Oct 2007)
- | 7 lines Add autoservice to several more functions which might
- delay in their responses. Also, make sure that func_odbc
- functions have a channel on which to set variables. Reported by
- russell Fixed by tilghman Closes issue #11099 ........
-
-2007-10-27 15:41 +0000 [r87233-87247] Russell Bryant <russell@digium.com>
-
- * configure, configure.ac: Update the configure script for the last
- libss7 API change
-
- * funcs/func_shell.c, funcs/func_lock.c: Make sure a channel exists
- before attempting to start or stop channel autoservice in
- func_lock and func_shell.
-
-2007-10-27 00:48 +0000 [r87231-87232] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Add Circuit Group Queury message code
-
- * channels/chan_zap.c: Make sure we turn on the DSP when we answer
- the call
-
-2007-10-26 22:21 +0000 [r87217] Mark Michelson <mmichelson@digium.com>
-
- * CHANGES: Forgot to update CHANGES when I committed the linear
- queue strategy. Thank you Russell, for pointing this out!
-
-2007-10-26 21:37 +0000 [r87202] Jason Parker <jparker@digium.com>
-
- * channels/chan_local.c, channels/chan_zap.c,
- channels/chan_agent.c, channels/chan_features.c,
- res/res_crypto.c, res/res_realtime.c, res/res_monitor.c:
- Correctly use defined return values in (some) load_module
- functions. (issue #11096) Patches: chan_agent.c.patch uploaded by
- eliel (license 64) chan_local.c.patch uploaded by eliel (license
- 64) chan_features.c.patch uploaded by eliel (license 64)
- chan_zap.c.patch uploaded by eliel (license 64)
- res_monitor.c.patch uploaded by eliel (license 64)
- res_realtime.c.patch uploaded by eliel (license 64)
- res_crypto.c.patch uploaded by eliel (license 64)
-
-2007-10-26 17:39 +0000 [r87187] Steve Murphy <murf@digium.com>
-
- * res/ael/pval.c, /, include/asterisk/pval.h, res/ael/ael.tab.c,
- res/ael/ael.y, pbx/pbx_ael.c, res/ael/ael_lex.c,
- res/ael/ael.tab.h, utils/ael_main.c,
- pbx/ael/ael-test/ref.ael-test16, res/ael/ael.flex,
- utils/conf2ael.c, pbx/ael/ael-test/ref.ael-test19: Merged
- revisions 87168 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r87168 | murf | 2007-10-26 10:34:02 -0600 (Fri, 26 Oct 2007) | 1
- line closes issue #11086 where a user complains that references
- to following contexts report a problem; The problem was REALLy
- that he was referring to empty contexts, which were being
- ignored. Reporter stated that empty contexts should be OK. I
- checked it out against extensions.conf, and sure enough, empty
- contexts ARE ok. So, I removed the restriction from AEL. This,
- though, highlighted a problem with multiple contexts of the same
- name. This should be OK, also. So, I added the extend keyword to
- AEL, and it can preceed the 'context' keyword (mixed with
- 'abstract', if nec.). This will turn off the warnings in AEL if
- the same context name is used 2 or more times. Also, I now call
- ast_context_find_or_create for contexts now, instead of just
- ast_context_create; I did this because pbx_config does this. The
- 'extend' keyword thus becomes a statement of intent. AEL can now
- duplicate the behavior of pbx_config, ........
-
-2007-10-26 15:19 +0000 [r87153-87154] Mark Michelson <mmichelson@digium.com>
-
- * configs/queues.conf.sample, apps/app_queue.c: Added queue
- strategy "linear". This strategy is useful for those who always
- wish for their phones to be rung in a specific order. (closes
- issue #7279, reported and initially patched by diLLec, patch
- reworked by me)
-
- * configs/queues.conf.sample: Remove information about the
- roundrobin strategy from trunk's queues.conf.sample since it no
- longer exists
-
-2007-10-26 14:00 +0000 [r87103-87121] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_curl.c, /: Merged revisions 87120 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r87120 | tilghman | 2007-10-26 08:54:30 -0500 (Fri, 26 Oct 2007)
- | 7 lines The addition of autoservice to func_curl additionally
- made func_curl dependent on the existence of a channel, with no
- real reason. This should make func_curl once again work without a
- channel. Reported by jmls. Fixed by tilghman. Closes issue #11090
- ........
-
- * include/asterisk/app.h, funcs/func_strings.c, funcs/func_cut.c,
- main/app.c: Use the same delimited character as the FILTER
- function in FIELDQTY and CUT.
-
-2007-10-25 23:11 +0000 [r87070] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/channel.c, /, include/asterisk/linkedlists.h: Merged
- revisions 87069 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r87069 | kpfleming | 2007-10-25 18:03:11 -0500 (Thu, 25 Oct 2007)
- | 2 lines appending one list to another should leave the first
- list empty, and not require the user to do that ........
-
-2007-10-25 18:59 +0000 [r87040] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c: Add support for a muted user to request to
- talk. The '2' option in the user menu will adjust this status if
- a user is muted. The talk request status will be reflected in the
- CLI commands as well as the manager interface. (closes issue
- #9418) Reported by: imesper Patches: app_meetme_v2.patch uploaded
- by imesper (license 275)
-
-2007-10-25 16:21 +0000 [r87024] Steve Murphy <murf@digium.com>
-
- * main/ast_expr2.y, res/res_config_sqlite.c, main/ast_expr2.c:
- closes issue #11045 - each file needs to define
- ASTERISK_FILE_VERSION, if you are going to set MTX_PROFILE in the
- compiler flags; the problem was that the fixes were getting made
- to the generated .c file, and erased the next time someone
- regenerated that file from the corresponding .y or .flex file.
- Moral of story: keep your eyes open and make mods to the .y (or
- flex input file) and re-run bison (or flex) as the Makefile
- directs for that file, and then check in both. Also,
- res_config_sqlite was kinda missed, and has the same issue.
-
-2007-10-24 21:26 +0000 [r86985] Mark Michelson <mmichelson@digium.com>
-
- * configs/queues.conf.sample, apps/app_queue.c: Adding the general
- option "shared_lastcall" to queues so that a member's wrapuptime
- may be used across multiple queues. (closes issue #9777, reported
- and patched by eliel)
-
-2007-10-24 20:59 +0000 [r86983] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 86982 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #11079) ........ r86982 | qwell | 2007-10-24 15:56:47 -0500
- (Wed, 24 Oct 2007) | 5 lines Correctly respect hidecalleridname
- configuration option. Simplify code slightly in the process.
- Issue 11079, reported by ddv2005 ........
-
-2007-10-24 13:21 +0000 [r86900-86967] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael-test/ref.ael-ntest22, pbx/ael/ael-test/ref.ael-test2,
- pbx/ael/ael-test/ref.ael-test3, res/ael/ael_lex.c,
- pbx/ael/ael-test/ref.ael-test4, res/ael/ael.flex: closes issue
- #11005, where #include uses the current dir instead of the config
- dir (/etc/asterisk) for relative path includes for AEL
-
- * /, res/ael/ael.tab.c, res/ael/ael.y: Merged revisions 86936 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r86936 | murf | 2007-10-23 22:14:28 -0600 (Tue, 23 Oct 2007) | 1
- line closes issue #11037 -- unable to specify app:spec in hint
- arguments ........
-
- * /, funcs/func_logic.c: Merged revisions 86902 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r86902 | murf | 2007-10-23 15:18:08 -0600 (Tue, 23 Oct 2007) | 1
- line closes issue #11052 -- where nothing after the ? will allow
- un-initialized variable values to corrupt and crash asterisk on
- 64-bit platforms ........
-
- * /, main/ast_expr2f.c: Merged revisions 86880 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r86880 | murf | 2007-10-23 14:20:54 -0600 (Tue, 23 Oct 2007) | 1
- line This should get rid of a really, really irritating warning
- generated by some 64-bit platforms from libc, where free(0) is
- frowned upon ........
-
- * /, main/Makefile: Merged revisions 86881 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r86881 | murf | 2007-10-23 14:22:25 -0600 (Tue, 23 Oct 2007) | 1
- line this update to Makefile corrects how ast_expr2f.c should be
- generated ........
-
-2007-10-22 21:37 +0000 [r86835-86839] Russell Bryant <russell@digium.com>
-
- * /, include/asterisk/lock.h: Merged revisions 86836 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r86836 | russell | 2007-10-22 16:36:12 -0500 (Mon, 22
- Oct 2007) | 9 lines If lock tracking is not enabled, then we can
- not attempt to log any mutex failures. If so, we could end up in
- infinite recursion. The only lock that is affected by this is a
- mutex in astmm.c used when MALLOC_DEBUG is enabled. (closes issue
- #11044) Reported by: ys Patches: lock.h.diff uploaded by ys
- (license 281) ........
-
- * apps/app_playback.c: Convert some spaces to tabs and make it so
- the CLI command is only registered once instead of 3 times.
- (closes issue #11053) Reported by: seanbright Patches:
- app_playback.patch uploaded by seanbright (license 71)
-
-2007-10-22 20:05 +0000 [r86820] Jason Parker <jparker@digium.com>
-
- * main/udptl.c, channels/chan_local.c, main/frame.c,
- res/res_features.c, main/threadstorage.c, channels/chan_iax2.c,
- main/astobj2.c, main/config.c, main/cli.c,
- channels/chan_skinny.c, main/http.c, pbx/pbx_ael.c,
- channels/chan_alsa.c, main/db.c, main/pbx.c,
- channels/chan_agent.c, channels/iax2-provision.c,
- apps/app_playback.c, channels/chan_misdn.c,
- channels/chan_features.c, res/res_indications.c,
- pbx/pbx_config.c, apps/app_mixmonitor.c, main/asterisk.c,
- res/res_odbc.c, apps/app_voicemail.c, apps/app_meetme.c,
- main/utils.c, channels/chan_gtalk.c, res/res_musiconhold.c,
- res/res_jabber.c, codecs/codec_zap.c, res/res_config_sqlite.c,
- main/channel.c, main/cdr.c, apps/app_osplookup.c, main/manager.c,
- res/res_agi.c, apps/app_minivm.c, main/logger.c,
- res/res_realtime.c, main/image.c, apps/app_rpt.c,
- channels/chan_mgcp.c, res/res_clioriginate.c,
- res/res_config_pgsql.c, main/dnsmgr.c, channels/chan_zap.c,
- channels/chan_sip.c, res/res_limit.c, main/translate.c,
- res/res_convert.c, res/res_crypto.c, include/asterisk/cli.h,
- apps/app_queue.c, channels/chan_oss.c, main/rtp.c,
- channels/chan_jingle.c, channels/chan_usbradio.c, main/file.c,
- channels/chan_h323.c, pbx/pbx_dundi.c, main/astmm.c,
- funcs/func_devstate.c: Switch from AST_CLI (formerly NEW_CLI) to
- AST_CLI_DEFINE, since the former didn't make much sense
-
-2007-10-22 17:40 +0000 [r86790] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/astmm.c: Merged revisions 86787 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r86787 | tilghman | 2007-10-22 12:38:13 -0500 (Mon, 22 Oct 2007)
- | 2 lines Minor FreeBSD build fix ........
-
-2007-10-22 16:36 +0000 [r86755-86757] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 86756 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r86756 | file | 2007-10-22 13:35:22 -0300 (Mon, 22 Oct 2007) | 4
- lines After reading online I have confirmed that Record-Route
- headers should be copied to 1xx responses as well. (closes issue
- #10113) Reported by: makoto ........
-
- * /, apps/app_controlplayback.c: Merged revisions 86754 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r86754 | file | 2007-10-22 13:15:18 -0300 (Mon, 22 Oct 2007) | 4
- lines Make sure res is a positive value before performing the
- check to determine whether the user stopped it or not. (closes
- issue #11023) Reported by: cfc ........
-
-2007-10-22 15:57 +0000 [r86734-86751] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /: Merged revisions 86750 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r86750 | russell | 2007-10-22 10:52:48 -0500 (Mon, 22 Oct 2007) |
- 8 lines Don't leak a frame in the case that an END frame is
- received and the time since the BEGIN is less than that of the
- defined minimum DTMF duration. (closes issue #11051) Reported by:
- casper Patches: channel.c.86664.diff uploaded by casper (license
- 55) ........
-
- * channels/chan_zap.c: There is a really fun game that you can play
- before committing code, and it's called "make". :)
-
- * /, include/asterisk/lock.h: Merged revisions 86726 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r86726 | russell | 2007-10-22 10:43:30 -0500 (Mon, 22
- Oct 2007) | 4 lines Update the static mutex initializer to
- include the initialization of the internal mutex used to protect
- the lock debugging data. (closes issue #11044, patch suggested by
- Ivan) ........
-
-2007-10-22 14:59 +0000 [r86697] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c, configs/zapata.conf.sample: resetinterval
- defaulting to something other than 'never' doesn't seem to
- accomplish any good and causes problems for plenty of people...
-
-2007-10-22 14:58 +0000 [r86696] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 86694 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r86694 | mmichelson | 2007-10-22 09:48:46 -0500 (Mon, 22 Oct
- 2007) | 5 lines Account for the fact that sometimes headers may
- be terminated with \r\n instead of just \n (closes issue #11043,
- reported by yehavi) ........
-
-2007-10-22 14:56 +0000 [r86695] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/loader.c: merging patches that don't compile is bad...
- mmkay?
-
-2007-10-22 14:28 +0000 [r86631-86664] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /: Merged revisions 86663 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r86663 | file | 2007-10-22 11:27:03 -0300 (Mon, 22 Oct 2007) | 6
- lines Move log message to before the frame it references is
- freed. (closes issue #11050) Reported by: slavon Patches:
- channel.c.86662.diff uploaded by casper (license 55) ........
-
- * /, pbx/pbx_dundi.c: Merged revisions 86661 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r86661 | file | 2007-10-22 11:05:26 -0300 (Mon, 22 Oct 2007) | 6
- lines Fix tab completion for dundi show peer. (closes issue
- #11041) Reported by: jsmith Patches:
- asterisk-dundicomplete.diff.txt uploaded by jamesgolovich
- (license 176) ........
-
- * /, main/acl.c, main/loader.c: Merged revisions 86630 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r86630 | file | 2007-10-22 10:33:23 -0300 (Mon, 22 Oct
- 2007) | 6 lines Fixes for building under OpenSolaris. (closes
- issue #11047) Reported by: snuffy Patches: 11047-fixes.diff
- uploaded by snuffy (license 35) ........
-
-2007-10-22 10:18 +0000 [r86616-86617] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged
- revisions 86598 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r86598 | crichter | 2007-10-22 11:21:15 +0200 (Mo, 22 Okt 2007) |
- 1 line we send DISCONNECT instead of RELEASE/RELEASE_COMPLETE if
- the dialplan does not match after an overlap call. Also added
- out_cause=1 ........
-
- * channels/misdn/isdn_lib.h, channels/chan_misdn.c,
- channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c:
- started to add some basic support for supplementary services like
- CallForwarding and so forth
-
-2007-10-21 22:52 +0000 [r86585] Russell Bryant <russell@digium.com>
-
- * /, include/asterisk/cli.h, main/asterisk.c, main/cli.c: Merged
- revisions 85532 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r85532 | russell | 2007-10-13 00:24:33 -0500 (Sat, 13 Oct 2007) |
- 8 lines Properly handle the case where read() may return the text
- for more than one CLI command at once for a remote console.
- (closes issue #10888) Reported by: jamesgolovich Patches:
- asterisk-climultiple.diff.txt uploaded by jamesgolovich (license
- 176) ........
-
-2007-10-20 19:56 +0000 [r86572] Matthew Fredrickson <creslin@digium.com>
-
- * configs/zapata.conf.sample: Improved comments and organization
- for zapata.conf (#10904)
-
-2007-10-19 18:46 +0000 [r86549] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Add better support for blocking and
- unblocking of CICs (#10965)
-
-2007-10-19 18:29 +0000 [r86534-86536] Jason Parker <jparker@digium.com>
-
- * main/udptl.c, channels/chan_local.c, main/frame.c,
- res/res_features.c, main/threadstorage.c, channels/chan_iax2.c,
- main/astobj2.c, main/config.c, main/cli.c,
- channels/chan_skinny.c, main/http.c, pbx/pbx_ael.c,
- channels/chan_alsa.c, main/db.c, main/pbx.c,
- channels/chan_agent.c, channels/iax2-provision.c,
- apps/app_playback.c, channels/chan_misdn.c,
- channels/chan_features.c, res/res_indications.c,
- pbx/pbx_config.c, apps/app_mixmonitor.c, main/asterisk.c,
- res/res_odbc.c, apps/app_voicemail.c, apps/app_meetme.c,
- main/utils.c, channels/chan_gtalk.c, res/res_musiconhold.c,
- res/res_jabber.c, codecs/codec_zap.c, res/res_config_sqlite.c,
- main/channel.c, main/cdr.c, apps/app_osplookup.c, main/manager.c,
- res/res_agi.c, apps/app_minivm.c, main/logger.c,
- res/res_realtime.c, main/image.c, apps/app_rpt.c,
- channels/chan_mgcp.c, res/res_clioriginate.c,
- res/res_config_pgsql.c, main/dnsmgr.c, channels/chan_zap.c,
- channels/chan_sip.c, res/res_limit.c, main/translate.c,
- res/res_convert.c, res/res_crypto.c, include/asterisk/cli.h,
- apps/app_queue.c, channels/chan_oss.c, main/rtp.c,
- channels/chan_jingle.c, channels/chan_usbradio.c, main/file.c,
- channels/chan_h323.c, pbx/pbx_dundi.c, main/astmm.c,
- funcs/func_devstate.c: Convert NEW_CLI to AST_CLI. Closes issue
- #11039, as suggested by seanbright.
-
- * channels/chan_usbradio.c, res/res_config_pgsql.c,
- channels/chan_misdn.c, channels/chan_h323.c,
- res/res_indications.c, channels/chan_iax2.c, codecs/codec_zap.c,
- res/res_config_sqlite.c, main/config.c, main/rtp.c: More changes
- to NEW_CLI. Also fixes a few cli messages and some minor
- formatting. (closes issue #11001) Reported by: seanbright
- Patches: newcli.1.patch uploaded by seanbright (license 71)
- newcli.2.patch uploaded by seanbright (license 71) newcli.4.patch
- uploaded by seanbright (license 71) newcli.5.patch uploaded by
- seanbright (license 71) newcli.6.patch uploaded by seanbright
- (license 71) newcli.7.patch uploaded by seanbright (license 71)
-
-2007-10-19 16:40 +0000 [r86470-86503] Joshua Colp <jcolp@digium.com>
-
- * /, main/app.c: Merged revisions 86502 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r86502 | file | 2007-10-19 13:38:29 -0300 (Fri, 19 Oct 2007) | 4
- lines When returning a DTMF digit from ast_control_streamfile
- cast it as a char so that 0 does not overlap with the success
- return code. (closes issue #11023) Reported by: cfc ........
-
- * /, channels/chan_sip.c: Merged revisions 86471 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r86471 | file | 2007-10-19 12:33:49 -0300 (Fri, 19 Oct 2007) | 6
- lines Fix two issues with domains and transfers. If a port was
- given in the hostname it was treated as part of the hostname. If
- domains were configured but external domains were not enabled all
- transfers would be considered remote. (closes issue #11027)
- Reported by: ramonpeek Patches: 11027-1.diff uploaded by
- ramonpeek (license 266) ........
-
- * /, channels/chan_sip.c: Merged revisions 86469 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r86469 | file | 2007-10-19 12:08:12 -0300 (Fri, 19 Oct 2007) | 4
- lines Set port number in received as information for
- registrations as well. (closes issue #11028) Reported by: brad-x
- ........
-
-2007-10-19 01:56 +0000 [r86439] TransNexus OSP Development <support@transnexus.com>
-
- * apps/app_osplookup.c: Fixed a buffer size issue.
-
-2007-10-18 22:03 +0000 [r86407-86408] Jason Parker <jparker@digium.com>
-
- * Makefile, /: Merged revisions 86405 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes
- issue #11029) ........ r86405 | qwell | 2007-10-18 16:58:44 -0500
- (Thu, 18 Oct 2007) | 4 lines Add documentation for options in
- asterisk.conf Issue 11029, patch by eserra ........
-
-2007-10-18 18:40 +0000 [r86350] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_zap.c: Fixing a segfault from tab-completing a "zap
- restart" CLI command. (patch made by seanbright, pointed out in
- #asterisk-dev on IRC)
-
-2007-10-18 18:06 +0000 [r86331] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /, include/asterisk/channel.h: Merged revisions
- 86330 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r86330 | russell | 2007-10-18 13:03:10 -0500 (Thu, 18 Oct 2007) |
- 10 lines The channel needs to stay locked while running timer
- callbacks, as they access and modify channel data that may change
- elsewhere. I went through every timer callback in the source tree
- to make sure that none of them did any additional locking that
- could introduce deadlocks, and all is well. (closes issue #10765)
- Reported by: Ivan Patches: ast_1_4_11_svn_patch_channel_rc.diff
- uploaded by Ivan (license 229) ........
-
-2007-10-18 17:40 +0000 [r86298-86329] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 86328 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r86328 | mmichelson | 2007-10-18 12:38:26 -0500 (Thu, 18 Oct
- 2007) | 5 lines If a non-existent file is specified to be played
- either as a periodic announcement or as a hold/position
- announcement, the caller would be kicked out of the queue. No
- longer does this happen. ........
-
- * apps/app_queue.c: Changed some spaces to tabs
-
-2007-10-18 15:57 +0000 [r86297] Russell Bryant <russell@digium.com>
-
- * /, codecs/codec_zap.c: Merged revisions 86296 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r86296 | russell | 2007-10-18 10:45:55 -0500 (Thu, 18 Oct 2007) |
- 3 lines Execute the RELEASE operation on transcoder channels in
- the destroy callback. (patch from jsloan) ........
-
-2007-10-18 07:23 +0000 [r86277-86278] Tilghman Lesher <tlesher@digium.com>
-
- * main/acl.c: Code cleanup of acl.c Reported by dimas Closes issue
- #10784
-
- * res/res_musiconhold.c: On reload, re-read the files in the
- specified moh directory (closes issue #10536)
-
-2007-10-18 04:41 +0000 [r86238] Russell Bryant <russell@digium.com>
-
- * /, main/utils.c: Merged revisions 86237 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r86237 | russell | 2007-10-17 23:40:52 -0500 (Wed, 17 Oct 2007) |
- 9 lines Revert a change that I made for issue #10979 which, as
- has been pointed out to me in issue #11018, doesn't really make
- sense. There is no reason to have the base64 decode function
- force a '\0' terminated buffer, when the result is almost always
- binary, anyway. In fact, this caused some breakage, as some code
- in res_crypto passed in a buffer exactly the right size to get
- its binary result, which got stomped on by this patch. (closes
- issue #11018, reported by dimas) ........
-
-2007-10-17 21:41 +0000 [r86208] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 86202 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r86202 | mmichelson | 2007-10-17 16:39:05 -0500 (Wed, 17 Oct
- 2007) | 6 lines Changing the strategy field of the call_queue
- struct to be signed instead of unsigned, since the code attempts
- to set the strategy to -1 if you specify a bogus strategy. While
- this isn't a huge issue in 1.4, it could be a problem for someone
- who, say, tries to use the roundrobin strategy in trunk (despite
- all the deprecation warnings in 1.4). ........
-
-2007-10-17 21:16 +0000 [r86195-86197] Tilghman Lesher <tlesher@digium.com>
-
- * main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2.c: Simplify
- some preprocessor logic by using #elif
-
- * CHANGES, configs/meetme.conf.sample: Document the changes made
- earlier today to meetme
-
-2007-10-17 20:06 +0000 [r86180-86182] Steve Murphy <murf@digium.com>
-
- * utils/hashtest2.c, utils/check_expr.c, utils/clicompat.c: and
- then, I noticed the clicompat stuff.
-
- * utils/check_expr.c: more stub routines to allow linkage in
- stand-alone environment, with thread debugs turned on
-
- * utils/hashtest2.c: more stub routines to allow linkage in
- stand-alone environment, with thread debugs turned on
-
-2007-10-17 18:01 +0000 [r86150] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 86149 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r86149 | russell | 2007-10-17 12:57:45 -0500 (Wed, 17 Oct 2007) |
- 8 lines If Asterisk is in the middle of shutting down, respond to
- OPTIONS with 503 Unavailable. (closes issue #10994) Reported by:
- eserra Patches: sip-options-503.patch uploaded by eserra (license
- 45) ........
-
-2007-10-17 17:06 +0000 [r86119] Tilghman Lesher <tlesher@digium.com>
-
- * main/term.c: Support color on certain platforms, even when
- started at boot (before TERM is set) Closes issue #9048
-
-2007-10-17 17:00 +0000 [r86118] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 86117 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r86117 | file | 2007-10-17 13:58:03 -0300 (Wed, 17 Oct 2007) | 4
- lines Whoops, forgot to remove the original sip_scheddestroy.
- (closes issue #11010) Reported by: vadim ........
-
-2007-10-17 16:09 +0000 [r86104] Jason Parker <jparker@digium.com>
-
- * channels/chan_usbradio.c, channels/xpmr/xpmr.c: Allow
- chan_usbradio to compile again. Closes issue #11014, patch by
- seanbright.
-
-2007-10-17 15:39 +0000 [r86079] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/asterisk.c: Merged revisions 86066 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r86066 | tilghman | 2007-10-17 10:23:51 -0500 (Wed, 17 Oct 2007)
- | 3 lines When runuser/rungroup is specified, a remote console
- could only be attained by root (Closes issue #9999) ........
-
-2007-10-17 15:30 +0000 [r86067] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_usbradio.c: Change dependency for chan_usbradio to
- asound. Let's keep everything uniform. (closes issue #11013)
- Reported by: seanbright
-
-2007-10-17 15:13 +0000 [r86065] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_meetme.c: Enhancements to realtime (closes issue #9609)
-
-2007-10-17 15:09 +0000 [r86064] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 86063 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r86063 | file | 2007-10-17 12:06:36 -0300 (Wed, 17 Oct 2007) | 4
- lines Don't schedule dialog destruction if a MESSAGE is received
- using an existing dialog. (closes issue #11010) Reported by:
- vadim ........
-
-2007-10-16 23:36 +0000 [r86029-86033] Mark Michelson <mmichelson@digium.com>
-
- * /, configs/queues.conf.sample: Merged revisions 86032 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r86032 | mmichelson | 2007-10-16 18:35:31 -0500 (Tue, 16 Oct
- 2007) | 3 lines Since monitor-join is deprecated now, remove the
- example from the sample queues.conf file ........
-
- * apps/app_queue.c: Removed the monitor-join option. If one wishes
- to mix audio, they should instead use monitor-type=mixmonitor.
- (related to issue #10885)
-
-2007-10-16 22:36 +0000 [r85995-85998] Russell Bryant <russell@digium.com>
-
- * /, include/asterisk/lock.h: Merged revisions 85997 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r85997 | russell | 2007-10-16 17:36:16 -0500 (Tue, 16
- Oct 2007) | 1 line really picky formatting tweak ... ........
-
- * /, include/asterisk/lock.h: Merged revisions 85994 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r85994 | russell | 2007-10-16 17:14:36 -0500 (Tue, 16
- Oct 2007) | 16 lines Some locking errors exposed the fact that
- the lock debugging code itself was not thread safe. How ironic!
- Anyway, these changes ensure that the code that is accessing the
- lock debugging data is thread-safe. Many thanks to Ivan for
- finding and fixing the core issue here, and also thanks to those
- that tested the patch and provided test results. (closes issue
- #10571) (closes issue #10886) (closes issue #10875) (might close
- some others, as well ...) Patches: (from issue #10571)
- ivan_ast_1_4_12_rel_patch_lock.h.diff uploaded by Ivan (license
- 229) - a few small changes by me ........
-
-2007-10-16 21:51 +0000 [r85959-85992] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Fixing the build.
-
- * apps/app_read.c: Fixing app_read so that if a timeout of less
- than 1 ms is specified, assume that 1 ms is desired. (closes
- issue #11000, reported and patched by michael-fig, with a warning
- line added by me)
-
- * /, apps/app_queue.c: Merged revisions 85958 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r85958 | mmichelson | 2007-10-16 16:14:34 -0500 (Tue, 16 Oct
- 2007) | 5 lines Trying to remove a non-dynamic queue member via
- dynamic means can lead to some interesting (read nasty)
- situations. This patch clears up the issue by making only dynamic
- queue members removable via dynamic methods. ........
-
-2007-10-16 20:55 +0000 [r85957] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Don't hangup the call for SS7 if we get an
- alarm
-
-2007-10-16 20:32 +0000 [r85944] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: This fixes SIP subscriptions in trunk. Don't
- improperly memset() over an ast_str. This was leftover from
- before it got changed to use ast_str. (closes issue #11003,
- reported by pj) (closes issue #10770, reported by yehavi)
- (patched by me)
-
-2007-10-16 19:47 +0000 [r85943] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/stdtime/localtime.c: Merged revisions 85921 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r85921 | tilghman | 2007-10-16 14:41:40 -0500 (Tue, 16
- Oct 2007) | 4 lines Also set up gmtoff (this is used in the %z
- gnu extension to strftime) Reported and fixed by jcmoore Closes
- issue #11002 ........
-
-2007-10-16 19:12 +0000 [r85897] Russell Bryant <russell@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 85896 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r85896 | russell | 2007-10-16 14:10:01 -0500 (Tue, 16 Oct 2007) |
- 2 lines Remove a pointless lock. ........
-
-2007-10-16 16:40 +0000 [r85853-85883] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Fix IMAP compilation error. (closes issue
- #10986, reported and patched by snuffy)
-
- * /: Blocking changes from previous commit
-
-2007-10-16 15:15 +0000 [r85819-85851] Joshua Colp <jcolp@digium.com>
-
- * /, funcs/func_vmcount.c: Merged revisions 85850 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r85850 | file | 2007-10-16 11:52:22 -0300 (Tue, 16 Oct 2007) | 4
- lines Check to make sure a value has been given to the VMCOUNT
- dialplan function. (closes issue #10996) Reported by: marsosa
- ........
-
- * main/threadstorage.c: Permit building under DEBUG_THREADLOCALS.
- Thanks snuff.
-
- * /, main/threadstorage.c: Merged revisions 85818 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r85818 | file | 2007-10-16 11:19:39 -0300 (Tue, 16 Oct 2007) | 6
- lines Fix memory allocation issue in threadstorage. (closes issue
- #10995) Reported by: snuffy Patches: new-patch.diff uploaded by
- snuffy (license 35) ........
-
-2007-10-16 10:38 +0000 [r85777-85787] Philippe Sultan <philippe.sultan@gmail.com>
-
- * channels/chan_jingle.c, channels/chan_gtalk.c: Fix CLI help
- output
-
- * channels/chan_jingle.c: Added two CLI functions, taken from
- chan_gtalk : - jingle reload ; - jingle show channels.
-
- * channels/chan_jingle.c: Make an audio path under the following
- call configuration : SIP Phone 1 --- [chan_sip]Asterisk
- 1[chan_jingle] --- [chan_jingle]Asterisk 2[chan_sip] --- SIP
- Phone 2 Modifications : - set bridge type to partial ; - process
- media candidates from the remote peer properly. Now we have
- Jingle audio, at least between two Asterisk Jingle clients.
-
-2007-10-15 23:20 +0000 [r85764] Jason Parker <jparker@digium.com>
-
- * configs/dundi.conf.sample, channels/chan_sip.c,
- channels/chan_h323.c, main/acl.c, UPGRADE.txt,
- channels/iax2-provision.c, doc/tex/qos.tex, pbx/pbx_dundi.c,
- channels/chan_iax2.c, channels/chan_mgcp.c: Switch dundi to new
- tos config format. Remove old unused defines for old style.
- Closes issue 10860, patch by IgorG.
-
-2007-10-15 21:11 +0000 [r85718-85721] Russell Bryant <russell@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 85720 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r85720 | russell | 2007-10-15 16:10:02 -0500 (Mon, 15 Oct 2007) |
- 3 lines Ensure that no pending state changes are leaked when the
- device state change thread gets stopped on module unload.
- ........
-
- * /, main/say.c: Merged revisions 85686 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r85686 | russell | 2007-10-15 15:21:27 -0500 (Mon, 15 Oct 2007) |
- 7 lines Add a small fix for the tw version of saying dates.
- (closes issue #7827) Reported by: sharkey Patches: say.nits.patch
- uploaded by sharkey (license 172) ........
-
-2007-10-15 20:16 +0000 [r85685] Jason Parker <jparker@digium.com>
-
- * Makefile, /: Merged revisions 85684 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #10938) ........ r85684 | qwell | 2007-10-15 15:15:51 -0500
- (Mon, 15 Oct 2007) | 5 lines Properly use DESTDIR in 'config'
- target. Do not try to run chkconfig or similar if using DESTDIR.
- Issue 10938, patch by cabal95. ........
-
-2007-10-15 20:09 +0000 [r85648-85683] Russell Bryant <russell@digium.com>
-
- * doc/tex/channelvariables.tex: add TOUCH_MONITOR_PREF to the
- channel var docs
-
- * res/res_features.c, CHANGES: Added support for reading the
- TOUCH_MONITOR_PREFIX channel variable. It allows you to configure
- a prefix for auto-monitor recordings. (closes issue #6353)
- Reported by: ivanfm Patches: asterisk_automon_v4.patch uploaded
- by ivanfm (original patch) - updated patch:
- 6353-touch_monitor_prefix.diff uploaded by qwell (license 4)
-
- * /, main/utils.c: Merged revisions 85649 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r85649 | russell | 2007-10-15 14:22:45 -0500 (Mon, 15 Oct 2007) |
- 2 lines Be pedantic about handling memory allocation failure.
- ........
-
- * /, main/utils.c: Merged revisions 85647 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r85647 | russell | 2007-10-15 14:11:38 -0500 (Mon, 15 Oct 2007) |
- 5 lines The loop in the handler for the "core show locks" could
- potentially block for some amount of time. Be a little bit more
- careful and prepare all of the output in an intermediary buffer
- while holding a global resource. Then, after releasing it, send
- the output to ast_cli(). ........
-
-2007-10-15 17:51 +0000 [r85633] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_strings.c: Document my changes from Friday
-
-2007-10-15 16:59 +0000 [r85605] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 85604 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r85604 | russell | 2007-10-15 11:54:57 -0500 (Mon, 15 Oct 2007) |
- 6 lines Make the default for the srvlookup option to be yes. It
- doesn't really make sense for it to default to off. The default
- configuration file has it on, and proper RFC behavior, as
- indicated by a comment in the code, is for it to be on. So, let's
- have it on by default to make lives easier. (closes issue #10954,
- suggested by jtodd) ........
-
-2007-10-15 16:41 +0000 [r85578] Joshua Colp <jcolp@digium.com>
-
- * /, configs/features.conf.sample: Merged revisions 85571 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r85571 | file | 2007-10-15 13:39:59 -0300 (Mon, 15 Oct 2007) | 4
- lines Document that DTMF based features only work when two
- channels are bridged together. (closes issue #10773) Reported by:
- pbayley ........
-
-2007-10-15 16:36 +0000 [r85562] Russell Bryant <russell@digium.com>
-
- * /, include/asterisk/strings.h: Merged revisions 85561 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r85561 | russell | 2007-10-15 11:34:13 -0500 (Mon, 15 Oct 2007) |
- 4 lines Make a few changes so that characters in the upper half
- of the ISO-8859-1 character set don't get stripped when reading
- configuration. (closes issue #10982, dandre) ........
-
-2007-10-15 16:23 +0000 [r85560] Joshua Colp <jcolp@digium.com>
-
- * /, main/rtp.c: Merged revisions 85559 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r85559 | file | 2007-10-15 13:22:02 -0300 (Mon, 15 Oct 2007) | 4
- lines Bring both DTMF begin and end frames up through to the core
- for DTMF feature handling. (closes issue #10826) Reported by:
- dimas ........
-
-2007-10-15 15:55 +0000 [r85557-85558] Russell Bryant <russell@digium.com>
-
- * pbx/dundi-parser.c: Simplify buffer handling in dundi-parser.c.
- This also makes the code a bit safer by removing various
- assumptions about sizes. (No vulnerabilities, though) (closes
- issue #10977) Reported by: dimas Patches: dundiparser.patch
- uploaded by dimas (license 88)
-
- * /, pbx/pbx_dundi.c: Merged revisions 85556 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r85556 | russell | 2007-10-15 10:40:45 -0500 (Mon, 15 Oct 2007) |
- 9 lines Ensure the buffer passed to ast_canmatch_extension() is
- properly initialized so that it is null terminated. (issue
- #10977) Reported by: dimas Patches: pbxdundi.patch uploaded by
- dimas (license 88) - small mods by me ........
-
-2007-10-15 15:26 +0000 [r85555] Philippe Sultan <philippe.sultan@gmail.com>
-
- * channels/chan_jingle.c: Allow RTP structure registration
-
-2007-10-15 15:07 +0000 [r85553-85554] Joshua Colp <jcolp@digium.com>
-
- * main/frame.c: Add packetization data for G.722. (closes issue
- #10900) Reported by: andrew Patches: frame.diff uploaded by
- andrew (license 240)
-
- * /, main/rtp.c: Merged revisions 85552 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r85552 | file | 2007-10-15 11:55:04 -0300 (Mon, 15 Oct 2007) | 4
- lines If Monitor or a spy was added to a P2P or native bridged
- channel bring the channel back to the generic bridging core so
- the monitor or spy operations work. (closes issue #10943)
- Reported by: julianjm ........
-
-2007-10-15 13:51 +0000 [r85551] Philippe Sultan <philippe.sultan@gmail.com>
-
- * res/res_jabber.c: Allocate more space for the base64 output we
- need to generate. Closes issue #10913, reported by tootai, who
- graciously granted us access to his Asterisk server, thanks!
- Daniel, feel free to reopen the bug in case you can reproduce
- this on 1.4.
-
-2007-10-15 13:44 +0000 [r85539-85550] Russell Bryant <russell@digium.com>
-
- * main/cli.c: Move the CLI commands that were in builtins[] into
- the cli_cli[] array of CLI commands and remove the cli_iterator
- struct. This gets tab completion working again. (closes issue
- #10970) Reported by: jamesgolovich Patches:
- asterisk-clicomplete.diff.txt uploaded by jamesgolovich (license
- 176)
-
- * doc/tex/jitterbuffer.tex, doc/tex/extensions.tex,
- doc/tex/channelvariables.tex, doc/tex/ael.tex,
- doc/tex/queues-with-callback-members.tex, doc/tex/realtime.tex,
- doc/tex/dundi.tex, doc/tex/security.tex,
- doc/tex/configuration.tex, doc/tex/ajam.tex,
- doc/tex/cliprompt.tex, doc/tex/manager.tex, doc/tex/misdn.tex,
- doc/tex/imapstorage.tex, doc/tex/privacy.tex, doc/tex/sla.tex,
- doc/tex/app-sms.tex, doc/tex/billing.tex, apps/app_zapateller.c,
- doc/tex/localchannel.tex, doc/tex/cdrdriver.tex,
- doc/tex/queuelog.tex: Another major doc directory update from
- IgorG. This patch includes - Many uses of the astlisting
- environment around verbatim text to ensure that it gets properly
- formatted and doesn't run off the page. - Update some things that
- have been deprecated. - Add escaping as needed - and more ...
- (closes issue #10978) Reported by: IgorG Patches:
- texdoc-85542-1.patch uploaded by IgorG (license 20)
-
- * /, main/asterisk.c: Merged revisions 85545 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r85545 | russell | 2007-10-15 08:05:45 -0500 (Mon, 15 Oct 2007) |
- 7 lines Make sure remote consoles unmute themselves again after
- reconnecting. (closes issue #10847) Reported by: atis Patches:
- console_unmute_on_reconnect.patch uploaded by atis (license 242)
- ........
-
- * /, main/utils.c: Merged revisions 85543 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r85543 | russell | 2007-10-15 07:48:10 -0500 (Mon, 15 Oct 2007) |
- 8 lines Make sure that the base64 decoder returns a terminated
- string. (closes issue #10979) Reported by: ys Patches:
- util.c.diff uploaded by ys (license 281) - small mods by me
- ........
-
- * configure, configure.ac: Change the configure script to check for
- a function that was recently added to libss7.
-
- * /, pbx/pbx_config.c: Merged revisions 85540 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r85540 | russell | 2007-10-14 10:24:52 -0500 (Sun, 14 Oct 2007) |
- 7 lines Don't create the context for users in users.conf until we
- know at least one user exists. (closes issue #10971) Reported by:
- dimas Patches: pbxconfig.patch uploaded by dimas (license 88)
- ........
-
- * doc/tex/backtrace.tex (added): When merging the last
- documentation update, I forgot to "svn add" a file. Here it is.
- (closes issue #10962)
-
-2007-10-13 08:38 +0000 [r85535] James Golovich <james@gnuinter.net>
-
- * main/cli.c: Fix compiling cli.c due to differences with new cli
- system (closes issue 0010966)
-
-2007-10-13 05:53 +0000 [r85534] Russell Bryant <russell@digium.com>
-
- * include/asterisk/logger.h, /, main/asterisk.c, main/cli.c: Merged
- revisions 85533 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r85533 | russell | 2007-10-13 01:48:10 -0400 (Sat, 13 Oct 2007) |
- 12 lines Fix an issue with console verbosity when running
- asterisk -rx to execute a command and retrieve its output. The
- issue was that there was no way for the main Asterisk process to
- know that the remote console was connecting in the -rx mode. The
- way that James has fixed this is to have all remote consoles
- muted by default. Then, regular remote consoles automatically
- execute a CLI command to unmute themselves when they first start
- up. (closes issue #10847) Reported by: atis Patches:
- asterisk-consolemute.diff.txt uploaded by jamesgolovich (license
- 176) ........
-
-2007-10-12 20:06 +0000 [r85527] Mark Michelson <mmichelson@digium.com>
-
- * configs/queues.conf.sample, apps/app_queue.c: Allow for the
- position announcement to be turned off if desired. (closes issue
- #8515, reported by bruno_rocha, initial patch by bruno_rocha,
- final patch by qwell)
-
-2007-10-12 19:41 +0000 [r85525-85526] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c, doc/tex/channelvariables.tex: Trying to
- finish the last of the charge_number patch up #10916
-
- * channels/chan_zap.c: Add support for receive charge number in
- dialplan #10916
-
-2007-10-12 18:37 +0000 [r85522-85524] Tilghman Lesher <tlesher@digium.com>
-
- * doc/asterisk-mib.txt, doc/PEERING, /, LICENSE: Merged revisions
- 85523 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r85523 | tilghman | 2007-10-12 13:30:55 -0500 (Fri, 12 Oct 2007)
- | 2 lines Change Digium address ........
-
- * funcs/func_strings.c: Enable ranges, hexadecimal, octal, and
- special backslashed characters for the FILTER function
-
-2007-10-12 15:50 +0000 [r85516-85519] Russell Bryant <russell@digium.com>
-
- * doc/tex/odbcstorage.tex, doc/tex/extensions.tex,
- doc/tex/channelvariables.tex, doc/tex/ael.tex,
- doc/tex/queues-with-callback-members.tex, doc/tex/dundi.tex,
- doc/tex/enum.tex, doc/tex/cliprompt.tex, doc/tex/manager.tex,
- doc/tex/privacy.tex, doc/tex/sla.tex, doc/tex/app-sms.tex,
- doc/tex/localchannel.tex, doc/tex/ices.tex,
- doc/tex/cdrdriver.tex, doc/tex/asterisk.tex: Many doc directory
- improvements, including: - Added development section
- (backtrace.tex) - Correct filesystem path formating - Replace all
- "|" argument separator to "," - Endless count of spaces at the
- end of line - Using astlisting to make listings do not take so
- much place - Take back ASTRISKVERSION on first page - Make
- localchannel.tex readable by inserting extra end of lines (closes
- issue #10962) Reported by: IgorG Patches: texdoc-85177-1.patch
- uploaded by IgorG (license 20)
-
- * res/res_smdi.c, /: Merged revisions 85517 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r85517 | russell | 2007-10-12 10:45:09 -0500 (Fri, 12 Oct 2007) |
- 3 lines Fix a spelling error in a log message. SMDI, not SDMI.
- (closes issue #10959) ........
-
- * /, pbx/pbx_realtime.c: Merged revisions 85515 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r85515 | russell | 2007-10-12 10:40:35 -0500 (Fri, 12 Oct 2007) |
- 7 lines Fix the potential use of an uninitialized buffer in a log
- message. (closes issue #10958) Reported by: dimas Patches:
- realtime.patch uploaded by dimas (license 88) ........
-
-2007-10-11 22:42 +0000 [r85474-85499] Matthew Fredrickson <creslin@digium.com>
-
- * apps/app_dial.c: Make sure we propogate ANI2 to the outbound
- channel
-
- * funcs/func_callerid.c: See if I can fix this borked ANI2 code I
- added
-
- * channels/chan_zap.c: Make sure we set the ANI2 field for PRI
-
- * funcs/func_callerid.c: Add ANI2 support to func_callerid
-
- * channels/chan_zap.c: Add SS7 ANI2 support tx and rx. #10916
-
- * channels/chan_zap.c: Add CCR test support #10916
-
-2007-10-11 19:03 +0000 [r85460] Russell Bryant <russell@digium.com>
-
- * main/udptl.c, main/threadstorage.c, res/res_limit.c,
- main/translate.c, res/res_crypto.c, res/res_convert.c,
- channels/iax2-provision.c, channels/chan_gtalk.c,
- channels/chan_oss.c, main/astobj2.c, main/cli.c, main/cdr.c,
- main/channel.c, apps/app_osplookup.c, channels/chan_skinny.c,
- pbx/pbx_ael.c, main/file.c, pbx/pbx_dundi.c, main/image.c,
- pbx/pbx_config.c, apps/app_mixmonitor.c, apps/app_rpt.c,
- main/asterisk.c, main/db.c, channels/chan_mgcp.c,
- res/res_clioriginate.c: 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)
-
-2007-10-11 17:17 +0000 [r85431-85444] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Let's hard code this until I fix it
-
- * channels/chan_zap.c: Make sure we are clean to build without
- libpri
-
-2007-10-11 04:40 +0000 [r85357] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c, /: Merged revisions 85356 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r85356 | tilghman | 2007-10-10 23:35:33 -0500 (Wed, 10 Oct 2007)
- | 2 lines A dollar sign by itself, not indicating a start of a
- variable or expression prematurely ends substitution (closes
- issue #10939) ........
-
-2007-10-10 16:01 +0000 [r85317] Russell Bryant <russell@digium.com>
-
- * include/asterisk/file.h, /: Merged revisions 85316 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r85316 | russell | 2007-10-10 10:56:23 -0500 (Wed, 10
- Oct 2007) | 6 lines I introduced a new member to the
- ast_filestream struct in 1.4.12, but put it in the middle of the
- struct, instead of at the end. One of the Debian folks, paravoid,
- pointed out that this breaks binary compatability with modules
- compiled against older headers. So, I'm moving the new member to
- the end of the struct to resolve the situation. ........
-
-2007-10-10 14:43 +0000 [r85281] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 85280 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r85280 | file | 2007-10-10 11:42:00 -0300 (Wed, 10 Oct 2007) | 4
- lines If devicestate is passed a port number strip it out.
- (closes issue #10930) Reported by: ibc ........
-
-2007-10-10 14:38 +0000 [r85279] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 85276 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r85276 | mmichelson | 2007-10-10 09:26:31 -0500 (Wed, 10 Oct
- 2007) | 5 lines A bunch of changes from sprintf to snprintf. See
- security advisory AST-2002-022 ........
-
-2007-10-10 14:30 +0000 [r85234-85278] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 85277 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r85277 | file | 2007-10-10 11:28:18 -0300 (Wed, 10 Oct 2007) | 6
- lines Add support for handling a 182 Queued response. (closes
- issue #10924) Reported by: ramonpeek Patches: queued-182.diff
- uploaded by ramonpeek (license 266) ........
-
- * /, apps/app_voicemail.c: Merged revisions 85242 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r85242 | file | 2007-10-10 11:14:56 -0300 (Wed, 10 Oct 2007) | 6
- lines Close voicemail message description file if duration did
- not meet the minimum, or else we will eventually run out of file
- descriptors. (closes issue #10918) Reported by: brak2718 Patches:
- vm1.4.12.1.patch uploaded by brak2718 (license 279) ........
-
- * main/logger.c: Process outstanding log messages before shutting
- down the logger thread. (closes issue #10933) Reported by:
- sperreault
-
-2007-10-10 06:48 +0000 [r85197] Luigi Rizzo <rizzo@icir.org>
-
- * bootstrap.sh: Adapt the autotools names to different versions of
- FreeBSD (and open the way to better adaptation for other
- platforms as well).
-
-2007-10-10 06:41 +0000 [r85196] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, include/asterisk/frame.h: Merged revisions 85195 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r85195 | kpfleming | 2007-10-10 08:24:41 +0200 (Wed, 10
- Oct 2007) | 2 lines use a macro instead of an inline function, so
- that backtraces will report the caller of ast_frame_free()
- properly ........
-
-2007-10-09 22:35 +0000 [r85177] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Patch to add one-touch parking for queues.
- (closes issue #10869, reported and patched by bluecrow76)
-
-2007-10-09 22:21 +0000 [r85140-85176] Tilghman Lesher <tlesher@digium.com>
-
- * main/channel.c, /, main/utils.c, include/asterisk/lock.h: Merged
- revisions 85158 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r85158 | tilghman | 2007-10-09 16:55:06 -0500 (Tue, 09 Oct 2007)
- | 5 lines This commit fixes the following issues: - Deadlock in
- ast_write (issue #10406) - Deadlock in ast_read (issue #10406) -
- Possible mutex initialization error in lock.h (issue #10571)
- ........
-
- * apps/app_dial.c, channels/chan_jingle.c, channels/chan_misdn.c,
- apps/app_festival.c, apps/app_minivm.c, apps/app_zapras.c,
- utils/astman.c, apps/app_adsiprog.c, utils/check_expr.c: Remove
- redundant includes (patch by snuffy) (Closes issue #10922)
-
-2007-10-09 15:12 +0000 [r85097-85098] Russell Bryant <russell@digium.com>
-
- * CHANGES: Note jitterbuffer support for chan_local in CHANGES
-
- * channels/chan_local.c, doc/tex/localchannel.tex: Add jitterbuffer
- support for chan_local. To enable it, you use the 'j' option in
- the Dial command. The 'j' option _must_ be used in conjunction
- with the 'n' option. This feature will allow you to use the
- existing jitterbuffer implementation to put a jitterbuffer on
- incoming SIP calls connecting to Asterisk applications by putting
- a local channel in the middle.
-
-2007-10-09 14:31 +0000 [r84991-85094] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 85093 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r85093 | file | 2007-10-09 11:30:16 -0300 (Tue, 09 Oct 2007) | 4
- lines Don't perform a reinvite if a transfer is in progress.
- (issue #10915) Reported by: ramonpeek ........
-
- * /, main/rtp.c: Merged revisions 85057 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r85057 | file | 2007-10-08 17:06:33 -0300 (Mon, 08 Oct 2007) | 4
- lines Only update codec information if the channel has a
- technology private structure. (issue #10915) Reported by:
- ramonpeek ........
-
- * res/res_limit.c, utils/hashtest2.c, utils/conf2ael.c,
- main/ast_expr2.c, utils/check_expr.c: Fix up tree so that it
- compiles when MTX Profiling is enabled. (closes issue #10898)
- Reported by: snuffy Patches: 10898-mtx_prof.diff uploaded by
- qwell (license 4)
-
- * /, main/rtp.c: Merged revisions 85023 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r85023 | file | 2007-10-08 12:37:46 -0300 (Mon, 08 Oct 2007) | 4
- lines Update codec information as well as address when doing hold
- reinvites. (issue #10868) Reported by: mavince ........
-
- * main/channel.c, /: Merged revisions 84990 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r84990 | file | 2007-10-08 12:03:07 -0300 (Mon, 08 Oct 2007) | 4
- lines Don't keep trying to native bridge if either of the
- channels are involved in a masquerade operation to be done.
- (closes issue #10696) Reported by: tbelder ........
-
-2007-10-08 03:29 +0000 [r84958] Russell Bryant <russell@digium.com>
-
- * /, Makefile.rules: Merged revisions 84957 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r84957 | russell | 2007-10-07 22:28:34 -0500 (Sun, 07 Oct 2007) |
- 6 lines Enable file dependency tracking for _all_ builds, and not
- just for builds with dev-mode enabled. I have seen enough
- problems caused by this that I don't think it's worth keeping. I
- want to continue to encourage anybody that is interested to
- continue to run Asterisk from svn. Furthermore, I do not want
- their systems to break when we change a structure definition in a
- header file. :) ........
-
-2007-10-07 16:28 +0000 [r84891-84939] Philippe Sultan <philippe.sultan@gmail.com>
-
- * configs/jabber.conf.sample, include/asterisk/jabber.h,
- res/res_jabber.c: Make the status and priority configurable.
- Closes issue #10785, patch by Luke-Jr, thanks!
-
- * /, res/res_jabber.c: Merged revisions 84902 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r84902 | phsultan | 2007-10-07 18:15:39 +0200 (Sun, 07 Oct 2007)
- | 5 lines Presence packets from a client who's connected with our
- Jabber ID are valid, therefore, those clients must be considered
- as buddies. The resource string helps us make the distinction
- between clients. Closes issue #10707, reported by yusufmotiwala.
- ........
-
- * res/res_jabber.c: Fix indentation
-
- * /, res/res_jabber.c: Merged revisions 84890 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r84890 | phsultan | 2007-10-07 17:52:44 +0200 (Sun, 07 Oct 2007)
- | 5 lines Prevent Asterisk from crashing when receiving a
- presence packet without resource from a buddy that is known to
- have a resource list. Revert a change I previously made, where
- Asterisk could point to a freed memory location. ........
-
-2007-10-05 19:48 +0000 [r84852] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/db.c: Merged revisions 84851 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r84851 | tilghman | 2007-10-05 14:42:21 -0500 (Fri, 05 Oct 2007)
- | 2 lines Log exactly why we can't open the database, if we fail
- (closes issue #10887) ........
-
-2007-10-05 18:57 +0000 [r84819] Joshua Colp <jcolp@digium.com>
-
- * /, main/rtp.c: Merged revisions 84818 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r84818 | file | 2007-10-05 15:55:36 -0300 (Fri, 05 Oct 2007) | 4
- lines Update the remembered RTP peer information when putting an
- endpoint on hold or taking it off hold so that the RTP stack does
- not initiate a needless reinvite. (closes issue #10868) Reported
- by: mavince ........
-
-2007-10-05 16:49 +0000 [r84743-84784] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 84783 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r84783 | russell | 2007-10-05 11:44:21 -0500 (Fri, 05 Oct 2007) |
- 4 lines Do deadlock avoidance in a couple more places. You can't
- lock two channels at the same time without doing extra work to
- make sure it succeeds. (closes issue #10895, patch by me)
- ........
-
- * main/manager.c, /: Merged revisions 84742 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r84742 | russell | 2007-10-04 20:39:07 -0500 (Thu, 04 Oct 2007) |
- 3 lines Fix a copy/paste error in the description of UpdateConfig
- that was pointed out by JerJer on #asterisk-dev ........
-
-2007-10-04 22:58 +0000 [r84693-84726] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: A two-in-one patch from the bugtracker 1) Fix
- some bad logic in the counting of statistics for QueueSummary
- manager event. Variables were not being reset for each additional
- queue, so cumulative totals were reported on each successive
- queue. 2) Add a longest hold time stat to QueueSummary manager
- event.
-
- * /, apps/app_queue.c: Merged revisions 84692 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r84692 | mmichelson | 2007-10-04 16:57:03 -0500 (Thu, 04 Oct
- 2007) | 5 lines Don't allocate space for queue members unless
- it's needed. You end up deleting dynamic members on a reload. Not
- good. closes issue (#10879, reported by dazza76, patched by me)
- ........
-
-2007-10-04 21:38 +0000 [r84691] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 84690 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r84690 | kpfleming | 2007-10-04 16:36:56 -0500 (Thu, 04 Oct 2007)
- | 2 lines callers of sig2str already add the word 'signalling' in
- the appropriate place, so don't duplicate it ........
-
-2007-10-04 16:56 +0000 [r84671] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_jabber.c: Update to current coding standards, also
- changing the argument delimiter to ',' (Closes issue #10876)
-
-2007-10-04 14:54 +0000 [r84613-84638] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 84637 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r84637 | file | 2007-10-04 11:51:57 -0300 (Thu, 04 Oct 2007) | 4
- lines Create a duplicate of the channel's member name as the tab
- completion stuff will free it. (closes issue #10884) Reported by:
- adamg ........
-
- * main/pbx.c: Don't register the exception function with module
- information. Since it is in the core there is none and it will
- explode.
-
-2007-10-03 23:05 +0000 [r84580-84582] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/rtp.c: Merged revisions 84581 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r84581 | tilghman | 2007-10-03 17:59:17 -0500 (Wed, 03 Oct 2007)
- | 2 lines When an RFC 2833 event is sent that we don't recognize,
- ignore it, don't queue a NULL digit (closes issue #10877)
- ........
-
- * main/pbx.c, doc/tex/extensions.tex, include/asterisk/pbx.h:
- Create a universal exception handling extension, "e" (closes
- issue #9785)
-
-2007-10-03 18:23 +0000 [r84512-84545] Steve Murphy <murf@digium.com>
-
- * /: blocked 84544 from trunk; it only applies to 1.4; 10870 -- the
- CUT in AEL
-
- * res/ael/pval.c, pbx/ael/ael-test/ref.ael-vtest17, /,
- pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
- pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-test5,
- pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18,
- pbx/ael/ael-test/ref.ael-test19,
- pbx/ael/ael-test/ref.ael-vtest13: Merged revisions 84511 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r84511 | murf | 2007-10-03 08:23:00 -0600 (Wed, 03 Oct 2007) | 1
- line closes issue #10834 ; where a null input to a switch
- statement results in a hangup; since switch is implemented with
- extensions, and the default case is implemented with a '.', and
- the '.' matches 1 or more remaining characters, the case where 0
- characters exist isn't matched, and the extension isn't matched,
- and the goto fails, and a hangup occurs. Now, when a default case
- is generated, it also generates a single fixed extension that
- will match a null input. That extension just does a goto to the
- default extension for that switch. I played with an alternate
- solution, where I just tack an extra char onto all the patterns
- and the goto, but not the default case's pattern. Then even a
- null input will still have at least one char in it. But it made
- me nervous, having that extra char in , even if that's a pretty
- secret and low-level issue. ........
-
-2007-10-02 20:07 +0000 [r84475] Russell Bryant <russell@digium.com>
-
- * Makefile, /, build_tools/prep_tarball: Merged revisions 84474 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r84474 | russell | 2007-10-02 15:06:07 -0500 (Tue, 02 Oct 2007) |
- 5 lines * Don't build the menuselect-tree for the tarball, as it
- requires running the configure script first * Change the Makefile
- to note that menuselect-tree depends on the configure script.
- ........
-
-2007-10-02 19:02 +0000 [r84432-84440] Jason Parker <jparker@digium.com>
-
- * /, res/res_features.c: Merged revisions 84410 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #10821) ........ r84410 | qwell | 2007-10-02 13:52:55 -0500
- (Tue, 02 Oct 2007) | 4 lines Finish up on transferee channel
- before return on failure. Issue 10821, patch by Ivan ........
-
-2007-10-02 18:12 +0000 [r84405] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c: Add MSet for people who prefer the old, deprecated
- syntax of Set (Closes issue #10549)
-
-2007-10-02 14:13 +0000 [r84371] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 84370 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r84370 | russell | 2007-10-02 09:12:35 -0500 (Tue, 02 Oct 2007) |
- 6 lines Use snprintf instead of sprintf in one place. There is no
- vulnerability here due to various buffer sizes around the code,
- but I still didn't like seeing a non length-limited copy of data
- coming off of the wire into a stack buffer, as this would be a
- problem in the future if buffer sizes elsewhere got changed or
- size limitations removed ... ........
-
-2007-10-02 13:58 +0000 [r84368] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Don't swap channel priority if using epoll as polling
- should/will only happen off the first channel. (closes issue
- #10867) Reported by: phsultan
-
-2007-10-01 23:33 +0000 [r84327-84331] Steve Murphy <murf@digium.com>
-
- * utils/check_expr.c: OK. THis a DEBUG_THREADS situation.
-
- * utils/check_expr.c: picky gcc versions... sigh.
-
- * utils/check_expr.c: This mod will allow check_expr to compile in
- the presence of DEBUG_THREAD situations. At least, it does for
- me. And it's less expensive than several other approaches I
- tried.
-
- * res/ael/pval.c, /, res/ael/ael.tab.c, res/ael/ael.y,
- pbx/pbx_ael.c: Merged revisions 84239 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r84239 | murf | 2007-10-01 14:27:52 -0600 (Mon, 01 Oct 2007) | 1
- line closes issue #10777 -- by returning a null for the parse
- tree when there's really nothing there, and making sure we don't
- try to do checking on a null tree. ........
-
-2007-10-01 21:54 +0000 [r84300] Jason Parker <jparker@digium.com>
-
- * Makefile, /, Makefile.rules, channels/Makefile: Merged revisions
- 84291 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r84291 | qwell | 2007-10-01 16:52:45 -0500 (Mon, 01 Oct 2007) | 6
- lines Add dist-clean support for subdirs. Change h323 to only
- remove the Makefile on a dist-clean, rather than a clean. This
- fixes a bug I found with trying to run make after a make clean
- ........
-
-2007-10-01 21:31 +0000 [r84275] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * main/channel.c, main/manager.c, /, channels/chan_agent.c: Merged
- revisions 84274 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r84274 | dhubbard | 2007-10-01 16:25:37 -0500 (Mon, 01 Oct 2007)
- | 1 line moved get_base_channel() code from action_redirect to
- ast_channel_masquerade() for issue 7706 and BE-160 ........
-
-2007-10-01 21:15 +0000 [r84207-84272] Russell Bryant <russell@digium.com>
-
- * /, main/utils.c, include/asterisk/lock.h: Merged revisions 84271
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r84271 | russell | 2007-10-01 16:07:06 -0500 (Mon, 01 Oct 2007) |
- 4 lines Fulfull a feature request from Qwell on the "core show
- locks" output. It will now note the lock type for each lock that
- a thread holds. (mutex, rdlock, or wrlock) ........
-
- * /, res/res_agi.c: Merged revisions 84236 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r84236 | russell | 2007-10-01 14:56:28 -0500 (Mon, 01 Oct 2007) |
- 5 lines Add another sanity check in the AGI read loop. We really
- don't care about EAGAIN unless we didn't read an entire line. If
- there is a newline at the end if the read buffer, break, because
- we got the whole thing. (reported and patched by bmd) ........
-
- * /, include/asterisk/lock.h: Merged revisions 84206 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r84206 | russell | 2007-10-01 14:34:12 -0500 (Mon, 01
- Oct 2007) | 2 lines Show rwlocks in the "core show locks" output.
- Before, it only showed mutexes. ........
-
-2007-10-01 15:57 +0000 [r84176] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Check to make sure a structure pointer is
- non-NULL before touching it... crashing is bad, mmmk? (closes
- issue #10831) Reported by: eliel Patches: chan_sip.c.patch
- uploaded by eliel (license 64)
-
-2007-10-01 15:34 +0000 [r84167-84174] Russell Bryant <russell@digium.com>
-
- * main/say.c: Change simple uses of snprintf to ast_copy_string.
- This was provided by mvanbaak as a part of issue #10843, but this
- part didn't apply because of a patch I applied right beforehand.
-
- * channels/chan_misdn.c, main/frame.c, res/res_config_odbc.c,
- apps/app_meetme.c, channels/chan_sip.c, channels/chan_skinny.c,
- main/say.c, apps/app_minivm.c, pbx/dundi-parser.c,
- channels/chan_iax2.c, channels/iax2-parser.c, main/asterisk.c,
- main/rtp.c, channels/chan_mgcp.c: 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)
-
- * main/say.c: Simplify code by using the -= and %= operators.
- (closes issue #10848) Reported by: opticron Patches: saymod.diff
- uploaded by opticron (license 267)
-
- * codecs/g722/Makefile, /, res/Makefile, channels/Makefile: The
- trunk version of this patch also includes a couple more small
- clean fixes from IgorG. Merged revisions 84170 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r84170 | russell | 2007-10-01 10:00:56 -0500 (Mon, 01 Oct 2007) |
- 3 lines Remove another file in "make clean". (closes issue
- #10814, paravoid) ........
-
- * main/cli.c: Don't set the full command string until after
- verifying that there is not another CLI command with the same
- command text registered. This prevents a crash if someone
- accidentally calls ast_cli_register() on the same CLI command
- data twice. This also fixes a small bug where the helpers list
- would get unlocked without being locked if building the full
- command failed. (closes issue #10858, reported by jamesgolovich,
- patched by me)
-
- * configs/musiconhold.conf.sample, res/res_musiconhold.c: Add a new
- option for files-based music on hold to ensure that the sort
- order of the files is alphabetical. (closes issue #10855)
- Reported by: jamesgolovich Patches:
- asterisk-mohsortalpha.diff.txt uploaded by jamesgolovich (license
- 176)
-
- * apps/app_dial.c, /: Merged revisions 84166 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r84166 | russell | 2007-10-01 09:24:49 -0500 (Mon, 01 Oct 2007) |
- 2 lines Simplify the CAN_EARLY_BRIDGE macro a bit. ........
-
-2007-10-01 14:21 +0000 [r84159-84165] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Add MP4 to part of the SDP code. (closes
- issue #10820) Reported by: ruikubo Patches: chan_sip.patch
- uploaded by ruikubo (license 250)
-
- * main/dnsmgr.c: Don't register the dnsmgr refresh CLI command
- twice. (closes issue #10856) Reported by: jamesgolovich Patches:
- asterisk-dnsmgrclireg.diff.txt uploaded by jamesgolovich (license
- 176)
-
- * /, res/res_musiconhold.c: Merged revisions 84160 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r84160 | file | 2007-10-01 10:57:42 -0300 (Mon, 01 Oct
- 2007) | 6 lines Fix randomness. save_pos was being set to 0
- initially instead of -1, causing it to jump to position 0 when
- moh started. (closes issue #10859) Reported by: jamesgolovich
- Patches: asterisk-mohpos2.diff.txt uploaded by jamesgolovich
- (license 176) ........
-
- * apps/app_dial.c, /: Merged revisions 84158 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r84158 | file | 2007-10-01 10:49:36 -0300 (Mon, 01 Oct 2007) | 4
- lines Only attempt early bridging if the options given to Dial()
- permit it. (closes issue #10861) Reported by: peekyb ........
-
-2007-09-30 20:06 +0000 [r84143-84147] Russell Bryant <russell@digium.com>
-
- * /, include/asterisk/module.h: Merged revisions 84146 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r84146 | russell | 2007-09-30 16:02:16 -0400 (Sun, 30
- Sep 2007) | 4 lines Fix the AST_MODULE_INFO macro for C++
- modules. The load and reload parameters were in the wrong place.
- (closes issue #10846, alebm) ........
-
- * funcs/func_lock.c: * The documentation for the LOCK() function
- says that it will block for up to 3 seconds while waiting on a
- lock when other locks are currently held to avoid deadlocks.
- Change the code to reflect this. * Since trying to grab a lock
- may block for some time, put the channel in autoservice so that
- audio is still read from the channel and that any active
- generators on the channel don't pause.
-
-2007-09-29 23:47 +0000 [r84134-84137] Steve Murphy <murf@digium.com>
-
- * /, res/ael/ael_lex.c, res/ael/ael.flex: Merged revisions 84133
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r84133 | murf | 2007-09-29 15:47:53 -0600 (Sat, 29 Sep 2007) | 1
- line This issue sort of closes 10786; All config files support
- #include with globbing (you know, *,[chars],?,{list,list},etc),
- so I've updated the AEL system to support this also. ........
-
- * pbx/ael/ael-test/ael-ntest22/t2 (added),
- pbx/ael/ael-test/ael-ntest22/t3 (added),
- pbx/ael/ael-test/ael-ntest22/extensions.ael (added),
- pbx/ael/ael-test/ael-ntest22 (added),
- pbx/ael/ael-test/ael-ntest22/t1/a.ael (added),
- pbx/ael/ael-test/ael-ntest22/t1/b.ael (added),
- pbx/ael/ael-test/ael-ntest22/t1/c.ael (added),
- pbx/ael/ael-test/ael-ntest22/t2/d.ael (added),
- pbx/ael/ael-test/ael-ntest22/t2/e.ael (added),
- pbx/ael/ael-test/ael-ntest22/t2/f.ael (added),
- pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-ntest22
- (added), pbx/ael/ael-test/ael-ntest22/t3/g.ael (added),
- pbx/ael/ael-test/ref.ael-test3,
- pbx/ael/ael-test/ael-ntest22/t3/h.ael (added),
- pbx/ael/ael-test/ref.ael-test4,
- pbx/ael/ael-test/ael-ntest22/t3/i.ael (added),
- pbx/ael/ael-test/ael-ntest22/t3/j.ael (added),
- pbx/ael/ael-test/ael-ntest22/qq.ael (added),
- pbx/ael/ael-test/ael-ntest22/t1 (added): the last commit for AEL
- affected a small number of tests. Added a regression test for
- glob'd includes
-
-2007-09-29 18:21 +0000 [r84130] Tilghman Lesher <tlesher@digium.com>
-
- * cdr/cdr_manager.c: Set enablecdr at the end of re-reading the
- config file (Closes issue #10852)
-
-2007-09-29 00:19 +0000 [r84115] Matthew Fredrickson <creslin@digium.com>
-
- * main/translate.c: Let's use process time instead of wall clock
- time for show translation
-
-2007-09-28 14:35 +0000 [r84050-84080] Tilghman Lesher <tlesher@digium.com>
-
- * configure, configure.ac: Autoconf requires version 2.60, not
- 2.59, to process (Closes issue #10842)
-
- * /, main/say.c: Merged revisions 84078 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r84078 | tilghman | 2007-09-28 09:13:47 -0500 (Fri, 28 Sep 2007)
- | 2 lines Correct pronunciations of numbers for .nl (Closes issue
- #10837) ........
-
- * main/channel.c, /: Merged revisions 84049 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r84049 | tilghman | 2007-09-28 00:30:22 -0500 (Fri, 28 Sep 2007)
- | 3 lines Avoid a deadlock with ALL of the locks in the
- masquerade function, not just the pairs of channels. (Closes
- issue #10406) ........
-
-2007-09-27 23:18 +0000 [r84019] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * main/manager.c, /, channels/chan_agent.c,
- include/asterisk/channel.h: Merged revisions 84018 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r84018 | dhubbard | 2007-09-27 18:12:25 -0500 (Thu, 27
- Sep 2007) | 1 line if an Agent is redirected, the base channel
- should actually be redirected. This was causing multiple issues,
- especially issue 7706 and BE-160 ........
-
-2007-09-27 00:08 +0000 [r83978-83986] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, channels/chan_alsa.c: Merged revisions 83974 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r83974 | kpfleming | 2007-09-26 16:53:03 -0700 (Wed, 26 Sep 2007)
- | 2 lines avoid the weird usage of assert() in the ALSA header
- files that gcc 4.2 wants to complain about ........
-
- * res/ael/ael.tab.c, res/ael/ael.y: deal with more gcc 4.2 const
- pointer warnings
-
-2007-09-27 00:02 +0000 [r83911-83977] Russell Bryant <russell@digium.com>
-
- * /, pbx/pbx_dundi.c: Merged revisions 83976 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r83976 | russell | 2007-09-26 19:01:29 -0500 (Wed, 26 Sep 2007) |
- 1 line remove a todo item that has been completed ........
-
- * /, channels/chan_sip.c: Merged revisions 83943 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r83943 | russell | 2007-09-26 16:35:23 -0500 (Wed, 26 Sep 2007) |
- 2 lines I changed my mind ... I think this should be a
- LOG_NOTICE. ........
-
- * /, channels/chan_sip.c: Merged revisions 83941 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r83941 | russell | 2007-09-26 16:15:15 -0500 (Wed, 26 Sep 2007) |
- 5 lines Add a log message that was requested by the masses in the
- developer tutorial session at Astricon. chan_sip did not output
- any message when a call was rejected because the extension was
- not found. This adds a verbose message (at verbose level 3) to
- note when this happens. ........
-
- * /: Merged revisions 83910 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r83910 | russell | 2007-09-26 15:50:09 -0500 (Wed, 26 Sep 2007) |
- 3 lines Fix building chan_misdn under dev-mode. (please run the
- configure script with --enable-dev-mode so this doesn't happen
- again ...) ........
-
-2007-09-26 18:43 +0000 [r83880] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 83879 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r83879 | tilghman | 2007-09-26 13:35:56 -0500 (Wed, 26 Sep 2007)
- | 2 lines Remove unused 4k of memory on the program stack (closes
- issue #10827) ........
-
-2007-09-26 06:53 +0000 [r83849-83864] Russell Bryant <russell@digium.com>
-
- * include/asterisk/event.h: fix a typo in a comment
-
- * include/asterisk/file.h: Change function documentation to use
- doxygen tags. (Really, I just needed to make some minor change in
- trunk to test something with automerge ...)
-
-2007-09-25 23:14 +0000 [r83834] Matthew Fredrickson <creslin@digium.com>
-
- * doc/ss7.txt: Fix typo in readme
-
-2007-09-25 21:06 +0000 [r83819] Russell Bryant <russell@digium.com>
-
- * include/asterisk/devicestate.h: Don't note that functions are
- deprecated in favor of themselves. This was found by showing a
- very poor example doxygen function in a presentation this
- morning. :)
-
-2007-09-25 16:34 +0000 [r83804] Philippe Sultan <philippe.sultan@gmail.com>
-
- * res/res_jabber.c: Added a CLI command that shows our buddy list,
- as suggested by Daniel McKeehan, thanks!
-
-2007-09-25 14:18 +0000 [r83774] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/app.c: Merged revisions 83773 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r83773 | tilghman | 2007-09-25 09:13:25 -0500 (Tue, 25 Sep 2007)
- | 2 lines jmls pointed out that unsetting the group and setting
- the group to the blank string aren't quite the same. ........
-
-2007-09-25 13:41 +0000 [r83758] Joshua Colp <jcolp@digium.com>
-
- * res/ael/pval.c: Fix minor memory leak in pval.c. Overwriting a
- value without freeing the previous result is bad, mmmk?
-
-2007-09-25 09:07 +0000 [r83743] Philippe Sultan <philippe.sultan@gmail.com>
-
- * channels/chan_jingle.c, include/asterisk/jingle.h: Comply with
- latest XEP-0166, XEP-0167, XEP-0176. No real Jingle
- implementation being available, testing was made using two
- Asterisk servers relaying SIP calls over their Jingle channels:
- SIP Phone 1 --- [chan_sip]Asterisk 1[chan_jingle] ---
- [chan_jingle]Asterisk 2[chan_sip] --- SIP Phone 2 Thus, it was
- possible to test the code in both ways, and make the Jingle
- channel comply with the latest specifications. No sound available
- yet. Main modifications include : - modified the
- 'jingle_candidate' structure and the 'jingle_create_candidates'
- function according to XEP-0176 ; - modified the 'jingle_action'
- function in order to properly terminate a Jingle session, in
- conformance with XEP-0166 ; - modified username format used in
- STUN requests ; - actually make the bindaddr configuration field
- useable. Todo : - set audio paths up (no native bridging) ; -
- make the CLI gtalk functions available to jingle ; - clean up the
- storage space used in strings.
-
-2007-09-25 08:09 +0000 [r83741] Russell Bryant <russell@digium.com>
-
- * utils/Makefile, utils: Add some files to the utils directory
- svn:ignore and Makefile clean target (closes issue #10808,
- reported by mvanbaak)
-
-2007-09-24 22:06 +0000 [r83696-83726] Tilghman Lesher <tlesher@digium.com>
-
- * Makefile, main/asterisk.c: Permit custom locations for astdb and
- the keys directory (though default to the current locations)
- (Closes issue #10267)
-
- * /, build_tools/make_defaults_h: Merged revisions 83695 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r83695 | tilghman | 2007-09-24 12:22:08 -0500 (Mon, 24 Sep 2007)
- | 4 lines In the source, keys are relative to the datadir, not
- varlib (which is the same in most cases, but it's good to be
- accurate). Closes issue #10811 ........
-
-2007-09-24 17:10 +0000 [r83671] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * channels/chan_sip.c, configs/sip.conf.sample: merged jcmoore's
- patch for configurable SDP origin-field username and session
- field, closes issue# 10795
-
-2007-09-24 17:00 +0000 [r83656] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: interface_exists_global was never returning 1.
- Most likely an error from my merge on Friday. (closes issue
- #10817, reported and patched by snar, patch simplified by me)
-
-2007-09-24 16:42 +0000 [r83654-83655] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/app.c: Merged revisions 83637 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r83637 | tilghman | 2007-09-24 10:17:06 -0500 (Mon, 24 Sep 2007)
- | 3 lines Making change to group splitting, as discussed on the
- -dev list. The main effect of this will be to permit
- Set(GROUP([cat])=), i.e. unsetting a group. ........
-
-2007-09-22 19:54 +0000 [r83575-83590] Steve Murphy <murf@digium.com>
-
- * res/ael/pval.c, /: Merged revisions 83589 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r83589 | murf | 2007-09-22 13:39:16 -0600 (Sat, 22 Sep 2007) | 1
- line This closes issue #10788 -- The exact same fixes are made
- here for the first arg in the for(arg1; arg2; arg3) {} statement,
- as were done for the 3rd arg. It can now be an assignment that
- will embedded in a Set() app, or a macro call, or an app call.
- ........
-
- * res/ael/pval.c, /, pbx/pbx_ael.c: Merged revisions 83558 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r83558 | murf | 2007-09-22 10:41:43 -0600 (Sat, 22 Sep 2007) | 1
- line This closes issue #10788 -- the 3rd arg in the for statement
- is now wrapped in Set() only if there's an '=' in that string.
- Otherwise, if it begins with '&', then a Macro call is generated;
- otherwise it is made into an app call. A bit more accomodating,
- keeps the new guys happy, and the guys with ael-1 code should be
- happy, too ........
-
-2007-09-22 17:37 +0000 [r83574] Matthew Fredrickson <creslin@digium.com>
-
- * doc/ss7.txt: Fix potential point of confusion
-
-2007-09-22 14:45 +0000 [r83517-83545] Tilghman Lesher <tlesher@digium.com>
-
- * utils/Makefile, utils/hashtest2.c, utils/clicompat.c (added): Fix
- build of check_expr and hashtest2 when DEBUG_THREADLOCAL is
- defined
-
- * main/manager.c, apps/app_meetme.c: Add the MeetmeList and Reload
- manager commands, which supplement the need to have Command
- privilege. (closes issue #10736)
-
- * configure, include/asterisk/autoconfig.h.in, main/ast_expr2.h,
- main/ast_expr2.y, configure.ac, main/ast_expr2.c: Fixes for
- FreeBSD... testing for every conceivable math function now
-
-2007-09-21 19:55 +0000 [r83500] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c: Fix compilation errors in CLI command
- updates to SS7 CLI commands
-
-2007-09-21 19:54 +0000 [r83499] Matthew Fredrickson <creslin@digium.com>
-
- * doc/ss7.txt (added): Add an SS7 readme for setup and use of
- libss7 and asterisk
-
-2007-09-21 18:41 +0000 [r83484] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_queue.c: Fix some areas where we were still using '|'
- for an argument delimiter (closes issue #10793)
-
-2007-09-21 18:27 +0000 [r83483] Russell Bryant <russell@digium.com>
-
- * apps/app_queue.c: Update app_queue to use commas as application
- argument separators. (closes issue #10793, snar)
-
-2007-09-21 17:36 +0000 [r83466] Tilghman Lesher <tlesher@digium.com>
-
- * cdr/cdr_manager.c: Fix cdr_manager, such that if the config file
- is created past load, it'll start logging (and conversely, if the
- config file is destroyed or deactivated, the logging is
- disabled). Reported by Juggie via IRC, fix by me.
-
-2007-09-21 14:40 +0000 [r83433] Russell Bryant <russell@digium.com>
-
- * res/res_config_pgsql.c, main/dnsmgr.c, /, channels/chan_sip.c,
- main/db1-ast/hash/hash.c, include/asterisk/channel.h,
- channels/chan_iax2.c, main/rtp.c, channels/misdn_config.c,
- main/cdr.c, main/channel.c, channels/chan_misdn.c,
- main/ast_expr2f.c, main/file.c, include/asterisk/sched.h,
- channels/chan_h323.c, utils/ael_main.c, pbx/pbx_dundi.c,
- main/sched.c, channels/chan_mgcp.c, main/ast_expr2.fl: Merged
- revisions 83432 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r83432 | russell | 2007-09-21 09:37:20 -0500 (Fri, 21 Sep 2007) |
- 4 lines gcc 4.2 has a new set of warnings dealing with cosnt
- pointers. This set of changes gets all of Asterisk (minus
- chan_alsa for now) to compile with gcc 4.2. (closes issue #10774,
- patch from qwell) ........
-
-2007-09-21 14:25 +0000 [r83431] Tilghman Lesher <tlesher@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in, main/ast_expr2.h,
- main/ast_expr2.y, configure.ac, main/ast_expr2.c: Check for the
- presence of trunc and round, and make the ISOC99 detection a
- little more sane (closes issue #10776)
-
-2007-09-20 23:14 +0000 [r83381] Jason Parker <jparker@digium.com>
-
- * apps/app_minivm.c, main/astmm.c, apps/app_playback.c: More
- NEW_CLI conversions. (issue #10724) Patches: app_playback.c.patch
- uploaded by moy (license 222) app_minivm.c.patch uploaded by
- eliel (license 64) astmm.c.patch uploaded by eliel (license 64)
-
-2007-09-20 21:37 +0000 [r83350-83351] Mark Michelson <mmichelson@digium.com>
-
- * /: Oops. Getting rid of svnmerge-integrated and automerge stuff
-
- * /, apps/app_queue.c: Merging changes from queue_refcount_trunk
- into trunk. Refcounted queues now in place.
-
-2007-09-20 21:17 +0000 [r83293-83349] Russell Bryant <russell@digium.com>
-
- * /, main/asterisk.c: Merged revisions 83348 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r83348 | russell | 2007-09-20 16:16:48 -0500 (Thu, 20 Sep 2007) |
- 4 lines When daemonizing, don't change working directory to "/".
- It makes it not be able to do a core dump when not running as
- uid=root. (closes issue #10766, xrg) ........
-
- * /, contrib/scripts/safe_asterisk: Merged revisions 83316 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r83316 | russell | 2007-09-20 16:01:20 -0500 (Thu, 20 Sep 2007) |
- 3 lines Change safe_asterisk to explicitly ask for /bin/bash, as
- it uses bashisms. (closes issue #10772, reported by culrich)
- ........
-
- * main/dsp.c: trivial formatting change
-
- * main/asterisk.c: trivial formatting change
-
- * main/app.c: minor spelling fixes in a comment
-
- * main/app.c: minor grammar fix
-
- * channels/chan_sip.c: fix spelling in a comment
-
- * main/asterisk.c: trivial formatting change
-
-2007-09-20 19:05 +0000 [r83251-83278] Jason Parker <jparker@digium.com>
-
- * doc/modules.txt: Fix a trivial typo, to test our new commit bot
-
- * /, apps/app_disa.c: Merged revisions 83246 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r83246 | qwell | 2007-09-20 12:09:14 -0500 (Thu, 20 Sep 2007) | 8
- lines If # is pressed after dialing an extension in DISA, stop
- trying to collect more digits. (closes issue #10754) Reported by:
- atis Patches: app_disa.c.branch.patch uploaded by atis (license
- 242) app_disa.c.trunk.patch uploaded by atis (license 242)
- ........
-
-2007-09-20 16:28 +0000 [r83234] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 83232 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r83232 | file | 2007-09-20 13:25:30 -0300 (Thu, 20 Sep 2007) | 7
- lines Make sure the minimum T1 timer value is obeyed in all
- cases. (closes issue #10768) Reported by: flefoll Patches:
- chan_sip.c.trunk.83071.retrans-patch uploaded by flefoll (license
- 244) chan_sip.c.br14.83070.retrans-patch uploaded by flefoll
- (license 244) ........
-
-2007-09-20 16:27 +0000 [r83233] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c: Don't start the event processing thread until
- after forking. (reported by Simon on the -dev list, thanks!)
-
-2007-09-20 16:19 +0000 [r83229-83231] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 83230 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r83230 | file | 2007-09-20 13:17:24 -0300 (Thu, 20 Sep 2007) | 7
- lines Fix a minor spelling error. (closes issue #10769) Reported
- by: flefoll Patches: chan_sip.c.trunk.83071.inita-patch uploaded
- by flefoll (license 244) chan_sip.c.br14.83070.inita-patch
- uploaded by flefoll (license 244) ........
-
- * pbx/pbx_dundi.c, cdr/cdr_pgsql.c, main/config.c: Fix memory leaks
- in pbx_dundi, cdr_pgsql, and the configuration file parser.
-
-2007-09-19 23:16 +0000 [r83213] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c, apps/app_meetme.c, apps/app_queue.c,
- apps/app_voicemail.c: More conversions to NEW_CLI (issue #10724)
- Patches: chan_zap.c.patch uploaded by moy (license 222)
- app_queue.c.patch uploaded by eliel (license 64)
- app_voicemail.c.patch uploaded by eliel (license 64)
- app_meetme.c.patch uploaded by eliel (license 64)
-
-2007-09-19 20:06 +0000 [r83182-83183] Joshua Colp <jcolp@digium.com>
-
- * cdr/cdr_csv.c: Clean up code in cdr_csv. (Are you sensing a theme
- for me today?)
-
- * res/res_adsi.c: Clean up code in res_adsi.
-
-2007-09-19 19:54 +0000 [r83176-83181] Russell Bryant <russell@digium.com>
-
- * funcs/func_shell.c: put the channel in autoservice when executing
- func_shell
-
- * /, apps/app_system.c: Merged revisions 83179 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r83179 | russell | 2007-09-19 14:50:48 -0500 (Wed, 19 Sep 2007) |
- 5 lines The System() and TrySystem() applications can take a
- substantial amount of time to execute while not servicing the
- channel. So, put the channel in autoservice while the command is
- being executed. (closes issue #10726, reported by mnicholson)
- ........
-
- * funcs/func_curl.c, /: Merged revisions 83177 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r83177 | russell | 2007-09-19 14:34:25 -0500 (Wed, 19 Sep 2007) |
- 4 lines Using curl can take a substantial amount of time, so the
- channel should be autoserviced while waiting for it to complete.
- (closes issue #10725, reported by mnicholson) ........
-
- * /, channels/chan_iax2.c: Merged revisions 83175 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r83175 | russell | 2007-09-19 14:13:29 -0500 (Wed, 19 Sep 2007) |
- 8 lines When handling a reload of chan_iax2, don't use an
- ao2_callback() to POKE all peers. Instead, use an iterator. By
- using an iterator, the peers container is not locked while the
- POKE is being done. It can cause a deadlock if the peers
- container is locked because poking a peer will try to lock pvt
- structs, while there is a lot of other code that will hold a pvt
- lock when trying to go lock the peers container. (reported to me
- directly by Loic Didelot. Thank you for the debug info!) ........
-
-2007-09-19 17:22 +0000 [r83155-83157] Joshua Colp <jcolp@digium.com>
-
- * apps/app_db.c: Fix indentation in app_db.
-
- * apps/app_authenticate.c: Clean up code in app_authenticate.
-
- * apps/app_adsiprog.c: Clean up code in app_adsiprog.
-
-2007-09-19 15:11 +0000 [r83126] Russell Bryant <russell@digium.com>
-
- * main/manager.c, /: Merged revisions 83121 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r83121 | russell | 2007-09-19 10:10:14 -0500 (Wed, 19 Sep 2007) |
- 4 lines Fix up another potential race condition. Do the loop
- decrementing use count on events with the eventq protected from
- being changed. (reported on IRC by Ivan) ........
-
-2007-09-19 15:08 +0000 [r83105-83114] Joshua Colp <jcolp@digium.com>
-
- * apps/app_disa.c: DISA only needs to know about the end of DTMF,
- not the beginning/duration.
-
- * apps/app_disa.c: Clean up app_disa code a bit.
-
-2007-09-19 13:55 +0000 [r83076] Philippe Sultan <philippe.sultan@gmail.com>
-
- * channels/chan_jingle.c: Replace Google namespace occurrences with
- Jingle. The former namespace is handled by chan_gtalk.
-
-2007-09-19 13:49 +0000 [r83073-83075] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 83074 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r83074 | file | 2007-09-19 10:47:59 -0300 (Wed, 19 Sep 2007) | 6
- lines Protect the CDR record from modification by pbx_exec so
- that the application data contains the Queue data. (closes issue
- #10761) Reported by: snar Patches: app-queue-mixmonitor.patch
- uploaded by snar (license 245) ........
-
- * main/manager.c: Extend manager show connected with additional
- information. (closes issue #10757) Reported by: outtolunc
- Patches: manager.c.sessionstart.diff uploaded by outtolunc
- (license 237)
-
-2007-09-19 13:29 +0000 [r83072] Philippe Sultan <philippe.sultan@gmail.com>
-
- * channels/chan_jingle.c: Remove namespaces in payload-type tags.
-
-2007-09-19 13:21 +0000 [r83071] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 83070 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r83070 | file | 2007-09-19 10:18:22 -0300 (Wed, 19 Sep 2007) | 6
- lines (closes issue #10760) Reported by: dimas Patches:
- chan_sip.patch uploaded by dimas (license 88) Read in
- subscribecontext option in general to be the default. ........
-
-2007-09-19 12:23 +0000 [r83055] Philippe Sultan <philippe.sultan@gmail.com>
-
- * channels/chan_jingle.c, include/asterisk/jingle.h: Transmit
- proper invitation, thus conforming to XEP-0166 (Jingle general
- specifications), XEP-0167 (Jingle Audio via RTP) and XEP-0176
- (Jingle ICE Transport).
-
-2007-09-19 09:48 +0000 [r83025] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /, channels/misdn/chan_misdn_config.h,
- channels/misdn_config.c: Merged revisions 83023-83024 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r83023 | crichter | 2007-09-19 11:31:55 +0200 (Mi, 19 Sep 2007) |
- 1 line added 'astdtmf' option to allow configuring the asterisk
- dtmf detector instead of the mISDN_dsp ones. also added the patch
- from irroot #10190, so that dtmf tones detected by the asterisk
- detector are passed outofband to asterisk, to make any use of
- dtmf tones at all. ........ r83024 | crichter | 2007-09-19
- 11:32:42 +0200 (Mi, 19 Sep 2007) | 1 line removed comment which
- violates the coding guidelines. ........
-
-2007-09-19 00:21 +0000 [r82993] Russell Bryant <russell@digium.com>
-
- * /, apps/app_flash.c: Merged revisions 82992 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82992 | russell | 2007-09-18 19:19:49 -0500 (Tue, 18 Sep 2007) |
- 4 lines Change the description of app_flash to note how it can be
- a useful tool instead of just saying that it is generally a
- worthless feature. (Thanks to Jim Van Meggelen for pointing it
- out and providing the proposed text) ........
-
-2007-09-18 23:42 +0000 [r82962] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 82961 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82961 | file | 2007-09-18 20:41:02 -0300 (Tue, 18 Sep 2007) | 2
- lines Initialize a variable to NULL to make the world happy.
- ........
-
-2007-09-18 22:46 +0000 [r82931] Russell Bryant <russell@digium.com>
-
- * include/asterisk/agi.h, /, res/res_agi.c: Merged revisions 82929
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82929 | russell | 2007-09-18 17:42:27 -0500 (Tue, 18 Sep 2007) |
- 11 lines Add a new patch to handle interrupting the fgets() call
- when using FastAGI. This version of the patch maintains the
- original behavior of the code when not using FastAGI. (closes
- issue #10553) Reported by: juggie Patches: res_agi_fgets-4.patch
- uploaded by juggie (license 24) res_agi_fgets_1.4svn.patch
- uploaded by juggie (license 24) Slight mods by me Tested by:
- juggie, festr ........
-
-2007-09-18 22:43 +0000 [r82871-82930] Jason Parker <jparker@digium.com>
-
- * main/pbx.c, main/frame.c, main/dnsmgr.c, channels/chan_local.c,
- channels/chan_sip.c, res/res_features.c, channels/chan_agent.c,
- res/res_musiconhold.c, res/res_jabber.c, main/manager.c,
- res/res_agi.c, channels/chan_features.c, main/logger.c,
- main/http.c, channels/chan_alsa.c, res/res_realtime.c,
- res/res_odbc.c: (issue #10724) Reported by: eliel Patches:
- res_features.c.patch uploaded by eliel (license 64)
- res_agi.c.patch uploaded by seanbright (license 71)
- res_musiconhold.c.patch uploaded by seanbright (license 71)
- pbx.c.patch uploaded by moy (license 222) logger.c.patch uploaded
- by moy (license 222) frame.c.patch uploaded by moy (license 222)
- manager.c.patch uploaded by moy (license 222) http.c.patch
- uploaded by moy (license 222) dnsmgr.c.patch uploaded by moy
- (license 222) res_realtime.c.patch uploaded by eliel (license 64)
- res_odbc.c.patch uploaded by seanbright (license 71)
- res_jabber.c.patch uploaded by eliel (license 64)
- chan_local.c.patch uploaded by eliel (license 64)
- chan_agent.c.patch uploaded by eliel (license 64)
- chan_alsa.c.patch uploaded by eliel (license 64)
- chan_features.c.patch uploaded by eliel (license 64)
- chan_sip.c.patch uploaded by eliel (license 64) RollUp.1.patch
- (includes all of the above patches) uploaded by seanbright
- (license 71) Convert many CLI commands to the NEW_CLI format.
-
- * configs/voicemail.conf.sample, apps/app_voicemail.c: (closes
- issue #10739) Reported by: ruffle Patches: app_voicemail.c.diff
- uploaded by ruffle (license 201) 10739-moveheard.diff uploaded by
- qwell (license 4) Tested by: callguy, ruffle Add an option to
- disable the automatic moving of "heard" messages to the Old
- folder.
-
-2007-09-18 20:59 +0000 [r82868] Russell Bryant <russell@digium.com>
-
- * main/manager.c, /: Merged revisions 82867 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82867 | russell | 2007-09-18 15:56:43 -0500 (Tue, 18 Sep 2007) |
- 10 lines Fix a memory leak that can occur on systems under higher
- load. The issue is that when events are appended to the master
- event queue, they use the number of active sessions as a use
- count so it will know when all active sessions at the time the
- event happened have consumed it. However, the handling of the
- number of sessions was not properly synchronized, so the use
- count was not always correct, causing an event to disappear
- early, or get stuck in the event queue for forever. (closes issue
- #9238, reported by bweschke, patch from Ivan, modified by me)
- ........
-
-2007-09-18 20:10 +0000 [r82866] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 82865 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82865 | mmichelson | 2007-09-18 15:09:02 -0500 (Tue, 18 Sep
- 2007) | 4 lines Moving the logic for handling an empty membername
- to the create_member function so that there is a common place
- where this occurs instead of being spread out to several
- different places. ........
-
-2007-09-18 19:06 +0000 [r82835] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 82834 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82834 | kpfleming | 2007-09-18 13:59:52 -0500 (Tue, 18 Sep 2007)
- | 2 lines there is no need for conditional logic to select
- ->interface or ->membername, snince ->membername will always be
- populated ........
-
-2007-09-18 16:34 +0000 [r82803] Russell Bryant <russell@digium.com>
-
- * /, pbx/pbx_dundi.c: Merged revisions 82802 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82802 | russell | 2007-09-18 11:31:01 -0500 (Tue, 18 Sep 2007) |
- 4 lines When copying the contents from the wildcard peer, do a
- deep copy instead of shallow copy so that it doesn't crash when
- beging destroyed. (closes issue #10546, patch by me) ........
-
-2007-09-18 16:16 +0000 [r82800] Jason Parker <jparker@digium.com>
-
- * configs/queues.conf.sample, apps/app_queue.c: (closes issue
- #10755) Reported by: snar Patches: app-queue-cdr-trunk.patch
- uploaded by snar (license 245) queues.conf.patch uploaded by snar
- (license 245) Add an updatecdr option to queues.conf, so that if
- a "member name" is specified, the cdr record will be updated with
- that, rather than the channel.
-
-2007-09-18 16:14 +0000 [r82776-82793] Russell Bryant <russell@digium.com>
-
- * include/asterisk/threadstorage.h: Make sure that libpthread
- doesn't try to call free() directly when MALLOC_DEBUG is enabled.
- If it does, Asterisk will crash as the address isn't the real
- beginning of the allocation.
-
- * channels/chan_zap.c: Don't use ast_channel_lock_both() here, it
- only exists in one of my branches. This is theoretically a
- potential deadlock, but it's the way it was before so I'm going
- to leave it this way for now.
-
-2007-09-18 15:29 +0000 [r82752] Jason Parker <jparker@digium.com>
-
- * /, configs/sip.conf.sample: Merged revisions 82751 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes
- issue #10753) ........ r82751 | qwell | 2007-09-18 10:28:21 -0500
- (Tue, 18 Sep 2007) | 4 lines Correct the allowexternaldomains
- option in SIP sample config. Issue 10753 ........
-
-2007-09-17 22:59 +0000 [r82728] Russell Bryant <russell@digium.com>
-
- * channels/chan_local.c, channels/chan_zap.c, apps/app_zapscan.c,
- channels/chan_agent.c, channels/chan_alsa.c,
- channels/chan_iax2.c, channels/chan_mgcp.c: convert various
- places that access the channel lock directly to use the channel
- lock wrappers
-
-2007-09-17 21:52 +0000 [r82710-82712] Jason Parker <jparker@digium.com>
-
- * cdr/cdr_sqlite3_custom.c: Don't try to continue loading
- cdr_sqlite3_custom on a module load failure (such as the config
- not existing) Closes issue #10749, patch by seanbright.
-
- * configs/http.conf.sample: Fix the sample redirect to point to a
- valid file in the Asterisk GUI. Closes issue #10748, patch by
- bkruse
-
-2007-09-17 20:24 +0000 [r82595-82679] Russell Bryant <russell@digium.com>
-
- * doc/res_config_sqlite.txt, res/res_config_sqlite.c: Add support
- for #include, var_metric, and cat_metric in res_config_sqlite
- (closes issue #10738, rbraun_proformatique)
-
- * /, main/stdtime/localtime.c, apps/app_voicemail.c: Merged
- revisions 82676 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82676 | russell | 2007-09-17 15:16:25 -0500 (Mon, 17 Sep 2007) |
- 4 lines Put a memset in ast_localtime() instead of a couple
- places in app_voicemail to prevent the problem everywhere instead
- of just a couple of places. (related to issue #10746) ........
-
- * /, apps/app_voicemail.c: Merged revisions 82644 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82644 | russell | 2007-09-17 15:00:32 -0500 (Mon, 17 Sep 2007) |
- 6 lines Initialize some memory to fix crashes when leaving
- voicemail. This problem was fixed by running Asterisk under
- valgrind. (closes issue #10746, reported by arcivanov, patched by
- me) *** IMPORTANT NOTE: We need to check to see if this same bug
- exists elsewhere. ........
-
- * apps/app_dial.c, res/ael/pval.c, include/asterisk/utils.h,
- apps/app_meetme.c, channels/chan_sip.c, channels/chan_skinny.c,
- res/res_features.c, apps/app_queue.c, channels/chan_iax2.c,
- pbx/pbx_config.c: 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
-
- * /, res/res_features.c: Merged revisions 82594 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82594 | russell | 2007-09-17 11:46:59 -0500 (Mon, 17 Sep 2007) |
- 5 lines Handle the case where there are multiple dynamic features
- with the same digit mapping, but won't always match the activated
- on/by access controls. In that case, the code needs to keep
- trying features for a match. (reported by Atis on the
- asterisk-dev list, patched by me) ........
-
-2007-09-17 16:44 +0000 [r82593] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 82590,82592 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r82590 | kpfleming | 2007-09-17 11:33:30 -0500 (Mon, 17
- Sep 2007) | 2 lines fix a couple of places where a logical member
- name (if specified) was not used, but instead the direct
- interface was listed ........ r82592 | kpfleming | 2007-09-17
- 11:40:12 -0500 (Mon, 17 Sep 2007) | 2 lines revert a change that
- wasn't supposed to be committed... doh! ........
-
-2007-09-17 14:58 +0000 [r82568] Doug Bailey <dbailey@digium.com>
-
- * main/http.c: Fix memory leak introduced when POST support was
- added.
-
-2007-09-17 02:20 +0000 [r82516-82546] Joshua Colp <jcolp@digium.com>
-
- * res/res_features.c: (closes issue #10715) Reported by:
- the-chopper Don't bother hanging up the new channel if it does
- not exist yet.
-
- * main/pbx.c, /: Merged revisions 82514 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82514 | file | 2007-09-16 23:00:59 -0300 (Sun, 16 Sep 2007) | 4
- lines (closes issue #10734) Reported by: asgaroth Instead of
- passing a NULL pointer into snprintf pass "". It makes Solaris
- much happier. ........
-
-2007-09-16 15:32 +0000 [r82496] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: Option maxmessage should be maxsecs
- per-folder, too (closes issue #10729)
-
-2007-09-14 21:30 +0000 [r82457] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, /: Merged revisions 82444 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82444 | murf | 2007-09-14 15:19:27 -0600 (Fri, 14 Sep 2007) | 1
- line closes issue #10668; thanks to arkadia for his patch; had to
- leave out the bit about ending the previous cdr in the fork; it
- would destroy current implementations. ........
-
-2007-09-14 21:21 +0000 [r82454] Russell Bryant <russell@digium.com>
-
- * /, configs/zapata.conf.sample: Merged revisions 82435 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82435 | russell | 2007-09-14 16:17:08 -0500 (Fri, 14 Sep 2007) |
- 3 lines Add a note to help clarify the value set with the
- echocancel option. (inspired by Malcolm's blog post on
- blogs.digium.com about HPEC) ........
-
-2007-09-14 19:49 +0000 [r82401] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c, configs/skinny.conf.sample: Add support
- in chan_skinny for sending RTP directly to the endpoints. Closes
- issue #9154, patch by DEA
-
-2007-09-14 18:37 +0000 [r82397-82400] Mark Michelson <mmichelson@digium.com>
-
- * /: Blocking revision 82398
-
- * /, apps/app_queue.c: Merged revisions 82396 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82396 | mmichelson | 2007-09-14 13:28:36 -0500 (Fri, 14 Sep
- 2007) | 5 lines Adding member name field to manager events where
- they were missing before (closes issue #10721, reported by snar)
- ........
-
-2007-09-14 17:51 +0000 [r82395] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 82394 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82394 | qwell | 2007-09-14 12:48:05 -0500 (Fri, 14 Sep 2007) | 5
- lines If a channel does not have an owner, do not try to set a
- channel variable. This will end up making the channel variable
- global, which is not right. Closes issue #10720, patch by
- flefoll. ........
-
-2007-09-14 17:29 +0000 [r82393] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/res_odbc.h, res/res_odbc.c: Add a direct execute
- method to res_odbc (closes issue #10722)
-
-2007-09-14 16:02 +0000 [r82386-82391] Russell Bryant <russell@digium.com>
-
- * channels/xpmr/xpmr.h, channels/xpmr/LICENSE (removed),
- channels/xpmr/sinetabx.h, channels/xpmr/xpmr.c,
- channels/xpmr/xpmr_coef.h: use the standard license header for
- the xpmr files
-
- * channels/chan_usbradio.c (added), channels/xpmr (added): Add
- chan_usbradio to trunk
-
- * /, build_tools/menuselect-deps.in, configure,
- include/asterisk/autoconfig.h.in, configure.ac, makeopts.in:
- Merged revisions 82385 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82385 | russell | 2007-09-14 10:50:49 -0500 (Fri, 14 Sep 2007) |
- 3 lines Add checking for libusb here, so nobody has to deal with
- conflicts in the chan_usbradio-1.4 branch every time the
- configure script gets changed ........
-
-2007-09-14 14:44 +0000 [r82377] Mark Michelson <mmichelson@digium.com>
-
- * doc/CODING-GUIDELINES, /: Merged revisions 82376 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r82376 | mmichelson | 2007-09-14 09:42:29 -0500 (Fri, 14
- Sep 2007) | 5 lines Fixing a typo in the coding guidelines
- (closes issue #10717, reported and patched by leedm777) ........
-
-2007-09-14 13:02 +0000 [r82373] Philippe Sultan <philippe.sultan@gmail.com>
-
- * channels/chan_jingle.c: Fix DTMF following what has been done in
- issue #9401. Thanks irroot.
-
-2007-09-13 23:12 +0000 [r82359] Jason Parker <jparker@digium.com>
-
- * pbx/pbx_spool.c, /: Merged revisions 82358 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82358 | qwell | 2007-09-13 18:11:27 -0500 (Thu, 13 Sep 2007) | 4
- lines Fix a small typo. retrytime > waittime ........
-
-2007-09-13 21:53 +0000 [r82347-82352] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Changed "in" to "queue" in "queue
- {pause|unpause} member" command to be more clear. Also added
- check to be sure that sixth argument is the word "reason" if full
- command is given
-
- * CHANGES, apps/app_queue.c: Added the ability to pause and unpause
- members via the CLI
-
- * /, apps/app_queue.c: Merged revisions 82346 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82346 | mmichelson | 2007-09-13 15:16:37 -0500 (Thu, 13 Sep
- 2007) | 4 lines Preemptively fixing a possible segfault. It is
- possible that queuename is NULL (meaning pause ALL queues), so
- use q->name instead. ........
-
-2007-09-13 20:13 +0000 [r82345] Jason Parker <jparker@digium.com>
-
- * /, cdr/cdr_csv.c: Merged revisions 82344 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82344 | qwell | 2007-09-13 15:11:40 -0500 (Thu, 13 Sep 2007) | 9
- lines Fix a crash that could occur in cdr_csv when mutliple
- threads tried to close the same file. Do we actually need the
- locking here? What happens if you open the same file twice, and
- two threads try to write to it at the same time? Is fputs() going
- to write out the entire line at once? I suspect that it could be
- possible for the second fopen to run during the first fputs, so
- the position could be in the middle of the previously written
- line... Issue 10347, initial patch by explidous (but I removed
- all of the paranoia stuff..) ........
-
-2007-09-13 19:16 +0000 [r82338-82341] Russell Bryant <russell@digium.com>
-
- * /, main/astobj2.c: Merged revisions 82339 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82339 | russell | 2007-09-13 13:57:08 -0500 (Thu, 13 Sep 2007) |
- 1 line resolve a warning when not building under dev mode
- ........
-
- * include/asterisk.h, /, main/astobj2.c, main/asterisk.c: Merged
- revisions 82337 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82337 | russell | 2007-09-13 13:45:59 -0500 (Thu, 13 Sep 2007) |
- 4 lines Only compile in tracking astobj2 statistics if dev-mode
- is enabled. Also, when dev mode is enabled, register the CLI
- command that can be used to run the astobj2 test and print out
- statistics. ........
-
-2007-09-13 18:13 +0000 [r82336] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, LICENSE: Merged revisions 82335 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r82335 | kpfleming | 2007-09-13 11:12:00 -0700
- (Thu, 13 Sep 2007) | 10 lines Merged revisions 82334 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r82334 | kpfleming | 2007-09-13 11:10:12 -0700 (Thu, 13
- Sep 2007) | 2 lines clarify the OpenSSL and OpenH323 license
- exceptions ........ ................
-
-2007-09-13 16:58 +0000 [r82329] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_zap.c, CHANGES, configs/zapata.conf.sample: Add
- setvar support to chan_zap. Just like you can in chan_sip and
- chan_iax2 you can now use it with zaptel channels. (done while in
- Montreal at the Asterisk bootcamp!)
-
-2007-09-13 16:27 +0000 [r82327] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 82326 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82326 | mmichelson | 2007-09-13 11:25:59 -0500 (Thu, 13 Sep
- 2007) | 7 lines Added logic to handle the unlikely case that
- someone has two queues with the same name. Asterisk will log a
- warning message letting the user know that one was already
- defined with that name and is it skipping all further instances.
- This also will work for realtime queues but in order for that to
- happen, the user would have to trigger a perfectly timed reload
- as a realtime queue is being looked up, which is highly unlikely
- (but taken care of nonetheless). ........
-
-2007-09-13 15:26 +0000 [r82321] Russell Bryant <russell@digium.com>
-
- * include/asterisk/doxyref.h, doc/res_config_sqlite.txt,
- res/res_config_sqlite.c, configs/res_config_sqlite.conf: Various
- code and documentation cleanups for res_config_sqlite (closes
- issue #10711, rbraun_proformatique)
-
-2007-09-13 15:25 +0000 [r82312-82320] Philippe Sultan <philippe.sultan@gmail.com>
-
- * channels/chan_jingle.c: Modify rule filters to match with the
- Jingle namespace constant
-
- * include/asterisk/jingle.h: Assign namespace properly
-
- * channels/chan_jingle.c, include/asterisk/jingle.h: Changed Jingle
- and Jingle DTMF namespaces. As both specifications are in the
- Experimental status, the namespaces specified therein shall be of
- the form "http://www.xmpp.org/extensions/xep-XXXX.html#ns". See
- the Namespace issuance section in XEP-0053 :
- http://www.xmpp.org/extensions/xep-0053.html#namespaces
-
- * channels/chan_jingle.c: Reflect Jingle DTMF specification changes
-
-2007-09-13 13:34 +0000 [r82311] Russell Bryant <russell@digium.com>
-
- * apps/app_queue.c: Fix a missing unref of a member struct. This
- was pointed out by Marta. Thanks! This function in 1.4 didn't
- have the problem.
-
-2007-09-13 11:54 +0000 [r82310] Philippe Sultan <philippe.sultan@gmail.com>
-
- * /, channels/chan_gtalk.c: Merged revisions 82309 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r82309 | phsultan | 2007-09-13 13:47:14 +0200 (Thu, 13
- Sep 2007) | 4 lines Closes issue #9401, reported and patched by
- irrot, with slight modifications by me. Handle DTMF sent by
- Asterisk properly. ........
-
-2007-09-12 21:57 +0000 [r82297] Russell Bryant <russell@digium.com>
-
- * /, res/res_agi.c: Merged revisions 82296 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82296 | russell | 2007-09-12 16:56:32 -0500 (Wed, 12 Sep 2007) |
- 3 lines Fix a check of the wrong pointer, as pointed out by an
- XXX comment left in the code. The problem was harmless, however.
- ........
-
-2007-09-12 21:55 +0000 [r82294] Jason Parker <jparker@digium.com>
-
- * channels/chan_iax2.c: After some discussions, we decided that the
- return values here were a bit messy. This also fixes a bug on
- reload, where peers may not have reregistered properly.
-
-2007-09-12 21:32 +0000 [r82290-82292] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/stdtime/tzfile.h: Merged revisions 82291 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r82291 | tilghman | 2007-09-12 16:28:33 -0500 (Wed, 12
- Sep 2007) | 2 lines Oops, wrong location for FreeBSD zone files
- ........
-
- * main/stdtime/private.h, /, main/stdtime/tzfile.h,
- funcs/func_strings.c, apps/app_sms.c,
- include/asterisk/localtime.h, main/stdtime/localtime.c: Merged
- revisions 82285 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82285 | tilghman | 2007-09-12 15:12:06 -0500 (Wed, 12 Sep 2007)
- | 4 lines Working on issue #10531 exposed a rather nasty 64-bit
- issue on ast_mktime, so we updated the localtime.c file from
- source. Next we'll have to write ast_strptime to match. ........
-
-2007-09-12 21:17 +0000 [r82289] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Removed an unneeded ao2_ref. This was a problem
- because unless get_member_status returned QUEUE_NORMAL, a NULL
- member would be unreferenced. While this didn't cause any crashes
- or anything terrible, it still is incorrect
-
-2007-09-12 20:50 +0000 [r82288] Steve Murphy <murf@digium.com>
-
- * main/config.c: This fix closes issue #10642 -- it's not perfect,
- but should retain most blank lines in config files, via
- read/write cycles.
-
-2007-09-12 20:47 +0000 [r82287] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 82286 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82286 | dhubbard | 2007-09-12 15:24:24 -0500 (Wed, 12 Sep 2007)
- | 1 line remove a race condition for the creation of
- recordthread's, and fix a small memory leak. This closes issue#
- 10636 ........
-
-2007-09-12 16:24 +0000 [r82283] Mark Michelson <mmichelson@digium.com>
-
- * main/pbx.c, main/app.c, main/asterisk.c: Fixes Solaris build
- warnings (closes issue #10698, reported and patched by snuffy)
-
-2007-09-12 15:53 +0000 [r82279-82282] Russell Bryant <russell@digium.com>
-
- * utils/hashtest2.c: Change the traversal to use ao2_callback()
- instead of an ao2_iterator. Using ao2_callback() is a much more
- efficient way of performing an operation on every item in the
- container. This change makes hashtest2 run in about 25% of the
- time it ran before on my system. In general, I would say that it
- makes the most sense to use an ao2_iterator if the operation
- being performed is going to take a long time and you don't want
- to keep the container locked while you work with each object.
- Otherwise, the use of ao2_callback is preferred.
-
- * /, main/asterisk.c: Merged revisions 82280 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82280 | russell | 2007-09-12 10:16:49 -0500 (Wed, 12 Sep 2007) |
- 4 lines Clean up the output of "asterisk -h". This tweaks the
- wording and wraps lines at 80 characters. (closes issue #10699,
- seanbright) ........
-
- * /, res/res_agi.c: Merged revisions 82278 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82278 | russell | 2007-09-12 10:11:11 -0500 (Wed, 12 Sep 2007) |
- 3 lines revert patch from issue #10553, as someone not using
- fastagi reported that this broke their system. ........
-
-2007-09-12 14:31 +0000 [r82275-82277] Mark Michelson <mmichelson@digium.com>
-
- * /: Blocking changes from revision 82276
-
- * /, apps/app_queue.c: Merged revisions 82274 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82274 | mmichelson | 2007-09-12 09:24:53 -0500 (Wed, 12 Sep
- 2007) | 6 lines We should only initialize a realtime queue when
- it is allocated, not every time we access it. This prevents the
- members ao2_container from being reallocated every time the queue
- is accessed. I also removed a debug message I had accidentally
- left in on a previous commit. ........
-
-2007-09-11 23:07 +0000 [r82273] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Fix to make sure we don't hangup a call when
- getting a RLC without sending REL. Found making sure we are Q.784
- (the SS7 test specification) compliant
-
-2007-09-11 22:38 +0000 [r82269-82270] Russell Bryant <russell@digium.com>
-
- * main/config.c: remove unused functions that made this file not
- build under dev mode
-
- * /, apps/app_queue.c: Merged revisions 82267 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82267 | russell | 2007-09-11 17:37:17 -0500 (Tue, 11 Sep 2007) |
- 3 lines Fix incorrect uses of ao2_find(). Every one of these
- calls was reading bogus memory ... ........
-
-2007-09-11 22:37 +0000 [r82268] Steve Murphy <murf@digium.com>
-
- * utils/Makefile, main/config.c: This solves an unreported solaris
- compile problem (missing -lnsl -lsocket).
-
-2007-09-11 21:43 +0000 [r82266] Joshua Colp <jcolp@digium.com>
-
- * /, codecs/gsm/src/long_term.c, codecs/gsm/src/lpc.c: Merged
- revisions 82265 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82265 | file | 2007-09-11 18:41:49 -0300 (Tue, 11 Sep 2007) | 4
- lines (closes issue #10679) Reported by: andrew Build under dev
- mode when K6OPTS is enabled. ........
-
-2007-09-11 20:50 +0000 [r82264] Russell Bryant <russell@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 82263 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82263 | russell | 2007-09-11 15:49:34 -0500 (Tue, 11 Sep 2007) |
- 5 lines Fix another missing unref of member objects. This one was
- pointed out by Marta. When building the outgoing list in
- try_calling(), a member reference is stored in each outgoing
- entry. However, when this list got destroyed, the reference was
- not released. ........
-
-2007-09-11 20:49 +0000 [r82262] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, /: Merged revisions 82261 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82261 | murf | 2007-09-11 14:36:15 -0600 (Tue, 11 Sep 2007) | 1
- line this change should fix issue # 10659 -- what I worry about
- is how many other bug reports it may generate. Hopefully, we can
- please the/a majority. Hopefully. We shall see. Calls not marked
- ANSWERED and with only one channel name will not be posted. This
- should eliminate the double CDR's. ........
-
-2007-09-11 18:37 +0000 [r82257-82258] Joshua Colp <jcolp@digium.com>
-
- * configs/sip.conf.sample: Lil' bit more documentation to keep
- folks happy.
-
- * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: (closes
- issue #9433) Reported by: junky Patches: register_trying.diff.txt
- uploaded by jcmoore Disable sending 100 Trying on REGISTER
- attempts and make it an option. This has been signed off by oej.
-
-2007-09-11 17:16 +0000 [r82256] Steve Murphy <murf@digium.com>
-
- * utils/Makefile: fixing up the pthread stuff for hashtest2
-
-2007-09-11 16:15 +0000 [r82254] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, channels/misdn/isdn_lib.c: Merged
- revisions 82249 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82249 | crichter | 2007-09-11 18:01:27 +0200 (Di, 11 Sep 2007) |
- 1 line fixed a hold/retrieve issue. ........
-
-2007-09-11 16:12 +0000 [r82253] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 82252 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82252 | mmichelson | 2007-09-11 11:05:56 -0500 (Tue, 11 Sep
- 2007) | 6 lines All instances of ao2_iterators which were just
- named 'i' have been renamed to 'mem_iter' so that when refcounted
- queues are merged into trunk, there will be little confusion
- regarding iterator names, especially when a queue and member
- iterator are used in the same function. ........
-
-2007-09-11 16:05 +0000 [r82251] Russell Bryant <russell@digium.com>
-
- * /, pbx/pbx_dundi.c: Merged revisions 82250 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82250 | russell | 2007-09-11 11:03:42 -0500 (Tue, 11 Sep 2007) |
- 4 lines The sample dundi.conf claims support for a wildcard peer
- entry - [*], but the code did not support it. This patch makes it
- work. (closes issue #10546, patch by dds, with some changes by
- me) ........
-
-2007-09-11 15:34 +0000 [r82248] Joshua Colp <jcolp@digium.com>
-
- * main/cdr.c: (closes issue #10666) Reported by: arkadia Patches:
- cdr_lockorder.patch uploaded by arkadia (license 233) Optimize
- CDR stuff a bit.
-
-2007-09-11 15:31 +0000 [r82246-82247] Russell Bryant <russell@digium.com>
-
- * res/res_agi.c: Remove an unused variable. I have no idea why this
- was marked with the unused attribute instead of just removing it.
- :)
-
- * /, res/res_agi.c: Merged revisions 82245 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82245 | russell | 2007-09-11 10:26:51 -0500 (Tue, 11 Sep 2007) |
- 9 lines (closes issue #10553) Reported by: juggie Patches:
- res_agi_fgets-2.patch uploaded by juggie (license 24) Tested by:
- juggie When using fastagi, fgets() can return before a full line
- is read. Add explicit handling for the case where it gets
- interrupted. ........
-
-2007-09-11 14:58 +0000 [r82242-82244] Joshua Colp <jcolp@digium.com>
-
- * /, pbx/pbx_dundi.c: Merged revisions 82243 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82243 | file | 2007-09-11 11:56:39 -0300 (Tue, 11 Sep 2007) | 6
- lines (closes issue #10577) Reported by: jamesgolovich Patches:
- asterisk-dundifree.diff.txt uploaded by jamesgolovich (license
- 176) Don't leak memory when unloading DUNDi. ........
-
- * apps/app_meetme.c: (closes issue #10560) Reported by: ruffle
- Patches: rb uploaded by ruffle (license 201) Show whether the
- conference is locked or not on the CLI.
-
-2007-09-11 14:35 +0000 [r82237-82241] Russell Bryant <russell@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 82240 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82240 | russell | 2007-09-11 09:34:12 -0500 (Tue, 11 Sep 2007) |
- 2 lines Add a couple more missing unrefs of queue member objects
- ........
-
- * /, apps/app_queue.c: Merged revisions 82238 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82238 | russell | 2007-09-11 09:21:17 -0500 (Tue, 11 Sep 2007) |
- 2 lines Add a missing unref of a queue member in an error
- handling block ........
-
- * /, apps/app_queue.c: Merged revisions 82236 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82236 | russell | 2007-09-11 09:09:43 -0500 (Tue, 11 Sep 2007) |
- 2 lines Document why membercount can not simply be replaced by
- ao2_container_count() ........
-
-2007-09-11 13:46 +0000 [r82231-82235] Joshua Colp <jcolp@digium.com>
-
- * utils/Makefile: Include string compatibility file in hashtest2.
-
- * utils/hashtest2.c: Include compat.h to hopefully make it
- compatible with FreeBSD.
-
- * utils/hashtest2.c: Fix building under FreeBSD. Make sure alloca.h
- exists before including it.
-
- * main/manager.c: (closes issue #10695) Reported by: junky Patches:
- count_showconn.diff uploaded by junky (license 177) Provide a
- count of connected users to manager.
-
- * main/minimime/minimime.c, main/minimime/tests/create.c,
- main/minimime/mm_mem.c, main/minimime/tests/parse.c: (closes
- issue #10692) Reported by: snuffy Patches: minivm.diff uploaded
- by snuffy (license 35) Instead of using err (which is not
- available under Solaris) use fdprintf with stderr.
-
-2007-09-10 20:03 +0000 [r82200] Tilghman Lesher <tlesher@digium.com>
-
- * UPGRADE.txt, channels/chan_iax2.c: Change the IAXPeers command to
- have manager-style output, instead of CLI-style output (closes
- issue #8254)
-
-2007-09-10 19:10 +0000 [r82185] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Fixing a problem where NULL channels would
- cause a crash when calling indisposed queue members (i.e. paused,
- wrapup time not completed, etc.)
-
-2007-09-10 18:32 +0000 [r82178] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 82155 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r82155 | tilghman | 2007-09-10 13:02:02 -0500 (Mon, 10 Sep 2007)
- | 2 lines Convert struct member to use refcounts (closes issue
- #10199) ........
-
-2007-09-10 17:39 +0000 [r82154] Jason Parker <jparker@digium.com>
-
- * main/db.c: Add a counter to the 'database deltree' CLI command.
- Note: this is slightly different than the initial patch, because
- I felt that using res <= 0 would be a change in behavior. Closes
- issue #10687, patch by junky
-
-2007-09-10 16:59 +0000 [r82140] Steve Murphy <murf@digium.com>
-
- * utils/Makefile, utils/hashtest2.c (added): Committing my test for
- astobj2, hashtest2.c, along with makefile changes in utils.
-
-2007-09-10 16:24 +0000 [r82125] Jason Parker <jparker@digium.com>
-
- * main/db.c: Add counter to 'database show' CLI command. (also a
- minor whitespace change that I found along the way) Closes issue
- #10683, patch by junky
-
-2007-09-10 16:19 +0000 [r82124] Steve Murphy <murf@digium.com>
-
- * main/astobj2.c: Changes applied from marta's team/marta/astobj2
- branch to solve a race condition
-
-2007-09-10 15:05 +0000 [r82092] Mark Michelson <mmichelson@digium.com>
-
- * /, configs/misdn.conf.sample: Merged revisions 82091 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r82091 | mmichelson | 2007-09-10 10:02:12 -0500 (Mon, 10
- Sep 2007) | 5 lines Removing non-existent options from misdn
- configuration sample. (closes issue #10678, reported and patched
- by IgorG) ........
-
-2007-09-10 14:26 +0000 [r82062-82077] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: (closes issue #10688) Reported by: casper
- Patches: chan_sip.c.82076.diff uploaded by casper (license 55)
- Remove double check for zombie flag and optimize things a bit.
-
- * res/res_agi.c: (closes issue #10684) Reported by: junky Patches:
- debug.diff uploaded by junky (license 177) Fix issue with debug
- always showing up.
-
- * apps/app_meetme.c: (closes issue #10686) Reported by: junky
- Patches: meet.diff uploaded by junky (license 177) Change NOTICE
- message to DEBUG.
-
-2007-09-09 02:45 +0000 [r82029] Tilghman Lesher <tlesher@digium.com>
-
- * /, include/asterisk/lock.h: Merged revisions 82028 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r82028 | tilghman | 2007-09-08 21:35:18 -0500 (Sat, 08
- Sep 2007) | 2 lines Fix inline compiles on really old compilers
- (who uses gcc 2.7 anymore, really?) (closes issue #10675)
- ........
-
-2007-09-08 19:01 +0000 [r81998-81999] Russell Bryant <russell@digium.com>
-
- * include/asterisk/slinfactory.h: Add doxygen documentation for
- slinfactory_destroy(), mainly just noting that it doesn't free
- the slinfactory itself. (This isn't related to a bug, i'm just
- looking over random code)
-
- * /, main/asterisk.c: Merged revisions 81997 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81997 | russell | 2007-09-08 13:41:32 -0500 (Sat, 08 Sep 2007) |
- 2 lines Fix a small memory leak. ast_unregister_atexit() did not
- free the entry it removed. ........
-
-2007-09-08 16:37 +0000 [r81984] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Make Callerid more consistent in IMAP mail
- headers (closes issue #10056, reported and patched by jaroth,
- with small modification by me)
-
-2007-09-08 13:45 +0000 [r81953] Russell Bryant <russell@digium.com>
-
- * /, .cleancount: Merged revisions 81952 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81952 | russell | 2007-09-08 08:42:26 -0500 (Sat, 08 Sep 2007) |
- 11 lines (closes issue #10672) Bump the cleancount so that a
- "make clean" will be forced. This is needed because my fix in
- revision 81599 made a change to a data structure in file.h, and
- since file dependency tracking is only on with dev-mode enabled,
- file format modules that don't get rebuilt may crash, as is the
- case with this issue. This makes me wonder - how much faster does
- the code build without the file dependency tracking enabled? If
- it doesn't make much of a difference, then it may be worth just
- keeping it on all of the time, or perhaps just not in release
- tarballs, so that this type of issue is avoided. ........
-
-2007-09-07 19:53 +0000 [r81910-81924] Jason Parker <jparker@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 81923 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #10671) ........ r81923 | qwell | 2007-09-07 14:48:00 -0500
- (Fri, 07 Sep 2007) | 5 lines Allow the MEMBERINTERFACE variable
- to be used as the mixmonitor filename. This moves the setting of
- the MEMBERINTERFACE variable to before mixmonitor. Issue 10671,
- patch by sim. ........
-
- * apps/app_queue.c: Add an optional reason parameter to
- PauseQueueMember/UnpauseQueueMember applications and manager
- events. Issue 8738, patch by rgollent
-
-2007-09-07 15:29 +0000 [r81891] Mark Michelson <mmichelson@digium.com>
-
- * /, configs/queues.conf.sample: Merged revisions 81886 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81886 | mmichelson | 2007-09-07 10:25:19 -0500 (Fri, 07 Sep
- 2007) | 3 lines Moving the explanation for joinempty to a more
- appropriate place ........
-
-2007-09-07 12:32 +0000 [r81858-81873] Joshua Colp <jcolp@digium.com>
-
- * configure, configure.ac: Don't check for epoll support when cross
- compiling.
-
- * main/channel.c, main/audiohook.c: Fix memory issue that crept up
- with Russell's testing. It is *not* proper to free the frame we
- get in ast_write.
-
-2007-09-06 22:32 +0000 [r81839-81849] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: fix the build ... oops
-
- * /, channels/chan_sip.c: Merged revisions 81832 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81832 | russell | 2007-09-06 17:28:57 -0500 (Thu, 06 Sep 2007) |
- 16 lines (closes issue #9724, closes issue #10374) Reported by:
- kenw Patches: 9724.txt uploaded by russell (license 2) Tested by:
- kenw, russell Resolve a deadlock that occurs when doing a SIP
- transfer to parking. I come across this type of deadlock fairly
- often it seems. It is very important to mind the boundary between
- the channel driver and the core in respect to the channel lock
- and the channel-pvt lock. Channel drivers lock to lock the pvt
- and then the channel once it calls into the core, while the core
- will do it in the opposite order. The way this is avoided is by
- having channel drivers either release their pvt lock while
- calling into the core, or such as in this case, unlocking the pvt
- just long enough to acquire the channel lock. ........
-
-2007-09-06 22:06 +0000 [r81827] Jason Parker <jparker@digium.com>
-
- * Makefile, /: Merged revisions 81826 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81826 | qwell | 2007-09-06 17:05:02 -0500 (Thu, 06 Sep 2007) | 1
- line We added COPTS for ASTCFLAGS additions, but not LDOPTS for
- ASTLDFLAGS. This adds LDOPTS ........
-
-2007-09-06 21:01 +0000 [r81814] Joshua Colp <jcolp@digium.com>
-
- * channels/iax2-parser.c: Initialize iax_frames variable to NULL,
- keeps valgrind happy.
-
-2007-09-06 20:54 +0000 [r81783-81813] Russell Bryant <russell@digium.com>
-
- * CHANGES, funcs/func_extstate.c (added): Add EXTENSION_STATE()
- function that can retrieve the state of an extension that has a
- hint. (closes issue #10635, adamgundy)
-
- * CHANGES: s/DEVSTATE/DEVICE_STATE/
-
- * funcs/func_devstate.c: Rename the DEVSTATE() function to
- DEVICE_STATE() to better conform to how other functions are
- named. (inspired by issue #10635)
-
- * CHANGES, funcs/func_devstate.c: Merge HINT() dialplan function
- from my sandbox branch into trunk. This function will let you
- retrieve the list of devices or name associated with a hint.
- (inspired by issue #10635)
-
-2007-09-06 20:16 +0000 [r81782] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_skinny.c, CHANGES: (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.
-
-2007-09-06 20:05 +0000 [r81781] Russell Bryant <russell@digium.com>
-
- * configs/extensions.conf.sample: Fix the syntax of declaring a
- hint with a name to be compatible with trunk
-
-2007-09-06 20:00 +0000 [r81779] Jason Parker <jparker@digium.com>
-
- * /, include/asterisk/astobj2.h: Merged revisions 81778 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81778 | qwell | 2007-09-06 14:59:07 -0500 (Thu, 06 Sep 2007) | 2
- lines This should fix a build issue that people building against
- uClibc were seeing with the addition of astobj2 ........
-
-2007-09-06 19:43 +0000 [r81777] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 81776 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81776 | file | 2007-09-06 16:40:37 -0300 (Thu, 06 Sep 2007) | 7
- lines (closes issue #10122) Reported by: stevefeinstein Patches:
- meetme-unmute-manager.diff uploaded by qwell (license 4) Tested
- by: stevefeinstein After looking over the code I agree with
- Qwell. Setting the file descriptor to conference each time just
- causes a fight back and forth. ........
-
-2007-09-06 17:00 +0000 [r81745] Philippe Sultan <philippe.sultan@gmail.com>
-
- * /, include/asterisk/jabber.h, channels/chan_gtalk.c: Merged
- revisions 81743 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81743 | phsultan | 2007-09-06 18:56:29 +0200 (Thu, 06 Sep 2007)
- | 1 line Various string length fixes. Removed an unused variable
- in aji_client structure (context) ........
-
-2007-09-06 16:57 +0000 [r81744] Tilghman Lesher <tlesher@digium.com>
-
- * contrib/scripts/safe_asterisk: Incorporate the ability to log
- output of safe_asterisk to syslog (closes issue #9882)
-
-2007-09-06 16:38 +0000 [r81742] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Patch on 10575. Add support for unequipped
- CIC (UCIC) message as well as improve some of our CIC flags in
- chan_zap
-
-2007-09-06 16:31 +0000 [r81730] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 81713 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81713 | mmichelson | 2007-09-06 11:25:40 -0500 (Thu, 06 Sep
- 2007) | 6 lines Fixes an issue where valid DTMF had to be pressed
- twice to exit a queue if a member's phone was ringing. (closes
- issue #10655, reported by strider2k, patched by me) ........
-
-2007-09-06 15:43 +0000 [r81712] Luigi Rizzo <rizzo@icir.org>
-
- * include/asterisk/astobj2.h, main/astobj2.c: various changes to
- the documentation, and redefinition of ao2_hash_fn and
- ao2_callback_fn typedefs, in preparation to more cleanup of the
- _search_flags Please do not merge this change to 1.4 yet - there
- are no functional changes anyways.
-
-2007-09-06 15:21 +0000 [r81683] Mark Michelson <mmichelson@digium.com>
-
- * /, res/res_features.c: Merged revisions 81682 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81682 | mmichelson | 2007-09-06 10:20:36 -0500 (Thu, 06 Sep
- 2007) | 5 lines Fixes a memory leak (closes issue #10658,
- reported and patched by Ivan) ........
-
-2007-09-06 14:24 +0000 [r81651] Philippe Sultan <philippe.sultan@gmail.com>
-
- * /, res/res_jabber.c: Merged revisions 81650 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81650 | phsultan | 2007-09-06 16:20:54 +0200 (Thu, 06 Sep 2007)
- | 3 lines According to both RFC 3920 - section 9.1.2 - and
- Google's XMPP server complaint, if set, the 'from' attribute must
- be set to the user's full JID. ........
-
-2007-09-05 21:59 +0000 [r81632] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Not having this epoll specific code in
- wait_for_answer was causing app_queue to infinitely loop. This
- makes it so it doesn't. Thanks to file for pointing out where the
- problem was and showing a similar function in app_dial as an
- example of how to fix it.
-
-2007-09-05 21:45 +0000 [r81631] Tilghman Lesher <tlesher@digium.com>
-
- * /, include/asterisk/lock.h: Merged revisions 81569 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r81569 | tilghman | 2007-09-05 12:18:24 -0500 (Wed, 05
- Sep 2007) | 2 lines Solaris x86 compatibility fix ........
-
-2007-09-05 20:58 +0000 [r81601] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * apps/app_zapateller.c: added ZAPATELLERSTATUS to app_zapateller
-
-2007-09-05 20:58 +0000 [r81600] Russell Bryant <russell@digium.com>
-
- * include/asterisk/file.h, /, main/say.c, res/res_features.c,
- main/file.c, include/asterisk/channel.h: Merged revisions 81599
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81599 | russell | 2007-09-05 15:53:41 -0500 (Wed, 05 Sep 2007) |
- 11 lines Fix an issue that can occur when you do an attended
- transfer to parking. If you complete the transfer before the
- announcement of the parking spot finishes, then the channel being
- parked will hear the remainder of the announcement. These changes
- make it so that will not happen anymore. Basically, res_features
- sets a flag on the channel is playing the announcement to so that
- the file streaming core knows that it needs to watch out for a
- channel masquerade, and if it occurs, to abort the announcement.
- (closes BE-182) ........
-
-2007-09-05 16:48 +0000 [r81568] Tilghman Lesher <tlesher@digium.com>
-
- * utils: Add two more generated files (requested by mvanbaak via
- irc)
-
-2007-09-05 16:31 +0000 [r81560] Jason Parker <jparker@digium.com>
-
- * include/asterisk/devicestate.h, res/res_config_odbc.c,
- channels/chan_sip.c, include/asterisk/audiohook.h, main/sha1.c,
- res/res_features.c, include/asterisk/astobj2.h, res/res_crypto.c,
- include/asterisk/strings.h, main/audiohook.c, res/res_jabber.c,
- res/res_config_sqlite.c, include/asterisk/sha1.h,
- include/asterisk/stringfields.h, include/asterisk/features.h:
- Doxygen cleanups/fixes. Closes issue #10654, patch by snuffy
-
-2007-09-05 15:32 +0000 [r81526-81535] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Weird. When I merged my changes from 1.4, they
- merged into the wrong function. This should fix the build for
- trunk.
-
- * /, apps/app_queue.c: Merged revisions 81525 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81525 | mmichelson | 2007-09-05 10:19:47 -0500 (Wed, 05 Sep
- 2007) | 4 lines Fixing the build... ........
-
-2007-09-05 15:16 +0000 [r81524] Jason Parker <jparker@digium.com>
-
- * channels/chan_phone.c, /: Merged revisions 81523 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #10651) ........ r81523 | qwell | 2007-09-05 10:14:30 -0500
- (Wed, 05 Sep 2007) | 5 lines Do not try to unregister a NULL
- channel tech. Also changed load_module function to use defines
- rather than numbers for return values. Issue 10651, patch by
- rbraun_proformatique, with additions by me. ........
-
-2007-09-05 15:04 +0000 [r81522] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 81520 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81520 | mmichelson | 2007-09-05 10:03:22 -0500 (Wed, 05 Sep
- 2007) | 6 lines Reverting behavior of QUEUE_MEMBER_COUNT to only
- count members who are logged in and available. (related to issue
- #10652, reported by wuwu) ........
-
-2007-09-05 14:47 +0000 [r81519] Steve Murphy <murf@digium.com>
-
- * include/asterisk/config.h, main/config.c: this set of changes
- fixes issue # 10643 by keeping track of the last object defined
- in a file, and attaching any accumulated comments to that object
- (category header or variable declaration). The file_save routine
- also had to be upgraded to output these trailing comments.
- Config.h was modified to include the trailing comment list on
- categories and variables.
-
-2007-09-05 13:13 +0000 [r81459-81493] Joshua Colp <jcolp@digium.com>
-
- * main/editline/sys.h: Finish up commit from revision 81452 by
- removing last remnants of strlcat/strlcpy checks.
-
-2007-09-04 20:59 +0000 [r81454-81456] Jason Parker <jparker@digium.com>
-
- * /, apps/app_followme.c: Merged revisions 81455 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #10634) ........ r81455 | qwell | 2007-09-04 15:54:51 -0500
- (Tue, 04 Sep 2007) | 4 lines Rather than attempt to play a file,
- we can just check whether it exists. Issue 10634, patch by me,
- testing by pabelanger, sanity checked by bweschke ........
-
- * /, configs/followme.conf.sample: Merged revisions 81453 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #10644) ........ r81453 | qwell | 2007-09-04 14:56:06 -0500
- (Tue, 04 Sep 2007) | 4 lines Change default followme config file
- to point to the correct files. Issue 10644, patch by pabelanger
- ........
-
-2007-09-04 19:51 +0000 [r81445-81452] Russell Bryant <russell@digium.com>
-
- * main/editline/configure, main/editline/configure.in: Don't check
- for and include strlcpy and strlcat in editline. We also include
- them directly in Asterisk. For platforms that need them (like my
- mac), you will get a linker error due to the functions being
- included twice.
-
- * /, include/asterisk/astobj2.h, channels/chan_iax2.c,
- main/astobj2.c: Merged revisions 81448 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81448 | russell | 2007-09-04 13:37:44 -0500 (Tue, 04 Sep 2007) |
- 4 lines Remove the typedefs on ao2_container and ao2_iterator.
- This is simply because we don't typedef objects anywhere else in
- Asterisk, so we might as well make this follow the same
- convention. ........
-
- * include/asterisk/logger.h: logger.h depends on options.h, so go
- ahead and include it
-
-2007-09-04 16:41 +0000 [r81443] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 81442 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81442 | kpfleming | 2007-09-04 11:40:39 -0500 (Tue, 04 Sep 2007)
- | 2 lines there is no point in sending 401 Unauthorized to a UAS
- that sent us a properly-formatted Authentication header with the
- expected username and nonce but an incorrect response (which
- indicates the shared secret does not match)... instead, let's
- send 403 Forbidden so that the UAS doesn't retry with the same
- authentication credentials repeatedly ........
-
-2007-09-04 14:28 +0000 [r81436-81441] Joshua Colp <jcolp@digium.com>
-
- * configs/extensions.ael.sample: (closes issue #10633) Reported by:
- pabelanger Patches: extensions.ael.sample.patch uploaded by
- pabelanger (license 224) Update extensions.ael.sample with
- voicemail and | changes.
-
- * /, channels/chan_iax2.c: Merged revisions 81439 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81439 | file | 2007-09-04 11:23:18 -0300 (Tue, 04 Sep 2007) | 6
- lines (closes issue #10632) Reported by: jamesgolovich Patches:
- asterisk-iaxfirmwareleak.diff.txt uploaded by jamesgolovich
- (license 176) Fix memory leak when unloading chan_iax2. The
- firmware files were not being freed. ........
-
- * main/channel.c, /: Merged revisions 81437 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81437 | file | 2007-09-04 10:46:23 -0300 (Tue, 04 Sep 2007) | 4
- lines (closes issue #10476) Reported by: mdu113 Only look for the
- end of a digit when waiting for a digit. This in turn disables
- emulation in the core. ........
-
- * /, main/dns.c: Merged revisions 81435 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81435 | file | 2007-09-04 10:10:56 -0300 (Tue, 04 Sep 2007) | 7
- lines (closes issue #10610) Reported by: john Patches:
- dns.c.patch uploaded by john (license 218) Tested by: mvanbaak
- Don't return a match if no SRV record actually exists. ........
-
-2007-09-03 18:59 +0000 [r81434] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 81433 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81433 | russell | 2007-09-03 13:57:53 -0500 (Mon, 03 Sep 2007) |
- 5 lines Remove a couple of calls to ast_string_field_free_pools()
- on peers in error handling blocks in the code for building peers.
- The peer object destructor does this and doing it twice will
- cause a crash. (closes issue #10625, reported by and patched by
- pnlarsson) ........
-
-2007-09-03 18:01 +0000 [r81430-81432] Tilghman Lesher <tlesher@digium.com>
-
- * main/config.c: Once we get past the file checks, we're loading,
- so clear the FILEUNCHANGED flag (fixes #include) (closes issue
- #10629)
-
- * /, funcs/func_logic.c: Merged revisions 81415 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81415 | tilghman | 2007-08-31 14:16:52 -0500 (Fri, 31 Aug 2007)
- | 2 lines The IF() function was not allowing true values that had
- embedded colons (closes issue #10613) ........
-
- * main/config.c: We shouldn't use a filename blindly without
- checking to make sure it's unused first
-
-2007-09-01 06:03 +0000 [r81427] Mark Michelson <mmichelson@digium.com>
-
- * /, include/asterisk/astobj2.h, main/astobj2.c: Merged revisions
- 81426 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81426 | mmichelson | 2007-09-01 01:02:06 -0500 (Sat, 01 Sep
- 2007) | 4 lines Making match_by_addr into ao2_match_by_addr and
- making it available everywhere since it could be a handy callback
- to have ........
-
-2007-08-31 21:29 +0000 [r81419] Russell Bryant <russell@digium.com>
-
- * /, include/asterisk/astobj2.h: Merged revisions 81418 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81418 | russell | 2007-08-31 16:27:49 -0500 (Fri, 31 Aug 2007) |
- 2 lines Remove references to a debugging parameter that does not
- exist ........
-
-2007-08-31 19:50 +0000 [r81417] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 81416 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81416 | mmichelson | 2007-08-31 14:48:55 -0500 (Fri, 31 Aug
- 2007) | 6 lines Fixed broken behavior of a reload on realtime
- queues. Prior to this patch, if a reload was issued and a
- realtime queue had callers waiting in it, then the queue would be
- removed from the queue list, but it would not actually be freed
- (in fact, a debug message warning about a memory leak would come
- up). With this patch, reloads do not touch realtime queues at
- all. ........
-
-2007-08-31 18:46 +0000 [r81413] Jason Parker <jparker@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 81412 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #10621) ........ r81412 | qwell | 2007-08-31 13:44:44 -0500
- (Fri, 31 Aug 2007) | 4 lines Re-order dial options to be in line
- with the existing alpha order. Issue 10621, initial patch by
- junky ........
-
-2007-08-31 17:43 +0000 [r81411] Philippe Sultan <philippe.sultan@gmail.com>
-
- * /, channels/chan_gtalk.c: Merged revisions 81410 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r81410 | phsultan | 2007-08-31 19:38:26 +0200 (Fri, 31
- Aug 2007) | 3 lines Make the 'gtalk show channels' CLI command
- available. Closes issue 10548, reported by keepitcool. ........
-
-2007-08-31 15:58 +0000 [r81408] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, codecs/codec_zap.c: Merged revisions 81405 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81405 | kpfleming | 2007-08-31 10:51:45 -0500 (Fri, 31 Aug 2007)
- | 2 lines add missing "transcoder show" (and deprecated "show
- transcoder") CLI commands that were in 1.2 but never added to 1.4
- ........
-
-2007-08-31 15:54 +0000 [r81402-81407] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_speech.c: Merged revisions 81406 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81406 | file | 2007-08-31 12:53:16 -0300 (Fri, 31 Aug 2007) | 2
- lines Make it the engine's responsible to check for the presence
- of results. ........
-
- * /, res/res_features.c: Merged revisions 81403 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81403 | file | 2007-08-31 11:38:59 -0300 (Fri, 31 Aug 2007) | 4
- lines (closes issue #10618) Reported by: dimas Don't pass through
- the stopped sounds frame.... just drop it. ........
-
- * /, res/res_features.c: Merged revisions 81401 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81401 | file | 2007-08-30 20:53:41 -0300 (Thu, 30 Aug 2007) | 4
- lines (closes issue #10009) Reported by: dimas Don't output a
- bridge failed warning message if it failed because one of the
- channels was part of the masquerade process. That is perfectly
- normal. ........
-
-2007-08-30 23:52 +0000 [r81400] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_zap.c: Add new queryable fields from zaptel to 'zap
- show status'
-
-2007-08-30 22:08 +0000 [r81398] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 81397 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81397 | mmichelson | 2007-08-30 17:05:56 -0500 (Thu, 30 Aug
- 2007) | 7 lines Removing an extraneous (and possibly misleading)
- log message. Firstly, if the announce file isn't found, the
- streaming functions will report it. Secondly, not all non-zero
- returns from play_file mean that the announce file wasn't found.
- Positive return values simply mean that a digit was pressed (most
- likely to skip through the announcement). (closes issue #10612,
- reported and patched by dimas) ........
-
-2007-08-30 21:25 +0000 [r81394-81396] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 81395 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81395 | file | 2007-08-30 18:23:50 -0300 (Thu, 30 Aug 2007) | 6
- lines (closes issue #10514) Reported by: casper Patches:
- chan_sip.c.80129.diff uploaded by casper (license 55) Remove
- needless check for AUTH_UNKNOWN_DOMAIN. It was impossible for it
- to ever be that value. ........
-
- * channels/chan_sip.c: (closes issue #10565) Reported by: tootai
- Make sure the external IP address has the standard SIP port set
- for when the user does not specify the port in the externip
- setting.
-
-2007-08-30 21:16 +0000 [r81393] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, /: Merged revisions 81392 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81392 | murf | 2007-08-30 15:11:48 -0600 (Thu, 30 Aug 2007) | 1
- line via issue 10599, where 'CDR already initialized' messages
- are being generated. Since all channels will have an init'd CDR
- attached at creation time, this message is now particularly
- useless. Removed. ........
-
-2007-08-30 20:55 +0000 [r81391] Joshua Colp <jcolp@digium.com>
-
- * apps/app_minivm.c: (closes issue #10336) Reported by: junky
- Patches: minivm_output2.diff uploaded by junky (license 177)
- Change console output of minivm show stats to be more simple for
- external parsing.
-
-2007-08-30 20:31 +0000 [r81389-81390] Tilghman Lesher <tlesher@digium.com>
-
- * main/sched.c: A schedule id of 0 is not possible and is used to
- flag that we want to add a new item
-
- * apps/app_readexten.c: Change wording as requested by Kevin
-
-2007-08-30 18:52 +0000 [r81388] Mark Michelson <mmichelson@digium.com>
-
- * configs/queues.conf.sample: Added note to sample queues.conf file
- to line up with most recent change regarding setinterfacevar.
- MEMBERREALTIME indicates whether a member is realtime.
-
-2007-08-30 17:51 +0000 [r81387] Tilghman Lesher <tlesher@digium.com>
-
- * main/logger.c: Always force reread of the config when we're
- rotating the log file (closes issue #10598)
-
-2007-08-30 15:40 +0000 [r81384] Russell Bryant <russell@digium.com>
-
- * /, channels/h323/ast_h323.cxx: Merged revisions 81383 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81383 | russell | 2007-08-30 10:38:29 -0500 (Thu, 30 Aug 2007) |
- 3 lines Add missing checks for the PTRACING define. (closes issue
- #10559, paravoid) ........
-
-2007-08-30 15:36 +0000 [r81382] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 81381 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81381 | mmichelson | 2007-08-30 10:35:51 -0500 (Thu, 30 Aug
- 2007) | 3 lines Changed some manager event messages to reflect
- whether a queue member is a realtime member or not ........
-
-2007-08-30 15:34 +0000 [r81380] Russell Bryant <russell@digium.com>
-
- * configs/modem.conf.sample (removed), /, configs/enum.conf.sample,
- configs/extensions.ael.sample: Merged revisions 81379 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81379 | russell | 2007-08-30 10:33:48 -0500 (Thu, 30 Aug 2007) |
- 3 lines Fix a typo, update a reload command, and remove an unused
- configuration file. (closes issue #10606, casper) ........
-
-2007-08-30 15:24 +0000 [r81378] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_readexten.c (added): Add ReadExten app and VALID_EXTEN
- function (closes issue #10082)
-
-2007-08-30 14:54 +0000 [r81376] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /: Merged revisions 81373 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r81373 | crichter | 2007-08-30 16:43:33 +0200 (Do, 30
- Aug 2007) | 1 line Fixed some warnings. ........
-
-2007-08-30 14:42 +0000 [r81370-81372] Joshua Colp <jcolp@digium.com>
-
- * main/pbx.c, CHANGES: (closes issue #10603) Reported by: jmls
- Patches: pbx.diff uploaded by jmls (license 141) Add REASON
- dialplan variable for when an originated call fails and the
- failed extension is executed.
-
- * /, res/res_features.c: Merged revisions 81369 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81369 | file | 2007-08-30 11:23:40 -0300 (Thu, 30 Aug 2007) | 4
- lines (issue #10599) Reported by: dimas Handle the -1 control
- subclass during feature dialing (it indicates to stop sounds).
- ........
-
-2007-08-30 08:50 +0000 [r81368] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged
- revisions 81367 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81367 | crichter | 2007-08-30 10:31:59 +0200 (Do, 30 Aug 2007) |
- 11 lines Fixed a severe issue where a misdn_read would lock the
- channel, but read would not return because it blocks. later
- chan_misdn would try to queue a frame like a AST_CONTROL_ANSWER
- which could result in a deadlock situation. misdn_read will now
- not block forever anymore, and we don't queue the ANSWER frame at
- all when we already was called with misdn_answer -> answer would
- be called twice. Also we don't explicitly send a RELEASE_COMPLETE
- on receiption of a RELEASE anymore, because mISDN does that for
- us, this resulted in a problem on some switches, which would
- block our port after some calls for a short while. ........
-
-2007-08-29 22:05 +0000 [r81365] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Added the MEMBERREALTIME variable when using
- setinterfacevar in queues.conf
-
-2007-08-29 21:55 +0000 [r81364] Joshua Colp <jcolp@digium.com>
-
- * include/asterisk/event.h: Make the event header file work under
- C++.
-
-2007-08-29 21:30 +0000 [r81363] Steve Murphy <murf@digium.com>
-
- * main/config.c: init newer so compile won't complain.
-
-2007-08-29 21:25 +0000 [r81362] Russell Bryant <russell@digium.com>
-
- * main/config.c: make trunk build again. murf will have to review
- this to see if it was the right fix, as it is related to his last
- change.
-
-2007-08-29 20:55 +0000 [r81361] Steve Murphy <murf@digium.com>
-
- * res/res_config_pgsql.c, channels/chan_sip.c,
- include/asterisk/config.h, channels/chan_iax2.c,
- channels/iax2-parser.c, res/res_config_sqlite.c, main/config.c,
- main/channel.c, res/res_config_odbc.c, pbx/pbx_spool.c,
- main/manager.c, channels/chan_skinny.c, apps/app_minivm.c,
- main/http.c, utils/extconf.c, apps/app_directory.c,
- apps/app_parkandannounce.c, apps/app_voicemail.c: 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.
-
-2007-08-29 19:41 +0000 [r81353-81356] Russell Bryant <russell@digium.com>
-
- * main/event.c: Try to clarify the rules on changing ast_event and
- ast_event_ie
-
- * main/event.c: Fix parenthesis from my last commit
-
- * main/event.c: Change pointer aritmetic on void * to char *
-
- * main/event.c: there is not actually code that sends these over
- the network in trunk yet
-
-2007-08-29 16:39 +0000 [r81350] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 81349 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81349 | mmichelson | 2007-08-29 11:35:29 -0500 (Wed, 29 Aug
- 2007) | 12 lines This patch, in essence, will correctly pause a
- realtime queue member and reflect those changes in the realtime
- engine. (issue #10424, reported by irroot, patch by me) This
- patch creates a new function called update_realtime_member_field,
- which is a generic function which will allow any one field of a
- realtime queue member to be updated. This patch only uses this
- function to update the paused status of a queue member, but it
- lays the foundation for persisting the state of a realtime member
- the same way that static members' state is maintained when using
- the persistentmembers setting ........
-
-2007-08-29 16:25 +0000 [r81348] Joshua Colp <jcolp@digium.com>
-
- * main/event.c: Return ast_event_get_ie_raw to using an iterator
- and fix logic in ast_event_iterator_next.
-
-2007-08-29 16:09 +0000 [r81347] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 81346 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81346 | mmichelson | 2007-08-29 11:08:09 -0500 (Wed, 29 Aug
- 2007) | 3 lines Changed some tabs to spaces ........
-
-2007-08-29 16:07 +0000 [r81344-81345] Joshua Colp <jcolp@digium.com>
-
- * main/event.c: This concludes bringing trunk back to a working
- state.
-
- * include/asterisk/event.h, main/event.c: To keep others happy...
- revert part of my additions so trunk works.
-
-2007-08-29 15:59 +0000 [r81343] Russell Bryant <russell@digium.com>
-
- * /, main/Makefile: Merged revisions 81342 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81342 | russell | 2007-08-29 10:57:29 -0500 (Wed, 29 Aug 2007) |
- 3 lines If chan_h323 is not being built, don't use g++ to do the
- final link of Asterisk. (in response to a question on the
- asterisk-dev list) ........
-
-2007-08-29 15:57 +0000 [r81341] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 81340 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81340 | mmichelson | 2007-08-29 10:52:42 -0500 (Wed, 29 Aug
- 2007) | 8 lines This fix creates a more accurate way of detecting
- whether realtime members were deleted. (closes issue 10541,
- reported by Alric, patched by me) The REALLY nice things about
- this patch is that queue members now have a "realtime" field
- which will be true if the member is a realtime member. This means
- we can check this value prior to certain processing if it should
- ONLY be done for realtime members. ........
-
-2007-08-29 15:21 +0000 [r81335] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_iax2.c: Changed one too many variable settings in
- issue #9315 (closes issue #10592)
-
-2007-08-29 15:19 +0000 [r81334] Joshua Colp <jcolp@digium.com>
-
- * include/asterisk/event.h, include/asterisk/event_defs.h,
- main/event.c: Add API calls for iterating through an event. This
- should allow events to have multiple information elements (while
- there was nothing preventing it before you could not actually
- access any except the first one).
-
-2007-08-29 14:19 +0000 [r81333] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_meetme.c: Changing a NOTICE to a DEBUG. (closes issue
- #10591, reported and patched by junky, with small modification by
- me)
-
-2007-08-29 14:16 +0000 [r81326-81332] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 81331 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81331 | file | 2007-08-29 11:13:55 -0300 (Wed, 29 Aug 2007) | 4
- lines (closes issue #9690) Reported by: mattv Make rtp timeouts
- work even if two RTP streams are directly bridged in the RTP
- stack. ........
-
- * include/asterisk/utils.h: Add inline function for signed linear
- subtraction.
-
-2007-08-28 21:39 +0000 [r81292] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 81291 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81291 | russell | 2007-08-28 16:38:26 -0500 (Tue, 28 Aug 2007) |
- 3 lines Change the message about receiving a mini-frame before
- the first full voice frame to a DEBUG message. ........
-
-2007-08-28 21:35 +0000 [r81290] Joshua Colp <jcolp@digium.com>
-
- * main/logger.c: Add some read/write locking magic to make logger
- reload operate again.
-
-2007-08-28 20:03 +0000 [r81277] Tilghman Lesher <tlesher@digium.com>
-
- * main/logger.c, UPGRADE.txt, configs/logger.conf.sample: Support
- better rotation of log files to be more like system logging
- (closes issue #10398)
-
-2007-08-28 19:12 +0000 [r81227-81264] Russell Bryant <russell@digium.com>
-
- * include/asterisk/audiohook.h: Change the audiohook lock and
- unlock wrappers to macros instead of inline functions. As inline
- functions, the lock debug information will show that these are
- always locked in audiohooks.h instead of the file where the lock
- was actually acquired.
-
- * funcs/func_enum.c, pbx/pbx_dundi.c: Add proper channel locking
- around the uses of datastore_add and _find. There are still more
- places in the tree that I have not yet changed if someone wants
- to go through and find the places they are used without the
- channel locked.
-
- * main/channel.c, funcs/func_volume.c, include/asterisk/channel.h:
- * Constify the uid field of channel datastores * Convert some
- spaces to tabs in func_volume * Add a note in channel.h making it
- clear that none of the datastore API calls lock the channel they
- are given, so the channel should be locked before calling the
- functions that take a channel argument.
-
- * include/asterisk/app.h, main/app.c, CHANGES, main/asterisk.c,
- doc/tex/asterisk-conf.tex: (closes issue #7852) Reported by:
- nic_bellamy Patches:
- 2006-10-03_svn_44249_voicemail_lockmode_v3.patch uploaded by
- nic_bellamy (license 213) Add support for configurable file
- locking methods. The default is "lockfile", which is the old
- behavior. There is an additional option, "flock", which is
- intended for use in situations where the lockfile method will not
- work, such as with SMB/CIFS mounts.
-
- * /, configs/indications.conf.sample: Merged revisions 81226 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81226 | russell | 2007-08-28 10:41:15 -0500 (Tue, 28 Aug 2007) |
- 2 lines Add Russian tones. (closes issue #7953, hanabana)
- ........
-
-2007-08-28 14:37 +0000 [r81210] Joshua Colp <jcolp@digium.com>
-
- * res/res_features.c: (closes issue #10579) Reported by: ornati
- Make sure the called channel during the attended transfer process
- becomes associated with the calling channel so that the
- ast_waitfor_* call works properly under epoll.
-
-2007-08-28 14:12 +0000 [r81121-81190] Mark Michelson <mmichelson@digium.com>
-
- * /, contrib/scripts/vmail.cgi: Merged revisions 81189 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r81189 | mmichelson | 2007-08-28 09:12:14 -0500 (Tue, 28
- Aug 2007) | 5 lines Fixes a forwarding problem when using
- res_config_mysql (closes issue #10573, reported by chrisvaughan,
- patch suggested by chrisvaughan as well) ........
-
- * /, apps/app_queue.c: Merged revisions 81158 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81158 | mmichelson | 2007-08-27 17:40:19 -0500 (Mon, 27 Aug
- 2007) | 5 lines Resolve a potential deadlock. In this case, a
- single queue is locked, then the queue list. In changethread(),
- the queue list is locked, and then each individual queue is
- locked. Under the right circumstances, this could deadlock. As
- such, I have unlocked the individual queue before locking the
- queue list, and then locked the queue back after the queue list
- is unlocked. ........
-
- * /, channels/chan_agent.c: Merged revisions 81120 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r81120 | mmichelson | 2007-08-27 16:08:48 -0500 (Mon, 27
- Aug 2007) | 7 lines DTMF begin frames should be ignored so that
- when an agent acks a call with the '#' key, he doesn't cause a
- queue's announce file to be interrupted. Also went ahead and did
- the same for the '*' key and for ending a call. (closes issue
- #10528, reported by deskhack, patched by me) ........
-
-2007-08-27 20:55 +0000 [r81118] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_directed_pickup.c: Enhance Pickup to do native
- pickupgroup pickup when no arguments are specified (closes issue
- #10404)
-
-2007-08-27 17:44 +0000 [r81043-81098] Russell Bryant <russell@digium.com>
-
- * /, pbx/pbx_dundi.c: This should have been trunk only, I guess. oh
- well ... it's harmless. Merged revisions 81065 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81065 | russell | 2007-08-27 11:38:33 -0500 (Mon, 27 Aug 2007) |
- 1 line explicity define a variable as a boolean ........
-
- * /, pbx/pbx_dundi.c: Merged revisions 81074 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81074 | russell | 2007-08-27 12:27:48 -0500 (Mon, 27 Aug 2007) |
- 3 lines Add a \todo to note that this module leaks most of the
- memory it allocates on unload and should be fixed (when I'm not
- in the middle of something else ...). ........
-
- * /, res/res_musiconhold.c: Merged revisions 81042 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r81042 | russell | 2007-08-27 11:16:25 -0500 (Mon, 27
- Aug 2007) | 11 lines (closes issue #10419) Reported by:
- mustardman Patches: asterisk-mohposition.diff.txt uploaded by
- jamesgolovich (license 176) This patch fixes a few problems with
- music on hold. * Fix issues with starting at the beginning of a
- file when it shouldn't. * Fix the inuse counter to be decremented
- even if the class had not been set to be deleted when not in use
- anymore * Don't arbitrarily limit the number of MOH files to 255
- ........
-
-2007-08-27 15:03 +0000 [r81013] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 81012 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81012 | file | 2007-08-27 12:01:59 -0300 (Mon, 27 Aug 2007) | 6
- lines (closes issue #10561) Reported by: jesselang Patches:
- chan_sip-ChannelReload-20080825.patch uploaded by jesselang
- (license 202) Remove an extra \r\n to make the ChannelReload
- event conform with every other event. ........
-
-2007-08-27 14:56 +0000 [r81011] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 81010 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r81010 | mmichelson | 2007-08-27 09:55:44 -0500 (Mon, 27 Aug
- 2007) | 3 lines Found a case where the queue's membercount is
- off. It does not take into account dynamic members on a reload.
- ........
-
-2007-08-27 13:35 +0000 [r80962-80991] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Remove places that say if no language is
- specified it will default to english... since on some setups this
- is untrue.
-
- * /, main/rtp.c: Merged revisions 80974 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r80974 | file | 2007-08-27 10:20:31 -0300 (Mon, 27 Aug 2007) | 4
- lines (closes issue #10562) Reported by: idkpmiller Correct
- jitter value output in the CLI to be as expected. ........
-
- * configs/sip.conf.sample: (closes issue #10569) Reported by: IgorG
- Patches: sip_conf-80933-1.patch uploaded by IgorG (license 20)
- Fix up sip.conf sample configuration.
-
-2007-08-26 18:12 +0000 [r80933] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 80932 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r80932 | russell | 2007-08-26 13:11:26 -0500 (Sun, 26 Aug 2007) |
- 3 lines Remove an extra signal_condition() for the scheduler
- thread. (closes issue #10564, patch from casper) ........
-
-2007-08-25 17:55 +0000 [r80821-80898] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 80895 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r80895 | russell | 2007-08-25 12:37:39 -0500 (Sat, 25 Aug 2007) |
- 7 lines Fix some issues with the handling of the scheduler in
- chan_iax2. Most of the places that scheduled items to be executed
- by the scheduler thread did not signal the scheduler thread to
- wake up so that it could recalculate the time until the next
- action. These changes will make the scheduler thread more
- responsive and ensure that actions get executed as close to when
- intended as possible instead of it being possible for very long
- delays. ........
-
- * pbx/pbx_dundi.c: localize a variable and remove a duplicate error
- message
-
- * apps/app_queue.c: use ast_strlen_zero
-
- * /, channels/chan_iax2.c: Merged revisions 80849 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r80849 | russell | 2007-08-24 16:22:50 -0500 (Fri, 24 Aug 2007) |
- 5 lines If dnsmgr is in use, and no DNS servers are available
- when Asterisk first starts, then don't give up on poking peers.
- Allow the poke to get rescheduled so that it will work once the
- dnsmgr is able to resolve the host. (closes issue #10521, patch
- by jamesgolovich) ........
-
- * /, main/dsp.c: Merged revisions 80820 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r80820 | russell | 2007-08-24 15:24:05 -0500 (Fri, 24 Aug 2007) |
- 7 lines Improve the debouncing logic in the DTMF detector to fix
- some reliability issues. Previously, this code used a shift
- register of hits and non-hits. However, if the start of the digit
- isn't clean, it is possible for the leading edge detector to miss
- the digit. These changes replace the flawed shift register logic
- and also does the debouncing on the trailing edge as well.
- (closes issue #10535, many thanks to softins for the patch)
- ........
-
-2007-08-24 20:21 +0000 [r80819] BJ Weschke <bweschke@btwtech.com>
-
- * apps/app_queue.c: Merged revisions 80818 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r80818 | bweschke | 2007-08-24 15:52:06 -0400 (Fri, 24 Aug 2007)
- | 3 lines A minor correction to the available logic of autofill.
- If a queue member is paused, they're not really "available" so
- don't count them as such. Somewhat related to issue #10155
- ........
-
-2007-08-24 19:50 +0000 [r80817] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c: Fix documentation for Set (closes issue #10549)
-
-2007-08-24 19:03 +0000 [r80790] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, /: Merged revisions 80789 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r80789 | murf | 2007-08-24 12:52:15 -0600 (Fri, 24 Aug 2007) | 1
- line From a complaint by jmls, I realize that the message in
- cdr_disposition is unnecessary. To get failure disposition, just
- return -1; no use having more than one case do that. ........
-
-2007-08-24 18:05 +0000 [r80778] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Add VMWI chan_zap support #9909
-
-2007-08-24 15:53 +0000 [r80751] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 80750 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r80750 | mmichelson | 2007-08-24 10:51:03 -0500 (Fri, 24 Aug
- 2007) | 3 lines Fix a possible crash in IMAP voicemail. ........
-
-2007-08-24 15:42 +0000 [r80748] Steve Murphy <murf@digium.com>
-
- * utils/conf2ael.c: fix up the MODULEINFO in conf2ael.c as well
-
-2007-08-24 15:29 +0000 [r80725] Russell Bryant <russell@digium.com>
-
- * /, utils/ael_main.c: Merged revisions 80722 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r80722 | russell | 2007-08-24 10:28:05 -0500 (Fri, 24 Aug 2007) |
- 3 lines Tweak the formatting of this MODULEINFO block. I think
- this would have caused a "*" to get in the menuselect-tree file.
- ........
-
-2007-08-24 14:55 +0000 [r80690-80718] Steve Murphy <murf@digium.com>
-
- * /, utils/ael_main.c, utils/conf2ael.c: Merged revisions 80717 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r80717 | murf | 2007-08-24 08:48:49 -0600 (Fri, 24 Aug 2007) | 1
- line This change addresses JerJer's complaint that aelparse
- builds and installs even if pbx_ael is unchecked in the
- menuselect stuff. ........
-
-2007-08-24 11:49 +0000 [r80662] Philippe Sultan <philippe.sultan@gmail.com>
-
- * /, channels/chan_gtalk.c: Merged revisions 80661 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r80661 | phsultan | 2007-08-24 13:42:46 +0200 (Fri, 24
- Aug 2007) | 9 lines Closes issue #10509 Googletalk calls are
- answered too early, which results in CDRs wrongly stating that a
- call was ANSWERED when the calling party cancelled a call before
- before being established. We must not answer the call upon
- reception of a 'transport-accept' iq packet, but this packet
- still needs to be acknowledged, otherwise the remote peer would
- close the call (like in #8970). ........
-
-2007-08-23 23:37 +0000 [r80649] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael-test/ref.ael-ntest10, res/ael/ael.tab.c,
- res/ael/ael.y, res/ael/ael.tab.h, pbx/ael/ael-test/ref.ael-test6,
- pbx/ael/ael-test/ref.ael-test7: an unreported crash I debugged,
- looked like it was backing up way too far after hitting the
- syntax error. An inspection of the code revealed that error
- tokens in lists were not rearranged when the rules were
- rearranged as part of a code neatening-up process. By moving the
- error tokens to where they should be, I also reduced the number
- of shift/reduce conflicts to 3 instead of 8. This introduces
- subtle differences in error messages, so the regressions had to
- be updated.
-
-2007-08-23 21:34 +0000 [r80510-80616] Russell Bryant <russell@digium.com>
-
- * apps/app_while.c: Use the comma separator in app_while. reported
- by blitzrage on irc, patched by me
-
- * /, res/res_features.c, include/asterisk/features.h: Merged
- revisions 80573 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r80573 | russell | 2007-08-23 15:16:41 -0500 (Thu, 23 Aug 2007) |
- 5 lines When executing a dynamic feature, don't look it up a
- second time by digit pattern after we already looked it up by
- name. This causes broken behavior if there is more than one
- feature defined with the same digit pattern. (closes issue
- #10539, reported by bungalow, patch by me) ........
-
- * /, funcs/func_timeout.c: Merged revisions 80547 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r80547 | russell | 2007-08-23 14:29:44 -0500 (Thu, 23 Aug 2007) |
- 3 lines Revert very broken fix for issue #10540 ... none of these
- values take ms so I don't know what I was thinking ........
-
- * /, funcs/func_timeout.c: Merged revisions 80539 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r80539 | russell | 2007-08-23 14:21:53 -0500 (Thu, 23 Aug 2007) |
- 4 lines Fix func_timeout to take values in floating point so 1.5
- actually means 1.5 seconds instead of being rounded. (closes
- issue #10540, reported by spendergrass, patch by me) ........
-
- * doc/asterisk-mib.txt, res/snmp/agent.c: Fix a typo in the
- Asterisk MIB and fix astNumChanBridged so it acts as a counter
- again (closes issue #10118, patch by jeffg)
-
-2007-08-23 17:18 +0000 [r80508] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 80501 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r80501 | kpfleming | 2007-08-23 12:08:25 -0500 (Thu, 23 Aug 2007)
- | 2 lines report the actual channel number that was unregistered,
- instead of assuming that the interface list consists of channels
- 1 through <x> with no gaps in the sequence ........
-
-2007-08-23 17:04 +0000 [r80470-80500] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 80499 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r80499 | russell | 2007-08-23 12:02:50 -0500 (Thu, 23 Aug 2007) |
- 3 lines Fix some code where it was possible for a reference to a
- peer to not get released when it should. Thank you to Marta
- Carbone for pointing this out! ........
-
- * /, res/res_agi.c: Merged revisions 80469 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r80469 | russell | 2007-08-23 10:49:28 -0500 (Thu, 23 Aug 2007) |
- 2 lines Revert res_agi fix that didn't quite work until we get it
- right ... ........
-
-2007-08-23 15:48 +0000 [r80453-80468] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: If no default language has been specified
- print out that it will default to english when using sip show
- peer or sip show user.
-
- * main/minimime/mm.h: Return trunk to a working state by including
- compat.h in minimime.
-
-2007-08-22 23:26 +0000 [r80428-80429] Jason Parker <jparker@digium.com>
-
- * main/minimime/mm_util.c, main/minimime/mm_codecs.c,
- main/minimime/mm_mem.h, main/minimime/mm_base64.c,
- main/minimime/mm.h: Convert minimime to use the proper uint*_t
- types, rather than u_int*_t
-
- * apps/app_minivm.c: Cast calls to getpid. This was done in 1.4
- already, this one was just new
-
-2007-08-22 22:54 +0000 [r80361-80427] Russell Bryant <russell@digium.com>
-
- * /, include/asterisk/astobj2.h: Merged revisions 80426 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r80426 | russell | 2007-08-22 17:54:03 -0500 (Wed, 22 Aug 2007) |
- 6 lines Add some more documentation on iterating ao2 containers.
- The documentation implies that is possible to miss an object or
- see an object twice while iterating. After looking through the
- code and talking with mmichelson, I have documented the exact
- conditions under which this can happen (which are rare and
- harmless in most cases). ........
-
- * /, main/astobj2.c: Merged revisions 80424 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r80424 | russell | 2007-08-22 17:40:27 -0500 (Wed, 22 Aug 2007) |
- 10 lines When converting this code to use the list macros, I
- changed it so objects are added to the head of a bucket instead
- of the tail. However, while looking over code with mmichelson, we
- noticed that the algorithm used in ao2_iterator_next requires
- that items are added to the tail. This wouldn't have caused any
- huge problem, but it wasn't correct. It meant that if an object
- was added to a container while you were iterating it, and it was
- added to the same bucket that the current element is in, then the
- new object would be returned by ao2_iterator_next, and any other
- objects in the bucket would be bypassed in the traversal.
- ........
-
- * channels/chan_iax2.c: allow peers and users to go into a hash
- table
-
- * /, channels/chan_sip.c: Merged revisions 80390 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r80390 | russell | 2007-08-22 16:00:44 -0500 (Wed, 22 Aug 2007) |
- 3 lines Don't crash when using realtime in chan_sip without an
- insecure setting in the database. (closes issue #10348, reported
- by link55, fixed by me) ........
-
- * channels/chan_iax2.c: Unsubscribe from MWI events in the peer
- destructor
-
- * /, main/Makefile, include/asterisk/astobj2.h (added),
- include/asterisk/strings.h, channels/chan_iax2.c, main/astobj2.c
- (added): Merged revisions 80362 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r80362 | russell | 2007-08-22 15:21:36 -0500 (Wed, 22 Aug 2007) |
- 34 lines Merge changes from team/russell/iax_refcount. This set
- of changes fixes problems with the handling of iax2_user and
- iax2_peer objects. It was very possible for a thread to still
- hold a reference to one of these objects while a reload operation
- tries to delete them. The fix here is to ensure that all
- references to these objects are tracked so that they can't go
- away while still in use. To accomplish this, I used the astobj2
- reference counted object model. This code has been in one of
- Luigi Rizzo's branches for a long time and was primarily
- developed by one of his students, Marta Carbone. I wanted to go
- ahead and bring this in to 1.4 because there are other problems
- similar to the ones fixed by these changes, so we might as well
- go ahead and use the new astobj if we're going to go through all
- of the work necessary to fix the problems. As a nice side benefit
- of these changes, peer and user handling got more efficient.
- Using astobj2 lets us not hold the container lock for peers or
- users nearly as long while iterating. Also, by changing a define
- at the top of chan_iax2.c, the objects will be distributed in a
- hash table, drastically increasing lookup speed in these
- containers, which will have a very big impact on systems that
- have a large number of users or peers. The use of the hash table
- will be made the default in trunk. It is not the default in 1.4
- because it changes the behavior slightly. Previously, since peers
- and users were stored in memory in the same order they were
- specified in the configuration file, you could influence peer and
- user matching order based on the order they are specified in the
- configuration. The hash table does not guarantee any order in the
- container, so this behavior will be going away. It just means
- that you have to be a little more careful ensuring that peers and
- users are matched explicitly and not forcing chan_iax2 to have to
- guess which user is the right one based on secret, host, and
- access list settings, instead of simply using the username. If
- you have any questions, feel free to ask on the asterisk-dev
- list. ........
-
- * /, res/res_agi.c: Merged revisions 80360 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r80360 | russell | 2007-08-22 14:53:30 -0500 (Wed, 22 Aug 2007) |
- 5 lines Juggie in #asterisk-dev was reporting problems where
- fgets would return without reading the whole line when using
- fastagi. When this happens, errno was set to EINTR or EAGAIN.
- This patch accounts for the possibility and lets fgets continue
- in that case. ........
-
-2007-08-22 18:54 +0000 [r80303-80331] Jason Parker <jparker@digium.com>
-
- * Makefile, build_tools/mkpkgconfig, /, build_tools/make_build_h,
- build_tools/strip_nonapi, build_tools/prep_moduledeps,
- build_tools/make_buildopts_h: Merged revisions 80330 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r80330 | qwell | 2007-08-22 13:53:18 -0500 (Wed, 22 Aug
- 2007) | 7 lines Fix a few build issues in Solaris (and likely
- others). Use GREP and ID variables from autoconf. Reported to me
- in #asterisk-dev I forgot who reported this - sorry. :( ........
-
- * Makefile, /: Merged revisions 80304 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r80304 | qwell | 2007-08-22 13:25:34 -0500 (Wed, 22 Aug 2007) | 2
- lines Change a syntax that the GNU make in Solaris dislikes.
- ........
-
- * /, build_tools/make_version: Merged revisions 80302 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r80302 | qwell | 2007-08-22 13:06:00 -0500 (Wed, 22 Aug
- 2007) | 3 lines Fix a bashism (we explicitly request /bin/sh).
- Remove some oddly placed quotes I found in passing. ........
-
-2007-08-22 16:27 +0000 [r80258-80262] Russell Bryant <russell@digium.com>
-
- * utils/check_expr.c: Ensure that the object code for
- ast_atomic_fetchadd_int() gets included in the check_expr binary
- when building with LOW_MEMORY defined. (reported by Brian Capouch
- on the asterisk-dev list, patch by me)
-
- * Makefile, /: Merged revisions 80257 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r80257 | russell | 2007-08-22 11:21:58 -0500 (Wed, 22 Aug 2007) |
- 4 lines Honor the contents of the COPTS variable as custom target
- CFLAGS. Apparently this is what openwrt does. (reported by Brian
- Capouch on the asterisk-dev list, patch by me) ........
-
-2007-08-22 16:16 +0000 [r80256] Joshua Colp <jcolp@digium.com>
-
- * /, main/rtp.c: Merged revisions 80255 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r80255 | file | 2007-08-22 13:14:38 -0300 (Wed, 22 Aug 2007) | 4
- lines (closes issue #10526) Reported by: sinistermidget Revert
- commit from issue #10355 and return timestamp skew to 640.
- ........
-
-2007-08-22 14:17 +0000 [r80241-80242] Steve Murphy <murf@digium.com>
-
- * /: blocking 80167
-
- * /, main/alaw.c: Merged revisions 80166 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r80166 | murf | 2007-08-21 10:36:34 -0600 (Tue, 21 Aug 2007) | 1
- line This patch solves problem 1 in 8126; it should not slow down
- the alaw codec, but should prevent signal degradation via
- multiple trips thru the codec. Fossil estimates the twice thru
- this codec will prevent fax from working. 4-6 times thru would
- result hearable, noticeable, voice degradation. ........
-
-2007-08-21 21:58 +0000 [r80226] Russell Bryant <russell@digium.com>
-
- * funcs/func_odbc.c: use ast_atomic_fetchadd_int for incrementing
- resultcount
-
-2007-08-21 20:55 +0000 [r80217] Steve Murphy <murf@digium.com>
-
- * res/ael/pval.c: As per 10472, mvanbaak thought the generated code
- would look better this way.
-
-2007-08-21 18:49 +0000 [r80184] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 80183 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r80183 | russell | 2007-08-21 13:42:15 -0500 (Tue, 21 Aug 2007) |
- 7 lines Don't record SIP dialog history if it's not turned on.
- Also, put an upper limit on how many history entires will be
- stored for each SIP dialog. It is currently set to 50, but can be
- increased if deemed necessary. (closes issue #10421, closes issue
- #10418, patches suggested by jmoldenhauer, patches updated by me)
- (Security implications documented in AST-2007-020) ........
-
-2007-08-21 15:51 +0000 [r80157] Joshua Colp <jcolp@digium.com>
-
- * main/audiohook.c: Minor tweak. Don't manipulate volume of the
- audio in the buffer if no audio is actually there.
-
-2007-08-21 15:23 +0000 [r80133] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_mgcp.c: Merged revisions 80132 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r80132 | russell | 2007-08-21 10:22:22 -0500 (Tue, 21 Aug 2007) |
- 3 lines Don't try to dereference the owner channel when it may
- not exist (issue #10507, maxper) ........
-
-2007-08-21 15:04 +0000 [r80131] Jason Parker <jparker@digium.com>
-
- * /, configs/cdr.conf.sample: Merged revisions 80130 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r80130 | qwell | 2007-08-21 10:03:45 -0500 (Tue, 21 Aug
- 2007) | 7 lines (closes issue #10510) Reported by: casper
- Patches: cdr.conf.diff uploaded by casper (license 55) Fix a few
- errors in sample cdr config file. ........
-
-2007-08-20 22:53 +0000 [r80113] Steve Murphy <murf@digium.com>
-
- * build_tools/cflags.xml, main/ulaw.c, codecs/slin_ulaw_ex.h,
- codecs/ulaw_slin_ex.h, include/asterisk/alaw.h, main/translate.c,
- include/asterisk/ulaw.h, main/alaw.c: This change set fixes bug
- 8126 in trunk. It is implemented via compile time options,
- activated via the menuselect stuff, which defaults to the old
- way. non-zero sample data added. Translate tables expressed in
- microseconds instead of milliseconds, with 5-digit data now
- instead of 3, giving 2 more digits of precision.
-
-2007-08-20 17:37 +0000 [r80075] Steve Murphy <murf@digium.com>
-
- * include/asterisk/lock.h, utils/extconf.c: Stephn Davies reports
- that this will help make things work on 64-bit machines
-
-2007-08-20 16:18 +0000 [r80050] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 80049 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r80049 | mmichelson | 2007-08-20 11:17:43 -0500 (Mon, 20 Aug
- 2007) | 4 lines Found a pointless ternary if. member->dynamic was
- set to 1 and has no opportunity to change between then and this
- line, so "dynamic" will ALWAYS be output. ........
-
-2007-08-20 16:12 +0000 [r80048] Jason Parker <jparker@digium.com>
-
- * /, configs/extensions.conf.sample: Merged revisions 80047 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r80047 | qwell | 2007-08-20 11:08:49 -0500 (Mon, 20 Aug 2007) | 7
- lines (closes issue #10499) Reported by: casper Patches:
- extensions.conf.sample.diff uploaded by casper (license 55)
- Update CLI examples in extensions.conf.sample to reflect command
- changes. ........
-
-2007-08-20 15:53 +0000 [r80046] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c: Remove remnants of last commit so trunk
- builds again.
-
-2007-08-20 15:37 +0000 [r80045] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 80044 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r80044 | mmichelson | 2007-08-20 10:34:43 -0500 (Mon, 20 Aug
- 2007) | 5 lines Ukrainian language voicemail support. (closes
- issue #10458, reported and patched by Oleh) ........
-
-2007-08-20 15:27 +0000 [r80037] Steve Murphy <murf@digium.com>
-
- * utils/pval.c (removed): pval.c should not be in svn, in the utils
- dir
-
-2007-08-20 15:10 +0000 [r80023-80033] Joshua Colp <jcolp@digium.com>
-
- * utils/pval.c: Bring pval.c in utils up to date with pval.c in
- res/ael.
-
- * channels/chan_zap.c: Fix random segfault issue when loading
- chan_zap. Trying to access a configuration structure that has
- already been destroyed is bad, mmmk?
-
-2007-08-20 02:46 +0000 [r79999] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 79998 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r79998 | tilghman | 2007-08-19 21:42:49 -0500 (Sun, 19 Aug 2007)
- | 2 lines Missing curly braces. Oops. (Reported by snuffy via
- IRC) ........
-
-2007-08-20 00:54 +0000 [r79988-79990] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: (closes issue #10495) Reported by:
- stevedavies Make sure context pointer is valid or else chan_iax2
- will go kaboom.
-
- * utils/Makefile: (closes issue #10496) Reported by: caio1982 Fix
- building on OSX.
-
- * channels/chan_h323.c: Fix building of trunk. I'm doing work on a
- Sunday night just to avoid watching Snakes on a Plane which my
- roommate is watching.
-
-2007-08-19 14:17 +0000 [r79980] Tilghman Lesher <tlesher@digium.com>
-
- * utils/Makefile: Add strcompat dependency for check_expr (needed
- for platforms that don't have strndup)
-
-2007-08-18 23:58 +0000 [r79972] Joshua Colp <jcolp@digium.com>
-
- * configure, configure.ac: Actually check the return value of
- epoll_create to make sure it works.
-
-2007-08-18 14:34 +0000 [r79940-79949] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 79947 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r79947 | tilghman | 2007-08-18 09:30:44 -0500 (Sat, 18 Aug 2007)
- | 3 lines Don't allocate vmu for messagecount when we could just
- use the stack instead (closes issue #10490) Also, remove a
- useless (and leaky) SQLAllocHandle (closes issue #10480) ........
-
- * channels/chan_zap.c, channels/chan_sip.c, channels/chan_h323.c,
- channels/chan_iax2.c: We weren't properly encapsulating the mtime
- ignores of config files (closes issue #10488)
-
-2007-08-17 21:19 +0000 [r79915] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: I broke the build. Now I'm fixing it.
-
-2007-08-17 21:04 +0000 [r79913] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 79912 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r79912 | russell | 2007-08-17 16:01:43 -0500 (Fri, 17 Aug 2007) |
- 4 lines Avoid a crash in the handling of DTMF based Caller ID. It
- is valid for ast_read to return NULL in the case that the channel
- has been hung up. (crash reported by anonymouz666 on IRC in
- #asterisk-dev) ........
-
-2007-08-17 19:16 +0000 [r79907] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 79906 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r79906 | mmichelson | 2007-08-17 14:14:05 -0500 (Fri, 17 Aug
- 2007) | 6 lines Patch allows for more seamless transition from
- file storage voicemail to ODBC storage voicemail. If a retrieval
- of a greeting from the database fails, but the file is found on
- the file system, then we go ahead an insert the greeting into the
- database. The result of this is that people who switch from file
- storage to ODBC storage do not need to rerecord their voicemail
- greetings. ........
-
-2007-08-17 19:13 +0000 [r79903-79905] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_sip.c, main/utils.c, include/asterisk/strings.h:
- Merged revisions 79904 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #10430) ........ r79904 | qwell | 2007-08-17 14:12:19 -0500
- (Fri, 17 Aug 2007) | 11 lines Don't send a semicolon over the
- wire in sip notify messages. Caused by fix for issue 9938. I
- basically took the code that existed before 9938 was fixed, and
- copied it into a new function - ast_unescape_semicolon There
- should be very few places this will be needed (pbx_config does
- NOT need this (see issue 9938 for details)) Issue 10430, patch by
- me, with help/ideas from murf (thanks murf). ........
-
- * channels/chan_local.c, /: Merged revisions 79902 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #10485) ........ r79902 | qwell | 2007-08-17 12:44:22 -0500
- (Fri, 17 Aug 2007) | 4 lines Re-add the setting of callerid name
- and number. Issue 10485, reported by and fix explained by
- paradise. ........
-
-2007-08-17 16:39 +0000 [r79901] Tilghman Lesher <tlesher@digium.com>
-
- * configs/logger.conf.sample: Documentation for %q in logger.conf,
- as suggested by jtodd (closes issue #10475)
-
-2007-08-17 16:04 +0000 [r79888-79894] Jason Parker <jparker@digium.com>
-
- * res/res_features.c: Fix Dial arguments in res_features. Closes
- issue #10484, patch by lunn.
-
- * pbx/pbx_dundi.c: Correct the argument separator for a Dial
- statement in pbx_dundi. Closes issue #10483, patch by lunn
-
-2007-08-17 14:41 +0000 [r79885] Tilghman Lesher <tlesher@digium.com>
-
- * main/config.c: Change this flag... might not otherwise unlock in
- an OOM situation
-
-2007-08-17 14:14 +0000 [r79861-79862] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Make use of ast_sched_replace() in some
- places in chan_iax2
-
- * channels/chan_sip.c, include/asterisk/sched.h, main/sched.c: This
- commit adds a scheduler API call, ast_sched_replace that can be
- used in place of a very common construct. I also used it in a
- number of places in chan_sip. if (id > -1) ast_sched_del(sched,
- id); id = ast_sched_add(sched, ...); changes to:
- ast_sched_replace(id, sched, ...);
-
-2007-08-17 13:45 +0000 [r79859-79860] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_config_odbc.c, res/res_config_sqlite.c: store and destroy
- implementations for sqlite (closes issue #10446) and odbc (closes
- issue #10447)
-
- * res/res_config_pgsql.c, funcs/func_lock.c: store and destroy
- implementations for realtime pgsql (closes issue #10372)
-
-2007-08-17 13:39 +0000 [r79858] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 79857 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r79857 | russell | 2007-08-17 08:37:08 -0500 (Fri, 17 Aug 2007) |
- 5 lines Fix some crashes in chan_sip. This patch changes various
- places that add items to the scheduler to ensure that they don't
- overwrite the ID of a previously scheduled item. If there is one,
- it should be removed. (closes issue #10391, closes issue #10256,
- probably others, patch by me) ........
-
-2007-08-17 08:29 +0000 [r79841] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /: Merged revisions 79833 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r79833 | crichter | 2007-08-17 10:22:36 +0200 (Fr, 17
- Aug 2007) | 1 line sometimes we don't need to signal dtmf tones
- to asterisk, we just want them to go through as inband. Otherwise
- they might be generated by the other channel partner and then
- there is a double tone. ........
-
-2007-08-17 01:19 +0000 [r79824] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_zap.c: Fix building of chan_zap under development
- mode without libpri and libss7 installed.
-
-2007-08-16 23:31 +0000 [r79813] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_lock.c: Revise dialplan locks to permit multiple locks
- per channel, but with deadlock avoidance
-
-2007-08-16 22:33 +0000 [r79764-79794] Russell Bryant <russell@digium.com>
-
- * /: Merged revisions 79792 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r79792 | russell | 2007-08-16 17:32:33 -0500 (Thu, 16 Aug 2007) |
- 4 lines Fix a little race condition that could cause a crash if
- two channels had MOH stopped at the same time that were using a
- class that had been marked for deletion when its use count hits
- zero. ........
-
- * /, res/res_musiconhold.c: Merged revisions 79778 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r79778 | russell | 2007-08-16 17:24:25 -0500 (Thu, 16
- Aug 2007) | 14 lines This patch fixes a bug where reloading the
- module with "module reload" did not delete classes from memory
- that were no longer in the config. This patch fixes that problem
- as well as another one. Previously, if you reloaded MOH using the
- "moh reload" CLI command, which behaved differently than "module
- reload ...", MOH had to be stopped on every channel and started
- again immediately. However, there was no way to tell what class
- was being used, so they would all fall back to the default class.
- (closes issue #10139) Reported by: blitzrage Patches:
- asterisk-10139-advanced.diff.txt uploaded by jamesgolovich
- (license 176) Tested by: jamesgolovich ........
-
- * /, channels/chan_iax2.c: Merged revisions 79756 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r79756 | russell | 2007-08-16 16:29:24 -0500 (Thu, 16 Aug 2007) |
- 11 lines Fix more deadlocks in chan_iax2 that were introduced by
- making frame handling and scheduling multi-threaded.
- Unfortunately, we have to do some expensive deadlock avoidance
- when queueing frames on to the ast_channel owner of the IAX2 pvt
- struct. This was already handled for regular frames, but
- ast_queue_hangup and ast_queue_control were still used directly.
- Making these changes introduced even more places where the IAX2
- pvt struct can disappear in the context of a function holding its
- lock due to calling a function that has to unlock/lock it to
- avoid deadlocks. I went through and fixed all of these places to
- account for this possibility. (issue #10362, patch by me)
- ........
-
-2007-08-16 21:28 +0000 [r79755] Joshua Colp <jcolp@digium.com>
-
- * /: Fix properties on trunk again.
-
-2007-08-16 21:21 +0000 [r79749] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 79748 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r79748 | mmichelson | 2007-08-16 16:16:40 -0500 (Thu, 16
- Aug 2007) | 8 lines Fixes a problem where agents would get stuck
- busy due to their wrapuptime being longer than the queue's
- wrapuptime and ringinuse=no for the queue. (closes issue #10215,
- reported by Doug, repaired by me) Special thanks to fkasumovic
- for pointing out the source of the problem and to bweschke for
- helping to come up with a solution! ........
-
-2007-08-16 21:09 +0000 [r79747] Tilghman Lesher <tlesher@digium.com>
-
- * main/udptl.c, cdr/cdr_sqlite3_custom.c, /, res/res_features.c,
- codecs/codec_adpcm.c, apps/app_alarmreceiver.c,
- cdr/cdr_adaptive_odbc.c, channels/chan_iax2.c, main/config.c,
- main/loader.c, res/res_smdi.c, channels/chan_skinny.c,
- main/http.c, apps/app_amd.c, channels/chan_alsa.c,
- cdr/cdr_odbc.c, cdr/cdr_manager.c, codecs/codec_g722.c,
- apps/app_privacy.c, codecs/codec_speex.c, channels/chan_agent.c,
- codecs/codec_g726.c, channels/iax2-provision.c,
- apps/app_playback.c, channels/iax2-provision.h,
- channels/chan_misdn.c, res/res_indications.c, pbx/pbx_config.c,
- main/asterisk.c, res/res_odbc.c, apps/app_voicemail.c,
- channels/chan_vpb.cc, res/res_snmp.c, apps/app_meetme.c,
- codecs/codec_gsm.c, res/res_musiconhold.c, channels/chan_gtalk.c,
- cdr/cdr_pgsql.c, apps/app_followme.c, res/res_jabber.c,
- cdr/cdr_radius.c, codecs/codec_zap.c, res/res_config_sqlite.c,
- main/enum.c, channels/misdn_config.c, cdr/cdr_csv.c, main/cdr.c,
- channels/chan_phone.c, res/res_config_odbc.c, main/manager.c,
- apps/app_osplookup.c, funcs/func_odbc.c, apps/app_minivm.c,
- main/logger.c, apps/app_directory.c, apps/app_rpt.c,
- cdr/cdr_custom.c, channels/chan_mgcp.c, codecs/codec_lpc10.c,
- res/res_config_pgsql.c, main/dnsmgr.c, channels/chan_zap.c,
- channels/chan_sip.c, apps/app_festival.c, codecs/codec_alaw.c,
- res/res_adsi.c, include/asterisk/config.h, apps/app_queue.c,
- channels/chan_oss.c, main/rtp.c, cdr/cdr_tds.c,
- channels/chan_jingle.c, channels/misdn/chan_misdn_config.h,
- channels/chan_h323.c, pbx/pbx_dundi.c, codecs/codec_ulaw.c: Don't
- reload a configuration file if nothing has changed.
-
-2007-08-16 19:40 +0000 [r79736] Steve Murphy <murf@digium.com>
-
- * utils/pval.c, utils/conf2ael.c: Many thanks to mvanbaak for his
- update to translate hints; I added the -d option for local
- testing purposes. This is from bug 10472
-
-2007-08-16 18:23 +0000 [r79724-79725] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * channels/chan_iax2.c: added counter for iax2 show registry CLI
- output, closes issue 10461, thanks junky
-
- * apps/app_voicemail.c: added counter for voicemail show users,
- issue 10462, thanks junky
-
-2007-08-16 17:34 +0000 [r79714-79719] Steve Murphy <murf@digium.com>
-
- * utils/conf2ael.c: mvanbaak asks: why did you include that twice?
- Answer: dunno. removed redundant include
-
- * utils/extconf.c, utils/conf2ael.c: svn did me dirty for some
- reason. Left 5 files out of the commit; Tilghman copied them in
- from the branch, but I had made changes to these. Here they are.
-
-2007-08-16 15:59 +0000 [r79691] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 79690 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r79690 | mmichelson | 2007-08-16 10:58:34 -0500 (Thu, 16 Aug
- 2007) | 5 lines base_encode is not trying to open a log file, so
- we should not call it a log file in the warning. (related to
- issue #10452, reported by bcnit) ........
-
-2007-08-16 15:29 +0000 [r79687-79688] Joshua Colp <jcolp@digium.com>
-
- * pbx/pbx_dundi.c: (closes issue #10467) Reported by: lunn Patches:
- pbx_dundi.diff uploaded by lunn (license 179) Don't print a
- warning saying an ethernet interface was found when it indeed
- was.
-
- * utils/conf2ael.c: Make conf2ael build on 64-bit systems.
-
-2007-08-16 09:45 +0000 [r79666] Philippe Sultan <philippe.sultan@gmail.com>
-
- * /, res/res_jabber.c: Merged revisions 79665 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r79665 | phsultan | 2007-08-16 11:37:10 +0200 (Thu, 16 Aug 2007)
- | 21 lines A fix for two critical problems detected while working
- with Daniel McKeehan in issue #10184. Upon priority change, the
- resource list is not NULL terminated when moving an item to the
- end of the list. This makes Asterisk endlessy loop whenever it
- needs to read the list. Jids with different resource and priority
- values, like in Gmail's and GoogleTalk's jabber clients put that
- problem in evidence. Upon reception of a 'from' attribute with an
- empty resource string, Asterisk crashes when trying to access the
- found->cap pointer if the resource list for the given buddy is
- not empty. This situation is perfectly valid and must be handled.
- The Gizmoproject's jabber client put that problem in evidence.
- Also added a few comments in the code as well as a handle for the
- capabilities from Gmail's jabber client, which are stored in a
- caps:c tag rather than the usual c tag. Closes issue #10184.
- ........
-
-2007-08-16 09:22 +0000 [r79660] Christian Richter <christian.richter@beronet.com>
-
- * /, channels/misdn/ie.c: Merged revisions 79642 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r79642 | crichter | 2007-08-16 10:21:21 +0200 (Do, 16 Aug 2007) |
- 1 line 0x80 + protocol is wrong for USERUSER when we want to send
- IA5 Chars. ........
-
-2007-08-16 06:52 +0000 [r79638] Olle Johansson <oej@edvina.net>
-
- * CHANGES: Doc change
-
-2007-08-15 22:53 +0000 [r79634] Jason Parker <jparker@digium.com>
-
- * res/res_musiconhold.c: Modify the names of functions/variables in
- res_musiconhold to be useful. Closes issue #10464, patch by
- caio1982
-
-2007-08-15 21:25 +0000 [r79623] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/pval.h (added), utils/pval.c (added),
- include/asterisk/extconf.h (added), utils/extconf.c (added),
- utils/conf2ael.c (added): Missing from murf's last trunk commit,
- which was why trunk won't compile
-
-2007-08-15 19:34 +0000 [r79611] Joshua Colp <jcolp@digium.com>
-
- * /: Remove properties that appeared from Steve's last branch
- merge. Automerge has already run so everyone's branches based off
- of trunk are probably toast by now.
-
-2007-08-15 19:21 +0000 [r79595] Steve Murphy <murf@digium.com>
-
- * /, pbx/ael/ael.y (removed), pbx/ael/ael-test/ref.ael-test11,
- res/Makefile, pbx/ael/ael-test/ref.ael-test14,
- pbx/ael/ael-test/ref.ael-test15, pbx/ael/ael-test/ref.ael-test16,
- pbx/ael/ael-test/ref.ael-test19, include/asterisk/ast_expr.h,
- pbx/ael/ael_lex.c (removed), pbx/pbx_ael.c, pbx/ael/ael.flex
- (removed), res/ael (added), main/pbx.c, UPGRADE.txt,
- res/res_ael_share.c (added), pbx/Makefile, CHANGES,
- utils/Makefile, pbx/ael/ael-test/ref.ael-ntest10,
- pbx/ael/ael.tab.c (removed), pbx/ael/ael-test/ref.ael-test1,
- pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3,
- pbx/ael/ael-test/ref.ael-test4, include/asterisk/ael_structs.h,
- pbx/ael/ael.tab.h (removed), pbx/ael/ael-test/ref.ael-test5,
- utils/ael_main.c, include/asterisk/pbx.h,
- pbx/ael/ael-test/ref.ael-test6, pbx/ael/ael-test/ref.ael-test7,
- utils/check_expr.c: This commit closes bug 7605, and half-closes
- 7638. The AEL code has been redistributed/repartitioned to allow
- code re-use both inside and outside of Asterisk. This commit
- introduces the utils/conf2ael program, and an external
- config-file reader, for both normal config files, and for
- extensions.conf (context, exten, prio); It provides an API for
- programs outside of asterisk to use to play with the dialplan and
- config files.
-
-2007-08-15 14:42 +0000 [r79558] Joshua Colp <jcolp@digium.com>
-
- * /, main/rtp.c: Merged revisions 79553 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r79553 | file | 2007-08-15 11:40:23 -0300 (Wed, 15 Aug 2007) | 6
- lines (closes issue #10440) Reported by: irroot (closes issue
- #10454) Reported by: flo_turc Increase maximum timestamp skew to
- 120. 20 was apparently far too low. ........
-
-2007-08-15 14:27 +0000 [r79529] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 79527 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r79527 | mmichelson | 2007-08-15 09:26:40 -0500 (Wed, 15 Aug
- 2007) | 5 lines Fixed an error in the Russian language voicemail
- intro. (issue #10458, reported and patched by Oleh) ........
-
-2007-08-15 14:20 +0000 [r79524] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 79523 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r79523 | file | 2007-08-15 11:18:44 -0300 (Wed, 15 Aug 2007) | 6
- lines (closes issue #10456) Reported by: irroot Patches:
- sip_timeout.patch uploaded by irroot (license 52) Change
- hardcoded timer value to defined value. I'm doing this in 1.4 as
- well so if it needs to be changed in the future this place would
- not have been forgotten. ........
-
-2007-08-15 11:27 +0000 [r79507] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /, channels/misdn/ie.c,
- channels/misdn/isdn_msg_parser.c: Merged revisions 78936 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r78936 | crichter | 2007-08-10 15:24:03 +0200 (Fr, 10 Aug 2007) |
- 1 line fixed a bug with the useruser information element. We send
- them now also in the disconnect message. ........
-
-2007-08-14 18:50 +0000 [r79437-79471] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 79470 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r79470 | russell | 2007-08-14 13:49:10 -0500 (Tue, 14 Aug 2007) |
- 2 lines Fix another spot where an iax2_peer would be leaked if
- realtime was in use. ........
-
- * /, channels/chan_iax2.c: Merged revisions 79436 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r79436 | russell | 2007-08-14 12:31:39 -0500 (Tue, 14 Aug 2007) |
- 3 lines Fix some memory leaks throughout chan_iax2 related to the
- use of realtime. I found these while working on iax2_peer object
- reference tracking. ........
-
-2007-08-14 15:30 +0000 [r79403] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_features.c: Merged revisions 79397 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r79397 | file | 2007-08-14 12:27:13 -0300 (Tue, 14 Aug 2007) | 4
- lines (closes issue #10415) Reported by: atis Revert fix for
- #10327 as it causes more issues then it solves. ........
-
-2007-08-14 14:32 +0000 [r79392] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael-test/ref.ael-vtest17, /,
- pbx/ael/ael-test/ref.ael-test11, pbx/ael/ael-test/ref.ael-test14,
- pbx/ael/ael-test/ael-test5/extensions.ael,
- pbx/ael/ael-test/ael-test6/extensions.ael,
- pbx/ael/ael-test/ref.ael-test19,
- pbx/ael/ael-test/ael-vtest21/extensions.ael,
- pbx/ael/ael-test/ael-vtest21 (added),
- pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
- pbx/ael/ael-test/ref.ael-test2, pbx/pbx_ael.c,
- pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-test4,
- utils/ael_main.c, pbx/ael/ael-test/ref.ael-test6,
- pbx/ael/ael-test/ref.ael-vtest21 (added),
- pbx/ael/ael-test/ref.ael-vtest13: Merged revisions 79255 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r79255 | murf | 2007-08-13 11:49:54 -0600 (Mon, 13 Aug 2007) | 1
- line This patch fixes bug 10411. I added a new regression test,
- some regression test cleanups ........
-
-2007-08-14 14:17 +0000 [r79379] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: (closes issue #10427) Reported by: pj Two of the
- three places ast_waitfor_nandfds could branch off to did not
- clear outfd and exception. If the calling function did not clear
- these there was a chance they could get a false positive on
- testing to see whether they were set.
-
-2007-08-14 13:46 +0000 [r79378] Steve Murphy <murf@digium.com>
-
- * main/channel.c, channels/chan_zap.c: Don't ask me why, but
- waitfordigit will immediately return a 1 on my system, unless the
- outfd is initialized to -1 before calling the nandfds func
-
-2007-08-13 21:59 +0000 [r79335] Joshua Colp <jcolp@digium.com>
-
- * /, include/asterisk/speech.h, res/res_speech.c,
- apps/app_speech_utils.c: Merged revisions 79334 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r79334 | file | 2007-08-13 18:57:20 -0300 (Mon, 13 Aug 2007) | 2
- lines Instead of accepting a single DTMF character accept a full
- string. ........
-
-2007-08-13 21:44 +0000 [r79333] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_odbc.c: Only use the sanitysql if it's not zero-len
-
-2007-08-13 20:40 +0000 [r79273-79306] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 79301 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r79301 | russell | 2007-08-13 15:37:50 -0500 (Mon, 13 Aug 2007) |
- 3 lines Don't call find_peer in registry_authrequest with the pvt
- lock held to avoid a deadlock. ........
-
- * /, channels/chan_iax2.c: Merged revisions 79276 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r79276 | russell | 2007-08-13 15:18:30 -0500 (Mon, 13 Aug 2007) |
- 4 lines Release the pvt lock before calling find_peer in
- register_verify to avoid a deadlock. Also, remove some
- unnecessary locking in auth_fail that was only done recursively.
- ........
-
- * /, channels/chan_iax2.c: Merged revisions 79274 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r79274 | russell | 2007-08-13 15:02:57 -0500 (Mon, 13 Aug 2007) |
- 3 lines Don't call find_peer within update_registry with a pvt
- lock held. This can cause a deadlock as the code will eventually
- call find_callno. ........
-
- * /, channels/chan_iax2.c: Merged revisions 79272 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r79272 | russell | 2007-08-13 14:27:39 -0500 (Mon, 13 Aug 2007) |
- 9 lines I am fighting deadlocks in chan_iax2. I have tracked them
- down to a single core issue. You can not call find_callno() while
- holding a pvt lock as this function has to lock another (every)
- other pvt lock. Doing so can lead to a classic deadlock. So, I am
- tracking down all of the code paths where this can happen and
- fixing them. The fix I committed earlier today was along the same
- theme. This patch fixes some code down the path of
- authenticate_reply. ........
-
-2007-08-13 15:39 +0000 [r79238] Mark Michelson <mmichelson@digium.com>
-
- * CHANGES, apps/app_queue.c: Allow non-realtime queues to have
- realtime members (issue #10424, reported and patched by irroot)
-
-2007-08-13 15:32 +0000 [r79222] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 79214 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r79214 | russell | 2007-08-13 10:28:13 -0500 (Mon, 13 Aug 2007) |
- 4 lines Fix a potential deadlock in socket_process.
- check_provisioning can eventually call find_callno. You can't
- hold a pvt lock while calling find_callno because it goes through
- and locks every single one looking for a match. ........
-
-2007-08-13 14:55 +0000 [r79208] Joshua Colp <jcolp@digium.com>
-
- * /, include/asterisk/speech.h, res/res_speech.c,
- apps/app_speech_utils.c: Merged revisions 79207 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r79207 | file | 2007-08-13 11:51:09 -0300 (Mon, 13 Aug 2007) | 2
- lines Add an API call to allow the engine to know that DTMF was
- received. ........
-
-2007-08-13 14:23 +0000 [r79176] Russell Bryant <russell@digium.com>
-
- * main/channel.c, include/asterisk/channel.h: constify the return
- value of reason2str
-
-2007-08-13 14:22 +0000 [r79175] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_jingle.c, channels/chan_phone.c,
- channels/chan_local.c, channels/chan_misdn.c,
- channels/chan_zap.c, /, channels/chan_sip.c,
- channels/chan_skinny.c, channels/chan_h323.c,
- channels/chan_gtalk.c, channels/chan_iax2.c, channels/chan_oss.c,
- channels/chan_mgcp.c: 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. ........
-
-2007-08-11 05:28 +0000 [r79147] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_odbc.c: Merged revisions 79142 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r79142 | tilghman | 2007-08-11 00:23:04 -0500 (Sat, 11 Aug 2007)
- | 2 lines Ensure the connection gets marked as used at allocation
- time (closes issue #10429, report and fix by mnicholson) ........
-
-2007-08-10 21:29 +0000 [r79109] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Use localized softkey labels. Add some
- information about localization "codes".
-
-2007-08-10 21:03 +0000 [r79100] Steve Murphy <murf@digium.com>
-
- * main/channel.c, pbx/pbx_spool.c, include/asterisk/channel.h:
- Merged revisions 79099 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r79099 | murf | 2007-08-10 14:53:43 -0600 (Fri, 10 Aug 2007) | 1
- line From a user complaint on #asterisk, I have forced pbx_spool
- to explain what reason codes mean, when they are logged ........
-
-2007-08-10 20:48 +0000 [r79098] Russell Bryant <russell@digium.com>
-
- * funcs/func_devstate.c: Store custom device states in astdb so
- that they will persist a restart. As a side benefit, this
- simplifies the code a bit, too.
-
-2007-08-10 18:37 +0000 [r79074] Joshua Colp <jcolp@digium.com>
-
- * main/dial.c: Bring up to date with poll changes.
-
-2007-08-10 18:35 +0000 [r79045-79068] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, /: Merged revisions 79049 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r79049 | murf | 2007-08-10 12:25:51 -0600 (Fri, 10 Aug 2007) | 1
- line Re bug behavior mentioned in #asterisk, made this tweak to
- code, to prevent hundreds of log messages from being generated
- ........
-
- * /: oops. forgot to commit the prop change on .
-
- * main/cdr.c: Merged revisions 79044 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r79044 | murf | 2007-08-10 11:43:49 -0600 (Fri, 10 Aug 2007) | 1
- line This will help debug; from a question asked on #asterisk
- ........
-
-2007-08-10 16:24 +0000 [r79005-79027] Russell Bryant <russell@digium.com>
-
- * include/asterisk/devicestate.h, apps/app_meetme.c,
- res/res_features.c, main/devicestate.c, main/event.c,
- funcs/func_devstate.c: Merge a set of device state improvements
- from team/russell/events. The way a device state change
- propagates is kind of silly, in my opinion. A device state
- provider calls a function that indicates that the state of a
- device has changed. Then, another thread goes back and calls a
- callback for the device state provider to find out what the new
- state is before it can go send it off to whoever cares. I have
- changed it so that you can include the state that the device has
- changed to in the first function call from the device state
- provider. This removes the need to have to call the callback,
- which locks up critical containers to go find out what the state
- changed to. This change set changes the "simple" device state
- providers to use the new method. This includes parking, meetme,
- and SLA. I have also mostly converted chan_agent in my branch,
- but still have some more things to think through before
- presenting the plan for converting channel drivers to ensure all
- of the right events get generated ...
-
- * /, include/asterisk/lock.h: Merged revisions 78995 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r78995 | russell | 2007-08-10 10:20:09 -0500 (Fri, 10
- Aug 2007) | 4 lines The last set of changes that I made to "core
- show locks" made it not able to track mutexes unless they were
- declared using AST_MUTEX_DEFINE_STATIC. Locks initialized with
- ast_mutex_init() were not tracked. It should work now. ........
-
-2007-08-10 14:17 +0000 [r78952-78956] Joshua Colp <jcolp@digium.com>
-
- * /, main/file.c: Merged revisions 78955 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r78955 | file | 2007-08-10 11:15:53 -0300 (Fri, 10 Aug 2007) | 2
- lines Don't bother having the core pass through or emulate begin
- DTMF frames when in an ast_waitstream. It only cares about the
- end of DTMF. ........
-
- * /, configs/queues.conf.sample: Merged revisions 78951 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r78951 | file | 2007-08-10 10:49:19 -0300 (Fri, 10 Aug 2007) | 4
- lines (closes issue #10422) Reported by: bhowell Add note to
- sample configuration about module load order and how it can cause
- perfectly good queue members to be marked as invalid. ........
-
-2007-08-09 23:49 +0000 [r78908] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 78907 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r78907 | mmichelson | 2007-08-09 18:47:00 -0500 (Thu, 09 Aug
- 2007) | 4 lines Improved a bit of logic regarding comma-separated
- mailboxes in has_voicemail. Also added some braces to some
- compound if statements since unbraced if statements scare me in
- general. ........
-
-2007-08-09 23:32 +0000 [r78906] Steve Murphy <murf@digium.com>
-
- * Makefile, /: Merged revisions 78891 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r78891 | murf | 2007-08-09 17:10:46 -0600 (Thu, 09 Aug 2007) | 1
- line This fixes bug 10416; thanks to mvanbaak for the pretty
- output ........
-
-2007-08-09 22:19 +0000 [r78861-78862] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 78859 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r78859 | mmichelson | 2007-08-09 16:51:17 -0500 (Thu, 09 Aug
- 2007) | 9 lines Quite a few changes regarding IMAP storage. 1.
- instead of using inboxcount as the core message counting
- function, we use messagecount instead. This makes it possible to
- count messages in folders besides just INBOX and Old. 2.
- inboxcount and hasvoicemail now use messagecount as their means
- of determining return values. 3. Added a copy_message function
- for IMAP storage. Unfortunately I don't have the means to test
- it, but it seems like a pretty straightforward function. 4.
- Removed a #ifndef IMAP_STORAGE and matching #endif from
- leave_voicemail for a couple of reasons. One, we want to support
- copying mail to multiple IMAP boxes, and two, IMAP was broken
- because a STORE macro had been moved into this section of code.
- ........
-
-2007-08-09 20:07 +0000 [r78829] Russell Bryant <russell@digium.com>
-
- * apps/app_minivm.c: Don't use strncpy for moving a chunk of memory
- to another that is overlapping. This was found by running
- Asterisk under valgrind.
-
-2007-08-09 19:35 +0000 [r78718-78824] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: When looking up a mailbox, use the default
- context if not specified as something else
-
- * channels/chan_sip.c: Restore the ability to have multiple
- mailboxes listed for the mailbox option in sip.conf. chan_sip now
- maintains separate internal MWI subscriptions for each one.
-
- * /, apps/app_voicemail.c: Merged revisions 78778 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r78778 | russell | 2007-08-09 12:58:31 -0500 (Thu, 09 Aug 2007) |
- 1 line add a comment to indicate that inboxcount for ODBC_STORAGE
- needs to be fixed to support multiple mailboxes ........
-
- * /, apps/app_voicemail.c: Merged revisions 78749 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r78749 | russell | 2007-08-09 12:24:40 -0500 (Thu, 09 Aug 2007) |
- 9 lines Fix subscriptions to multiple mailboxes for ODBC_STORAGE.
- Also, leave a comment for this to be fixed for IMAP_STORAGE, as
- well. I left IMAP alone since I know MarkM was working on this
- code right now for another reason. This is broken even worse in
- trunk, but for a different reason. The fact that the mailbox
- option supported multiple mailboxes is completely not obvious
- from the code in the channel drivers. Anyway, I will fix that in
- another commit ... ........
-
- * channels/chan_zap.c, channels/chan_sip.c,
- include/asterisk/event_defs.h, channels/chan_iax2.c,
- channels/chan_mgcp.c, apps/app_voicemail.c: Fix a problem that I
- had introduced into MWI handling. I had ignored the mailbox
- context. Now, all related MWI event dealings pay attention to the
- context as well.
-
- * /, apps/app_meetme.c: Merged revisions 78717 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r78717 | russell | 2007-08-09 11:12:57 -0500 (Thu, 09 Aug 2007) |
- 7 lines Fix a problem with the combination of the 'F' option to
- pass DTMF through a conference and options that use DTMF to
- activate various features. The problem was that the BEGIN frame
- would be passed through, but the END frame would get intercepted
- to activate a feature. Then, the other conference members would
- hear DTMF for forever, which they didn't seem to like very much.
- (closes issue #10400, reported by stevefeinstein, fixed by me)
- ........
-
-2007-08-08 22:05 +0000 [r78649-78686] Joshua Colp <jcolp@digium.com>
-
- * configure: Regenerate configure script. This actually just
- updated the revision number... since my last merge changed it to
- an older number, while it was in fact generated from a much newer
- revision.
-
- * channels/chan_skinny.c: Minor fix for building under dev mode
- when byteswapping macro header files are not available.
-
- * apps/app_dial.c, channels/chan_zap.c, channels/chan_sip.c,
- include/asterisk/autoconfig.h.in, channels/chan_agent.c,
- configure.ac, include/asterisk/channel.h, channels/chan_gtalk.c,
- channels/chan_oss.c, main/rtp.c, main/channel.c,
- channels/chan_jingle.c, channels/chan_phone.c,
- channels/chan_misdn.c, channels/chan_skinny.c, configure,
- channels/chan_features.c, channels/chan_h323.c,
- channels/chan_alsa.c, channels/chan_nbs.c, channels/chan_mgcp.c:
- 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.
-
- * channels/chan_zap.c: HAVEL_SS7 should be HAVE_SS7. Reported by
- kwallace.
-
- * main/channel.c, include/asterisk/audiohook.h (added),
- funcs/func_volume.c (added), main/Makefile, main/slinfactory.c,
- include/asterisk/chanspy.h (removed), include/asterisk/channel.h,
- main/audiohook.c (added), apps/app_chanspy.c,
- apps/app_mixmonitor.c, include/asterisk/slinfactory.h: Merge
- audiohooks branch into trunk. This is a new API for developers to
- listen and manipulate the audio going through a channel.
-
-2007-08-08 19:30 +0000 [r78648] Jason Parker <jparker@digium.com>
-
- * /, doc/jabber.txt: Merged revisions 78646 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r78646 | qwell | 2007-08-08 14:29:42 -0500 (Wed, 08 Aug 2007) | 2
- lines Fix mogs email address. ........
-
-2007-08-08 19:03 +0000 [r78637] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: Correct spelling. s/threaads/threads/
-
-2007-08-08 18:34 +0000 [r78590-78635] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 78575 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r78575 | mmichelson | 2007-08-08 09:26:36 -0500 (Wed, 08 Aug
- 2007) | 4 lines Changing a bit of logic so that someone will
- NEVER exit the queue on timeout unless they have enabled the 'n'
- option. This commit relates to issue #10320. Thanks to
- jfitzgibbon for detailing the idea behind this code change.
- ........
-
-2007-08-08 13:52 +0000 [r78570] Joshua Colp <jcolp@digium.com>
-
- * /, configs/sip.conf.sample: Merged revisions 78569 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r78569 | file | 2007-08-08 10:51:01 -0300 (Wed, 08 Aug
- 2007) | 4 lines (closes issue #10335) Reported by: adamgundy
- Update sip.conf to include another scenario where directrtpsetup
- will fail. ........
-
-2007-08-07 23:04 +0000 [r78541] Russell Bryant <russell@digium.com>
-
- * main/pbx.c, pbx/pbx_spool.c, main/sha1.c, res/res_features.c,
- res/res_crypto.c, utils/smsq.c, include/asterisk/features.h: Add
- another big set of doxygen documentation improvements from
- snuffy. (closes issue #9892) (closes issue #10395)
-
-2007-08-07 22:13 +0000 [r78521] Joshua Colp <jcolp@digium.com>
-
- * main/manager.c, include/asterisk/manager.h: Use the linkedlists.h
- macros for the manager action list.
-
-2007-08-07 21:00 +0000 [r78489] Russell Bryant <russell@digium.com>
-
- * res/res_config_odbc.c, /: Merged revisions 78488 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r78488 | russell | 2007-08-07 15:57:54 -0500 (Tue, 07
- Aug 2007) | 2 lines Fix the build of this module on 64-bit
- platforms ........
-
-2007-08-07 19:44 +0000 [r78451] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 78450 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r78450 | mmichelson | 2007-08-07 14:43:57 -0500 (Tue, 07 Aug
- 2007) | 5 lines The logic behind inboxcount's return value was
- reversed in has_voicemail and message_count. (closes issue
- #10401, reported by st1710, patched by me) ........
-
-2007-08-07 19:36 +0000 [r78442] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_odbc.c: Merged revisions 78437 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r78437 | tilghman | 2007-08-07 14:34:25 -0500 (Tue, 07 Aug 2007)
- | 2 lines Don't free the environment handle when the connection
- fails, because other connections might be depending upon it
- ........
-
-2007-08-07 19:14 +0000 [r78417] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_config_odbc.c, /, apps/app_directory.c,
- apps/app_voicemail.c: Merged revisions 78415 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r78415 | tilghman | 2007-08-07 14:09:38 -0500 (Tue, 07 Aug 2007)
- | 2 lines Reconnection doesn't happen automatically when a DB
- goes down (fixes issue #9389) ........
-
-2007-08-07 18:26 +0000 [r78378] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_skinny.c: 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) ........
-
-2007-08-07 17:46 +0000 [r78372] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 78371 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r78371 | russell | 2007-08-07 12:45:30 -0500
- (Tue, 07 Aug 2007) | 12 lines Merged revisions 78370 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r78370 | russell | 2007-08-07 12:44:04 -0500 (Tue, 07
- Aug 2007) | 4 lines Revert patch committed for issue #9660. It
- broke E&M trunks. (closes issue #10360) (closes issue #10364)
- ........ ................
-
-2007-08-07 16:17 +0000 [r78346-78347] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_zap.c: Can't forget outsignaling!
-
- * channels/chan_zap.c: Just for jsmith... make signaling a valid
- option that acts like signalling.
-
-2007-08-07 16:04 +0000 [r78342] Russell Bryant <russell@digium.com>
-
- * res/res_eventtest.c (removed): Remove some test code from trunk
- as it doesn't need to be here. I'm just going to keep it with a
- bunch of other changes i have sitting in a branch.
-
-2007-08-07 15:40 +0000 [r78338] Joshua Colp <jcolp@digium.com>
-
- * main/frame.c: (closes issue #10225) Reported by: klaus3000 Clean
- up AST_FORMAT_LIST list. It may have mattered in the old days to
- have undefined entries but these days it does not.
-
-2007-08-06 23:00 +0000 [r78312] Jason Parker <jparker@digium.com>
-
- * channels/chan_agent.c: Add a TalkingToChan to the response of the
- "agents" manager action. This is similar to the existing "talking
- to" that you see what using the "agent show" CLI command. Closes
- issue #10102
-
-2007-08-06 21:59 +0000 [r78276-78279] Joshua Colp <jcolp@digium.com>
-
- * apps/app_senddtmf.c: Fix bug where a NULL timeout would make
- things explode if SendDTMF was called with it.
-
- * apps/app_dial.c, main/channel.c, include/asterisk/app.h,
- res/res_features.c, apps/app_test.c, main/app.c,
- include/asterisk/channel.h, apps/app_senddtmf.c: Extend the
- ast_senddigit and ast_dtmf_stream API calls to allow the duration
- of the DTMF digit(s) to be specified and make the SendDTMF
- application have the capability to use it.
-
- * main/channel.c, /: Merged revisions 78275 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r78275 | file | 2007-08-06 18:41:13 -0300 (Mon, 06 Aug 2007) | 2
- lines Add additional DTMF log messages to help when debugging
- issues. ........
-
-2007-08-06 20:45 +0000 [r78243] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 78242 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r78242 | russell | 2007-08-06 15:44:09 -0500 (Mon, 06 Aug 2007) |
- 4 lines Fix an issue where dynamic threads can get free'd, but
- still exist in the dynamic thread list. (closes issue #10392,
- patch from Mihai, with credit to his colleague, Pete) ........
-
-2007-08-06 19:52 +0000 [r78227] Doug Bailey <dbailey@digium.com>
-
- * main/tdd.c, include/asterisk/fskmodem.h, main/callerid.c,
- main/fskmodem.c: Change the fsk filter used in CID and TDD decode
- to an integer based implementation
-
-2007-08-06 17:51 +0000 [r78186-78192] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: Fixing a compiler warning which warns that a
- variable may be used unitialized. Thanks to mvanbaak for pointing
- this out.
-
- * /, channels/chan_sip.c, include/asterisk/config.h, main/config.c:
- Merged revisions 78103 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r78103 | mmichelson | 2007-08-03 15:25:22 -0500 (Fri, 03 Aug
- 2007) | 7 lines Changed the behavior of sip's realtime_peer
- function to match the corresponding way of matching for
- non-realtime peers. Now matches are made on both the IP address
- and port number, or if the insecure setting is set to "port" then
- just match on the IP address. In order to accomplish this, I also
- added a new API call, ast_category_root, which returns the first
- variable of an ast_category struct ........
-
-2007-08-06 16:51 +0000 [r78185] Russell Bryant <russell@digium.com>
-
- * /, include/asterisk/linkedlists.h: Merged revisions 78184 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r78184 | russell | 2007-08-06 11:50:54 -0500 (Mon, 06 Aug 2007) |
- 5 lines Fix the return value of AST_LIST_REMOVE(). This shouldn't
- be causing any problems, though, because the only code that uses
- the return value only checks to see if it is NULL. (closes issue
- #10390, pointed out by mihai) ........
-
-2007-08-06 16:34 +0000 [r78183] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 78182 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r78182 | file | 2007-08-06 13:32:44 -0300 (Mon, 06 Aug 2007) | 2
- lines It is possible for a transfer to occur before the remote
- device has our tag in which case they send none in the transfer.
- In this case we need to not fail the transfer dialog lookup.
- ........
-
-2007-08-06 16:31 +0000 [r78179-78181] Jason Parker <jparker@digium.com>
-
- * /, main/config.c: Merged revisions 78180 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #9938) ........ r78180 | qwell | 2007-08-06 11:30:51 -0500
- (Mon, 06 Aug 2007) | 5 lines Fix an issue with using UpdateConfig
- (manager action) where escaped semicolons in a config would be
- converted to just semicolons (\; to ;) Issue 9938 ........
-
- * channels/chan_skinny.c, configs/skinny.conf.sample: Implement
- setvar functionality in chan_skinny Closes issue #10379, patch by
- mvanbaak.
-
-2007-08-06 15:28 +0000 [r78167-78173] Joshua Colp <jcolp@digium.com>
-
- * /, main/rtp.c: Merged revisions 78172 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r78172 | file | 2007-08-06 12:27:24 -0300 (Mon, 06 Aug 2007) | 4
- lines (closes issue #10355) Reported by: wdecarne Now that we
- pass through RTP timestamp information we need to make the
- allowed timestamp skew considerably less. There are situations
- where a source may change and due to the timestamp difference the
- receiver will experience an audio gap since we did not indicate
- by setting the marker bit that the source changed. ........
-
- * apps/app_externalivr.c: (closes issue #10381) Reported by: yehavi
- Use the filename we parsed using the standard parsing when
- launching the application specified to ExternalIVR.
-
- * /, configure, configure.ac: Merged revisions 78166 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r78166 | file | 2007-08-06 11:18:20 -0300 (Mon, 06 Aug
- 2007) | 4 lines (closes issue #10383) Reported by: rizzo Include
- stdlib.h so NULL gets defined for gethostbyname_r checks.
- ........
-
-2007-08-05 04:16 +0000 [r78142-78144] Russell Bryant <russell@digium.com>
-
- * /, include/asterisk/lock.h: Merged revisions 78143 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r78143 | russell | 2007-08-04 23:15:31 -0500 (Sat, 04
- Aug 2007) | 2 lines Fix compilation failure when MALLOC_DEBUG is
- enabled, but DEBUG_THREADS is not ........
-
- * apps/app_exec.c: Make this module build on my mac
-
-2007-08-05 03:42 +0000 [r78140] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 78139 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r78139 | tilghman | 2007-08-04 22:29:01 -0500 (Sat, 04 Aug 2007)
- | 2 lines If peer is not found, the error message is misleading
- (should be peer not found, not ACL failure) ........
-
-2007-08-05 03:14 +0000 [r78138] Russell Bryant <russell@digium.com>
-
- * include/asterisk/linkedlists.h: Fix building res_crypto on
- systems that init locks with constructors. The problem was that
- res_crypto now has a RWLIST named "keys". The macro for defining
- this list defines a function used as a constructor for the list
- called "init_keys". However, there was another function called
- init_keys in this module for a CLI command. The fix is just to
- prepend the generated functions with underscores.
-
-2007-08-03 20:21 +0000 [r78029-78102] Russell Bryant <russell@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 78101 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r78101 | russell | 2007-08-03 15:14:06 -0500 (Fri, 03 Aug 2007) |
- 10 lines (closes issue #10194) Reported by: blitzrage Patches:
- bug0010194 uploaded by vovochka Tested by: blitzrage Fix a
- problem when you call Voicemail() with multiple mailboxes
- specified and ODBC_STORAGE is in use. The audio part of the
- message was only given to the first mailbox specified. ........
-
- * /, main/utils.c, include/asterisk/lock.h, main/astmm.c: Merged
- revisions 78095 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r78095 | russell | 2007-08-03 14:39:49 -0500 (Fri, 03 Aug 2007) |
- 28 lines Add some improvements to lock debugging. These changes
- take effect with DEBUG_THREADS enabled and provide the following:
- * This will keep track of which locks are held by which thread as
- well as which lock a thread is waiting for in a thread-local data
- structure. A reference to this structure is available on the
- stack in the dummy_start() function, which is the common entry
- point for all threads. This information can be easily retrieved
- using gdb if you switch to the dummy_start() stack frame of any
- thread and print the contents of the lock_info variable. * All of
- the thread-local structures for keeping track of this lock
- information are also stored in a list so that the information can
- be dumped to the CLI using the "core show locks" CLI command.
- This introduces a little bit of a performance hit as it requires
- additional underlying locking operations inside of every
- lock/unlock on an ast_mutex. However, the benefits of having this
- information available at the CLI is huge, especially considering
- this is only done in DEBUG_THREADS mode. It means that in most
- cases where we debug deadlocks, we no longer have to request
- access to the machine to analyze the contents of ast_mutex_t
- structures. We can now just ask them to get the output of "core
- show locks", which gives us all of the information we needed in
- most cases. I also had to make some additional changes to astmm.c
- to make this work when both MALLOC_DEBUG and DEBUG_THREADS are
- enabled. I disabled tracking of one of the locks in astmm.c
- because it gets used inside the replacement memory allocation
- routines, and the lock tracking code allocates memory. This
- caused infinite recursion. ........
-
- * /, channels/chan_iax2.c: Merged revisions 78063 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r78063 | russell | 2007-08-03 12:01:07 -0500 (Fri, 03 Aug 2007) |
- 4 lines Only pass through HOLD and UNHOLD control frames when the
- mohinterpret option is set to "passthrough". This was pointed out
- by Kevin in the middle of a training session. ........
-
- * /, channels/chan_iax2.c: Merged revisions 78028 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r78028 | russell | 2007-08-02 21:04:22 -0500 (Thu, 02 Aug 2007) |
- 6 lines Don't reuse the timespec that was set to 0 in the
- previous timedwait as it will just return immediately. Also, fix
- some logic so the thread's lock isn't unlocked twice in the weird
- case of dynamic threads getting acquired right after a timeout.
- (pointed out by SteveK) ........
-
-2007-08-02 21:54 +0000 [r77994-77997] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_skinny.c, configs/skinny.conf.sample: Merged
- revisions 77996 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #9779) ........ r77996 | qwell | 2007-08-02 16:53:39 -0500
- (Thu, 02 Aug 2007) | 5 lines Make sure we actually allow 6 chars
- to be sent. Also make note of the "A" option of date format.
- Issue 9779, modifications by DEA, wedhorn, and myself. ........
-
- * /, channels/chan_skinny.c: Merged revisions 77993 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #10325) ........ r77993 | qwell | 2007-08-02 15:22:40 -0500
- (Thu, 02 Aug 2007) | 5 lines If a device disconnects, the session
- will go away. If this happens during call setup, we need to give
- up. Issue 10325. ........
-
-2007-08-02 19:26 +0000 [r77950] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 77949 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r77949 | russell | 2007-08-02 14:25:14 -0500 (Thu, 02 Aug 2007) |
- 5 lines Fix the case where a dynamic thread times out waiting for
- something to do during the first time it runs. This shouldn't
- ever happen, but we should account for it anyway. (pointed out by
- pete, who works with mihai) ........
-
-2007-08-02 18:43 +0000 [r77948] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_skinny.c: Merged revisions 77947 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #10299) ........ r77947 | qwell | 2007-08-02 13:42:36 -0500
- (Thu, 02 Aug 2007) | 5 lines Make sure we clear the prompt status
- message on a hangup. Also rearrange messages to better fit with
- what a wireshark trace shows it should be. Issue 10299, initial
- patch and solution by sbisker, modified by me to fit with
- wireshark trace. ........
-
-2007-08-02 18:32 +0000 [r77946] Steve Murphy <murf@digium.com>
-
- * /, main/fskmodem.c: Merged revisions 77945 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r77945 | murf | 2007-08-02 12:21:40 -0600 (Thu,
- 02 Aug 2007) | 9 lines Merged revisions 77942 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r77942 | murf | 2007-08-02 11:56:37 -0600 (Thu, 02 Aug 2007) | 1
- line This patch hopefully solves 10141; The user is running with
- it, and it doesn't appear to harm asterisk's operation, and may
- prevent a crash. I'll store it in 1.2, as we have shut down
- support on 1.2, but since I developed the patch before support
- finished, and it might affect 1.4 and trunk, I'm going ahead with
- it. ........ ................
-
-2007-08-02 18:05 +0000 [r77940-77944] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 77943 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r77943 | russell | 2007-08-02 13:04:43 -0500 (Thu, 02 Aug 2007) |
- 9 lines Fix another race condition in the handling of dynamic
- threads. If the dynamic thread timed out waiting for something to
- do, but was acquired to perform an action immediately afterwords,
- then wait on the condition again to give the other thread a
- chance to finish setting up the data for what action this thread
- should perform. Otherwise, if it immediately continues, it will
- perform the wrong action. (reported on IRC by mihai, patch by me)
- (related to issue #10289) ........
-
- * channels/chan_iax2.c: Fix an issue that Simon pointed out to me
- on IRC. There were cases in the trunk version of
- find_idle_thread() where the old full frame processing
- information was not cleared out. This would have caused full
- frames to get deferred for processing by threads that weren't
- actually processing frames for that call. Nice catch!!
-
- * /, channels/chan_iax2.c: Merged revisions 77939 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r77939 | russell | 2007-08-02 11:56:04 -0500 (Thu, 02 Aug 2007) |
- 4 lines Add another sanity check to vnak_retransmit(). This check
- ensures that frames that have already been marked for deletion
- don't get retransmitted. (closes issue #10361, patch from mihai)
- ........
-
-2007-08-02 15:16 +0000 [r77891-77895] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_skinny.c: Merged revisions 77894 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #10358) ........ r77894 | qwell | 2007-08-02 10:15:45 -0500
- (Thu, 02 Aug 2007) | 5 lines Make sure that we show the correct
- extension if dialed from a macro "From: 5555" rather than "From:
- s" Issue 10358, initial patch by DEA, reworked by me to use S_OR,
- tested by sbisker ........
-
- * /, channels/chan_skinny.c: Merged revisions 77890 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #10291) ........ r77890 | qwell | 2007-08-01 17:28:56 -0500
- (Wed, 01 Aug 2007) | 4 lines Put in some additional debug
- information for softkey/stimulus messages. Issue 10291, patch by
- DEA. ........
-
-2007-08-01 22:24 +0000 [r77889] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 77887 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r77887 | russell | 2007-08-01 17:16:17 -0500 (Wed, 01 Aug 2007) |
- 23 lines Fix some race conditions which have been causing weird
- problems in chan_iax2. The most notable problem is that people
- have been seeing storms of VNAK frames being sent due to really
- old frames mysteriously being in the retransmission queue and
- never getting removed. It was possible that a dynamic thread got
- created, but did not acquire its lock before the thread that
- created it signals it to perform an action. When this happens,
- the thread will sleep until it hits a timeout, and then get
- destroyed. So, the action never gets performed and in some cases,
- means a frame doesn't get transmitted and never gets freed since
- the scheduler never gets a chance to reschedule transmission.
- Another less severe race condition is in the handling of a
- timeout for a dynamic thread. It was possible for it to be
- acquired to perform at action at the same time that it hit a
- timeout. When this occurs, whatever action it was acquired for
- would never get performed. (patch contributed by Mihai and
- SteveK) (closes issue #10289) (closes issue #10248) (closes issue
- #10232) (possibly related to issue #10359) ........
-
-2007-08-01 22:19 +0000 [r77888] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 77886 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r77886 | tilghman | 2007-08-01 17:14:47 -0500 (Wed, 01 Aug 2007)
- | 2 lines Voicemail with ODBC_STORAGE defined does not compile
- cleanly (missing def) ........
-
-2007-08-01 21:12 +0000 [r77879-77884] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_skinny.c: Merged revisions 77883 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r77883 | qwell | 2007-08-01 16:08:42 -0500 (Wed, 01 Aug
- 2007) | 7 lines Fix an issue that caused one-way audio on some
- newer devices (specifically the 7921), due to sending packets in
- the wrong order during hangup. Also make sure we clear
- tones/messages on the correct line/instance. Issue 10291, patch
- by DEA, tested by sbisker and myself. ........
-
- * apps/app_queue.c, doc/tex/queuelog.tex: Add the Ring time in the
- CONNECT on the queue_log and on the Manager event AgentConnect
- Closes issue #10349, patch by eliel
-
-2007-08-01 19:37 +0000 [r77864-77878] Joshua Colp <jcolp@digium.com>
-
- * main/pbx.c, configure, configure.ac, main/asterisk.c: Instead of
- adding the SOLARIS check to each HAVE_SYSINFO check let's just
- make the sysinfo autoconf logic a bit pickier about what it
- considers a usable sysinfo.
-
- * main/pbx.c, main/asterisk.c: Solaris does not have a sysinfo like
- we know of on Linux.
-
- * configure, configure.ac: Don't look for /dev/urandom when cross
- compiling. Just assume it is not available.
-
- * /, utils/smsq.c, channels/chan_iax2.c,
- include/asterisk/threadstorage.h, channels/chan_mgcp.c,
- apps/app_voicemail.c: Merged revisions 77869 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r77869 | file | 2007-08-01 14:56:59 -0300 (Wed, 01 Aug 2007) | 2
- lines Add some fixes for building on Solaris. ........
-
- * /, main/utils.c: Merged revisions 77867 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r77867 | file | 2007-08-01 14:52:11 -0300 (Wed, 01 Aug 2007) | 2
- lines Whoops, I meant R_5 not R5. ........
-
- * /, configure, configure.ac: Merged revisions 77865 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r77865 | file | 2007-08-01 14:42:52 -0300 (Wed, 01 Aug
- 2007) | 2 lines And for my last trick... make sure that if
- gethostbyname_r is exported by a library that it is used.
- ........
-
- * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
- main/utils.c: Merged revisions 77863 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r77863 | file | 2007-08-01 14:22:35 -0300 (Wed, 01 Aug 2007) | 2
- lines Extend autoconf logic to determine which version of
- gethostbyname_r is on the system. ........
-
-2007-08-01 15:39 +0000 [r77858] Russell Bryant <russell@digium.com>
-
- * apps/app_dial.c, main/autoservice.c, main/pbx.c,
- apps/app_osplookup.c, channels/chan_local.c,
- channels/chan_vpb.cc, apps/app_meetme.c, res/res_features.c,
- apps/app_zapras.c, apps/app_macro.c, pbx/pbx_dundi.c,
- apps/app_queue.c: Convert code that checks the _softhangup member
- of ast_channel directory to use the ast_check_hangup() funciton.
- This function takes scheduled hangups into account. (closes issue
- #10230, patch by Juggie)
-
-2007-08-01 15:28 +0000 [r77857] Joshua Colp <jcolp@digium.com>
-
- * main/cli.c: Convert CLI helpers list to rwlist.
-
-2007-08-01 14:09 +0000 [r77853-77855] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 77854 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r77854 | mmichelson | 2007-08-01 09:08:57 -0500 (Wed, 01 Aug
- 2007) | 8 lines Fixes an issue I introduced to queues wherein a
- queue with joinempty=yes would kick people out of the queue
- because of erroneously thinking the 'n' option was in use.
- (closes issue #10320, reported by jfitzgibbon, patched by me,
- tested by blitzrage and me) Thank you blitzrage for all the
- testing you've done lately with queues! It's much appreciated!
- ........
-
- * /, apps/app_queue.c: Merged revisions 77852 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r77852 | mmichelson | 2007-08-01 08:59:59 -0500 (Wed, 01 Aug
- 2007) | 7 lines If a queue uses dynamic realtime members, then
- the member list should be updated after each attempt to call the
- queue. This fixes an issue where if a caller calls into a queue
- where no one is logged in, they would wait forever even if a
- member logged in at some point. (closes issue #10346, reported by
- and tested by blitzrage, patched by me) ........
-
-2007-08-01 04:36 +0000 [r77851] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_agi.c: Twould help if we actually defined ->mod before
- comparing against it (reported and fixed by Juggie via IRC).
-
-2007-07-31 21:33 +0000 [r77847] Steve Murphy <murf@digium.com>
-
- * /, contrib/scripts/ast_grab_core: Merged revisions 77844 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r77844 | murf | 2007-07-31 14:59:10 -0600 (Tue,
- 31 Jul 2007) | 9 lines Merged revisions 77842 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r77842 | murf | 2007-07-31 13:19:35 -0600 (Tue, 31 Jul 2007) | 1
- line This probably isn't super-general, but it's a first stab at
- using kill -11 to generate a core file instead of gcore. ........
- ................
-
-2007-07-31 18:50 +0000 [r77834-77838] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_lock.c, CHANGES: Add some documentation detailing an
- aspect of dialplan functions, as requested by Russell
-
- * funcs/func_lock.c (added), UPGRADE.txt: Add func_lock, which
- creates dialplan mutexes, and note that the Macro apps are now
- deprecated. (Closes issue #10264)
-
-2007-07-31 16:21 +0000 [r77833] Joshua Colp <jcolp@digium.com>
-
- * /, include/asterisk/speech.h, res/res_speech.c: Merged revisions
- 77831 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r77831 | file | 2007-07-31 13:17:09 -0300 (Tue, 31 Jul 2007) | 2
- lines Add a flag to the speech API that allows an engine to set
- whether it received results or not. ........
-
-2007-07-31 15:59 +0000 [r77829] Steve Murphy <murf@digium.com>
-
- * channels/chan_sip.c: thanks to Russel, for pointing out that the
- dialoglist_lock/unlock routines also need to be macros if
- DETECT_DEADLOCKS is set
-
-2007-07-31 15:54 +0000 [r77828] Kevin P. Fleming <kpfleming@digium.com>
-
- * build_tools/cflags.xml, /: Merged revisions 77827 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r77827 | kpfleming | 2007-07-31 10:53:42 -0500 (Tue, 31
- Jul 2007) | 2 lines DETECT_DEADLOCKS can't be enabled without
- DEBUG_THREADS or it does nothing ........
-
-2007-07-31 15:22 +0000 [r77825] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 77824 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r77824 | mmichelson | 2007-07-31 10:21:22 -0500 (Tue, 31 Jul
- 2007) | 6 lines This patch makes Asterisk send 100 Trying
- provisional responses upon receipt of re-invites. This makes it
- so that if there are two or more Asterisk servers between
- endpoints, the Asterisk servers will not keep retransmitting the
- re-invites. (closes issue #10274, reported by cstadlmann, patched
- by me with approval from file) ........
-
-2007-07-31 15:01 +0000 [r77819-77821] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_sip.c: there is no use in having functions that
- have no code in them, and hide the locking info when
- DEBUG_THREADS is enabled... i could have fixed this to be
- dependent on DEBUG_THREADS, but it would be just as easy for
- someone to add their test/debugging code to the macros as it
- would have been to the functions
-
- * channels/chan_sip.c: use a different method for overriding the
- send_digit_begin pointer, as the old one fails to compile on my
- 64-bit system with gcc-4.1 and --enable-dev-mode turned on
-
- * apps/app_senddtmf.c: umm... let's build with --enable-dev-mode,
- mmkay?
-
-2007-07-31 03:32 +0000 [r77810] Steve Murphy <murf@digium.com>
-
- * channels/chan_sip.c: Discovered in experiments on core files: if
- you wrap the lock and unlock calls with sip_pvt_lock and
- sip_pvt_unlock, you lose the tracing info you would normally get
- via DETECT_DEADLOCKS; so I turn these two functions into macros
- when DETECT_DEADLOCKS is called. This way, you get meaningful
- stuff in the file and func slots in the lock_info struct.
-
-2007-07-31 01:10 +0000 [r77808] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_meetme.c, apps/app_dictate.c, apps/app_record.c,
- apps/app_authenticate.c, apps/app_sayunixtime.c,
- apps/app_userevent.c, apps/app_chanisavail.c, apps/app_image.c,
- apps/app_followme.c, apps/app_controlplayback.c,
- funcs/func_enum.c, funcs/func_odbc.c, apps/app_minivm.c,
- res/res_agi.c, apps/app_amd.c, apps/app_url.c,
- apps/app_directory.c, apps/app_rpt.c, apps/app_parkandannounce.c,
- apps/app_read.c, funcs/func_timeout.c, apps/app_page.c,
- apps/app_festival.c, apps/app_privacy.c,
- apps/app_waitforsilence.c, apps/app_disa.c, apps/app_transfer.c,
- apps/app_talkdetect.c, apps/app_queue.c, apps/app_playback.c,
- res/res_monitor.c, apps/app_speech_utils.c, funcs/func_curl.c,
- funcs/func_channel.c, funcs/func_cdr.c, apps/app_sendtext.c,
- apps/app_macro.c, apps/app_sms.c, apps/app_senddtmf.c,
- apps/app_mixmonitor.c, apps/app_chanspy.c, apps/app_stack.c,
- apps/app_voicemail.c: Mostly cleanup of documentation to
- substitute the pipe with the comma, but a few other formatting
- cleanups, too.
-
-2007-07-30 20:42 +0000 [r77801] Joshua Colp <jcolp@digium.com>
-
- * main/dial.c, include/asterisk/dial.h: Add support for call
- forwarding and timeouts to the dialing API.
-
-2007-07-30 20:36 +0000 [r77797-77800] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Change another unnecessary use of the
- increment operator to explicitly set the var to 1
-
- * channels/chan_iax2.c: Explicitly set a variable to 1 instead of
- using the increment operator.
-
- * /, channels/chan_iax2.c: Merged revisions 77794 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r77794 | russell | 2007-07-30 15:16:43 -0500 (Mon, 30 Jul 2007) |
- 8 lines Fix an issue that could potentially cause corruption of
- the global iax frame queue. In the network_thread() loop, it
- traverses the list using the AST_LIST_TRAVERSE_SAFE macro.
- However, to remove an element of the list within this loop, it
- used AST_LIST_REMOVE, instead of AST_LIST_REMOVE_CURRENT, which I
- believe could leave some of the internal variables of the SAFE
- macro invalid. Mihai says that he already made this change in his
- local copy and it didn't help his VNAK storm issues, but I still
- think it's wrong. :) ........
-
-2007-07-30 20:19 +0000 [r77796] Jason Parker <jparker@digium.com>
-
- * /, main/say.c: Merged revisions 77795 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #10083) ........ r77795 | qwell | 2007-07-30 15:17:08 -0500
- (Mon, 30 Jul 2007) | 6 lines Applications like SayAlpha() should
- not hang up the channel if you request an "unknown" character
- such as a comma. Instead, skip the character and move on. Issue
- 10083, initial patch by jsmith, modified by me. ........
-
-2007-07-30 19:42 +0000 [r77793] Luigi Rizzo <rizzo@icir.org>
-
- * main/channel.c: print formats as 0x%x instead of %d in a warning
- message. Being bitmasks, it is a lot easier to read this way.
-
-2007-07-30 19:39 +0000 [r77789-77792] Russell Bryant <russell@digium.com>
-
- * res/res_agi.c: Fix the return value of ast_agi_fdprintf() to
- include the result from ast_carefulwrite()
-
- * res/res_agi.c: Improve ast_agi_fdprintf() by using the ast_str()
- API. * Use a thread local ast_str for building the string that
- will be written out to the console for debug, and to the FD for
- the AGI itself, instead of allocating a buffer on the heap every
- time the function is called. * Use the information contained
- within the ast_str to determine how many bytes need to be written
- instead of calling strlen().
-
- * main/manager.c: Remove an XXX comment noting that it would be
- nice for a declaration to be inside of a function. (Yes, it
- would!) Replace it with a note that explains why it can't be done
- using the way that the AST_THREADSTORAGE macro is currently
- defined.
-
- * include/asterisk/agi.h, /, res/res_agi.c: Merged revisions 77788
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r77788 | russell | 2007-07-30 14:13:31 -0500 (Mon, 30 Jul 2007) |
- 10 lines (closes issue #10279) Reported by: seanbright Patches:
- res_agi.carefulwrite.1.4.07252007.patch uploaded by seanbright
- (license 71) res_agi.carefulwrite.trunk.07252007.patch uploaded
- by seanbright (license 71) Allow the "agi_network: yes" line to
- be printed out in the AGI debug output. Also, allow partial
- writes to be handled when writing out this line just like it is
- for all of the others. ........
-
-2007-07-30 19:11 +0000 [r77787] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/agi.h, res/res_agi.c: Cleanup of res_agi,
- ensuring thread safety (closes issue #10288)
-
-2007-07-30 18:56 +0000 [r77786] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /: Merged revisions 77785 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r77785 | russell | 2007-07-30 13:55:15 -0500 (Mon, 30 Jul 2007) |
- 3 lines file and I both committed changes for issue #10301.
- Remove a duplicated assignment to restore the original value of
- the previous channel. ........
-
-2007-07-30 18:45 +0000 [r77784] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_agi.c: Merged revisions 77783 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r77783 | tilghman | 2007-07-30 13:43:55 -0500
- (Mon, 30 Jul 2007) | 10 lines Merged revisions 77782 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r77782 | tilghman | 2007-07-30 13:40:54 -0500 (Mon, 30
- Jul 2007) | 2 lines Revert change in revision 71656, even though
- it fixed a bug, because many people were depending upon the
- (broken) behavior. ........ ................
-
-2007-07-30 17:31 +0000 [r77781] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /: Merged revisions 77780 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r77780 | russell | 2007-07-30 12:29:43 -0500 (Mon, 30 Jul 2007) |
- 16 lines (closes issue #10301) Reported by: fnordian Patches:
- asterisk-1.4.9-channel.c.patch uploaded by fnordian (license 110)
- Additional changes by me Fix some problems in
- channel_find_locked() which can cause an infinite loop. The
- reference to the previous channel is set to NULL in some cases.
- These changes ensure that the reference to the previous channel
- gets restored before needing it again. I'm not convinced that the
- code that is setting it to NULL is really the right thing to do.
- However, I am making these changes to fix the obvious problem and
- just leaving an XXX comment that it needs a better explanation
- that what is there now. ........
-
-2007-07-30 17:12 +0000 [r77772-77779] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_features.c: Merged revisions 77778 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r77778 | file | 2007-07-30 14:11:02 -0300 (Mon, 30 Jul 2007) | 4
- lines (closes issue #10327) Reported by: kkiely Instead of
- directly mucking with the extension/context/priority of the
- channel we are transferring when it has a PBX simply call
- ast_async_goto on it. This will ensure that the channel gets
- handled properly and sent to the right place. ........
-
- * apps/app_followme.c: Minor clean up of app_followme.
-
- * main/channel.c, /: Merged revisions 77771 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r77771 | file | 2007-07-30 12:47:52 -0300 (Mon, 30 Jul 2007) | 6
- lines (closes issue #10301) Reported by: fnordian Patches:
- asterisk-1.4.9-channel.c.patch uploaded by fnordian (license 110)
- Restore previous behavior where if we failed to lock the channel
- we wanted we would return to exactly the same point as if we had
- just reentered the function. ........
-
-2007-07-30 15:22 +0000 [r77770] Russell Bryant <russell@digium.com>
-
- * cdr/cdr_adaptive_odbc.c: Resolve some compiler warnings so that I
- can build under dev mode
-
-2007-07-30 14:53 +0000 [r77769] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_macro.c: Merged revisions 77768 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r77768 | file | 2007-07-30 11:51:44 -0300 (Mon,
- 30 Jul 2007) | 12 lines Merged revisions 77767 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r77767 | file | 2007-07-30 11:50:02 -0300 (Mon, 30 Jul 2007) | 4
- lines (closes issue #10334) Reported by: ramonpeek Pass through
- the return value from macro_exec through the MacroIf application.
- ........ ................
-
-2007-07-30 10:55 +0000 [r77616-77766] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_sip.c: minor code rearrangements: + place the link
- field at the beginning of struct sip_pvt, and not somewhere in
- the middle; + in __sip_reliable_xmit, remove a duplicate
- assignment, and put the statements in a more logical order (i.e.
- first copy the payload and associated info, then copy arguments
- from the caller, then finish initializing the headers...) nothing
- to backport.
-
- * channels/chan_sip.c: rename handle_request() to
- handle_incoming(), as the former was misleading - the function
- deals with all incoming packets, be them requests or responses.
-
- * channels/chan_sip.c: move some dialog-only flags to proper
- variables, namely SIP_NOVIDEO, SIP_DIALOG_ANSWEREDELSEWHERE,
- SIP_PAGE2_NOTEXT, SIP_PAGE2_OUTGOING_CALL These are seldom used
- so the diff is relatively small. Note that 'OUTGOING_CALL' is
- dangerously similar to another dialog flag, 'SIP_OUTGOING', so
- the description will need to clarify the different meaning of the
- two. Also note that the description of NOTEXT is a bit unclear -
- does it mean we don't support it, or 'not requested or not
- supported' ? On passing fix a comment referring to video instead
- of text. Finally, mark with XXX a possibly misleading debugging
- message. (maybe the latter is worth backporting).
-
- * channels/chan_sip.c: use a function, cli_yesno(), to produce the
- output Yes or No for CLI lines. This helps maintaining
- consistency on output, slightly improves readability, and maybe
- one day will make it easier to translate the output in other
- languages (though i have a hard time believing that a CLI user
- who needs 'yes' and 'no' to be translated can actually figure out
- what he/she is doing!)
-
- * channels/chan_sip.c: move the two remaining peer flags to proper
- variables.
-
- * channels/chan_sip.c: move RT_FROMCONTACT to a proper sip_peer
- field.
-
- * channels/chan_sip.c: Move some global 'flags' to individual
- variables. Start putting these variables in a single struct
- (called 'sip_cfg' for the time being, but it could as well be
- 'global' or some other name) so it is easy, when reading the
- code, to figure out what they are for. The downside of using
- struct fields instead of individual global variables is that the
- compiler cannot tell if there are unused fields. But the
- advantage of not cluttering the namespace and manilpulating all
- these variables at once certainly overcome the disadvantagess.
- Nothing to backport, again.
-
- * channels/chan_sip.c: minor simplification of a conditional
- statement
-
- * channels/chan_sip.c: build the version of sip_tech with no
- send_digit_begin at load time instead of duplicating the
- initializer. This should remove the risk of forgetting fields in
- the initializer.
-
- * channels/chan_sip.c: remove bit position from description of
- SIP_* flags. use AST_FORMAT_AUDIO_MASK instead of playing with
- AST_FORMAT_MAX_AUDIO to determine audio formats. There is a
- dubious use of AST_FORMAT_MAX_AUDIO in sip_request_call() which
- surely needs fixing, namely: /* mask request with some set of
- allowed formats. * XXX this needs to be fixed. * The original
- code uses AST_FORMAT_AUDIO_MASK, but it is * unclear what to use
- here. We have global_capabilities, which is * configured from
- sip.conf, and sip_tech.capabilities, which is * hardwired to all
- audio formats. */ The latter is possibly something to backport
- when fixed.
-
- * channels/chan_sip.c: back on cleaning up the usage of flags. Move
- together flags used in the same way (e.g. dialog only,
- dialog-peer, ...) so it will become easier to deal with them in a
- more systematic way. This is being done in stages so it will be
- easier to detect breakage, if any should occur.
-
- * channels/chan_sip.c: more documentation on internal
- representation of incoming SIP messages. Remove definitions for
- now-unused flags, and add references to print routines for other
- flags.
-
- * channels/chan_sip.c: make register_unref() return NULL so it is
- easy to cleanup the original pointer while calling the function.
- on passing add some comments on one of the places where it is
- used, and explain why it is safe there. again, a no-op for
- practical purposes.
-
- * channels/chan_sip.c: add some documentation to auto_congest(),
- and some dialog_ref/unref (they are a no-op at the moment). Also
- clean a pointer after freeing memory to avoid dangling
- references, and write a for() loop in canonical form. In
- practice, everything in this commit is a no-op.
-
- * channels/chan_sip.c: more dialog_ref()/dialog_unref() calls
-
- * channels/chan_sip.c: more dialog_ref()/dialog_unref() calls
-
- * channels/chan_sip.c: start introducing hooks for reference counts
- on dialog descriptors. This commit is, for all practical
- purposes, a no-op, as it only introduces the dialog_ref() and
- dialog_unref() methods, and uses them in a few places (not all
- the places where they would be needed). The goal is to start
- annotating the code with these calls, so the transition to a
- proper container will be easier. Nothing to backport.
-
- * channels/chan_sip.c: remove an unused string
-
- * channels/chan_sip.c: simplify a conditional expression using S_OR
-
- * channels/chan_sip.c: make use of received= and rport= fields in
- sip replies. In a nutshell, these fields are used to tell a sip
- entity the address and port its request came from, and are
- extremely useful in the presence of NATs, especially with
- symmetric NATs where STUN is totally ineffective. This patch
- stores the address and port in the 'ourip' field of the dialog
- descriptor, so they can be reused in subsequent transactions. As
- it is, it works well for things like REGISTER requiring
- authentication, because the second REGISTER request (with auth
- credentials) will carry the correct address. Maybe it can also be
- useful, in case of an address change, to do one or both of the
- following: + propagate the new address to the parent user/peer
- descriptor so that new dialogs will use the correct address from
- the beginning. This is trivial to implement, I am just waiting
- for feedback on this. + re-issue a request in case of an address
- change. This a lot less trivial, maybe unnecessary, and probably
- covered by the previous item. I would seriously consider this
- patch for addition to 1.4 and 1.2. The code is very little
- intrusive, and it would solve in a correct way the nat traversal
- problems for which externip/externaddr/stunaddr are only a
- partial and expensive workaround.
-
-2007-07-27 23:21 +0000 [r77572-77603] Tilghman Lesher <tlesher@digium.com>
-
- * cdr/cdr_adaptive_odbc.c: Some ODBC drivers don't set the
- CHAR_OCTET_LENGTH field correctly.
-
- * Makefile: Target asterisk.pdf stopped building when the build was
- moved to the doc directory.
-
- * /, res/res_odbc.c: Merged revisions 77571 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r77571 | tilghman | 2007-07-27 13:15:58 -0500 (Fri, 27 Jul 2007)
- | 2 lines Missing newline ........
-
-2007-07-27 17:05 +0000 [r77537-77541] Joshua Colp <jcolp@digium.com>
-
- * /, cdr/cdr_pgsql.c: Merged revisions 77540 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r77540 | file | 2007-07-27 14:04:08 -0300 (Fri, 27 Jul 2007) | 6
- lines (closes issue #10310) Reported by: prashant_jois Patches:
- cdr_pgsql.patch uploaded by prashant (license 114) Finish the
- Postgresql connection after the log messages are printed so we
- don't access invalid memory. ........
-
- * channels/chan_sip.c: Turn 4 lines of code into 1 line that does
- the same thing.
-
- * /, channels/chan_sip.c: Merged revisions 77536 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r77536 | file | 2007-07-27 13:27:16 -0300 (Fri, 27 Jul 2007) | 6
- lines (closes issue #10323) Reported by: julianjm Patches:
- chan_sip_device_state_hold_fix.v1.diff.txt uploaded by julianjm
- (license 99) Clear ONHOLD flag when decrementing the onHold peer
- count. If we did not do this the count may keep decreasing.
- ........
-
-2007-07-27 16:20 +0000 [r77534] Tilghman Lesher <tlesher@digium.com>
-
- * pbx/pbx_config.c: 'dialplan save' shouldn't be converting '|'
- back to ',' anymore.
-
-2007-07-27 15:46 +0000 [r77520] Steve Murphy <murf@digium.com>
-
- * apps/app_dial.c, pbx/pbx_ael.c: These fixes take care of two
- problems: a complaint in asterisk-dev that goto's aren't working
- in trunk, a side effect of the move to commas as arg seps in apps
- and funcs; and a problem I spotted myself with dial's 'e' option,
- where gotos were off by one, because I forgot to set the AUTOLOOP
- flag in the peer channel.
-
-2007-07-27 14:31 +0000 [r77491] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 77490 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r77490 | mmichelson | 2007-07-27 09:30:43 -0500 (Fri, 27 Jul
- 2007) | 3 lines "re-invite" was misspelled ........
-
-2007-07-26 23:20 +0000 [r77461] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /: Merged revisions 77460 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r77460 | file | 2007-07-26 20:19:04 -0300 (Thu, 26 Jul 2007) | 4
- lines (closes issue #10302) Reported by: litnialex If a DTMF end
- frame comes from a channel without a begin and it is going to a
- technology that only accepts end frames (aka INFO) then use the
- minimum DTMF duration if one is not in the frame already.
- ........
-
-2007-07-26 22:17 +0000 [r77432] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, doc/tex/mp3.tex, sounds/Makefile: Merged revisions 77424,77429
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r77424 | kpfleming | 2007-07-26 17:14:21 -0500 (Thu, 26 Jul 2007)
- | 2 lines use new canonical name for download server ........
- r77429 | kpfleming | 2007-07-26 17:16:42 -0500 (Thu, 26 Jul 2007)
- | 2 lines change protocol for downloads as well ........
-
-2007-07-26 21:24 +0000 [r77411] Russell Bryant <russell@digium.com>
-
- * Makefile, /: Merged revisions 77410 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r77410 | russell | 2007-07-26 16:23:23 -0500 (Thu, 26 Jul 2007) |
- 10 lines AST_DEVMODE was defined in trunk, but not in 1.4. When
- Asterisk is compiled under dev mode, AST_DEVMODE will get defined
- in buildopts.h. Change 1.4 to define it in the same way that
- trunk does. Also, revert the change that added this define in the
- Makefile The advantage to doing it this way is that buildopts.h
- gets installed when you install Asterisk. Then, when building any
- out of tree modules, or building asterisk-addons, these modules
- know which options the rest of Asterisk was built with. ........
-
-2007-07-26 20:39 +0000 [r77381] Mark Michelson <mmichelson@digium.com>
-
- * Makefile, /, main/logger.c: Merged revisions 77380 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r77380 | mmichelson | 2007-07-26 15:35:17 -0500 (Thu, 26
- Jul 2007) | 7 lines Fixes to get ast_backtrace working properly.
- The AST_DEVMODE macro was never defined so the majority of
- ast_backtrace never attempted compilation. The makefile now
- defines AST_DEVMODE if configure was run with --enable-dev-mode.
- Also, changes were made to acccomodate 64 bit systems in
- ast_backtrace. Thanks to qwell, kpfleming, and Corydon76 for
- their roles in allowing me to get this committed ........
-
-2007-07-26 19:33 +0000 [r77349-77351] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/logger.c: Merged revisions 77350 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r77350 | tilghman | 2007-07-26 14:32:17 -0500 (Thu, 26 Jul 2007)
- | 2 lines Missed one ........
-
- * /, main/logger.c: Merged revisions 77348 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r77348 | tilghman | 2007-07-26 14:27:18 -0500 (Thu, 26 Jul 2007)
- | 2 lines Oops, that builtin define should be all-lowercase.
- ........
-
-2007-07-26 18:31 +0000 [r77319] Mark Michelson <mmichelson@digium.com>
-
- * /, cdr/cdr_pgsql.c: Merged revisions 77318 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r77318 | mmichelson | 2007-07-26 13:30:29 -0500 (Thu, 26 Jul
- 2007) | 8 lines Two consecutive calls to PQfinish could occur,
- meaning free gets called on the same variable twice. This patch
- sets the connection to NULL after calls to PQfinish so that the
- problem does not occur. Also in this patch, prashant_jois
- informed me that it is safe to pass a null pointer to PQfinish,
- so I have removed the check for conn's existence from
- my_unload_module. (closes issue 10295, reported by junky, patched
- by me with input from prashant_jois) ........
-
-2007-07-26 15:49 +0000 [r77268-77299] Russell Bryant <russell@digium.com>
-
- * main/udptl.c, res/res_features.c, main/say.c,
- codecs/codec_adpcm.c, apps/app_alarmreceiver.c,
- cdr/cdr_adaptive_odbc.c, channels/chan_iax2.c,
- main/indications.c, main/config.c, main/loader.c, res/res_smdi.c,
- pbx/pbx_spool.c, channels/chan_skinny.c, apps/app_zapscan.c,
- apps/app_zapras.c, pbx/pbx_realtime.c, channels/chan_alsa.c,
- apps/app_amd.c, cdr/cdr_odbc.c, res/res_speech.c,
- apps/app_dial.c, codecs/codec_g722.c, funcs/func_timeout.c,
- codecs/codec_speex.c, channels/chan_agent.c, codecs/codec_g726.c,
- channels/iax2-provision.c, apps/app_db.c, channels/chan_misdn.c,
- main/srv.c, apps/app_waitforring.c, apps/app_macro.c,
- apps/app_chanspy.c, apps/app_voicemail.c, channels/chan_vpb.cc,
- apps/app_meetme.c, res/res_snmp.c, codecs/codec_gsm.c,
- res/res_musiconhold.c, apps/app_followme.c, codecs/codec_zap.c,
- res/res_jabber.c, main/channel.c, main/cdr.c,
- channels/chan_phone.c, main/dial.c, res/res_config_odbc.c,
- main/manager.c, funcs/func_odbc.c, res/res_agi.c, main/app.c,
- main/image.c, apps/app_rpt.c, apps/app_parkandannounce.c,
- channels/chan_mgcp.c, apps/app_adsiprog.c, apps/app_while.c,
- codecs/codec_lpc10.c, res/res_config_pgsql.c, main/dnsmgr.c,
- channels/chan_zap.c, apps/app_read.c, channels/chan_sip.c,
- main/translate.c, codecs/codec_alaw.c, apps/app_waitforsilence.c,
- res/res_crypto.c, apps/app_queue.c, apps/app_getcpeid.c,
- channels/chan_oss.c, main/rtp.c, apps/app_flash.c,
- main/abstract_jb.c, main/file.c, channels/chan_h323.c,
- codecs/codec_ulaw.c, pbx/pbx_dundi.c, apps/app_sms.c,
- pbx/pbx_gtkconsole.c: Do a massive conversion for using the
- ast_verb() macro (closes issue #10277, patches by mvanbaak)
- Basically, this changes ... if (option_verbose > 2)
- ast_verbose(VERBOSE_PREFIX_3, "Something\n"); to ... ast_verb(3,
- "Something\n");
-
- * doc/tex/odbcstorage.tex, doc/tex/hardware.tex, doc/tex/mp3.tex,
- doc/tex/channelvariables.tex, doc/tex/qos.tex,
- doc/tex/queues-with-callback-members.tex, doc/tex/realtime.tex,
- doc/tex/dundi.tex, doc/tex/enum.tex, doc/tex/asterisk-conf.tex,
- doc/tex/manager.tex, doc/tex/misdn.tex, doc/tex/imapstorage.tex,
- doc/tex/privacy.tex, LICENSE, doc/tex/app-sms.tex,
- doc/tex/cdrdriver.tex, doc/tex/asterisk.tex: Merge a big batch of
- documentation fixes for escaping, marking URLs, places where
- verbatim text went off the end of the page on the PDF, and
- various other improvements (closes issue #10307, IgorG)
-
- * channels/chan_sip.c: Revert some changes to call abs() on the
- result of ast_random(). * random() is defined to return a
- positive result, and now ast_random() will always do so as well
-
- * main/utils.c: Ensure that the read from /dev/urandom returns a
- positive result (closes issue #10308, reported by yehavi, patched
- by me)
-
-2007-07-26 13:19 +0000 [r77267] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_sip.c: Things expecting a positive result from
- ast_random() should not be surprised (closes #10308)
-
-2007-07-26 13:10 +0000 [r77266] Russell Bryant <russell@digium.com>
-
- * main/rtp.c: Add a link to the list of assigned RTP payload types
- for convenience.
-
-2007-07-26 05:35 +0000 [r77233-77248] Luigi Rizzo <rizzo@icir.org>
-
- * main/rtp.c: document how the RTP marker bit is passed for video
- frames, and why this does not overwrite useful information.
-
- * main/rtp.c: add an entry for h263plus in an empty slot of the rtp
- types.
-
-2007-07-26 01:33 +0000 [r77217-77218] Steve Murphy <murf@digium.com>
-
- * /, pbx/pbx_ael.c: The upgrade of application argument separators
- to comma has an effect on AEL; I commented out the code that
- substitutes commas with vertbars, so we can get apps to parse
- their args correctly.
-
- * apps/app_meetme.c: Merged revisions 77191 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r77191 | murf | 2007-07-25 16:39:27 -0600 (Wed, 25 Jul 2007) | 1
- line This fix solves problem with intense squelch noise when
- someone joins conf in bug 9430; We repro'd the problem with
- meetme opts of 'CciMo'; Josh Colp supplied this patch, and I'm
- applying it. It looks like playing the recorded username will
- louse up the next thing played into the channel. Josh rearranged
- the code so as to start things over before playing data directly
- into the conference. ........
-
-2007-07-25 22:18 +0000 [r77182] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_speech_utils.c: Merged revisions 77176 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r77176 | file | 2007-07-25 19:16:10 -0300 (Wed, 25 Jul
- 2007) | 4 lines (closes issue #10303) Reported by: jtodd Add
- SPEECH_DTMF_TERMINATOR variable so the user can specify the digit
- to terminate a DTMF string with. If none is specified then no
- terminator will be used. ........
-
-2007-07-25 21:58 +0000 [r77156] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_iax2.c: silence a warning in ast-devmode on a
- potentially uninitialized var. At first sight (but the function
- is very large so i am not 100% sure) the code seems correct, so
- maybe my compiler is just not smart enough to figure that out at
- the optimization level it has. Not worthwhile merging to 1.4 i
- believe.
-
-2007-07-25 21:53 +0000 [r77155] Mark Michelson <mmichelson@digium.com>
-
- * main/channel.c, /: Merged revisions 77154 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r77154 | mmichelson | 2007-07-25 16:52:47 -0500 (Wed, 25 Jul
- 2007) | 3 lines chan->emulate_dtmf_duration is an unsigned int,
- not a signed int, so use %u instead of %d in the format string
- ........
-
-2007-07-25 17:16 +0000 [r77072] Joshua Colp <jcolp@digium.com>
-
- * /, configure, acinclude.m4: Merged revisions 77071 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r77071 | file | 2007-07-25 14:14:14 -0300 (Wed, 25 Jul
- 2007) | 2 lines Fix autoconf logic for finding OpenH323 when it
- is not in the first place searched (/usr/share/openh323).
- ........
-
-2007-07-25 14:13 +0000 [r77023-77054] Luigi Rizzo <rizzo@icir.org>
-
- * main/translate.c: change the debug level to 3 for an exceedingly
- annoying message (3-deep nested loop)
-
- * main/rtp.c: Merged revisions 77022 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r77022 | rizzo | 2007-07-25 11:34:01 +0200 (Wed, 25 Jul 2007) | 3
- lines set the sequence number in a frame for all frame types
- ........
-
-2007-07-25 01:06 +0000 [r76985] Russell Bryant <russell@digium.com>
-
- * CHANGES: remove a couple of entries that got duplicated and snuck
- into the SIP section. Also, align the NAT/STUN entry with the
- others.
-
-2007-07-25 00:34 +0000 [r76984] Steve Murphy <murf@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 76983 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r76983 | murf | 2007-07-24 18:18:32 -0600 (Tue,
- 24 Jul 2007) | 9 lines Merged revisions 76978 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r76978 | murf | 2007-07-24 18:07:24 -0600 (Tue, 24 Jul 2007) | 1
- line this fixes bug 10293, where the error message because
- defaultzone or loadzone was not defined was confusing ........
- ................
-
-2007-07-24 22:13 +0000 [r76874-76940] Tilghman Lesher <tlesher@digium.com>
-
- * /, include/asterisk/lock.h: Merged revisions 76937 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r76937 | tilghman | 2007-07-24 17:12:43 -0500
- (Tue, 24 Jul 2007) | 10 lines Merged revisions 76934 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r76934 | tilghman | 2007-07-24 17:11:33 -0500 (Tue, 24
- Jul 2007) | 2 lines Oops, res contains the error code, not errno.
- I was wondering why a mutex was reporting "No such file or
- directory"... ........ ................
-
- * build_tools/cflags.xml: Add the flag to trigger an intentional
- crash on mutex errors
-
- * doc/tex/manager.tex, doc/tex/misdn.tex, doc/tex/jitterbuffer.tex,
- doc/tex/odbcstorage.tex, doc/tex/hardware.tex,
- doc/tex/privacy.tex, doc/tex/billing.tex, doc/tex/ael.tex,
- doc/tex/channelvariables.tex, doc/tex/qos.tex,
- doc/tex/realtime.tex, doc/tex/asterisk.tex, doc/tex/queuelog.tex:
- Fix escaping and some of the formattting (closes issue #10285)
-
-2007-07-24 17:43 +0000 [r76841-76852] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Revert trivial whitespace change (for
- testing)
-
- * channels/chan_skinny.c: Trivial whitespace change to test
- comitting...
-
-2007-07-24 17:05 +0000 [r76807] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 76803 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r76803 | qwell | 2007-07-24 11:32:20 -0500 (Tue, 24 Jul 2007) | 3
- lines Don't create the Asterisk channel until we are starting the
- PBX on it. (ASA-2007-018) ........
-
-2007-07-24 16:42 +0000 [r76804] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 76801 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r76801 | mmichelson | 2007-07-24 11:26:58 -0500 (Tue, 24 Jul
- 2007) | 13 lines Added a membercount variable to call_queue
- struct which keeps track of the number of logged in members in a
- particular queue. This makes it so that the 'n' option for
- Queue() can act properly depending on which strategy is used. If
- the strategy is roundrobin, rrmemory, or ringall, we want to ring
- each phone once before moving on in the dialplan. However, if any
- other strategy is used, we will only ring one phone since it
- cannot be guaranteed that a different phone will ring on
- subsequent attempts to ring a phone. As a side effect of this,
- the QUEUE_MEMBER_COUNT dialplan function now just reads the
- membercount variable instead of traversing through the member
- list to figure out how many members there are. Special thanks to
- blitzrage for helping to test this out. (closes issue #10127,
- reported by bcnit, patched by me, tested by blitzrage) ........
-
-2007-07-24 16:09 +0000 [r76791] Joshua Colp <jcolp@digium.com>
-
- * sounds/Makefile: Don't download/install the sound packages if
- already installed.
-
-2007-07-24 15:35 +0000 [r76785] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: The chan_skinny Dial() syntax was funky.
- You had to do Dial(Skinny/line@device) This allows you to just
- Dial(Skinny/line), as long as line isn't ambiguous. Note that
- this does not remove or deprecate the "old" syntax, as it's still
- quite useful - even moreso if shared lines get implemented.
- Initial patch by me, with some changes and suggestions from
- wedhorn. (closes issue #10263)
-
-2007-07-24 14:49 +0000 [r76755-76770] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_sip.c: two small fixes when using stun (reported by
- Marta Carbone): + externexpire was not initialized properly; +
- stunaddr was not handled properly on a sip reload
-
- * CHANGES: add documentation on nat/stun support in chan_sip
-
-2007-07-24 02:59 +0000 [r76710-76712] Joshua Colp <jcolp@digium.com>
-
- * main/manager.c: Move manager users list over to an rwlist.
-
- * res/res_agi.c: You need to put static in front of a static RWLIST
- declaration to make it really static... and don't call
- AST_RWLIST_HEAD_DESTROY on a statically declared list.
-
- * main/manager.c: Don't bother calling AST_RWLIST_EMPTY on a list
- before AST_RWLIST_TRAVERSE, it's just a double check.
-
-2007-07-23 22:41 +0000 [r76707-76709] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 76708 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r76708 | tilghman | 2007-07-23 17:38:06 -0500 (Mon, 23 Jul 2007)
- | 4 lines It was our stated intention for 1.4 that files created
- in app_voicemail should depend upon the umask. Unfortunately,
- mkstemp() creates files with mode 0600, regardless of the umask.
- This corrects that deficiency. ........
-
- * include/asterisk/agi.h, res/res_agi.c: Enhance AGI with several
- fixes: - Makes the structures handling external AGI commands a
- bit more thread-safe - Makes AGI transparently work with both
- live and hungup channels - DeadAGI is hence no longer necessary
- and is deprecated - CLI bug fixes - Commands will refuse to run
- if the channel is dead and the command is nonsensical for dead
- channels.
-
-2007-07-23 21:42 +0000 [r76706] Joshua Colp <jcolp@digium.com>
-
- * res/res_crypto.c: Clean up res_crypto module. It now uses an
- rwlist to keep the keys and it should also be thread safe now.
-
-2007-07-23 20:27 +0000 [r76703-76704] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_agi.c, UPGRADE.txt: Missed one conversion to comma
- delimiter (thanks, Juggie) and add documentation on the change to
- the Local channel name.
-
- * funcs/func_rand.c, apps/app_readfile.c, channels/chan_local.c,
- apps/app_record.c, funcs/func_env.c, funcs/func_strings.c,
- funcs/func_vmcount.c, include/asterisk/aes.h, funcs/func_logic.c,
- apps/app_exec.c, apps/app_controlplayback.c, funcs/func_odbc.c,
- apps/app_skel.c, apps/app_zapras.c, apps/app_url.c,
- apps/app_externalivr.c, apps/app_parkandannounce.c,
- apps/app_dial.c, main/pbx.c, apps/app_page.c,
- apps/app_softhangup.c, UPGRADE.txt, funcs/func_cut.c,
- apps/app_talkdetect.c, apps/app_queue.c, funcs/func_realtime.c,
- include/asterisk/app.h, apps/app_channelredirect.c,
- apps/app_macro.c, pbx/pbx_config.c, apps/app_verbose.c,
- apps/app_chanspy.c, funcs/func_callerid.c, apps/app_voicemail.c:
- Merge the dialplan_aesthetics branch. Most of this patch simply
- converts applications using old methods of parsing arguments to
- using the standard macros. However, the big change is that the
- really old way of specifying application and arguments separated
- by a comma will no longer work (e.g. NoOp,foo|bar). Instead, the
- way that has been recommended since long before 1.0 will become
- the only method available (e.g. NoOp(foo,bar).
-
-2007-07-23 19:00 +0000 [r76657] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_skinny.c: Merged revisions 76656 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r76656 | qwell | 2007-07-23 13:59:28 -0500 (Mon, 23 Jul
- 2007) | 3 lines Fix some incorrect softkey labels in messages.
- Don't try to play dialtone in some unimplemented features.
- ........
-
-2007-07-23 18:31 +0000 [r76655] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 76654 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r76654 | file | 2007-07-23 15:29:48 -0300 (Mon,
- 23 Jul 2007) | 12 lines Merged revisions 76653 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r76653 | file | 2007-07-23 15:28:13 -0300 (Mon, 23 Jul 2007) | 4
- lines (closes issue #5866) Reported by: tyler Do not force
- channel format changes when a generator is present. The generator
- may have changed the formats itself and changing them back would
- cause issues. ........ ................
-
-2007-07-23 17:58 +0000 [r76621] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_skinny.c: Merged revisions 76620 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #10276) ........ r76620 | qwell | 2007-07-23 12:57:53 -0500
- (Mon, 23 Jul 2007) | 4 lines Don't try to queue up hold/unhold
- frames on a non-existent channel. Issue 10276. ........
-
-2007-07-23 17:49 +0000 [r76619] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_morsecode.c: Merged revisions 76618 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r76618 | file | 2007-07-23 14:48:51 -0300 (Mon, 23 Jul 2007) | 2
- lines Allow app_morsecode to build on PPC Linux by putting the
- value of the digit char in an int. ........
-
-2007-07-23 14:45 +0000 [r76564] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_sip.c: add two missing entries in the replica of
- the sip_tech that does not use DTMF BEGIN frames. 1.4 seems
- correct (it does not have the two fields). However, as this bug
- shows, the current way of creating the sip_tech replica is too
- error-prone, one can easily forget to update one of the two
- entries. Perhaps it would be better to create sip_tech_info
- expliclty at module load, by doing sip_tech_info = sip_tech;
- sip_tech_info.send_digit_begin = NULL (in this case, this is
- something applicable to 1.4 as well).
-
-2007-07-23 14:38 +0000 [r76563] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 76561 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r76561 | file | 2007-07-23 11:34:21 -0300 (Mon,
- 23 Jul 2007) | 14 lines Merged revisions 76560 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r76560 | file | 2007-07-23 11:32:07 -0300 (Mon, 23 Jul 2007) | 6
- lines (closes issue #10236) Reported by: homesick Patches:
- rpid_1.4_75840.patch uploaded by homesick (license 91) Accept
- Remote Party ID on guest calls. ........ ................
-
-2007-07-23 14:37 +0000 [r76555-76562] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Mark str2dtmfmode() as currently unused to
- resolve a compiler warning and allow building under dev mode
-
- * include/asterisk.h, res/res_snmp.c, channels/chan_sip.c,
- res/res_crypto.c, res/res_convert.c, main/devicestate.c,
- include/jitterbuf.h, res/res_config_sqlite.c, main/enum.c,
- res/res_monitor.c, include/asterisk/file.h,
- include/asterisk/doxyref.h, res/res_config_odbc.c,
- res/res_indications.c, main/asterisk.c, res/res_clioriginate.c:
- (closes issue #10271) Reported by: snuffy Patches:
- doxygen-updates.diff uploaded by snuffy (license 35) Another big
- batch of doxygen documentation updates
-
- * CHANGES: note the debug and verbose changes in CHANGES
-
- * include/asterisk/logger.h, main/pbx.c, main/logger.c,
- include/asterisk/options.h, main/asterisk.c, main/cli.c: (closes
- issue #10192) Reported by: bbryant Patches:
- 20070720__core_debug_by_file.patch uploaded by bbryant (license
- 36) (with some modifications by me) Tested by: russell, bbryant
- This set of changes introduces the ability to set the core debug
- or verbose levels on a per-file basis. Interestingly enough, in
- 1.4, you have the ability to set core debug for a single file,
- but that functionality was accidentally lost in the conversion of
- the CLI commands to the new format. This patch improves upon what
- was in 1.4 by letting you set it for more than 1 file, and by
- also supporting verbose. *** Janitor Project *** This patch also
- introduces a new macro, ast_verb(), which is similar to
- ast_debug(). Setting the per file verbose value only works for
- messages that use this macro. Converting existing uses of
- ast_verbose() can be done like: if (option_debug > 2)
- ast_verbose(VERBOSE_PREFIX_3 "Something useful\n"); ...
- ast_verb(3, "Something useful\n");
-
-2007-07-23 14:18 +0000 [r76547] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_sip.c: introduce two functions, map_x_s() and
- map_s_x(), to map between integers and strings using a single
- translation table, and use them in a few places instead of ad-hoc
- routines that duplicate the table. On passing, note that
- REFER_CONFIRMED is never used, and add a few comments. Nothing to
- backport here.
-
-2007-07-23 14:02 +0000 [r76524] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Remove an unused function to resolve a
- compiler warning
-
-2007-07-23 13:46 +0000 [r76523] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_skinny.c, configure,
- include/asterisk/autoconfig.h.in, configure.ac: Use autoconf
- logic to determine byte swapping macro presence. This should now
- also use other macros if present.
-
-2007-07-23 13:29 +0000 [r76521] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_sip.c: move "sip prunte realtime ..." and "sip set
- debug ... " to NEW_CLI style.
-
-2007-07-23 13:24 +0000 [r76520] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_skinny.c: Merged revisions 76519 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r76519 | file | 2007-07-23 10:23:09 -0300 (Mon, 23 Jul
- 2007) | 6 lines (closes issue #10268) Reported by: mvanbaak
- Patches: chan_skinny_openbsd.diff uploaded by mvanbaak (license
- 7) Add another OS that has to use the Macros for byte ordering.
- ........
-
-2007-07-23 12:29 +0000 [r76486] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 76485 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r76485 | russell | 2007-07-23 07:25:01 -0500 (Mon, 23 Jul 2007) |
- 6 lines Use a signed integer for storing the number of bytes in
- the packet read from the network. Using an unsigned value here
- made it impossible to handle an error returned from recvfrom().
- Furthermore, in the case that recvfrom() did return an error,
- this would cause a crash due to a heap overflow. (closes issue
- #10265, reported by and fix suggested by timrobbins) ........
-
-2007-07-23 03:10 +0000 [r76313-76467] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_sip.c: Add some documentation on the sipregistry
- states and the handling of the sip_register structures. This
- commit only changes comments and whitespace.
-
- * channels/chan_sip.c: add a bit of comments on internal functions.
-
- * channels/chan_sip.c: rewrite "sip show {channels|subscriptions}"
- CLI handler using the new-style cli format. No functional
- changes, nothing to backport.
-
- * channels/chan_sip.c: Make sip_destroy() return NULL so the caller
- can do things like foo = sip_destroy(foo); and reduce the chance
- of bugs due to dangling pointers. Also remove a duplicate
- prototype for the function. nothing to backport.
-
- * channels/chan_sip.c: add two comment blocks, one on reusing
- nonces, and one on the handling of an 'authpeer' local variable.
-
- * channels/chan_sip.c: comment and slightly restructure
- handle_request() in the part that handles responses, so that
- there is a common exit point. Mark two places where probably we
- could return -1 instead of 0 to report an error to the caller.
- (change triggered by investigations on how the 'SIP_PKT_IGNORE'
- field was used). nothing to backport from this commit
-
- * channels/chan_sip.c: remove unused argument from
- handle_invite_replaces(), and also leftover SIP_PKT_* stuff from
- the previous commit.
-
- * channels/chan_sip.c: Cleanup of flags used in struct sip_request,
- moving them to individual variables. Apart from SIP_PKT_IGNORE
- which was used a zillion times, the other two are used seldom. On
- passing: - move the arrays to the end of struct sip_request, so a
- (small) buffer overflow is less likely to overwrite the other
- fields; - note that the 'ignore' argument to
- handle_invite_replaces() is not used and should be removed (will
- be done in a separate commit). Nothing to backport in this
- change.
-
- * channels/chan_sip.c: move two per-packet flags to proper
- variables.
-
- * channels/chan_sip.c: minor clarification on the usage of SIP_*
- flags. Also correct some items that were misclassified.
-
- * channels/chan_sip.c: document the way sipdebug works, and
- implement it through variables and not flags. NOTE: The old
- behaviour (preserved in this commit) is that if sipdebug is set
- in the config file, it can only be disabled by reloading the
- config. I am not sure if this is accidental or voluntary, but it
- is really unconvenient and I think it should be handled in the
- same way as other options i.e. consider requests from the config
- file or the cli (or the command line) to be fully equivalent and
- act on the same status variable.
-
- * channels/chan_sip.c: move the SIP_REALTIME flag to a field in the
- user/peer structure.
-
- * channels/chan_sip.c: Add a note to document how the temporary
- 'pvt' should be initialized before using it. I am unclear on the
- details right now so i hope someone can comment more. The obvious
- (and lazy) approach would be to bzero() all of it (except for the
- string pool), but isn't that too much work ? Feedback wanted
- here...
-
-2007-07-21 14:39 +0000 [r76296] Joshua Colp <jcolp@digium.com>
-
- * include/asterisk/utils.h, configure,
- include/asterisk/autoconfig.h.in, configure.ac, main/utils.c: Add
- support for using /dev/urandom to get random numbers on systems
- that support it.
-
-2007-07-21 09:35 +0000 [r76229-76279] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_sip.c: whoops... was setting needdestroy on the
- wrong dialog. (spotted by a diff with my own branch)
-
- * channels/chan_sip.c: more two more flags to proper variables:
- ALREADYGONE and NEEDDESTROY.
-
- * channels/chan_sip.c: use explicit variables for things that don't
- need to be stored in ast_flags. First victim is 'SIP_NO_HISTORY'
- replaced by a 'do_history' field in the sip_pvt structure.
-
- * channels/chan_sip.c: Use ast_str_append() instead of
- ast_build_string() to construct the sdp messages. Overall the
- code is slightly more readable (because the string is fully
- described by a single pointer), and more efficient (because the
- length is stored explicitly so you don't need to do strlen()). (I
- have been using this code for almost a year now.) I wish we had
- infix string operators to do this sort of things! Nothing to
- backport from this change.
-
-2007-07-21 01:25 +0000 [r76224] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_sip.c: We have two 'technology' descriptors for a
- SIP channel, so define and use a macro to determine whether we
- are pointing to one of them, so when one goes away (or a new one
- appears) we don't have to touch all the code.
-
-2007-07-21 01:08 +0000 [r76222] Steve Murphy <murf@digium.com>
-
- * apps/app_queue.c: One small documentation update made to
- accompany 10154, the upgrading of the queue ringing to allow
- periodic announcments
-
-2007-07-21 01:01 +0000 [r76221] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_sip.c, configs/sip.conf.sample: Enhance NAT support
- as discussed on the -dev list, i.e.: + extensive documentation
- changes both in sip.conf.sample and in the source; + allow
- "externip" and "externhost" to include a port number as well; +
- allow "bindaddr" to have a port number (making bindport
- unnecessary, even though it is still present for backward
- compatibility); + introduce the new "stunaddr" parameter to
- specify an STUN server to be used from the main SIP socket; +
- extend the "sip show settings" output to show all the above.
- Internally: + change related data structures from struct in_addr
- to struct sockaddr_in to store the port numbers as well; +
- reorganize ast_sip_ouraddrfor() (should also be renamed to
- sip_ouraddrfor() because it is not a generic API, though it might
- become so if called with a socket as an additional argument, in
- which case it can be moved elsewhere). As mentioned in the
- documentation, media sessions still do not use STUN so the port
- numbers may still be incorrect when Asterisk is behind a NAT On
- passing, some of the debugging messages printing media addresses
- are probably using the wrong values, but this will be
- checked/fixed in a subsequent commit if needed. Part of the
- following chunk in the function that handles a "sip reload" is
- probably needed on previous versions as well, to avoid leaking
- the memory used for the "localaddr" list: @@ -17244,13 +17274,17
- @@ /* Reset IP addresses */ memset(&bindaddr, 0,
- sizeof(bindaddr)); + memset(&stunaddr, 0, sizeof(stunaddr)); +
- memset(&internip, 0, sizeof(internip)); + /* Free memory for
- local network address mask */ + ---> ast_free_ha(localaddr);
- <----- memset(&localaddr, 0, sizeof(localaddr));
- memset(&externip, 0, sizeof(externip)); memset(&default_prefs, 0
- , sizeof(default_prefs));
-
-2007-07-21 00:57 +0000 [r76220] Steve Murphy <murf@digium.com>
-
- * apps/app_queue.c: This update was supplied in 10154; to allow
- announcemnts if the 'r' option (ringing) is provided.
-
-2007-07-20 22:25 +0000 [r76216] Jason Parker <jparker@digium.com>
-
- * configs/say.conf.sample, apps/app_playback.c: Add support for
- default "say mode" (whether to use the "old" method or "new"
- method. "new" method being config file) Add support for
- autocomplete of "say load" CLI command. Patch by IgorG (closes
- issue #10243)
-
-2007-07-20 21:41 +0000 [r76213] Steve Murphy <murf@digium.com>
-
- * /, sounds/Makefile: Merged revisions 76211 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r76211 | murf | 2007-07-20 15:36:05 -0600 (Fri, 20 Jul 2007) | 1
- line This patch from 10249 is worth applying! It prevents
- downloading sound files if they are already downloaded. Darn
- Practical, if you ask me ........
-
-2007-07-20 21:04 +0000 [r76175-76179] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_skinny.c: Merged revisions 76174 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r76174 | qwell | 2007-07-20 15:32:55 -0500 (Fri, 20 Jul
- 2007) | 2 lines It's possible for sub->owner to be NULL here if
- you cancel the call immediately after/during sending a digit.
- ........
-
-2007-07-20 18:44 +0000 [r76140] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_directory.c: Merged revisions 76139 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r76139 | mmichelson | 2007-07-20 13:42:27 -0500 (Fri, 20 Jul
- 2007) | 6 lines When using users.conf for the entries in the
- directory, if multiple users had the same last name, only the
- first user listed would be available in the directory. (closes
- issue #10200, reported by mrskippy, patched by me) ........
-
-2007-07-20 18:28 +0000 [r76138] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /: Merged revisions 76132 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r76132 | russell | 2007-07-20 13:22:24 -0500 (Fri, 20 Jul 2007) |
- 6 lines Use the define that specifies the default length of an
- artificially created DTMF digit in the ast_senddigit() function.
- The define is set to 100ms by default, which is the same thing
- that this function was using. But, using the define lets changes
- take effect in this case, as well as the others where it was
- already used. ........
-
-2007-07-20 17:21 +0000 [r76055-76091] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 76087 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r76087 | file | 2007-07-20 14:20:09 -0300 (Fri,
- 20 Jul 2007) | 14 lines Merged revisions 76080 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r76080 | file | 2007-07-20 14:16:48 -0300 (Fri, 20 Jul 2007) | 6
- lines (closes issue #10247) Reported by: fkasumovic Patches:
- chan_sip.patch uploaded by fkasumovic (license #101) Drop any
- peer realm authentication entries when reloading so multiple
- entries do not get added to the peer. ........ ................
-
- * /, res/res_convert.c: Merged revisions 76067 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r76067 | file | 2007-07-20 14:10:17 -0300 (Fri, 20 Jul 2007) | 6
- lines (closes issue #10246) Reported by: fkasumovic Patches:
- res_conver.patch uploaded by fkasumovic (license #101) Use the
- last occurance of . to find the extension, not the first
- occurance. ........
-
- * channels/chan_sip.c: It is impossible for the externhost variable
- to not exist, it is however possible for it to be empty.
-
-2007-07-20 15:06 +0000 [r76034-76037] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_sip.c: Don't use a field size for the last argument
- of printf format, because in this case the string is left-aligned
- and it is not truncated anyways. Omitting the field size prevents
- the generation of trailing whitespace, which makes the string fit
- in smaller windows.
-
- * channels/chan_sip.c: Extend the 'network settings' section with
- indication on the localnet settings (requires the change in SVN
- 76034), and also give an indication on whether/why/how the
- remapping of addresses in SIP message is done or not. I think
- this is especially useful for debugging the configuration, as the
- address remapping depends on a combination of at least 3
- parameters (localnet, externhost, externip) and successful DNS
- lookup. An example of the output of this section is below:
- Network Settings: --------------------------- SIP address
- remapping: Enabled using externhost Externhost: foo.dyndns.net
- Externip: 80.64.128.23:0 Externrefresh: 10 Internal IP:
- 12.34.56.78:5060 Localnet: 192.168.0.0/255.255.0.0
- 10.0.0.0/255.0.0.0 I leave to the community the judgement if the
- above info is a useful addition for 1.4. It is not a bugfix, but
- it is neither a new feature, only a useful diagnostic tool. Note
- that I would like to move there also the bindaddress/port
- information, in the usual addr:port format e.g. Bindaddress:
- 0.0.0.0:5060 so that network information is all in one place.
-
- * include/asterisk/acl.h, main/acl.c: expose struct ast_ha so
- external code can do things such as printing it (e.g. chan_sip.c
- in a subsequent commit). Obviously exposing the internals of a
- data structure is far from ideal (especially in a case like this
- where the implementation is very inefficient and will need to be
- changed at some point). On the other hand, it was also unclear
- what additional APIs should we provide instead, and because
- exposing the stucture has no impact on source and binary
- compatibility, this seemed to me the best option at this time.
-
-2007-07-20 01:54 +0000 [r76015] Tilghman Lesher <tlesher@digium.com>
-
- * main/logger.c: Reduce some logging contention by switching
- several locks over to rwlocks
-
-2007-07-19 23:24 +0000 [r75982-75983] Steve Murphy <murf@digium.com>
-
- * apps/app_dial.c, include/asterisk/utils.h, channels/chan_local.c,
- channels/chan_sip.c, include/asterisk/dundi.h,
- res/res_features.c, include/asterisk/chanspy.h,
- include/asterisk/speech.h, channels/iax2-provision.c,
- include/asterisk/cdr.h, include/asterisk/channel.h,
- res/res_musiconhold.c, channels/chan_iax2.c, main/rtp.c,
- channels/iax2-provision.h, main/loader.c,
- include/asterisk/abstract_jb.h, include/asterisk/features.h,
- main/channel.c, include/asterisk/app.h, funcs/func_odbc.c,
- include/asterisk/module.h, include/asterisk/jabber.h,
- apps/app_minivm.c, main/app.c, pbx/pbx_dundi.c,
- apps/app_mixmonitor.c, apps/app_voicemail.c: After some study,
- thought, comparing, etc. I've backed out the previous universal
- mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit
- version of ast_flags (ast_flags64), and 64-bit versions of the
- test-flag, set-flag, etc. macros, and an app_parse_options64
- routine, and I use these in app_dial alone, to eliminate the
- 30-option limit it had grown to meet. There is room now for 32
- more options and flags. I was heavily tempted to implement some
- of the other ideas that were presented, but this solution does
- not intro any new versions of dial, doesn't have a different API,
- has a minimal/zero impact on code outside of dial, and doesn't
- seriously (I hope) affect the code structure of dial. It's the
- best I can think of right now. My goal was NOT to rewrite dial. I
- leave that to a future, coordinated effort.
-
- * apps/app_queue.c: This repairs a 'warning: ISO C90 forbids mixed
- declarations and code' message that cripples my dev-mode enabled
- build
-
-2007-07-19 19:02 +0000 [r75977-75979] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 75978 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r75978 | mmichelson | 2007-07-19 13:59:30 -0500 (Thu, 19 Jul
- 2007) | 3 lines The diff on this looks pretty big but all I did
- was remove a pointless if statement (always evaluates true).
- ........
-
- * /, apps/app_queue.c: Merged revisions 75969 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r75969 | mmichelson | 2007-07-19 11:26:10 -0500 (Thu, 19 Jul
- 2007) | 10 lines Changes in handling return values of several
- functions in app_queue. This all started as a fix for issue
- #10008 but now includes all of the following changes: 1.
- Simplifying the code to handle positive return values from ast
- API calls. 2. Removing the background_file function. 3. The fix
- for issue #10008 (closes issue #10008, reported and patched by
- dimas) ........
-
-2007-07-19 15:59 +0000 [r75911-75930] Russell Bryant <russell@digium.com>
-
- * res/res_agi.c: (closes issue #10210, reported and patched by
- juggie) This merges the trunk only part of the patches from this
- issue. In 1.4, res_agi will issue a warning if you try to use
- DeadAGI on a channel that is not hung up. Now, in trunk, it just
- plain won't let you do it.
-
- * /, channels/chan_iax2.c: Merged revisions 75928 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r75928 | russell | 2007-07-19 10:53:15 -0500
- (Thu, 19 Jul 2007) | 14 lines Merged revisions 75927 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r75927 | russell | 2007-07-19 10:49:42 -0500 (Thu, 19
- Jul 2007) | 6 lines When processing full frames, take sequence
- number wraparound into account when deciding whether or not we
- need to request retransmissions by sending a VNAK. This code
- could cause VNAKs to be sent erroneously in some cases, and to
- not be sent in other cases when it should have been. (closes
- issue #10237, reported and patched by mihai) ........
- ................
-
- * main/acl.c: Remove some debug code that was added in revision
- 75894, which removed some other debug code. :)
-
-2007-07-19 12:38 +0000 [r75873-75894] Luigi Rizzo <rizzo@icir.org>
-
- * main/acl.c: comment out some terribly expensive debugging code in
- the body of ast_apply_ha()
-
- * channels/chan_sip.c: print more of the network settings
- (externip, externhost etc.) in the "sip show settings" cli
- output. I have put these in a separate section, probably even
- bindaddr and SIP port should go there. There are more things to
- add here e.g. localnet and so on.
-
- * channels/chan_sip.c: document the use of externip, externhost and
- other nat-related options, as well as the handling of the sip
- socket.
-
- * channels/chan_sip.c: ast_sip_ouraddrfor() never fails, so make it
- void and remove the code that would never be called.
-
- * channels/chan_sip.c: portability fix: use %f instead of %lf when
- printing double. The l is useless.
-
-2007-07-19 04:45 +0000 [r75841-75857] Tilghman Lesher <tlesher@digium.com>
-
- * channels/misdn/ie.c, channels/misdn/isdn_lib.c: Allow chan_misdn
- to build in dev-mode
-
- * apps/app_rpt.c: Fix trunk where I broke it earlier (for
- ast_strftime branch)
-
-2007-07-18 23:00 +0000 [r75808] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_skinny.c: Merged revisions 75807 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r75807 | qwell | 2007-07-18 17:59:18 -0500 (Wed, 18 Jul
- 2007) | 1 line Need to make sure we set milliseconds and
- timestamp - pointed out by the recent ast_ time stuff from
- Tilghman ........
-
-2007-07-18 22:52 +0000 [r75806] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: I thought I noticed a memory leak earlier
- when I saw that the contents of this list were not destroyed when
- the module is unloaded. However, after reading the code related
- to the use of this list a lot today, I realized that it isn't
- necessary. So, I have added a comment to explain why it isn't
- necessary.
-
-2007-07-18 22:40 +0000 [r75805] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_iax2.c: Change IAX variables to use datastores
- (closes issue #9315)
-
-2007-07-18 21:10 +0000 [r75761] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 75759 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r75759 | russell | 2007-07-18 16:09:46 -0500
- (Wed, 18 Jul 2007) | 13 lines Merged revisions 75757 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r75757 | russell | 2007-07-18 16:09:13 -0500 (Wed, 18
- Jul 2007) | 5 lines When traversing the queue of frames for
- possible retransmission after receiving a VNAK, handle sequence
- number wraparound so that all frames that should be retransmitted
- actually do get retransmitted. (issue #10227, reported and
- patched by mihai) ........ ................
-
-2007-07-18 20:43 +0000 [r75750] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 75749 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r75749 | tilghman | 2007-07-18 15:40:18 -0500
- (Wed, 18 Jul 2007) | 10 lines Merged revisions 75748 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r75748 | tilghman | 2007-07-18 15:31:36 -0500 (Wed, 18
- Jul 2007) | 2 lines Store prior to copy (closes issue #10193)
- ........ ................
-
-2007-07-18 20:18 +0000 [r75714-75734] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_skinny.c: Merged revisions 75732 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r75732 | qwell | 2007-07-18 15:17:27 -0500 (Wed, 18 Jul
- 2007) | 1 line Umm, why are we transmitting dialtone on cfwdall?
- ........
-
- * /, channels/chan_skinny.c: Merged revisions 75711 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #9245) ........ r75711 | qwell | 2007-07-18 14:54:32 -0500
- (Wed, 18 Jul 2007) | 4 lines Fixes for 7935/7936 conference
- phones. Issue 9245, patch by slimey. ........
-
-2007-07-18 19:51 +0000 [r75710] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_skinny.c: Merged revisions 75707 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #9887) ........ r75707 | qwell | 2007-07-18 14:48:12 -0500
- (Wed, 18 Jul 2007) | 4 lines Fix issues with new 79x1 phones.
- Issue 9887, patches by DEA ........
-
-2007-07-18 19:50 +0000 [r75709] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: convert some lines indented with spaces to
- tabs
-
-2007-07-18 19:47 +0000 [r75706] Tilghman Lesher <tlesher@digium.com>
-
- * main/say.c, funcs/func_strings.c, main/utils.c,
- apps/app_alarmreceiver.c, include/asterisk/localtime.h,
- cdr/cdr_pgsql.c, cdr/cdr_radius.c, channels/chan_iax2.c,
- main/loader.c, main/cli.c, cdr/cdr_csv.c, main/cdr.c,
- channels/chan_phone.c, main/manager.c, channels/chan_skinny.c,
- cdr/cdr_sqlite.c, apps/app_minivm.c, channels/misdn/ie.c,
- main/logger.c, main/http.c, main/stdtime/localtime.c,
- cdr/cdr_odbc.c, apps/app_rpt.c, include/asterisk/options.h,
- channels/chan_mgcp.c, cdr/cdr_manager.c, main/pbx.c,
- channels/chan_zap.c, funcs/func_timeout.c, channels/chan_sip.c,
- channels/chan_agent.c, channels/iax2-parser.c,
- apps/app_playback.c, cdr/cdr_tds.c, main/callerid.c,
- res/snmp/agent.c, apps/app_sms.c, include/asterisk/strings.h,
- main/asterisk.c, apps/app_voicemail.c: Merge in ast_strftime
- branch, which changes timestamps to be accurate to the
- microsecond, instead of only to the second
-
-2007-07-18 17:59 +0000 [r75659] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 75658 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r75658 | dhubbard | 2007-07-18 12:56:30 -0500
- (Wed, 18 Jul 2007) | 9 lines Merged revisions 75657 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r75657 | dhubbard | 2007-07-18 12:48:33 -0500 (Wed, 18
- Jul 2007) | 1 line removed the word 'pissed' from ast_log(...)
- function call for BE-90 ........ ................
-
-2007-07-18 15:45 +0000 [r75586-75624] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 75623 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r75623 | file | 2007-07-18 12:44:02 -0300 (Wed, 18 Jul 2007) | 2
- lines Few more places that needs to check for onhold state.
- ........
-
- * /, channels/chan_sip.c: Merged revisions 75621 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r75621 | file | 2007-07-18 12:41:06 -0300 (Wed, 18 Jul 2007) | 5
- lines (closes issue #10165) Reported by: elandivar It is possible
- for hold status to exist without call limits set, so we need to
- ensure update_call_counter is executed regardless. ........
-
- * /, channels/chan_h323.c: Merged revisions 75619 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r75619 | file | 2007-07-18 12:25:45 -0300 (Wed, 18 Jul 2007) | 2
- lines Don't bother reloading chan_h323 if it did not load
- successfully in the first place. This would otherwise cause a
- crash. ........
-
- * funcs/func_curl.c: Clean up func_curl a bit.
-
-2007-07-18 14:35 +0000 [r75585] Steve Murphy <murf@digium.com>
-
- * main/channel.c, channels/chan_sip.c, res/res_features.c,
- pbx/pbx_dundi.c, main/rtp.c, apps/app_voicemail.c: This corrects
- the problem with flags and %lld formats on 64-bit machines, where
- uint64_t is NOT acceptable for %lld, and also works on 32-bit
- machines. At least, with gcc.
-
-2007-07-18 14:20 +0000 [r75566-75584] Joshua Colp <jcolp@digium.com>
-
- * /, pbx/pbx_dundi.c: Merged revisions 75583 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r75583 | file | 2007-07-18 11:18:53 -0300 (Wed, 18 Jul 2007) | 5
- lines (closes issue #10224) Reported by: irroot Record the
- threadid of each running thread before shutting them down as the
- thread themselves may change the value. ........
-
- * channels/chan_sip.c, channels/chan_agent.c, pbx/pbx_realtime.c,
- apps/app_voicemail.c: Minor code tweaks. Variables were being
- checked wrong in some situations and didn't need to be checked in
- others.
-
-2007-07-18 12:38 +0000 [r75530] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 75529 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r75529 | tilghman | 2007-07-18 07:29:41 -0500 (Wed, 18 Jul 2007)
- | 2 lines Using a freed frame causes crashes (closes issue #9317)
- ........
-
-2007-07-17 21:52 +0000 [r75505] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_ael.c: Spotted this bug today myself, trying to reproduce
- a BE bug. Use a vert bar instead of a comma, when calling RAND.
-
-2007-07-17 20:58 +0000 [r75446-75451] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_skinny.c: Merged revisions 75450 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r75450 | russell | 2007-07-17 15:57:56 -0500
- (Tue, 17 Jul 2007) | 11 lines Merged revisions 75449 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r75449 | russell | 2007-07-17 15:57:09 -0500 (Tue, 17
- Jul 2007) | 3 lines Properly check for the length in the skinny
- packet to prevent an invalid memcpy. (ASA-2007-016) ........
- ................
-
- * channels/iax2-parser.h, /, channels/chan_iax2.c,
- channels/iax2-parser.c: Merged revisions 75445 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r75445 | russell | 2007-07-17 15:48:21 -0500
- (Tue, 17 Jul 2007) | 13 lines Merged revisions 75444 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r75444 | russell | 2007-07-17 15:45:27 -0500 (Tue, 17
- Jul 2007) | 5 lines Ensure that when encoding the contents of an
- ast_frame into an iax_frame, that the size of the destination
- buffer is known in the iax_frame so that code won't write past
- the end of the allocated buffer when sending outgoing frames.
- (ASA-2007-014) ........ ................
-
-2007-07-17 20:42 +0000 [r75438-75442] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 75441 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r75441 | russell | 2007-07-17 15:42:12 -0500
- (Tue, 17 Jul 2007) | 12 lines Merged revisions 75440 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r75440 | russell | 2007-07-17 15:41:41 -0500 (Tue, 17
- Jul 2007) | 4 lines After parsing information elements in IAX
- frames, set the data length to zero, so that code later on does
- not think it has data to copy. (ASA-2007-015) ........
- ................
-
-2007-07-17 20:05 +0000 [r75406] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 75405 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r75405 | mmichelson | 2007-07-17 15:03:48 -0500 (Tue, 17 Jul
- 2007) | 6 lines Fixing an error I made earlier. ast_fileexists
- can return -1 on failure, so I need to be sure that we only enter
- the if statement if it is successful. Related to my fix to issue
- #10186 ........
-
-2007-07-17 20:01 +0000 [r75402-75404] Russell Bryant <russell@digium.com>
-
- * main/pbx.c, /: Merged revisions 75403 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r75403 | russell | 2007-07-17 15:01:12 -0500 (Tue, 17 Jul 2007) |
- 12 lines (closes issue #10209) Reported by: juggie Patches:
- 10209-trunk-2.patch uploaded by juggie Tested by: juggie,
- blitzrage In ast_pbx_run(), mark a channel as hung up after an
- application returned -1, or when it runs out of extensions to
- execute. This is so that code can detect that this channel has
- been hung up for things like making sure DeadAGI is used on
- actual dead channels, and is beneficial for other things, like
- making sure someone doesn't try to start spying on a channel that
- is about to go away. ........
-
- * /, res/res_agi.c: Merged revisions 75401 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r75401 | russell | 2007-07-17 14:45:07 -0500 (Tue, 17 Jul 2007) |
- 3 lines Remove a duplicated newline character in AGI debug
- output. (closes issue #10207, patch by seanbright) ........
-
-2007-07-17 19:40 +0000 [r75400] Steve Murphy <murf@digium.com>
-
- * apps/app_dial.c, include/asterisk/utils.h, channels/chan_local.c,
- channels/chan_sip.c, include/asterisk/dundi.h,
- res/res_features.c, include/asterisk/chanspy.h,
- include/asterisk/speech.h, channels/iax2-provision.c,
- include/asterisk/cdr.h, include/asterisk/channel.h,
- res/res_musiconhold.c, channels/chan_iax2.c, main/rtp.c,
- channels/iax2-provision.h, main/loader.c,
- include/asterisk/features.h, include/asterisk/abstract_jb.h,
- main/channel.c, funcs/func_odbc.c, include/asterisk/module.h,
- include/asterisk/jabber.h, apps/app_minivm.c, utils/ael_main.c,
- pbx/pbx_dundi.c, apps/app_mixmonitor.c, utils/check_expr.c,
- apps/app_voicemail.c: via 10206, I have added an option (e) to
- Dial to allow the h exten to get run on peer. Had to upgrade
- ast_flag stuff to 64 bits to do this.
-
-2007-07-17 14:48 +0000 [r75381] Joshua Colp <jcolp@digium.com>
-
- * include/asterisk/config.h: Make trunk build once again.
-
-2007-07-17 14:32 +0000 [r75365-75379] Luigi Rizzo <rizzo@icir.org>
-
- * include/asterisk/config.h, main/config.c: Introduce
- ast_parse_arg() , a generic function to parse strings in a
- consistent way. This is meant to replace the custom code which is
- repeated all over the place in the various files when parsing
- config files, CLI entries and other string information. Right now
- the code supports parsing int32, uint32 and sockaddr_in with
- optional default values and bound checks. It contains minimal
- error checking, but that can be easily extended as the need
- arises. Being a new API i am introducing this only in trunk,
- though I believe that once the interface has been ironed out it
- might become a worthwhile addition to 1.4 as well - basically,
- the first time we will need to fix a piece of argument parsing
- code, we might as well bring in this change and use the new API
- instead.
-
- * apps/app_minivm.c: Initialize a variable to avoid a warning when
- the compiler (and/or the optimization level) may think it is used
- uninitialized. The code was indeed correct, but unfortunately the
- result of some compiler checks such as -Wunused and
- -Wuninitialized depends heavily on the optimization level.
-
-2007-07-17 12:01 +0000 [r75351] Jason Parker <jparker@digium.com>
-
- * apps/app_dial.c: Fix an incorrect parenthesization (TODO: Find a
- better word) in app_dial Pointed out by Fanzhou Zhao Closes issue
- #10216
-
-2007-07-16 20:58 +0000 [r75307] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, main/dns.c: Merged revisions 75306 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r75306 | kpfleming | 2007-07-16 15:53:24 -0500
- (Mon, 16 Jul 2007) | 11 lines Merged revisions 75304 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r75304 | kpfleming | 2007-07-16 15:46:58 -0500 (Mon, 16
- Jul 2007) | 3 lines provide proper copyright/license attribution
- for this structure that was copied from a BSD-licensed header
- file long, long ago... ........ ................
-
-2007-07-16 18:38 +0000 [r75255-75260] Joshua Colp <jcolp@digium.com>
-
- * main/pbx.c, include/asterisk/pbx.h: Change the function name
- slightly... just for kpfleming!
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac: Add in
- check for the GCC attribute deprecated. It may be used soon!
-
- * funcs/func_enum.c, funcs/func_rand.c, main/pbx.c,
- funcs/func_curl.c, funcs/func_version.c, funcs/func_cut.c,
- funcs/func_vmcount.c, include/asterisk/pbx.h,
- funcs/func_realtime.c: For my next trick I will make it so
- dialplan functions no longer need to call ast_module_user_add and
- ast_module_user_remove. These are now called in the ast_func_read
- and ast_func_write functions outside of the module.
-
-2007-07-16 18:18 +0000 [r75254] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 75253 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r75253 | mmichelson | 2007-07-16 13:16:15 -0500 (Mon, 16 Jul
- 2007) | 8 lines Restoring functionality from 1.2 wherein
- Retrydial will not exit if there is no announce file specified.
- This change makes it so that if there is no announce file
- specified, the application will continue until finished (or
- caller hangs up). If a bogus announce file is specified, then a
- warning message will be printed saying that the file could not be
- found, but execution will still continue. (closes issue #10186,
- reported by jon, patched by me) ........
-
-2007-07-16 15:57 +0000 [r75183-75227] Joshua Colp <jcolp@digium.com>
-
- * apps/app_verbose.c: I found this sillyness when I did my
- ast_module_user conversion. Return immediately if no data was
- passed to the Verbose application.
-
- * apps/app_readfile.c, apps/app_record.c, apps/app_sayunixtime.c,
- apps/app_test.c, apps/app_alarmreceiver.c, apps/app_image.c,
- apps/app_chanisavail.c, apps/app_ices.c, apps/app_exec.c,
- apps/app_skel.c, apps/app_zapscan.c, apps/app_dumpchan.c,
- apps/app_zapras.c, apps/app_amd.c, apps/app_url.c,
- apps/app_externalivr.c, apps/app_milliwatt.c, apps/app_dial.c,
- main/pbx.c, apps/app_page.c, apps/app_privacy.c, apps/app_echo.c,
- apps/app_softhangup.c, apps/app_disa.c, apps/app_morsecode.c,
- apps/app_talkdetect.c, apps/app_transfer.c, apps/app_db.c,
- apps/app_playback.c, apps/app_speech_utils.c,
- apps/app_zapbarge.c, apps/app_waitforring.c, apps/app_sendtext.c,
- apps/app_macro.c, apps/app_zapateller.c, apps/app_chanspy.c,
- apps/app_mixmonitor.c, apps/app_cdr.c, apps/app_voicemail.c,
- apps/app_meetme.c, apps/app_dictate.c, apps/app_authenticate.c,
- apps/app_userevent.c, apps/app_followme.c,
- apps/app_controlplayback.c, apps/app_osplookup.c,
- apps/app_setcallerid.c, apps/app_minivm.c, apps/app_mp3.c,
- apps/app_directory.c, apps/app_rpt.c, apps/app_ivrdemo.c,
- apps/app_parkandannounce.c, apps/app_adsiprog.c,
- apps/app_while.c, apps/app_nbscat.c, apps/app_read.c,
- apps/app_festival.c, apps/app_system.c, apps/app_getcpeid.c,
- apps/app_queue.c, apps/app_channelredirect.c, apps/app_forkcdr.c,
- apps/app_flash.c, apps/app_directed_pickup.c, apps/app_sms.c,
- include/asterisk/pbx.h, apps/app_senddtmf.c, apps/app_stack.c,
- apps/app_verbose.c: Applications no longer need to call
- ast_module_user_add and ast_module_user_remove. This is now taken
- care of in the pbx_exec function outside of the application.
-
- * apps/app_readfile.c, res/res_features.c, apps/app_record.c,
- apps/app_sayunixtime.c, apps/app_test.c,
- apps/app_alarmreceiver.c, apps/app_image.c,
- apps/app_chanisavail.c, apps/app_ices.c, apps/app_exec.c,
- apps/app_zapscan.c, apps/app_dumpchan.c, apps/app_zapras.c,
- apps/app_amd.c, apps/app_url.c, apps/app_externalivr.c,
- apps/app_milliwatt.c, apps/app_dial.c, apps/app_page.c,
- apps/app_privacy.c, apps/app_echo.c, apps/app_softhangup.c,
- apps/app_disa.c, apps/app_morsecode.c, funcs/func_cut.c,
- apps/app_talkdetect.c, apps/app_transfer.c, apps/app_playback.c,
- apps/app_speech_utils.c, funcs/func_curl.c, apps/app_zapbarge.c,
- apps/app_waitforring.c, apps/app_sendtext.c, apps/app_macro.c,
- apps/app_zapateller.c, apps/app_mixmonitor.c, apps/app_chanspy.c,
- apps/app_cdr.c, apps/app_voicemail.c, apps/app_meetme.c,
- apps/app_authenticate.c, apps/app_userevent.c,
- funcs/func_vmcount.c, apps/app_followme.c, funcs/func_enum.c,
- res/res_config_odbc.c, apps/app_setcallerid.c,
- apps/app_osplookup.c, apps/app_minivm.c, res/res_agi.c,
- apps/app_mp3.c, res/res_realtime.c, apps/app_rpt.c,
- apps/app_ivrdemo.c, apps/app_parkandannounce.c,
- apps/app_adsiprog.c, apps/app_while.c, apps/app_nbscat.c,
- res/res_config_pgsql.c, apps/app_read.c, apps/app_festival.c,
- apps/app_waitforsilence.c, apps/app_system.c, apps/app_queue.c,
- apps/app_getcpeid.c, funcs/func_realtime.c, apps/app_forkcdr.c,
- apps/app_channelredirect.c, apps/app_flash.c,
- funcs/func_blacklist.c, apps/app_sms.c, apps/app_senddtmf.c,
- apps/app_stack.c, apps/app_verbose.c: It is no longer required
- for each module that deals with a channel to call
- ast_module_user_hangup_all in it's unload function. The loader
- will automatically perform this action for it.
-
-2007-07-16 02:51 +0000 [r75163-75164] Russell Bryant <russell@digium.com>
-
- * include/asterisk/devicestate.h, include/asterisk/dundi.h,
- include/asterisk/enum.h, include/asterisk/config.h,
- include/asterisk/io.h, include/asterisk/cli.h,
- include/asterisk/channel.h, include/asterisk/cdr.h,
- include/asterisk/manager.h, include/asterisk/tdd.h,
- include/asterisk/abstract_jb.h, include/asterisk/file.h,
- include/asterisk/res_odbc.h, include/asterisk/adsi.h,
- include/asterisk/crypto.h, include/asterisk/doxyref.h,
- include/asterisk/image.h, include/asterisk/musiconhold.h,
- include/asterisk/jabber.h, include/asterisk/linkedlists.h,
- include/asterisk/module.h, include/asterisk/strings.h,
- include/asterisk/pbx.h, include/asterisk/frame.h,
- include/asterisk/say.h, include/asterisk/translate.h: Merge a
- bunch of doxygen updates to header files. This includes changes
- to use the \retval tag for documenting return values, fixing
- various warnings when generating the documentation, and various
- other things. (closes issue #10203, snuffy)
-
- * funcs/func_iconv.c: Cast the 2nd argument to iconv() to a void *,
- as some systems define it as a (const char *), while others
- define it as (char *). This is done to suppress compiler warnings
- about it.
-
-2007-07-13 20:37 +0000 [r75109] Russell Bryant <russell@digium.com>
-
- * /: Merged revisions 75108 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r75108 | russell | 2007-07-13 15:36:16 -0500
- (Fri, 13 Jul 2007) | 11 lines Merged revisions 75107 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r75107 | russell | 2007-07-13 15:35:22 -0500 (Fri, 13
- Jul 2007) | 3 lines Fix a couple potential minor memory leaks.
- load_moh_classes() could return without destroying the loaded
- configuration. ........ ................
-
-2007-07-13 20:16 +0000 [r75082] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_chanspy.c: Merged revisions 75078 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r75078 | mmichelson | 2007-07-13 15:15:30 -0500
- (Fri, 13 Jul 2007) | 13 lines Merged revisions 75066 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r75066 | mmichelson | 2007-07-13 15:10:39 -0500 (Fri, 13
- Jul 2007) | 5 lines Fixed an issue where chanspy flags were
- uninitialized if no options were passed. What triggered this
- investigation was an IRC chat where some people's quiet flags
- were set while others' weren't even though none of them had
- specified the q option. ........ ................
-
-2007-07-13 20:15 +0000 [r75054-75077] Russell Bryant <russell@digium.com>
-
- * main/rtp.c: resolve a compiler warning
-
- * /, res/res_musiconhold.c: Merged revisions 75067 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r75067 | russell | 2007-07-13 15:10:40 -0500
- (Fri, 13 Jul 2007) | 14 lines Merged revisions 75059 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r75059 | russell | 2007-07-13 15:07:21 -0500 (Fri, 13
- Jul 2007) | 6 lines Ensure that adding a user to the list of
- users of a specific music on hold class is not done at the same
- time as any of the other operations on this list to prevent list
- corruption. Using the global moh_data lock for this is not ideal,
- but it is what is used to protect these lists everywhere else in
- the module, and I am only changing what is necessary to fix the
- bug. ........ ................
-
- * channels/chan_zap.c, /: Merged revisions 75053 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r75053 | russell | 2007-07-13 14:11:26 -0500
- (Fri, 13 Jul 2007) | 20 lines Merged revisions 75052 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r75052 | russell | 2007-07-13 14:10:00 -0500 (Fri, 13
- Jul 2007) | 12 lines (closes issue #9660) Reported by: mmacvicar
- Patches submitted by: bbryant, russell Tested by: mmacvicar,
- marco, arcivanov, jmhunter, explidous When using a TDM400P (and
- probably other analog cards) there was a chance that you could
- hang up and pick the phone back up where it has been long enough
- to be not considered a flash hook, but too soon such that the
- device reports that it is busy and the person on the phone will
- only hear silence. This patch makes chan_zap more tolerant of
- this and gives the device a couple of seconds to succeed so the
- person on the phone happily gets their dialtone. ........
- ................
-
-2007-07-13 16:22 +0000 [r75034] Luigi Rizzo <rizzo@icir.org>
-
- * include/asterisk/rtp.h, main/rtp.c: Small improvement to the STUN
- support so it can be used by sockets other than RTP ones. The
- main change is a new API function in main/rtp.c (see there for a
- description) int ast_stun_request(int s, struct sockaddr_in *dst,
- const char *username, struct sockaddr_in *answer) which can be
- used to send an STUN request on a socket, and optionally wait for
- a reply and store the STUN_MAPPED_ADDRESS into the 'answer'
- argument (obviously, the version that waits for a reply is
- blocking, but this is no different from DNS resolutions).
- Internally there are minor modifications to let
- stun_handle_packet() be somewhat configurable on how to parse the
- body of responses. At the moment i am not committing any change
- to the clients, but adding STUN client support is extremely
- simple, e.g. chan_sip.c could do something like this: + add a
- variable to store the stun server address; static struct
- sockaddr_in stunaddr = { 0, }; /*!< stun server address */ + add
- code to parse a config file of the form
- "stunaddr=my.stun.server.org:3478" (not shown for brevity); +
- right after binding the main sip socket, talk to the stun server
- to determine the externally visible address if
- (stunaddr.sin_addr.s_addr != 0) ast_stun_request(sipsock,
- &stunaddr, NULL, &externip); so now 'externip' is set with the
- externally visible address. so it is really trivial. Similarly
- ast_stun_request could be called when creating the RTP socket
- (possibly adding a struct sockaddr_in field in the struct ast_rtp
- to store the externalip).
-
-2007-07-12 23:02 +0000 [r74999] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 74997 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ ........
-
-2007-07-12 20:46 +0000 [r74956] Steve Murphy <murf@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 74955 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r74955 | murf | 2007-07-12 14:42:08 -0600 (Thu, 12 Jul 2007) | 1
- line This patch resolves 10143; thanks to irroot for the patch;
- looked acceptable. Let the community decide if it messes things
- up ........
-
-2007-07-12 19:19 +0000 [r74891-74923] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /: Merged revisions 74922 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r74922 | file | 2007-07-12 16:17:59 -0300 (Thu, 12 Jul 2007) | 2
- lines Whoops... didn't want this to be returned to 0 each
- iteration. ........
-
- * main/channel.c, /: Merged revisions 74888 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r74888 | file | 2007-07-12 14:16:28 -0300 (Thu, 12 Jul 2007) | 2
- lines When waiting for a digit ensure that a begin frame was
- received with it, not just an end frame. (issue #10084 reported
- by rushowr) ........
-
-2007-07-12 16:54 +0000 [r74865-74867] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_skinny.c: Merged revisions 74866 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r74866 | qwell | 2007-07-12 11:53:35 -0500 (Thu, 12 Jul
- 2007) | 1 line It helps if I actually add this stuff for the 7921
- too - otherwise it won't actually do much of anything. ........
-
- * /, channels/chan_skinny.c: Merged revisions 74864 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r74864 | qwell | 2007-07-12 11:48:49 -0500 (Thu, 12 Jul
- 2007) | 1 line Add device ID for 7921 wireless skinny phone
- ........
-
-2007-07-12 16:21 +0000 [r74850] Luigi Rizzo <rizzo@icir.org>
-
- * main/rtp.c: more cleanup, this time to stun_handle_packet().
- Among other things: + mark a potentially dangerous
- write-past-end-of-buffer + localize some variables in the block
- generating stun replies. As before, not ready yet for a merge to
- 1.4
-
-2007-07-12 15:55 +0000 [r74816] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 74815 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r74815 | file | 2007-07-12 12:53:55 -0300 (Thu,
- 12 Jul 2007) | 10 lines Merged revisions 74814 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r74814 | file | 2007-07-12 12:51:24 -0300 (Thu, 12 Jul 2007) | 2
- lines Only print out a warning for situations where it is
- actually helpful. (issue #10187 reported by denke) ........
- ................
-
-2007-07-12 15:42 +0000 [r74813] Luigi Rizzo <rizzo@icir.org>
-
- * main/rtp.c: a little bit of code cleanup to rtp.c, mostly to
- function ast_rtp_new_with_bindaddr(): 1. add comments to the
- logic of the main loop; 2. use a common exit point on failure so
- the cleanup is done only in one place; 3. handle failures in
- rtp_socket() in the main loop of the function; No functional
- changes except for #3 above, so it is not yet worthwhile merging
- this and other changes to 1.4 Once the cleanup work on this file
- will be complete (which among other things should include some
- extensions to the stun support) it might be a good thing to push
- all the changes to 1.4
-
-2007-07-11 23:05 +0000 [r74769] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 74767 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r74767 | russell | 2007-07-11 17:57:07 -0500
- (Wed, 11 Jul 2007) | 13 lines Merged revisions 74766 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r74766 | russell | 2007-07-11 17:53:26 -0500 (Wed, 11
- Jul 2007) | 5 lines The function make_trunk() can fail and return
- -1 instead of a valid new call number. Fix the uses of this
- function to handle this instead of treating it as the new call
- number. This would cause a deadlock and memory corruption.
- (possible cause of issue #9614 and others, patch by me) ........
- ................
-
-2007-07-11 21:15 +0000 [r74726] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 74722 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r74722 | mmichelson | 2007-07-11 16:14:09 -0500
- (Wed, 11 Jul 2007) | 13 lines Merged revisions 74719 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r74719 | mmichelson | 2007-07-11 16:12:30 -0500 (Wed, 11
- Jul 2007) | 5 lines The cli command "agent logoff Agent/x soft"
- did not work...at all. Now it does. (closes issue #10178,
- reported and patched by makoto, with slight modification for 1.4
- and trunk by me) ........ ................
-
-2007-07-11 21:09 +0000 [r74703-74713] Joshua Colp <jcolp@digium.com>
-
- * res/res_agi.c: Code cleanup of res_agi
-
- * res/res_smdi.c: Code cleanup of res_smdi
-
- * pbx/pbx_spool.c: Clean up pbx_spool. So many nested if
- statements...
-
- * main/udptl.c, include/asterisk/udptl.h: Use linkedlist macros for
- UDPTL protocol list.
-
-2007-07-11 18:35 +0000 [r74658] Russell Bryant <russell@digium.com>
-
- * res/res_config_odbc.c: Merged revisions 74657 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r74657 | russell | 2007-07-11 13:34:51 -0500
- (Wed, 11 Jul 2007) | 12 lines Merged revisions 74656 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r74656 | russell | 2007-07-11 13:33:23 -0500 (Wed, 11
- Jul 2007) | 4 lines Make sure that the ESCAPE immediately follows
- the condition that uses LIKE. This fixes realtime extensions with
- ODBC. (closes issue #10175, reported by stuarth, patch by me)
- ........ ................
-
-2007-07-11 18:21 +0000 [r74636-74648] Steve Murphy <murf@digium.com>
-
- * Makefile, /: Merged revisions 74642 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r74642 | murf | 2007-07-11 12:18:42 -0600 (Wed, 11 Jul 2007) | 1
- line This fixes 10172, where the entire man8 dir gets removed
- during an uninstall of asterisk ........
-
- * /: blocking 74628 from trunk... only applied to 1.4
-
-2007-07-11 17:34 +0000 [r74575-74616] Joshua Colp <jcolp@digium.com>
-
- * include/asterisk/speech.h, res/res_speech.c,
- apps/app_speech_utils.c: Use the linkedlists.h AST_LIST_NEXT
- macro for modifying the list of results.
-
- * channels/chan_phone.c, /, configure,
- include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
- 74572 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r74572 | file | 2007-07-11 14:03:08 -0300 (Wed, 11 Jul 2007) | 2
- lines Instead of figuring out kernel versions that have
- compiler.h and not... let's just use autoconf to check for it's
- presence. (issue #10174 reported by francesco_r) ........
-
-2007-07-11 16:24 +0000 [r74571] Luigi Rizzo <rizzo@icir.org>
-
- * main/rtp.c: add a bit of documentation on what the stun code in
- rtp.c does (which is very little, at the moment). Eventually,
- when the functionality is extended, the changes can be merged
- back to 1.4. At the moment this is pointless. Note, this change
- is whitespace only.
-
-2007-07-11 16:19 +0000 [r74516-74570] Joshua Colp <jcolp@digium.com>
-
- * include/asterisk/speech.h, res/res_speech.c,
- apps/app_speech_utils.c: Allow the native formats of a channel to
- influence the audio that is going to the engine. The best format
- will try to be chosen with an ultimate fallback to signed linear
- if possible.
-
- * res/res_speech.c: Can't forget to remember what format is in use
- for writing.
-
- * include/asterisk/speech.h, res/res_speech.c: Change the speech
- API to allow passing the format through to the engine.
-
- * channels/misdn/isdn_lib_intern.h: Change header a bit to get rid
- of a doxygen parse error. (issue #10177 reported by snuffy)
-
- * channels/chan_phone.c, /: Merged revisions 74515 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r74515 | file | 2007-07-11 11:09:13 -0300 (Wed, 11 Jul
- 2007) | 2 lines Only check if we need to do a SIGMA based tone
- generation if we have a card. (issue #10179 reported by mikowhy)
- ........
-
-2007-07-10 23:34 +0000 [r74477] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 74476 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r74476 | mmichelson | 2007-07-10 18:32:52 -0500 (Tue, 10 Jul
- 2007) | 5 lines Forwarding a message with IMAP storage was
- storing the message in the sender's box instead of the forwarded
- mailbox. (closes issue #10138, reported and patched by jaroth)
- ........
-
-2007-07-10 20:02 +0000 [r74375-74429] Jason Parker <jparker@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 74428 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #10158) ................ r74428 | qwell | 2007-07-10
- 14:58:53 -0500 (Tue, 10 Jul 2007) | 14 lines Merged revisions
- 74427 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r74427 | qwell | 2007-07-10 14:57:20 -0500 (Tue, 10 Jul 2007) | 6
- lines Fix an issue where it was possible to have a service level
- of over 100% Between the time recalc_holdtime and update_queue
- was called, it was possible that the call could have been hungup.
- Move both additions to the same place, so this won't happen.
- Issue 10158, initial patch by makoto, modified by me. ........
- ................
-
- * /, main/dns.c: Merged revisions 74388 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r74388 | qwell | 2007-07-10 14:10:36 -0500 (Tue, 10 Jul 2007) | 4
- lines Don't use #if to check if something is defined - use #ifdef
- instead. Pointed out by kpfleming ........
-
- * /, channels/chan_agent.c: Merged revisions 74379 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #10169) ................ r74379 | qwell | 2007-07-10
- 14:06:24 -0500 (Tue, 10 Jul 2007) | 12 lines Merged revisions
- 74376 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r74376 | qwell | 2007-07-10 14:03:45 -0500 (Tue, 10 Jul 2007) | 4
- lines Fix an issue with wrapuptime not working when using
- AgentLogin. Issue 10169, patch by makoto, with a minor mod by me
- to not re-break issue 9618 ........ ................
-
- * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
- main/dns.c: Merged revisions 74374 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #10133) ................ r74374 | qwell | 2007-07-10
- 13:39:30 -0500 (Tue, 10 Jul 2007) | 13 lines Merged revisions
- 74373 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r74373 | qwell | 2007-07-10 13:37:23 -0500 (Tue, 10 Jul 2007) | 5
- lines Use res_ndestroy on systems that have it. Otherwise, use
- res_nclose. This prevents a memleak on NetBSD - and possibly
- others. Issue 10133, patch by me, reported and tested by scw
- ........ ................
-
-2007-07-10 16:01 +0000 [r74324] Russell Bryant <russell@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 74323 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r74323 | russell | 2007-07-10 11:00:11 -0500 (Tue, 10
- Jul 2007) | 1 line fix an uninitialized variable ........
-
-2007-07-10 15:41 +0000 [r74318-74319] Jason Parker <jparker@digium.com>
-
- * /: svn revert != svn resolved Fix merged property...
-
- * apps/app_voicemail.c: Merged revisions 74317 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes
- issue #10170) ................ r74317 | qwell | 2007-07-10
- 10:38:32 -0500 (Tue, 10 Jul 2007) | 12 lines Merged revisions
- 74316 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r74316 | qwell | 2007-07-10 10:37:54 -0500 (Tue, 10 Jul 2007) | 4
- lines Fix a small typo in description in of Voicemail()
- application. Issue 10170, patch by casper. ........
- ................
-
-2007-07-10 15:32 +0000 [r74315] Russell Bryant <russell@digium.com>
-
- * res/res_config_odbc.c, /: Merged revisions 74314 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r74314 | russell | 2007-07-10 10:31:41 -0500
- (Tue, 10 Jul 2007) | 11 lines Merged revisions 74313 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r74313 | russell | 2007-07-10 10:30:20 -0500 (Tue, 10
- Jul 2007) | 3 lines Only use ESCAPE when LIKE is used. (issue
- #10075, this part reported by jmls on IRC, patch by me) ........
- ................
-
-2007-07-10 15:07 +0000 [r74272] Jason Parker <jparker@digium.com>
-
- * channels/chan_agent.c, include/asterisk/monitor.h,
- apps/app_queue.c, res/res_monitor.c: Fix building that was broken
- by recent monitor.h changes. Thanks Russell for pointing this out
- (and pointing out what I probably did to prevent gcc from fixing
- it - don't ctrl-C builds)
-
-2007-07-10 14:51 +0000 [r74263-74266] Joshua Colp <jcolp@digium.com>
-
- * /, main/app.c: Merged revisions 74265 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r74265 | file | 2007-07-10 11:50:00 -0300 (Tue,
- 10 Jul 2007) | 10 lines Merged revisions 74264 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r74264 | file | 2007-07-10 11:48:00 -0300 (Tue, 10 Jul 2007) | 2
- lines Ensure the group information category exists before trying
- to do a string comparison with it. (issue #10171 reported by
- mlegas) ........ ................
-
-2007-07-09 21:32 +0000 [r74212] Russell Bryant <russell@digium.com>
-
- * /, configure, configure.ac: Merged revisions 74211 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r74211 | russell | 2007-07-09 16:31:30 -0500 (Mon, 09
- Jul 2007) | 5 lines Update the configure script to check for a
- required function that is not present in the 1.2 version of
- libpri. This will prevent the configure script from thinking that
- it has compatible libpri support for Asterisk 1.4, when it
- actually does not because the installed version is from 1.2.
- ........
-
-2007-07-09 20:58 +0000 [r74164] Jason Parker <jparker@digium.com>
-
- * include/asterisk/monitor.h, res/res_monitor.c: (closes issue
- #7596) Reported by: julien23 Patches submitted by: julien23 Add
- the ability to disable recording the input or output streams in
- res_monitor.
-
-2007-07-09 20:54 +0000 [r74163] Russell Bryant <russell@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 74162 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r74162 | russell | 2007-07-09 15:53:46 -0500 (Mon, 09
- Jul 2007) | 9 lines (closes issue #10123) Reported by: blitzrage
- Patches submitted by: juggie, qwell, me Tested by: blitzrage When
- trying to find a music on hold class to use, try all of the
- options, instead of only the first one that is set. Also, change
- the MusicOnHold applications to not hang up on the channel when a
- class can not be found. ........
-
-2007-07-09 20:21 +0000 [r74160] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 74159 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 Closes issue
- #9186 ................ r74159 | qwell | 2007-07-09 15:19:28 -0500
- (Mon, 09 Jul 2007) | 16 lines Merged revisions 74158 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r74158 | qwell | 2007-07-09 15:18:15 -0500 (Mon, 09 Jul
- 2007) | 8 lines Several chan_zap options were not working on
- reload because they were arbitrarily disallowed when reloading
- some/most PRI options (such as signalling) was disallowed.
- Options such as polarityonanswerdelay and answeronpolarityswitch
- can safely be changed on a reload. This corrects that behavior.
- Issue 9186, patch by tzafrir. ........ ................
-
-2007-07-09 18:58 +0000 [r74125] Russell Bryant <russell@digium.com>
-
- * channels/chan_agent.c: remove an unused variable
-
-2007-07-09 18:43 +0000 [r74121-74123] Mark Michelson <mmichelson@digium.com>
-
- * /: Merged revisions 74122 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r74122 | mmichelson | 2007-07-09 13:38:28 -0500 (Mon, 09 Jul
- 2007) | 3 lines Forgot to get rid of an extraneous debug message.
- ........
-
- * /, apps/app_queue.c: Merged revisions 74120 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r74120 | mmichelson | 2007-07-09 13:32:50 -0500 (Mon, 09 Jul
- 2007) | 6 lines The n option for Queue should make the queue exit
- immediately after failure to reach any members and should not be
- dependent on the timeout value passed to Queue (closes issue
- #10127, reported by bcnit, repaired by me) ........
-
-2007-07-09 16:35 +0000 [r74084] Russell Bryant <russell@digium.com>
-
- * apps/app_queue.c: Add Queue and DestinationChannel headers to the
- AgentCalled manager event to be more like the rest of the events
- in this module. (closes issue #10114, patch by kwakwaversal)
-
-2007-07-09 15:34 +0000 [r74083] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_skinny.c: Merged revisions 74082 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r74082 | file | 2007-07-09 12:32:43 -0300 (Mon, 09 Jul
- 2007) | 2 lines Only destroy the scheduler context if it was
- allocated. (issue #10124 reported by gzero) ........
-
-2007-07-09 14:58 +0000 [r74048] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 74047 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r74047 | mmichelson | 2007-07-09 09:57:41 -0500 (Mon, 09 Jul
- 2007) | 4 lines Fixed a logic error in leave_voicemail. Pass the
- mailbox instead of the context to inbox_count when the context is
- "default." (closes issue #10135, reported by yannj, repaired by
- me) ........
-
-2007-07-09 14:50 +0000 [r74044-74046] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_skinny.c, pbx/pbx_dundi.c: Merged revisions
- 74045 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r74045 | file | 2007-07-09 11:49:05 -0300 (Mon, 09 Jul 2007) | 2
- lines Few minor thread synchronization tweaks. (issue #10124
- reported by gzero) ........
-
- * /, configure, acinclude.m4: Merged revisions 74043 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r74043 | file | 2007-07-09 11:34:33 -0300 (Mon, 09 Jul
- 2007) | 2 lines Use AC_CHECK_HEADER to check for ptlib/openh323
- to allow for cross compiling. (issue #9675 reported by zandbelt)
- ........
-
-2007-07-09 08:30 +0000 [r74024-74025] Olle Johansson <oej@edvina.net>
-
- * CHANGES: Update with new features
-
- * apps/app_dial.c, channels/chan_local.c, channels/chan_sip.c,
- include/asterisk/channel.h: Implementation of a feature that will
- disable "missed calls" counters on SIP phones. If the call is
- answered by another phone, other phones won't display the call as
- "missed". You can also add an option to the dial command so that
- you can have a "followme" scenario and not count the calls as
- "missed" when you cancel the call. Thanks to Ramon and Frank for
- feedback on this feature.
-
-2007-07-09 04:09 +0000 [r73994] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/app.h, /, channels/chan_sip.c,
- main/ast_expr2f.c, include/asterisk/channel.h,
- funcs/func_devstate.c, apps/app_voicemail.c: Merged revisions
- 73985 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r73985 | tilghman | 2007-07-08 23:03:20 -0500 (Sun, 08 Jul 2007)
- | 2 lines Doxygen formatting fixes; fixes errors while 'make
- progdocs'. (Closes issue #10104) ........
-
-2007-07-09 03:14 +0000 [r73931-73983] Joshua Colp <jcolp@digium.com>
-
- * main/cdr.c, /: Merged revisions 73980 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r73980 | file | 2007-07-09 00:13:19 -0300 (Mon, 09 Jul 2007) | 2
- lines Give Agent channel names priority when doing CDR merging.
- (issue #10011 reported by krtorio) ........
-
- * res/res_features.c: Use linkedlist macros for parking.
-
- * main/manager.c: Make sure the idText variable is empty, and put
- it in the right place for the manager ack packet. (issue #10152
- reported by srt)
-
- * /, pbx/pbx_config.c: Merged revisions 73930 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r73930 | file | 2007-07-08 22:13:57 -0300 (Sun, 08 Jul 2007) | 2
- lines Add a few sanity checks when writing out the dialplan.
- (issue #10157 reported by dome) ........
-
-2007-07-08 21:01 +0000 [r73911] Tilghman Lesher <tlesher@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in, main/ast_expr2.h,
- main/ast_expr2.y, configure.ac, main/ast_expr2.c: Restore EXP2
- and LOG2 functions, by providing mathematical identify functions,
- when the underlying C functions are not available.
-
-2007-07-08 13:22 +0000 [r73886] Russell Bryant <russell@digium.com>
-
- * res/res_features.c: ast_exists_extension() does not return an
- ast_device_state, so change this function to explicitly check for
- the int return value. Also, make a few other minor changes such
- as removing a variable.
-
-2007-07-08 09:49 +0000 [r73850] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 73849 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r73849 | oej | 2007-07-08 11:47:31 +0200 (Sun, 08 Jul 2007) | 2
- lines While tracking down a bug, I need some more history.
- Dumphistory is very useful, indeed. ........
-
-2007-07-07 16:44 +0000 [r73821] Steve Murphy <murf@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in, main/ast_expr2.y,
- configure.ac, bootstrap.sh, main/ast_expr2.c: These changes fix
- 10145 and 10150, a prob with BSD and exp2/log2 not existing, as
- well as the bootstrap needing a small upgrade for openbsd. Many
- thanks to mvanbaak
-
-2007-07-06 23:05 +0000 [r73771] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 73769 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r73769 | russell | 2007-07-06 18:02:58 -0500
- (Fri, 06 Jul 2007) | 12 lines Merged revisions 73768 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r73768 | russell | 2007-07-06 18:01:22 -0500 (Fri, 06
- Jul 2007) | 4 lines If a sip_pvt struct has already registered an
- extension state callback, remove the old one before adding a new
- one. If this isn't done, Asterisk will crash. (issue #10120)
- ........ ................
-
-2007-07-06 16:39 +0000 [r73728] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 73727 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r73727 | mmichelson | 2007-07-06 11:36:17 -0500 (Fri, 06 Jul
- 2007) | 8 lines Fixing a rare case which causes voicemail to
- crash when compiled with IMAP storage. inboxcount has the
- possibility of finding an "interactive" vm_state when no
- persistent "non-interactive" vm_state exists for that mailbox. If
- this should happen when someone attempts to leave a message, it
- results in a crash. This patch, along with my commit in revision
- 72670 fix issue 10053, reported by jaroth. closes issue #10053
- ........
-
-2007-07-06 16:30 +0000 [r73726] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/minimime/mimeparser.yy.c, main/minimime/mimeparser.h,
- main/minimime/mimeparser.tab.c, main/minimime/mimeparser.y,
- main/minimime/Makefile, main/minimime/mimeparser.l,
- main/minimime/mimeparser.tab.h, main/minimime/mm_parse.c:
- eliminate another batch of compiler warnings (and a bug, although
- in code we aren't using)... note that this required manually
- editing the lexer output code (generated by flex), so some of
- them will come back if the lexer is rebuilt
-
-2007-07-06 16:14 +0000 [r73680-73701] Russell Bryant <russell@digium.com>
-
- * res/res_config_odbc.c, /: Merged revisions 73696 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r73696 | russell | 2007-07-06 11:12:51 -0500
- (Fri, 06 Jul 2007) | 16 lines Merged revisions 73684 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r73684 | russell | 2007-07-06 11:06:27 -0500 (Fri, 06
- Jul 2007) | 8 lines (closes issue #10075) Reported by: apsaras
- Patches submitted by: Corydon76 Tested by: apsaras Fix a problem
- with MSSQL 2005 by explicitly stating that '\' is being used as
- an escape character. ........ ................
-
- * /, channels/chan_sip.c: Merged revisions 73679 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r73679 | russell | 2007-07-06 10:57:25 -0500
- (Fri, 06 Jul 2007) | 15 lines Merged revisions 73678 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r73678 | russell | 2007-07-06 10:55:41 -0500 (Fri, 06
- Jul 2007) | 7 lines (closes issue #10125) Reported by: makoto
- Patches submitted by: makoto This fixes a crash in chan_sip that
- happens when the bindaddr setting is not valid on Asterisk
- startup, gets fixed, and then a reload gets issued. ........
- ................
-
-2007-07-06 15:47 +0000 [r73677] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/busy.h (added), channels/ringtone.h (added),
- channels/Makefile, channels: it really seems pointless to run
- gentone to create these header files every time we build
- Asterisk...
-
-2007-07-06 15:28 +0000 [r73676] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 73675 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r73675 | mmichelson | 2007-07-06 10:27:28 -0500
- (Fri, 06 Jul 2007) | 13 lines Merged revisions 73674 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r73674 | mmichelson | 2007-07-06 10:26:40 -0500 (Fri, 06
- Jul 2007) | 5 lines Fixed a bug wherein agents get stuck busy.
- (issue 9618, reported by jiddings, patched by moi) closes issue
- #9618 ........ ................
-
-2007-07-06 03:48 +0000 [r73557-73633] Russell Bryant <russell@digium.com>
-
- * CHANGES: Redistribute a lot of the items that were in the Misc.
- section
-
- * CHANGES: note TLS support for manager and HTTP in CHANGES
-
- * CREDITS: Philippe was listed twice
-
- * /, BUGS: Merged revisions 73629 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r73629 | russell | 2007-07-05 22:34:46 -0500 (Thu, 05 Jul 2007) |
- 1 line fix a little spelling error ........
-
- * /, channels/chan_sip.c: Merged revisions 73598 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r73598 | russell | 2007-07-05 18:59:22 -0500 (Thu, 05 Jul 2007) |
- 3 lines Fix a crash in chan_sip. Don't try to stop the monitor
- thread if it was never started. (closes issue #10124, reported by
- gzero, fixed by me) ........
-
- * /, channels/chan_iax2.c: Merged revisions 73555 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r73555 | russell | 2007-07-05 18:05:33 -0500 (Thu, 05 Jul 2007) |
- 3 lines copy from the correct buffer when deferring a full frame
- (related to issue #9937) ........
-
-2007-07-05 22:48 +0000 [r73553] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/minimime/mm_contenttype.c, main/minimime/mm_envelope.c,
- main/minimime/mm_mimepart.c, main/minimime/mm_param.c,
- main/minimime/mm_context.c, main/minimime/mm_mimeutil.c: comment
- out some code that is not used and does not have prototypes
-
-2007-07-05 22:32 +0000 [r73552] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 73551 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r73551 | russell | 2007-07-05 17:31:31 -0500 (Thu, 05 Jul 2007) |
- 6 lines * Store the call number that a thread is processing
- without the full frame bit set to ease debugging * When deferring
- a full frame for processing, stick it into the queue for the
- thread that is processing frames for that call, not the one that
- read the current frame and is about to go back into the idle list
- (related to issue #9937) ........
-
-2007-07-05 22:29 +0000 [r73550] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 73548 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r73548 | kpfleming | 2007-07-05 17:20:44 -0500
- (Thu, 05 Jul 2007) | 10 lines Merged revisions 73547 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r73547 | kpfleming | 2007-07-05 17:11:51 -0500 (Thu, 05
- Jul 2007) | 2 lines we shouldn't allow G.723.1 endpoints to use
- VAD, just like we don't support it for G.729 ........
- ................
-
-2007-07-05 22:23 +0000 [r73549] Jason Parker <jparker@digium.com>
-
- * apps/app_queue.c: Add the ability to play an announcement to
- queue caller just before bridging Issue 7479, patch by
- tristan_mahe.
-
-2007-07-05 20:52 +0000 [r73513-73514] Russell Bryant <russell@digium.com>
-
- * main/ast_expr2.y, main/ast_expr2.c: resolve a compiler warning so
- i can build in dev mode
-
- * /, res/res_features.c: Merged revisions 73512 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r73512 | russell | 2007-07-05 15:50:08 -0500 (Thu, 05 Jul 2007) |
- 5 lines Pass HOLD and UNHOLD frames to the other channel when
- they are returned from a native bridge function. This fixes a
- problem where when two zap channels are natively bridged and one
- does a flash hook, the other channel did not receive music on
- hold. (Reported to me directly by Doug Bailey at Digium) ........
-
-2007-07-05 19:20 +0000 [r73468] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 73467 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r73467 | file | 2007-07-05 16:18:02 -0300 (Thu,
- 05 Jul 2007) | 10 lines Merged revisions 73466 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r73466 | file | 2007-07-05 16:15:18 -0300 (Thu, 05 Jul 2007) | 2
- lines Copy language information to the dialog structure when
- calling a peer for situations where a PBX may be started on the
- dialed channel. (issue #10121 reported by clegall_proformatique)
- ........ ................
-
-2007-07-05 18:15 +0000 [r73449] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, utils/expr2.testinput, main/ast_expr2.h,
- main/ast_expr2.y, main/ast_expr2f.c, include/asterisk/ast_expr.h,
- pbx/pbx_ael.c, UPGRADE.txt, doc/tex/channelvariables.tex,
- utils/ael_main.c, main/ast_expr2.fl, main/ast_expr2.c,
- utils/check_expr.c: In regards to changes for 9508, expr2 system
- choking on floating point numbers, I'm adding this update to
- round out (no pun intended) and make this FP-capable version of
- the Expr2 stuff interoperate better with previous integer-only
- usage, by providing Functions syntax, with 20 builtin functions
- for floating pt to integer conversions, and some general floating
- point math routines that might commonly be used also. Along with
- this, I made it so if a function was not a builtin, it will try
- and find it in the ast_custom_function list, and if found,
- execute it and collect the results. Thus, you can call system
- functions like CDR(), CHANNEL(), etc, from within $\[..\] exprs,
- without having to wrap them in $\{...\} (curly brace) notation.
- Did a valgrind on the standalone and made sure there's no mem
- leaks. Looks good. Updated the docs, too.
-
-2007-07-05 17:21 +0000 [r73432] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: Remove directory creation of directories
- we've never used.
-
-2007-07-05 16:05 +0000 [r73402] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 73400 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r73400 | mmichelson | 2007-07-05 10:59:41 -0500 (Thu, 05 Jul
- 2007) | 5 lines Correcting a minor CLI bug I found. When issuing
- the queue show command, if you type queue show and then press
- tab, you can continue pressing tab and it will keep
- auto-completing queue names even though only 1 queue can be used
- as an argument. ........
-
-2007-07-05 15:29 +0000 [r73399] Russell Bryant <russell@digium.com>
-
- * channels/chan_vpb.cc, /, channels/Makefile: Merged revisions
- 73398 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r73398 | russell | 2007-07-05 10:28:27 -0500 (Thu, 05 Jul 2007) |
- 2 lines Make this module build for me in dev-mode ........
-
-2007-07-05 14:22 +0000 [r73317-73359] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /, apps/app_chanspy.c: Merged revisions 73355 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r73355 | file | 2007-07-05 11:21:44 -0300 (Thu,
- 05 Jul 2007) | 10 lines Merged revisions 73349 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r73349 | file | 2007-07-05 11:19:14 -0300 (Thu, 05 Jul 2007) | 2
- lines Tweak spy locking. (issue #9951 reported by welles)
- ........ ................
-
- * channels/chan_local.c, /: Merged revisions 73319 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r73319 | file | 2007-07-05 10:27:40 -0300 (Thu,
- 05 Jul 2007) | 10 lines Merged revisions 73318 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r73318 | file | 2007-07-05 10:26:02 -0300 (Thu, 05 Jul 2007) | 2
- lines Actually check to make sure a PBX was started on one of the
- Local channels instead of blindly assuming it was. (issue #10112
- reported by makoto) ........ ................
-
- * /, apps/app_queue.c: Merged revisions 73316 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r73316 | file | 2007-07-05 10:22:13 -0300 (Thu,
- 05 Jul 2007) | 10 lines Merged revisions 73315 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r73315 | file | 2007-07-05 10:19:17 -0300 (Thu, 05 Jul 2007) | 2
- lines Reset ServicelevelPerf variable back to 0 if we are unable
- to calculate it each time... otherwise we will get previous
- values. (issue #10117 reported by noriyuki) ........
- ................
-
-2007-07-05 07:45 +0000 [r73209-73298] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, channels/misdn/chan_misdn_config.h,
- configs/misdn.conf.sample, channels/misdn_config.c: added general
- Jitterbuffer Implementation. #9960
-
- * /, channels/misdn/isdn_lib.c: Merged revisions 73253 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r73253 | crichter | 2007-07-04 16:53:48 +0200
- (Mi, 04 Jul 2007) | 9 lines Merged revisions 73252 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r73252 | crichter | 2007-07-04 16:50:58 +0200 (Mi, 04
- Jul 2007) | 1 line bchannel configurations like echocancel and
- volume control, need to be setuped on inbound calls too. ........
- ................
-
- * channels/chan_misdn.c, /: Merged revisions 73208 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r73208 | crichter | 2007-07-04 10:27:44 +0200
- (Mi, 04 Jul 2007) | 9 lines Merged revisions 73207 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r73207 | crichter | 2007-07-04 10:20:54 +0200 (Mi, 04
- Jul 2007) | 1 line bad bug in overlapdial case, we called
- start_pbx multiple times, because the state wasn't changed..
- ........ ................
-
-2007-07-03 22:17 +0000 [r73191] Steve Murphy <murf@digium.com>
-
- * /: blocking 73143 (revert of 9508 bug fix for 1.4) -- don't want
- it backed out of trunk, too
-
-2007-07-03 21:44 +0000 [r73144-73175] Jason Parker <jparker@digium.com>
-
- * apps/app_voicemail.c: mkstemp doesn't specify a file mode, so we
- should chmod it to VOICEMAIL_FILE_MODE Taken from a larger patch
- by ltd - the rest of which is no longer necessary in trunk.
- Closes issue #9231
-
- * apps/app_meetme.c: Fix a build warning, and potential issue if
- option p is not set at all.
-
- * apps/app_meetme.c: Add support for changing the exit key from #
- to any DTMF. This does not break existing configs - the arguments
- to p are optional. Issue 8827, initial patch by junky, mostly
- rewritten by fw to re-use option p, further modified by me.
-
-2007-07-03 18:25 +0000 [r73127] Russell Bryant <russell@digium.com>
-
- * apps/app_queue.c: Fix up the device state processing thread in
- app_queue so that it's not possible for there to be entries in
- the queue and the thread is just sleeping (Thanks to mmichelson
- for bringing the problem to my attention)
-
-2007-07-03 12:40 +0000 [r73054] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 73053 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r73053 | tilghman | 2007-07-03 07:38:53 -0500
- (Tue, 03 Jul 2007) | 10 lines Merged revisions 73052 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r73052 | tilghman | 2007-07-03 07:34:14 -0500 (Tue, 03
- Jul 2007) | 2 lines RetryDial should accept a 0 argument, but it
- does not, because atoi does not distinguish between 0 and error
- (closes issue #10106) ........ ................
-
-2007-07-03 08:22 +0000 [r73006] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /: Merged revisions 73005 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r73005 | crichter | 2007-07-03 10:17:06 +0200
- (Di, 03 Jul 2007) | 9 lines Merged revisions 73004 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r73004 | crichter | 2007-07-03 10:04:35 +0200 (Di, 03
- Jul 2007) | 1 line fixed issue, that misdn_l2l1_check could only
- be called from mISDN Source channels.. #9449 ........
- ................
-
-2007-07-03 05:21 +0000 [r73003] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: Typo (closes issue 10105)
-
-2007-07-03 02:51 +0000 [r72987] Jason Parker <jparker@digium.com>
-
- * res/res_jabber.c: Correct an issue where the wrong type was being
- used to start sasl. Pointed out by and patch provided by mog.
-
-2007-07-02 23:02 +0000 [r72982-72986] Russell Bryant <russell@digium.com>
-
- * main/pbx.c, doc/tex/ast_funcdocs.tex (removed), main/manager.c,
- doc/tex/ast_cli_commands.tex (removed), res/res_agi.c,
- doc/tex/ast_appdocs.tex (removed), doc/tex/asterisk.tex,
- doc/tex/ast_manager_actiondocs.tex (removed),
- doc/tex/ast_agi_commands.tex (removed), main/cli.c: After some
- discussion on the asterisk-dev list, we determined that this
- approach for extracting application, function, manager, and agi
- documentation is the wrong one to take. The most severe problem
- is that the output depends on which modules are loaded as well as
- compile time options, which both determine which parts are
- available.
-
- * doc/jitterbuffer.tex (removed), doc/extensions.tex (removed),
- doc/tex/ast_cli_commands.tex (added), doc/tex/ast_appdocs.tex
- (added), doc/tex/realtime.tex (added), doc/qos.tex (removed),
- doc/queues-with-callback-members.tex (removed), doc/tex/dundi.tex
- (added), doc/ajam.tex (removed), doc/tex/cliprompt.tex (added),
- doc/misdn.tex (removed), doc/manager.tex (removed),
- doc/tex/chaniax.tex (added), doc/sla.tex (removed),
- doc/billing.tex (removed), doc/tex/app-sms.tex (added),
- build_tools/prep_tarball, doc/tex/ices.tex (added),
- doc/localchannel.tex (removed), doc/cdrdriver.tex (removed),
- doc/tex/asterisk.tex (added), doc/tex/queuelog.tex (added),
- doc/freetds.tex (removed), doc/odbcstorage.tex (removed),
- doc/tex/hardware.tex (added), doc/tex/mp3.tex (added), doc/tex
- (added), doc/channelvariables.tex (removed), doc/ael.tex
- (removed), doc/enum.tex (removed), doc/tex/configuration.tex
- (added), doc/security.tex (removed), doc/tex/asterisk-conf.tex
- (added), Makefile, doc/imapstorage.tex (removed),
- doc/tex/ast_funcdocs.tex (added), doc/privacy.tex (removed),
- doc/tex/ast_manager_actiondocs.tex (added),
- doc/ast_agi_commands.tex (removed), doc/tex/jitterbuffer.tex
- (added), doc/ast_cli_commands.tex (removed),
- doc/tex/extensions.tex (added), doc/ast_appdocs.tex (removed),
- doc/tex/queues-with-callback-members.tex (added), doc/tex/qos.tex
- (added), doc/realtime.tex (removed), doc/dundi.tex (removed),
- doc/tex/ajam.tex (added), doc/cliprompt.tex (removed),
- doc/tex/manager.tex (added), doc/tex/misdn.tex (added),
- doc/chaniax.tex (removed), doc/tex/README.txt (added),
- doc/tex/sla.tex (added), doc/app-sms.tex (removed),
- doc/tex/billing.tex (added), doc/ices.tex (removed),
- doc/tex/localchannel.tex (added), doc/tex/cdrdriver.tex (added),
- doc/asterisk.tex (removed), doc/queuelog.tex (removed),
- doc/tex/odbcstorage.tex (added), doc/tex/freetds.tex (added),
- doc/hardware.tex (removed), doc/mp3.tex (removed),
- doc/tex/channelvariables.tex (added), doc/tex/ael.tex (added),
- doc/tex/enum.tex (added), doc/configuration.tex (removed),
- doc/tex/security.tex (added), doc/asterisk-conf.tex (removed),
- doc/tex/imapstorage.tex (added), doc/ast_funcdocs.tex (removed),
- doc/tex/privacy.tex (added), doc/tex/Makefile (added),
- doc/ast_manager_actiondocs.tex (removed),
- doc/tex/ast_agi_commands.tex (added): * Move LaTeX docs into a
- tex/ subdirectory of the doc/ dir * Add a Makefile in doc/tex/
- for generating PDF and HTML * Add a README.txt file to doc/tex/
- to document which tools are used and what web sites to visit for
- getting them. * Update build_tools/prep_tarball to put the proper
- Asterisk version string in the automatically generated PDF for
- release tarballs
-
-2007-07-02 21:50 +0000 [r72940] Steve Murphy <murf@digium.com>
-
- * utils/expr2.testinput, /, main/Makefile, main/ast_expr2.h,
- main/ast_expr2.y, main/ast_expr2f.c, UPGRADE.txt,
- main/ast_expr2.fl, main/ast_expr2.c: Merged revisions 72933 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r72933 | murf | 2007-07-02 14:16:31 -0600 (Mon, 02 Jul 2007) | 1
- line support for floating point numbers added to ast_expr2
- $\[...\] exprs. Fixes bug 9508, where the expr code fails with fp
- numbers. The MATH function returns fp numbers by default, so this
- fix is considered necessary. ........
-
-2007-07-02 20:45 +0000 [r72937-72939] Russell Bryant <russell@digium.com>
-
- * res/res_agi.c, doc/ast_agi_commands.tex: Fix up the AGI doc dump
- CLI command and update the AGI commands tex file to not include a
- bunch of empty entries.
-
- * doc/ast_cli_commands.tex (added), doc/asterisk.tex: Add CLI
- commands to the docs
-
- * main/cli.c: Add a CLI command to output docs on CLI commands to a
- file
-
-2007-07-02 20:35 +0000 [r72935-72936] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: Yet another Solaris tweak...
-
- * res/res_limit.c: Fix building under Solaris.
-
-2007-07-02 19:31 +0000 [r72920-72932] Russell Bryant <russell@digium.com>
-
- * doc/asterisk.tex, doc/ast_agi_commands.tex (added): Add AGI
- commands to the documentation
-
- * res/res_agi.c: Add a CLI command to export the AGI command docs
-
- * res/res_agi.c: Add a note that the AGI commands array is not
- handled in a thread-safe way
-
- * doc/asterisk.tex, doc/ast_manager_actiondocs.tex (added): Update
- the documentation to include a manager action reference
-
- * main/manager.c: Add a CLI command to dump the built-in manager
- action documentation
-
- * main/manager.c, /: Merged revisions 72926 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r72926 | russell | 2007-07-02 13:18:46 -0500 (Mon, 02 Jul 2007) |
- 3 lines Remove a bogus comment and add proper locking to the
- handler function for the CLI command to show information on
- manager actions. ........
-
- * doc/ast_funcdocs.tex (added), doc/asterisk.tex: update
- documentation to include dialplan functions
-
- * main/pbx.c: Add "core dump funcdocs" CLI command
-
- * main/pbx.c: change the "core dump appdocs" CLI command to use the
- new API for creating CLI commands
-
- * doc/ast_appdocs.tex: update application documentation dump
-
-2007-07-02 14:39 +0000 [r72889] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /: Merged revisions 72888 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r72888 | file | 2007-07-02 11:32:59 -0300 (Mon, 02 Jul 2007) | 2
- lines Added additional DTMF debug messages for when emulation
- occurs. ........
-
-2007-07-02 09:34 +0000 [r72867-72869] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged
- revisions 72852 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r72852 | crichter | 2007-07-02 10:41:08 +0200
- (Mo, 02 Jul 2007) | 9 lines Merged revisions 72585 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r72585 | crichter | 2007-06-29 15:08:26 +0200 (Fr, 29
- Jun 2007) | 1 line check if the bchannel stack id is already
- used, if so don't use it a second time. Also added a release_chan
- lock, so that the same chan_list object cannot be freed twice.
- chan_misdn does not crash anymore on heavy load with these
- changes. ........ ................
-
- * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
- channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c:
- Merged revisions 72851 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r72851 | crichter | 2007-07-02 10:27:19 +0200
- (Mo, 02 Jul 2007) | 9 lines Merged revisions 72099 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r72099 | crichter | 2007-06-27 15:22:37 +0200 (Mi, 27
- Jun 2007) | 1 line simplified generation for dummy bchannels,
- also we mark them as dummies, so they are not used later as
- real-bchannels, optimized the RESTART mechanisms, we block a
- channel now on cause:44, and send out a RESTART automatically,
- then on reception of RESTART_ACKNOWLEDGE we unblock the channel
- again. ........ ................
-
- * channels/misdn/isdn_lib.h, /, channels/misdn/isdn_lib.c: Merged
- revisions 72850 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r72850 | crichter | 2007-07-02 10:14:43 +0200
- (Mo, 02 Jul 2007) | 9 lines Merged revisions 72087 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r72087 | crichter | 2007-06-27 11:26:53 +0200 (Mi, 27
- Jun 2007) | 1 line simplified channel finding and locking a lot.
- removed unnecessary #ifdefed areas. ........ ................
-
-2007-07-01 23:53 +0000 [r72807] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_spool.c, /: Merged revisions 72806 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r72806 | russell | 2007-07-01 18:52:45 -0500
- (Sun, 01 Jul 2007) | 13 lines Merged revisions 72805 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r72805 | russell | 2007-07-01 18:51:34 -0500 (Sun, 01
- Jul 2007) | 5 lines When appending lines to call files to keep
- track of retries, write a leading newline just in case the
- original call file did not have a newline at the end. This fix is
- in response to a problem I saw reported on the asterisk-users
- mailing list. ........ ................
-
-2007-06-30 16:53 +0000 [r72767] Russell Bryant <russell@digium.com>
-
- * /, configure, configure.ac: Merged revisions 72766 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r72766 | russell | 2007-06-30 11:50:40 -0500 (Sat, 30
- Jun 2007) | 3 lines Tweak the configure script so that error
- output isn't spewed to the console when searching for GTK2 libs,
- and they aren't found. ........
-
-2007-06-29 21:37 +0000 [r72741] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c, configs/skinny.conf.sample: Add support
- for regcontext and regexten to chan_skinny Issue 9762, patch by
- mvanbaak.
-
-2007-06-29 21:24 +0000 [r72738] Russell Bryant <russell@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac,
- main/http.c: Fix my recent change for sending large files via the
- http server. This code *must* write the file to the FILE *, and
- not the raw fd. Otherwise, it breaks TLS support. Thanks to rizzo
- for catching this!
-
-2007-06-29 21:14 +0000 [r72727] Luigi Rizzo <rizzo@icir.org>
-
- * main/minimime/Makefile: As the comment in the code says: Use
- weaker error checking because we have some automatically
- generated files. However just mask out -Werror, because other
- warnings below: -Wundef -Wstrict-prototypes
- -Wmissing-declarations -Wmissing-prototypes may actually be
- important and spot out real bugs.
-
-2007-06-29 20:56 +0000 [r72701-72706] Russell Bryant <russell@digium.com>
-
- * /, formats/format_pcm.c: Merged revisions 72705 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r72705 | russell | 2007-06-29 15:56:18 -0500 (Fri, 29 Jun 2007) |
- 1 line give format_pcm a more concise destription ........
-
- * include/asterisk/http.h, main/manager.c, configure,
- include/asterisk/autoconfig.h.in, configure.ac, main/http.c:
- Merge changes from team/russell/http_filetxfer Handle
- transferring large files from the built-in http server.
- Previously, the code attempted to malloc a block as large as the
- file itself. Now it uses the sendfile() system call so that the
- file isn't copied into userspace at all if it is available.
- Otherwise, it just uses a read/write of small chunks at a time.
-
-2007-06-29 20:33 +0000 [r72700] Luigi Rizzo <rizzo@icir.org>
-
- * main/Makefile: Make sure that we properly recurse in
- subdirectories to check dependencies for libraries. Because these
- targets (e.g. minimime/libmmime.a) are real ones, declaring them
- .PHONY would cause them to be rebuilt every time (see e.g. SVN
- 64355). As a workaround I am using the following CHECK_SUBDIR
- target: CHECK_SUBDIR: # do nothing, just make sure that we
- recurse in the subdir/ minimime/libmmime.a: CHECK_SUBDIR @cd
- minimime && $(MAKE) libmmime.a which seems to do a better job
- than .PHONY (probably because .PHONY forces the rebuild even if
- the recursive make does not think it is necessary). If this turns
- out to be the correct approach, we can then merge it back into
- 1.4
-
-2007-06-29 20:02 +0000 [r72670] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Found a grievous logical error in
- get_vm_state_by_imapuser. The imapuser being passed in was never
- getting compared to imapusers of any of the vm_states in the
- vmstates list. I also found some places in the code where I used
- my typical brace style and changed it to match the typical
- Asterisk brace style.
-
-2007-06-29 19:09 +0000 [r72666] Luigi Rizzo <rizzo@icir.org>
-
- * /: 72665 not applicable to trunk
-
-2007-06-29 04:56 +0000 [r72555-72557] Tilghman Lesher <tlesher@digium.com>
-
- * main/manager.c, /: Merged revisions 72556 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r72556 | tilghman | 2007-06-28 23:47:11 -0500 (Thu, 28 Jun 2007)
- | 2 lines Issue 10055 - Change memory allocation to use the heap
- for a command, since the output has the potential to overflow the
- stack (as it did here) ........
-
-2007-06-28 21:31 +0000 [r72539] Jason Parker <jparker@digium.com>
-
- * Makefile, configure, configure.ac, makeopts.in: Apparently some
- builds of gcc don't have declaration-after-statement. This checks
- for it in configure, and only uses it if it's available. If it's
- wrong, somebody please yell at me and tell me why.
-
-2007-06-28 20:52 +0000 [r72524] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * funcs/func_math.c: Added AND, OR, and XOR bitwise operations to
- MATH for issue 9891, thanks jcmoore
-
-2007-06-28 19:41 +0000 [r72492] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_config_pgsql.c, res/res_config_odbc.c,
- include/asterisk/strings.h: Remove the ill-advised ast_restrdupa
- API call and related structures
-
-2007-06-28 19:35 +0000 [r72490-72491] Jason Parker <jparker@digium.com>
-
- * channels/chan_sip.c: Fix building with
- -Wdeclaration-after-statement, here too
-
- * res/res_jabber.c: Fix building with -Wdeclaration-after-statement
-
-2007-06-28 19:07 +0000 [r72452-72466] Luigi Rizzo <rizzo@icir.org>
-
- * /: 72462 is not applicable to trunk
-
- * res/res_features.c, apps/app_sms.c: move variable declarations to
- the beginning of a block. Not applicable to previous branches.
-
- * channels/chan_skinny.c: move variable declarations to the
- beginning of the block
-
- * apps/app_minivm.c: move variable declarations to the beginning of
- a block. Not applicable to previous branches
-
- * /: 72453 was already applied to trunk some time ago
-
- * Makefile: Add -Wdeclaration-after-statement to AST_DEVMODE to
- detect declarations in the middle of a block. Approved by:
- Russel, Kevin The fallout will be fixed in separate commits. I am
- doing this only on trunk only for the time being, because 1.4
- still requires a bit more polishing to give a clean compile (at
- least on FreeBSD).
-
-2007-06-28 16:35 +0000 [r72437] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Fix bug where point code gets corrupted on
- CPG
-
-2007-06-27 23:30 +0000 [r72384] Brett Bryant <bbryant@digium.com>
-
- * /, main/asterisk.c: Merged revisions 72383 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r72383 | bbryant | 2007-06-27 18:29:14 -0500
- (Wed, 27 Jun 2007) | 11 lines Merged revisions 72373 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r72373 | bbryant | 2007-06-27 18:22:13 -0500 (Wed, 27
- Jun 2007) | 3 lines Reinstating patch. This actually fixes the
- problem, however I was running a development branch without it
- and mistakenly thought it wasn't fixed. Fixes issue #10010, and
- #9654: 100% CPU usage caused by an asterisk console losing it's
- controlling terminal. ........ ................
-
-2007-06-27 23:26 +0000 [r72354-72382] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_mixmonitor.c: Merged revisions 72381 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r72381 | file | 2007-06-27 19:25:12 -0400 (Wed,
- 27 Jun 2007) | 10 lines Merged revisions 72378 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72378 | file | 2007-06-27 19:24:01 -0400 (Wed, 27 Jun 2007) | 2
- lines Update documentation to clarify variable usage with
- MixMonitor. (issue #9494 reported by netoguy) ........
- ................
-
- * channels/chan_jingle.c: Silly jingle...
-
- * channels/chan_sip.c, CHANGES: Add SIPREFERRINGCONTEXT and
- SIPREFERREDBYHDR variables when a transfer takes place. (issue
- #8378 reported by jcovert)
-
-2007-06-27 23:04 +0000 [r72337] Brett Bryant <bbryant@digium.com>
-
- * /, main/asterisk.c: Merged revisions 72335 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r72335 | bbryant | 2007-06-27 18:03:01 -0500
- (Wed, 27 Jun 2007) | 10 lines Merged revisions 72333 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r72333 | bbryant | 2007-06-27 17:58:53 -0500 (Wed, 27
- Jun 2007) | 2 lines Reverted changes for earlier revisions 72259
- to 72261. Issue #9654, #10010 ........ ................
-
-2007-06-27 22:58 +0000 [r72330-72332] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_gtalk.c: Merged revisions 72331 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r72331 | file | 2007-06-27 18:58:02 -0400 (Wed, 27 Jun
- 2007) | 2 lines Make payload IDs for iLBC/Speex match to our
- list. Since these are dynamic payloads the other side shouldn't
- care. (issue #9426 reported by irroot) ........
-
- * /, apps/app_queue.c: Merged revisions 72328 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r72328 | file | 2007-06-27 18:45:49 -0400 (Wed,
- 27 Jun 2007) | 10 lines Merged revisions 72327 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72327 | file | 2007-06-27 18:43:11 -0400 (Wed, 27 Jun 2007) | 2
- lines Fix issue where queue log events might be missing. (issue
- #7765 reported by mtryfoss) ........ ................
-
-2007-06-27 22:47 +0000 [r72329] Mark Michelson <mmichelson@digium.com>
-
- * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c:
- Added ability to customize which buttons control forward,
- reverse, pause, and stop during message playback. (closes issue
- 9474, reported and patched by jaroth with modifications by me)
-
-2007-06-27 22:27 +0000 [r72325-72326] Jason Parker <jparker@digium.com>
-
- * main/cli.c: Fix a segfault when trying to tab complete the "core
- show uptime" command. Reported in #asterisk-dev on IRC by
- jcmoore, fixed by me.
-
- * main/say.c: Add support for Thai language in say.c Issue 9417,
- patch by dome, with some cleanup done by me.
-
-2007-06-27 21:44 +0000 [r72304] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Let's NOT create a deadlock scenario here
-
-2007-06-27 21:09 +0000 [r72274] Russell Bryant <russell@digium.com>
-
- * /, pbx/pbx_config.c: Merged revisions 72272 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r72272 | russell | 2007-06-27 16:08:34 -0500
- (Wed, 27 Jun 2007) | 13 lines Merged revisions 72267 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r72267 | russell | 2007-06-27 16:06:45 -0500 (Wed, 27
- Jun 2007) | 5 lines Fix a minor issue with parsing the priority
- number. You could have as much whitespace as you want around a
- numeric priority, but you couldn't have any whitespace around a
- special priority like "n" or "hint". (issue #10039, reported by
- mitheloc, fixed by me) ........ ................
-
-2007-06-27 20:47 +0000 [r72261] Brett Bryant <bbryant@digium.com>
-
- * /, main/asterisk.c: Merged revisions 72260 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r72260 | bbryant | 2007-06-27 15:46:12 -0500
- (Wed, 27 Jun 2007) | 12 lines Merged revisions 72259 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r72259 | bbryant | 2007-06-27 15:43:53 -0500 (Wed, 27
- Jun 2007) | 4 lines Fixes 100% load when controlling terminal
- disappears. Issue #9654, #10010 ........ ................
-
-2007-06-27 20:26 +0000 [r72233-72258] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /: Merged revisions 72257 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r72257 | file | 2007-06-27 16:25:24 -0400 (Wed,
- 27 Jun 2007) | 10 lines Merged revisions 72256 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72256 | file | 2007-06-27 16:23:24 -0400 (Wed, 27 Jun 2007) | 2
- lines I may possibly get shot for doing this... but... defer CDR
- processing until after the channel has been dealt with. This
- should eliminate all of the issues with channels going funky
- (SIP/PRI) when you are posting CDRs to a database that is either
- slow or unavailable and do not want to enable batching. ........
- ................
-
- * /: Fix up properties.
-
- * main/logger.c: Fix -T option. (issue #10073 reported by xylome)
-
-2007-06-27 19:50 +0000 [r72232] Mark Michelson <mmichelson@digium.com>
-
- * /, configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c:
- Adding feature to support the storage and retrieval of voicemail
- greetings using IMAP storage. This feature may be turned on by
- adding imapgreetings=yes to the general section of voicemail.conf
- voicemail.conf.sample has details on the options added. As a
- result, IMAP storage now has RETRIEVE and DISPOSE macros defined.
- In addition to the IMAP greeting changes, I also have added an
- enum for the voicemail folders and so now the code should be
- easier to understand and maintain when it comes to this area.
-
-2007-06-27 19:13 +0000 [r72207] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 72205 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r72205 | kpfleming | 2007-06-27 14:13:21 -0500 (Wed, 27 Jun 2007)
- | 2 lines use the proper type for storing group number bits so
- that if someone specifies 'group=42' it will actually work
- instead of being silently ignored ........
-
-2007-06-27 18:37 +0000 [r72183] Jason Parker <jparker@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 72182 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r72182 | qwell | 2007-06-27 13:36:56 -0500 (Wed, 27 Jun 2007) | 4
- lines Fix another problem in voicemail with missing symbols.
- Issue 10074, patch by kryptolus, extended to include #if 0'd
- blocks (just in case) ........
-
-2007-06-27 17:34 +0000 [r72149] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /: Merged revisions 72148 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r72148 | file | 2007-06-27 13:31:50 -0400 (Wed, 27 Jun 2007) | 2
- lines Make the ast_read_noaudio API call behave better under
- circumstances where DTMF emulation was happening and a generator
- was setup. (issue #10065 reported by stevefeinstein) ........
-
-2007-06-27 17:14 +0000 [r72134] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_gtalk.c: Merged revisions 72125 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r72125 | qwell | 2007-06-27 12:10:32 -0500 (Wed, 27 Jun
- 2007) | 4 lines Don't modify a variable that we don't want
- modified. Make a copy of it instead. Issue 10029, patch by
- phsultan with slight modifications by me (to remove needless
- casts). Note: chan_jingle in trunk does not appear to have the
- same bug. ........
-
-2007-06-27 16:38 +0000 [r72113] Russell Bryant <russell@digium.com>
-
- * /, main/rtp.c: Merged revisions 72112 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r72112 | russell | 2007-06-27 11:34:24 -0500 (Wed, 27 Jun 2007) |
- 3 lines Only output debug information related to RTCP timestamps
- when RTCP debug is turned on (issue #10066, patch by me) ........
-
-2007-06-27 08:08 +0000 [r72052] Christian Richter <christian.richter@beronet.com>
-
- * /, channels/misdn/isdn_lib.c: Merged revisions 72042 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r72042 | crichter | 2007-06-27 09:58:06 +0200
- (Mi, 27 Jun 2007) | 13 lines Merged revisions 72040-72041 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72040 | crichter | 2007-06-27 09:49:27 +0200 (Mi, 27 Jun 2007) |
- 1 line for inbound TE calls, we setup the bchannel when we get
- the CONNECT_ACKNOWLEDGE, to make sure mISDN has everything ready.
- removed some #if 0 areas which weren't used anymore. ........
- r72041 | crichter | 2007-06-27 09:54:30 +0200 (Mi, 27 Jun 2007) |
- 1 line isdn_lib.c didn't compile ........ ................
-
-2007-06-27 01:00 +0000 [r71988-72007] Joshua Colp <jcolp@digium.com>
-
- * /, pbx/pbx_dundi.c: Merged revisions 72006 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r72006 | file | 2007-06-26 20:58:35 -0400 (Tue, 26 Jun 2007) | 2
- lines Make unloading of pbx_dundi actually work. ........
-
- * channels/chan_sip.c, funcs/func_channel.c, CHANGES: Add rtpdest
- option to SIP CHANNEL() dialplan function to return the IP
- address and port that RTP (be it audio/video/text) is going to.
-
-2007-06-26 23:03 +0000 [r71952-71954] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 71953 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r71953 | mmichelson | 2007-06-26 18:02:09 -0500 (Tue, 26 Jun
- 2007) | 4 lines Removing a pointless line. This variable was
- already set earlier and between then and this line, there is no
- way that the values on the right side of the assignment could
- have changed. ........
-
- * apps/app_voicemail.c: The variable msgnum was being overwritten
- if IMAP storage was enabled. Put necessary #ifndef's around the
- line which would overwrite.
-
-2007-06-26 20:36 +0000 [r71916] Jason Parker <jparker@digium.com>
-
- * /, main/rtp.c: Merged revisions 71915 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r71915 | qwell | 2007-06-26 15:36:09 -0500 (Tue, 26 Jun 2007) | 4
- lines Don't dereference a pointer that may be NULL here. Issue
- 10017. ........
-
-2007-06-26 20:34 +0000 [r71883-71914] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_record.c: Create directory if it does not exist. (Closes
- issue 10061, Reported and patched by eliel)
-
- * /, apps/app_voicemail.c: Merged revisions 71877 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r71877 | mmichelson | 2007-06-26 14:00:05 -0500 (Tue, 26 Jun
- 2007) | 11 lines A few changes, the ultimate goal of which is to
- keep better track of the number of messages that a mailbox
- currently has. A description of the changes: 1. Changed the
- "updated" field of the vm_state struct to act more as a binary
- semaphore than a counting semaphore, since its current
- implementation made the inboxcount function not work properly.
- This change falls in line with a change made by UPenn with their
- IMAP setup and helps to sync our changes with theirs. 2.
- Eliminated some redundant calls to get_vm_state_by_mailbox inside
- leave_voicemail 3. Use the play_folder variable to keep track of
- the number of old and new messages in a mailbox as the messages
- are deleted 4. Added an increment to the number of new messages
- that was not there previously in the leave_voicemail function
- ........
-
-2007-06-26 16:39 +0000 [r71830] Jason Parker <jparker@digium.com>
-
- * res/res_jabber.c: Simplify some code in res_jabber relating to
- SASL support. Issue 9988, patch by phsultan.
-
-2007-06-26 15:50 +0000 [r71797] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 71796 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r71796 | mmichelson | 2007-06-26 10:47:31 -0500 (Tue, 26 Jun
- 2007) | 5 lines Fixing bug where the authuser was mistakenly
- pulled from the mailbox string instead of the IMAP user. (closes
- issue 10054, reported and patched by jaroth) ........
-
-2007-06-26 12:30 +0000 [r71752] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 71751 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r71751 | tilghman | 2007-06-26 07:27:47 -0500
- (Tue, 26 Jun 2007) | 10 lines Merged revisions 71750 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r71750 | tilghman | 2007-06-26 07:25:58 -0500 (Tue, 26
- Jun 2007) | 2 lines Issue 10062 - Trying to move a message
- without selecting one first results in memory corruption ........
- ................
-
-2007-06-26 00:10 +0000 [r71721-71732] Mark Michelson <mmichelson@digium.com>
-
- * configure, configure.ac: Fixes a problem where Asterisk would not
- compile if IMAP_STORAGE was enabled. Needed to add a space
- between file name and options.
-
- * apps/app_voicemail.c: In my commit earlier today, I accidentally
- left a prototype that isn't defined. This gets rid of that
- prototype.
-
-2007-06-25 19:20 +0000 [r71688] Russell Bryant <russell@digium.com>
-
- * doc/imapstorage.tex, configure, configure.ac,
- apps/app_voicemail.c: Allow compilation off app_voicemail with
- IMAP_STORAE against a system installed version of the c-client
- library. (issue #10047, jcollie)
-
-2007-06-25 18:20 +0000 [r71658] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_agi.c: Merged revisions 71657 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r71657 | tilghman | 2007-06-25 13:14:59 -0500
- (Mon, 25 Jun 2007) | 10 lines Merged revisions 71656 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r71656 | tilghman | 2007-06-25 13:12:37 -0500 (Mon, 25
- Jun 2007) | 2 lines Issue 10035 - handle_exec returns a result
- inconsistent with all of the other AGI commands ........
- ................
-
-2007-06-25 16:43 +0000 [r71637] Steve Murphy <murf@digium.com>
-
- * main/cdr.c: Luigi's suggestion to move the llfrom decl was a good
- one. Done.
-
-2007-06-25 16:13 +0000 [r71630] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Using inboxcount instead of countmessages.
-
-2007-06-25 15:35 +0000 [r71577-71613] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Tweak CLI command completion and some help
- text. (issue #10049 reported by IgorG)
-
- * /, channels/chan_h323.c: Merged revisions 71576 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r71576 | file | 2007-06-25 10:13:45 -0400 (Mon, 25 Jun 2007) | 2
- lines Build a peer as well when hash323 is enabled in users.conf
- (issue #9599 reported by asagage) ........
-
-2007-06-25 13:42 +0000 [r71557] Russell Bryant <russell@digium.com>
-
- * main/say.c, main/rtp.c, main/sched.c: Convert so more logging to
- ast_debug (issue #10045, dimas)
-
-2007-06-25 13:04 +0000 [r71521-71525] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 71522 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r71522 | file | 2007-06-25 09:03:03 -0400 (Mon, 25 Jun
- 2007) | 2 lines Minor tweak for queueing up the unhold frame...
- this will teach me to do bugs while half asleep. (issue #10046
- reported by dimas) ........
-
- * res/res_agi.c: Minor header inclusion tweak for new usage of
- stat()
-
-2007-06-25 12:40 +0000 [r71520] Russell Bryant <russell@digium.com>
-
- * doc/asterisk-mib.txt, /: Merged revisions 71519 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r71519 | russell | 2007-06-25 07:40:06 -0500 (Mon, 25 Jun 2007) |
- 2 lines Fix a typo in the Asterisk mib. (issue #10048, Matti)
- ........
-
-2007-06-25 09:46 +0000 [r71475-71500] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
- channels/misdn/isdn_lib.c: Merged revisions 71214 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r71214 | crichter | 2007-06-23 00:44:42 +0200
- (Sa, 23 Jun 2007) | 9 lines Merged revisions 70341 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r70341 | crichter | 2007-06-20 17:29:09 +0200 (Mi, 20
- Jun 2007) | 1 line fixed a bug that was introduced by copy and
- paste in the last commit ..bchannels weren't cleaned properly.
- ........ ................
-
- * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged
- revisions 71123 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r71123 | crichter | 2007-06-22 17:38:08 +0200
- (Fr, 22 Jun 2007) | 9 lines Merged revisions 70672 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r70672 | crichter | 2007-06-21 15:11:29 +0200 (Do, 21
- Jun 2007) | 1 line we activate the bchannels in TE mode on
- incoming calls only when we want to connect the call. ........
- ................
-
- * /, channels/misdn/isdn_lib.c: Merged revisions 71122 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r71122 | crichter | 2007-06-22 17:34:31 +0200
- (Fr, 22 Jun 2007) | 9 lines Merged revisions 70342 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r70342 | crichter | 2007-06-20 17:42:39 +0200 (Mi, 20
- Jun 2007) | 1 line forgot one place .. ........ ................
-
- * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
- channels/misdn/isdn_lib.c: Merged revisions 71121 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r71121 | crichter | 2007-06-22 17:32:54 +0200
- (Fr, 22 Jun 2007) | 9 lines Merged revisions 70311 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r70311 | crichter | 2007-06-20 16:47:59 +0200 (Mi, 20
- Jun 2007) | 1 line on receiption of cause:44 we mark the channel
- as in use and inform the user about the situation, we need to
- test the RESTART stuff then. Also shuffled the
- empty_chan_in_stack function after the bchannel cleaning
- functions, to avoid race conditions. ........ ................
-
- * channels/chan_misdn.c, /: Merged revisions 71120 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r71120 | crichter | 2007-06-22 17:30:08 +0200
- (Fr, 22 Jun 2007) | 9 lines Merged revisions 69887 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r69887 | crichter | 2007-06-19 15:23:04 +0200 (Di, 19
- Jun 2007) | 1 line when we send out a SETUP, but get no response,
- we should cleanup everything after reception of a hangup.
- ........ ................
-
- * /, channels/misdn/isdn_msg_parser.c: Merged revisions 71118 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r71118 | crichter | 2007-06-22 17:27:53 +0200
- (Fr, 22 Jun 2007) | 9 lines Merged revisions 69053 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r69053 | crichter | 2007-06-13 11:55:54 +0200 (Mi, 13
- Jun 2007) | 1 line restart indicator 0x80 is correct, at least
- that's what libpri does. ........ ................
-
- * channels/chan_misdn.c, /: Merged revisions 71106 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r71106 | crichter | 2007-06-22 17:22:06 +0200
- (Fr, 22 Jun 2007) | 9 lines Merged revisions 68887 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r68887 | crichter | 2007-06-12 10:35:22 +0200 (Di, 12
- Jun 2007) | 1 line if the bridged partner is mISDN too we should
- not send dtmf tones, they are transmitted inband always ........
- ................
-
- * channels/chan_misdn.c, /: Merged revisions 71096 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r71096 | crichter | 2007-06-22 17:17:04 +0200
- (Fr, 22 Jun 2007) | 9 lines Merged revisions 68874 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r68874 | crichter | 2007-06-12 09:48:52 +0200 (Di, 12
- Jun 2007) | 1 line if we have already some digits, we just stop
- the tones. ........ ................
-
-2007-06-25 01:11 +0000 [r71413-71434] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 71430 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r71430 | file | 2007-06-24 21:10:06 -0400 (Sun,
- 24 Jun 2007) | 10 lines Merged revisions 71414 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r71414 | file | 2007-06-24 21:02:49 -0400 (Sun, 24 Jun 2007) | 2
- lines Ignore other URIs after the first in a 300 Multiple Choice
- response. (issue #10041 reported by homesick) ........
- ................
-
- * main/cdr.c, /: Merged revisions 71422 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r71422 | file | 2007-06-24 21:07:31 -0400 (Sun, 24 Jun 2007) | 2
- lines Fix it so 1.4 actually compiles on my box. ........
-
- * /, channels/chan_agent.c: Merged revisions 71412 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r71412 | file | 2007-06-24 20:49:21 -0400 (Sun, 24 Jun
- 2007) | 2 lines Check to make sure the channel pointer is present
- before queueing up an unhold frame on it. (issue #10046 reported
- by dimas) ........
-
-2007-06-24 20:17 +0000 [r71338-71372] Russell Bryant <russell@digium.com>
-
- * /, build_tools/prep_tarball: Merged revisions 71371 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r71371 | russell | 2007-06-24 15:16:32 -0500 (Sun, 24
- Jun 2007) | 3 lines Include the menuselect-tree file in tarballs
- to make builds from tarballs a little bit faster ........
-
- * /, main/asterisk.c: Merged revisions 71362 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r71362 | russell | 2007-06-24 15:06:31 -0500
- (Sun, 24 Jun 2007) | 10 lines Merged revisions 71358 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r71358 | russell | 2007-06-24 15:04:21 -0500 (Sun, 24
- Jun 2007) | 2 lines Revert the patch from issue 9654 due to an
- unexpected side effect ........ ................
-
- * main/udptl.c, apps/app_meetme.c, main/say.c, main/translate.c,
- main/jitterbuf.c, apps/app_test.c, main/rtp.c, main/loader.c,
- main/io.c, main/manager.c, apps/app_skel.c, apps/app_minivm.c,
- main/logger.c, main/http.c, apps/app_rpt.c, main/sched.c:
- Conversions to ast_debug() (issue #9984, patches from eliel and
- dimas)
-
-2007-06-24 17:51 +0000 [r71268-71292] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_features.c: Merged revisions 71291 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r71291 | tilghman | 2007-06-24 12:50:24 -0500 (Sun, 24 Jun 2007)
- | 2 lines Issue 10044 - chan->cdr is NULL here, so peer->cdr is
- what we really wanted to use ........
-
- * main/manager.c, /, main/db.c: Merged revisions 71289 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r71289 | tilghman | 2007-06-24 12:39:34 -0500
- (Sun, 24 Jun 2007) | 10 lines Merged revisions 71288 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r71288 | tilghman | 2007-06-24 12:32:21 -0500 (Sun, 24
- Jun 2007) | 2 lines Issue 10043 - There is a legitimate need to
- be able to set variables to the empty string. ........
- ................
-
- * apps/app_mixmonitor.c: Issue 9970 - Ensure directory exists
- before trying to write an output file
-
-2007-06-23 03:32 +0000 [r71231] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, /, res/res_features.c: Merged revisions 71230 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r71230 | murf | 2007-06-22 21:29:48 -0600 (Fri, 22 Jun 2007) | 1
- line This patch is meant to fix 8433; where clid and src are lost
- via bridging. ........
-
-2007-06-22 19:53 +0000 [r71190] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_sms.c: Code cleanups
-
-2007-06-22 16:19 +0000 [r71146-71158] Joshua Colp <jcolp@digium.com>
-
- * res/res_agi.c: Use stat to determine whether the file exists or
- not. (issue #10038 reported by Mike Anikienko)
-
- * main/rtp.c: Behold the magic of casting!
-
-2007-06-22 15:15 +0000 [r71093] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, /, main/rtp.c: Merged revisions 71063 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r71063 | murf | 2007-06-22 08:10:24 -0600 (Fri, 22 Jun
- 2007) | 1 line My conditions for merging amaflags info was naive;
- DOCUMENTATION is the default, although null is possible; theft of
- user-settable fields is not good. Just copy them, leave them
- alone. This is for bug 10016. (plus a small fix to rtp, to elim a
- compiler warning (dev mode)) ........
-
-2007-06-22 15:03 +0000 [r71069] Jason Parker <jparker@digium.com>
-
- * /, res/res_agi.c, main/file.c, apps/app_speech_utils.c: Merged
- revisions 71068 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r71068 | qwell | 2007-06-22 10:00:30 -0500 (Fri,
- 22 Jun 2007) | 12 lines Merged revisions 71065 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r71065 | qwell | 2007-06-22 09:52:18 -0500 (Fri, 22 Jun 2007) | 4
- lines Fix a few silly usages of ast_playstream() - it only ever
- returns 0... Issue 10035 ........ ................
-
-2007-06-22 14:56 +0000 [r71067] Brett Bryant <bbryant@digium.com>
-
- * /, main/asterisk.c: Merged revisions 71066 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r71066 | bbryant | 2007-06-22 09:53:08 -0500
- (Fri, 22 Jun 2007) | 18 lines Merged revisions 71064 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r71064 | bbryant | 2007-06-22 09:39:34 -0500 (Fri, 22
- Jun 2007) | 10 lines Fixed infinite loop when controlling
- terminal was lost and return value of input function wasn't
- checked for errors. This would cause 100% cpu to be taken up.
- (closes issue #9654, issue #10010) Reported by: mnicholson, and
- eserra Idea for the patch from mnicholson, patched by me ........
- ................
-
-2007-06-22 04:35 +0000 [r71040] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_dial.c, include/asterisk/utils.h, pbx/pbx_spool.c,
- apps/app_dictate.c, apps/app_minivm.c, apps/app_test.c,
- main/logger.c, main/utils.c, apps/app_sms.c, res/res_monitor.c,
- apps/app_voicemail.c: Issue 9990 - New API ast_mkdir, which
- creates parent directories as necessary (and is faster than an
- outcall to mkdir -p)
-
-2007-06-22 04:13 +0000 [r71024] Jason Parker <jparker@digium.com>
-
- * build_tools/cflags.xml, main/asterisk.c: Nothing to see here.
-
-2007-06-22 03:15 +0000 [r71004] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 71003 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r71003 | russell | 2007-06-21 22:14:41 -0500 (Thu, 21 Jun 2007) |
- 3 lines Fix a small typo which ... well ... completely broke
- chan_iax2. oops! (issue #9937, patch by me) ........
-
-2007-06-21 23:07 +0000 [r70961] Jason Parker <jparker@digium.com>
-
- * main/manager.c, configs/manager.conf.sample,
- include/asterisk/manager.h, main/rtp.c: Add manager events for
- RTCP statistics. Also adds a new "reporting" permission for
- manager, since it can be incredibly spammy. This permission was
- discussed on the -dev mailing list some months back. Issue 8613,
- patch by johann8384, with some minor changes by me.
-
-2007-06-21 22:41 +0000 [r70951] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, /: Merged revisions 70949 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r70949 | murf | 2007-06-21 16:34:41 -0600 (Thu,
- 21 Jun 2007) | 9 lines Merged revisions 70948 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r70948 | murf | 2007-06-21 16:29:50 -0600 (Thu, 21 Jun 2007) | 1
- line This little fix is in response to bug 10016, but may not
- cure it. The code is wrong, clearly. In a situation where you set
- the CDR's amaflags, and then ForkCDR, and then set the new CDR's
- amaflags to some other value, you will see that all CDRs have had
- their amaflags changed. This is not good. So I fixed it. ........
- ................
-
-2007-06-21 21:41 +0000 [r70900] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 70899 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r70899 | file | 2007-06-21 17:40:19 -0400 (Thu,
- 21 Jun 2007) | 10 lines Merged revisions 70898 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r70898 | file | 2007-06-21 17:37:55 -0400 (Thu, 21 Jun 2007) | 2
- lines Don't explode if the gain option is specified without a
- value. (issue #9274 reported by mfarver) ........
- ................
-
-2007-06-21 21:16 +0000 [r70877-70887] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 70883 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r70883 | russell | 2007-06-21 16:14:53 -0500 (Thu, 21 Jun 2007) |
- 3 lines Put the thread reading from the socket back in the idle
- list if it deferred the processing of a full frame to another
- thread ........
-
- * /, channels/chan_iax2.c: Merged revisions 70866 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r70866 | russell | 2007-06-21 16:07:04 -0500 (Thu, 21 Jun 2007) |
- 5 lines If a full frame is received while one of the iax2 threads
- is in the middle of handling a full frame for the same call,
- queue it up for processing by that same thread later instead of
- dropping it. (issue #9937, patch by me) ........
-
-2007-06-21 20:28 +0000 [r70857] Steve Murphy <murf@digium.com>
-
- * /, cdr/cdr_custom.c: Merged revisions 70841 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r70841 | murf | 2007-06-21 14:19:36 -0600 (Thu,
- 21 Jun 2007) | 9 lines Merged revisions 70804 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r70804 | murf | 2007-06-21 13:13:17 -0600 (Thu, 21 Jun 2007) | 1
- line it was pointed out that the cdr_custom config load could get
- a lock, and under certain circumstances, would never release it.
- I also noted that the situation where more than one mapping spec
- was warned about, but did not ignore further mappings as it had
- promised. I think I have fixed both situations. ........
- ................
-
-2007-06-21 19:54 +0000 [r70809] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 70808 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r70808 | mmichelson | 2007-06-21 14:49:44 -0500 (Thu, 21 Jun
- 2007) | 4 lines When volgain is used don't leave a temporary file
- behind. (Closes Issue 8514, Reported and patched by ulogic, code
- reviewed by Jason Parker) ........
-
-2007-06-21 19:08 +0000 [r70794] Kevin P. Fleming <kpfleming@digium.com>
-
- * build_tools/make_buildopts_h: when we are building modules that
- other modules depend on, create preprocessor defines (in
- buildopts.h) marking that those modules were built
-
-2007-06-21 18:40 +0000 [r70783] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c: Merge changes from team/russell/sla_reload *
- Add support for the reload of sla.conf (closes issue #9481, patch
- by me)
-
-2007-06-21 18:03 +0000 [r70769] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Remove deprecated function call
-
-2007-06-21 15:58 +0000 [r70729-70731] Joshua Colp <jcolp@digium.com>
-
- * res/res_agi.c: Expand AGISTATUS variable to include NOTFOUND
- which is set when the AGI file could not be found. (issue #9285
- reported by srdjan)
-
- * /, main/rtp.c: Merged revisions 70727 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r70727 | file | 2007-06-21 11:22:39 -0400 (Thu, 21 Jun 2007) | 2
- lines Do not Packet2Packet bridge if packetization settings do
- not allow it. (issue #9117 reported by phsultan) ........
-
-2007-06-21 15:23 +0000 [r70728] Russell Bryant <russell@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 70726 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r70726 | russell | 2007-06-21 10:21:16 -0500 (Thu, 21 Jun 2007) |
- 2 lines Remove a couple of duplicate unlocks ........
-
-2007-06-21 14:00 +0000 [r70678] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 70677 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r70677 | file | 2007-06-21 09:58:36 -0400 (Thu, 21 Jun 2007) | 2
- lines Fix building with ODBC storage enabled. (issue #10025
- reported by denisgalvao) ........
-
-2007-06-21 13:18 +0000 [r70676] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, /: Merged revisions 70656 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r70656 | murf | 2007-06-21 07:00:39 -0600 (Thu, 21 Jun 2007) | 1
- line Via complaints aired in asterisk-users, I submit these
- changes, which allow cdr updates to see macro context/exten,
- whether hung up or not ........
-
-2007-06-20 23:33 +0000 [r70613] Jason Parker <jparker@digium.com>
-
- * /, cdr/cdr_pgsql.c: Merged revisions 70612 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r70612 | qwell | 2007-06-20 18:32:39 -0500 (Wed, 20 Jun 2007) | 4
- lines Fix some potential memory leaks in cdr_pgsql. Issue 10020,
- patch by me, with credit to prashant_jois for pointing out the
- problem. ........
-
-2007-06-20 23:31 +0000 [r70611] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Removed an extraneous debug message I'd
- left in my previous commit
-
-2007-06-20 23:31 +0000 [r70610] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c, apps/app_queue.c: Fix trunk brokenness; also,
- optimize application registration
-
-2007-06-20 23:26 +0000 [r70607] Steve Murphy <murf@digium.com>
-
- * apps/app_dial.c, main/pbx.c, apps/app_queue.c: Cleaning up a
- small disaster I created earlier
-
-2007-06-20 22:55 +0000 [r70555-70561] Jason Parker <jparker@digium.com>
-
- * /, cdr/cdr_pgsql.c: Merged revisions 70560 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r70560 | qwell | 2007-06-20 17:55:21 -0500 (Wed, 20 Jun 2007) | 1
- line Fix a stupid mistake in my last cdr_pgsql race condition fix
- ........
-
- * /, cdr/cdr_pgsql.c: Merged revisions 70554 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r70554 | qwell | 2007-06-20 17:31:35 -0500 (Wed, 20 Jun 2007) | 4
- lines Fix a race condition in cdr_pgsql that can occur when
- reloading the module. Issue 10022, patch by me, with credit to
- prashant_jois for finding the bug. ........
-
-2007-06-20 22:24 +0000 [r70553] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 70552 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r70552 | file | 2007-06-20 18:22:20 -0400 (Wed,
- 20 Jun 2007) | 10 lines Merged revisions 70551 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r70551 | file | 2007-06-20 18:20:16 -0400 (Wed, 20 Jun 2007) | 2
- lines Don't overwrite the configured username setting upon a
- REGISTER. (issue #8565 reported by jsmith) ........
- ................
-
-2007-06-20 21:38 +0000 [r70531] Steve Murphy <murf@digium.com>
-
- * apps/app_dial.c, apps/app_queue.c: As per 9228, now app_queue
- should have the proper machinery to do gosubs.
-
-2007-06-20 21:31 +0000 [r70530] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Main fix: Fixing a bug which caused
- VoiceMailMain to always report that you had 0 messages when using
- IMAP storage. Secondary fixes: adding locks to list access in
- several places Big thanks to Russell Bryant for helping out with
- this.
-
-2007-06-20 20:54 +0000 [r70493-70495] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_skinny.c: Merged revisions 70494 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r70494 | qwell | 2007-06-20 15:53:16 -0500 (Wed, 20 Jun
- 2007) | 4 lines Make sure we clear the previously dialed number
- if it did not exist. Issue 9958. ........
-
- * main/http.c: Revert the change made in revision 45474, since this
- causes other issues. Issue 10021.
-
-2007-06-20 20:10 +0000 [r70461] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/ael/ael_lex.c,
- pbx/pbx_ael.c, doc/ael.tex, include/asterisk/ael_structs.h,
- pbx/ael/ael.tab.h, CHANGES, pbx/ael/ael.flex: This finishes the
- changes for making Macro args LOCAL to the call, and allowing
- users to declare local variables.
-
-2007-06-20 19:30 +0000 [r70446] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 70445 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r70445 | tilghman | 2007-06-20 14:29:23 -0500
- (Wed, 20 Jun 2007) | 10 lines Merged revisions 70444 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r70444 | tilghman | 2007-06-20 14:25:54 -0500 (Wed, 20
- Jun 2007) | 2 lines Issue 9997 - Timelimit times out the wrong
- channel ........ ................
-
-2007-06-20 18:48 +0000 [r70398] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 70397 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r70397 | russell | 2007-06-20 13:46:49 -0500
- (Wed, 20 Jun 2007) | 13 lines Merged revisions 70396 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r70396 | russell | 2007-06-20 13:45:38 -0500 (Wed, 20
- Jun 2007) | 5 lines Fix a problem where an established call would
- not be properly disconnected when a PRI disconnect is received
- depending on which cause code was received. (issue #9588,
- original patch by softins, updated patch from jtexter3, and some
- additional feedback from mhardeman) ........ ................
-
-2007-06-20 17:55 +0000 [r70361] Joshua Colp <jcolp@digium.com>
-
- * main/frame.c, /, main/rtp.c: Merged revisions 70360 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r70360 | file | 2007-06-20 13:52:57 -0400 (Wed, 20 Jun
- 2007) | 2 lines Put the speex packetization values back in but
- disable it when setting up the smoother. ........
-
-2007-06-20 17:35 +0000 [r70358] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_dial.c, pbx/pbx_ael.c: Merge work to make U(...) option
- work for Dial
-
-2007-06-20 14:33 +0000 [r70310] Olle Johansson <oej@edvina.net>
-
- * channels/chan_zap.c: Show TDD status in "zap show channels"
- Inspired by work at Omnitor in Sweden
-
-2007-06-20 13:00 +0000 [r70253-70291] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_stack.c: Oops, shouldn't have taken that last shortcut
- (also add some checks)
-
- * apps/app_stack.c: Another method of doing local variables,
- hopefully a little closer to what codefreeze had in mind
-
- * apps/app_stack.c: Local variables for codefreeze
-
-2007-06-20 02:13 +0000 [r70234] Russell Bryant <russell@digium.com>
-
- * /, contrib/scripts/ast_grab_core: Merged revisions 70164 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r70164 | russell | 2007-06-19 19:03:22 -0500 (Tue, 19 Jun 2007) |
- 2 lines don't delete the backtrace in ast_grab_core ........
-
-2007-06-20 00:26 +0000 [r70199] Joshua Colp <jcolp@digium.com>
-
- * main/frame.c, /: Merged revisions 70198 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r70198 | file | 2007-06-19 20:24:36 -0400 (Tue, 19 Jun 2007) | 2
- lines Don't do packetization/smoother stuff with speex, it
- doesn't work. ........
-
-2007-06-19 23:38 +0000 [r70122-70162] Steve Murphy <murf@digium.com>
-
- * CHANGES: Added a little verbage to CHANGES
-
- * apps/app_dial.c, apps/app_queue.c, apps/app_rpt.c: Via bug9228,
- no way to create macros via AEL, and some of the apps allow you
- to call macros..., I modded the apps that allow macro calls to
- allow gosubs calls also, to make them AEL compliant.
-
- * UPGRADE.txt, CHANGES: Moved those comments from UPGRADE.txt to
- CHANGES. Ooops.
-
- * UPGRADE.txt: Some UPGRADE.txt comments to cover some enhancements
- added today.
-
- * configs/cdr_manager.conf.sample, cdr/cdr_manager.c: This
- enhancement provided via bug 9993, a patch to upgrade cdr_manager
- to have cdr_custom capabilities. Many thanks to eserra for this
- contribution
-
-2007-06-19 19:15 +0000 [r70088] Russell Bryant <russell@digium.com>
-
- * channels/chan_jingle.c, /, channels/chan_gtalk.c: Merged
- revisions 70084 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r70084 | russell | 2007-06-19 14:13:45 -0500 (Tue, 19 Jun 2007) |
- 3 lines Only attempt to queue a hangup on the owner channel if it
- actually exists. (issue #9795, patch from zandbelt) ........
-
-2007-06-19 18:31 +0000 [r70063] Steve Murphy <murf@digium.com>
-
- * main/channel.c, /: Merged revisions 70062 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r70062 | murf | 2007-06-19 12:23:23 -0600 (Tue,
- 19 Jun 2007) | 9 lines Merged revisions 70053 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r70053 | murf | 2007-06-19 12:07:59 -0600 (Tue, 19 Jun 2007) | 1
- line This fixes 9246, where channel variables are not available
- in the 'h' exten, on a 'ZOMBIE' channel. The fix is to
- consolidate the channel variables during a masquerade, and then
- copy the merged variables back onto the clone, so the zombie has
- the same vars that the 'original' has. ........ ................
-
-2007-06-19 17:09 +0000 [r70006] Joshua Colp <jcolp@digium.com>
-
- * /, main/rtp.c: Merged revisions 70003 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r70003 | file | 2007-06-19 13:07:40 -0400 (Tue,
- 19 Jun 2007) | 10 lines Merged revisions 69992 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r69992 | file | 2007-06-19 13:00:58 -0400 (Tue, 19 Jun 2007) | 2
- lines Handle the CC field in the RTP header. (issue #9384
- reported by DoodleHu) ........ ................
-
-2007-06-19 17:07 +0000 [r70001] Steve Murphy <murf@digium.com>
-
- * include/asterisk/callerid.h, channels/chan_zap.c,
- doc/India-CID.txt (added), configs/zapata.conf.sample: These
- changes were submitted via bug 6683, to allow CID detection in
- India, with carriers that do Polarity/DTMF CID signalling.
-
-2007-06-19 16:25 +0000 [r69988] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /: Merged revisions 69987 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r69987 | file | 2007-06-19 12:24:31 -0400 (Tue,
- 19 Jun 2007) | 10 lines Merged revisions 69986 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r69986 | file | 2007-06-19 12:21:29 -0400 (Tue, 19 Jun 2007) | 2
- lines Update BRIDGEPEER variable if set to the new channel name
- when a masquerade happens. (issue #9699 reported by dimas)
- ........ ................
-
-2007-06-19 15:27 +0000 [r69945] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 69944 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r69944 | russell | 2007-06-19 10:22:36 -0500 (Tue, 19 Jun 2007) |
- 10 lines Fix a crash that could occur when handing device state
- changes. When the state of a device changes, the device state
- thread tells the extension state handling code that it changed.
- Then, the extension state code calls the callback in chan_sip so
- that it can update subscriptions to that extension. A pointer to
- a sip_pvt structure is passed to this function as the call which
- needs a NOTIFY sent. However, there was no locking done to ensure
- that the pvt struct didn't disappear during this process. (issue
- #9946, reported by tdonahue, patch by me, patch updated to trunk
- to use the sip_pvt lock wrappers by eliel) ........
-
-2007-06-19 15:14 +0000 [r69943] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c, configs/zapata.conf.sample: Add support for
- setting nature of address, presentation, and other related SS7
- number options (#10000)
-
-2007-06-19 13:56 +0000 [r69850-69896] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 69895 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r69895 | file | 2007-06-19 09:55:25 -0400 (Tue,
- 19 Jun 2007) | 10 lines Merged revisions 69894 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r69894 | file | 2007-06-19 09:54:03 -0400 (Tue, 19 Jun 2007) | 2
- lines Perform an extra hangup check just in case. (issue #9589
- reported by bcnit) ........ ................
-
- * /, res/res_features.c: Merged revisions 69847 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r69847 | file | 2007-06-19 09:00:57 -0400 (Tue,
- 19 Jun 2007) | 10 lines Merged revisions 69846 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r69846 | file | 2007-06-19 08:57:55 -0400 (Tue, 19 Jun 2007) | 2
- lines Add parked call extension AFTER the parking slot has been
- announced, otherwise two threads will try to handle the same
- channel and it will go kaboom. (issue #9191 reported by japple)
- ........ ................
-
-2007-06-18 23:28 +0000 [r69808-69809] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Undoing my last commit. I misread the code
- before.
-
- * apps/app_voicemail.c: Cleaned up a section where there were two
- consecutive identical if statements. Combined the bodies of the
- two into one if. I blame svn merging for this.
-
-2007-06-18 22:23 +0000 [r69807] Brett Bryant <bbryant@digium.com>
-
- * apps/app_queue.c: Fixed issue where 'stop gracfeully' was hanging
- ...
-
-2007-06-18 21:58 +0000 [r69806] Joshua Colp <jcolp@digium.com>
-
- * /, main/callerid.c: Merged revisions 69805 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r69805 | file | 2007-06-18 17:57:10 -0400 (Mon, 18 Jun 2007) | 2
- lines Fix for building on PowerPC under Linux. ........
-
-2007-06-18 19:52 +0000 [r69797] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 69796 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r69796 | tilghman | 2007-06-18 14:48:17 -0500 (Mon, 18 Jun 2007)
- | 2 lines Issue 10005 - Segfault with missing arguments, plus fix
- a missing define for SIP INFO channels ........
-
-2007-06-18 19:02 +0000 [r69779-69795] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 69794 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r69794 | file | 2007-06-18 15:00:50 -0400 (Mon, 18 Jun 2007) | 2
- lines Don't count RTP timeout when involved in a T38 fax session.
- (issue #9222 reported by ivoc) ........
-
- * /, channels/chan_sip.c: Merged revisions 69775 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r69775 | file | 2007-06-18 14:18:12 -0400 (Mon,
- 18 Jun 2007) | 10 lines Merged revisions 69765 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r69765 | file | 2007-06-18 14:13:03 -0400 (Mon, 18 Jun 2007) | 2
- lines Set the peer name on the dialog to the one configured in
- sip.conf and NOT the username to be used for authentication
- attempts. (issue #9967 reported by achauvin) ........
- ................
-
-2007-06-18 17:50 +0000 [r69745-69746] Tilghman Lesher <tlesher@digium.com>
-
- * /, contrib/scripts/safe_asterisk: Merged revisions 69744 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r69744 | tilghman | 2007-06-18 12:46:40 -0500
- (Mon, 18 Jun 2007) | 10 lines Merged revisions 69743 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r69743 | tilghman | 2007-06-18 12:45:15 -0500 (Mon, 18
- Jun 2007) | 2 lines Issue 9998 - Remove SIG prefix, since it's
- not supported by ksh ........ ................
-
- * apps/app_rpt.c: Janitor for ast_localtime
-
-2007-06-18 16:56 +0000 [r69705-69709] Joshua Colp <jcolp@digium.com>
-
- * main/dnsmgr.c, /: Merged revisions 69708 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r69708 | file | 2007-06-18 12:51:36 -0400 (Mon, 18 Jun 2007) | 2
- lines Remember the DNS lookup done when dnsmgr is called for the
- first time so that it does not needlessly spit out changed
- messages when the host really didn't change. ........
-
- * main/cdr.c, main/dnsmgr.c, main/asterisk.c: Few more rwlist
- conversions... why not.
-
-2007-06-18 16:35 +0000 [r69691-69703] Russell Bryant <russell@digium.com>
-
- * res/res_config_odbc.c, /, build_tools/menuselect-deps.in,
- configure, funcs/func_odbc.c, include/asterisk/autoconfig.h.in,
- configure.ac, cdr/cdr_odbc.c, res/res_odbc.c,
- apps/app_voicemail.c: Merged revisions 69702 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r69702 | russell | 2007-06-18 11:35:02 -0500 (Mon, 18 Jun 2007) |
- 6 lines To prevent 92138749238754 more reports of "I have
- unixodbc installed, but still can't build *_odbc.so!", check for
- ltdl directly, instead of just listing it as another library to
- include in the unixodbc check in the configure script. This also
- makes ltdl show up as a dependency in menuselect so people know
- what to go install. (related to issue #9989, patch by me)
- ........
-
- * /, build_tools/prep_moduledeps: Merged revisions 69689 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r69689 | russell | 2007-06-18 11:15:12 -0500 (Mon, 18 Jun 2007) |
- 5 lines Change the use of "echo -e" to "printf". On systems where
- /bin/sh is not bash, most of the lines in menuselect-tree were
- getting a "-e" at the beginning of every line. I'm surprised
- nobody noticed this, but I think the XML parser was being very
- nice and ignoring them. ........
-
-2007-06-18 16:06 +0000 [r69663-69672] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 69668 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r69668 | file | 2007-06-18 12:04:55 -0400 (Mon, 18 Jun 2007) | 2
- lines Don't defer the BYE till later on a transfer when the
- transfer itself goes kaboom and has no hope of working. ........
-
- * /, channels/chan_sip.c: Merged revisions 69661 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r69661 | file | 2007-06-18 11:46:32 -0400 (Mon, 18 Jun 2007) | 2
- lines Few minor transfer tweaks. We can't unlock something we
- never locked, and better handle a specific scenario with doing an
- attended transfer between two non-bridged calls. ........
-
-2007-06-18 15:46 +0000 [r69662] Russell Bryant <russell@digium.com>
-
- * Makefile, /: Merged revisions 69660 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r69660 | russell | 2007-06-18 10:46:14 -0500 (Mon, 18 Jun 2007) |
- 2 lines Tweak paths for BSD systems (issue #10001, stuarth)
- ........
-
-2007-06-18 13:57 +0000 [r69626] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 69625 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r69625 | file | 2007-06-18 09:55:00 -0400 (Mon, 18 Jun 2007) | 2
- lines Fix issue where it would be possible for the negotiated
- codecs to get set back to nothing. (issue #9992 reported by
- yehavi) ........
-
-2007-06-15 20:21 +0000 [r69583] Russell Bryant <russell@digium.com>
-
- * /: This was only an issue in 1.4. This issue was fixed in trunk
- as a part of bbryant's patch to support named dynamic feature
- groups. Merged revisions 69579 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r69579 | russell | 2007-06-15 15:18:58 -0500 (Fri, 15 Jun 2007) |
- 5 lines Fix a silly deadlock in res_features that I found while
- debugging on one of blitzrage's test machines. It was one of the
- situations where he was seeing hung channels, and may be the
- cause of some of the reports from other people. (related to issue
- #9235) ........
-
-2007-06-15 19:25 +0000 [r69559] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_speech_utils.c: Merged revisions 69558 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r69558 | file | 2007-06-15 15:23:45 -0400 (Fri,
- 15 Jun 2007) | 2 lines Add support for setting the maximum length
- of acceptable DTMF in SpeechBackground.
-
-2007-06-15 15:36 +0000 [r69519] Russell Bryant <russell@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 69518 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r69518 | russell | 2007-06-15 10:27:34 -0500 (Fri, 15 Jun 2007) |
- 5 lines The SLATRUNK_STATUS variable indicated "SUCCESS" for both
- an answer of the incoming call on the trunk, or if the trunk
- reached its ring timeout. This patch changes the variable to say
- "RINGTIMEOUT" in that case. (issue #9973, reported by n00dle,
- patch by me) ........
-
-2007-06-14 23:23 +0000 [r69471] Jason Parker <jparker@digium.com>
-
- * /, main/config.c: Merged revisions 69470 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r69470 | qwell | 2007-06-14 18:22:51 -0500 (Thu,
- 14 Jun 2007) | 12 lines Merged revisions 69469 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r69469 | qwell | 2007-06-14 18:21:45 -0500 (Thu, 14 Jun 2007) | 4
- lines Fix an issue where the line number in an unterminated
- comment block error message would show the wrong line number.
- "Reported" to me on #asterisk (somebody posted an error message,
- and I happened to catch it) ........ ................
-
-2007-06-14 23:01 +0000 [r69436] Russell Bryant <russell@digium.com>
-
- * main/pbx.c, channels/chan_vpb.cc, apps/app_meetme.c,
- res/res_features.c, channels/iax2-provision.c, main/enum.c,
- res/res_monitor.c, apps/app_speech_utils.c, main/loader.c,
- main/cli.c, main/channel.c, channels/chan_misdn.c,
- apps/app_minivm.c, main/http.c, main/file.c,
- channels/chan_h323.c, res/res_indications.c,
- apps/app_directory.c, main/asterisk.c: Convert uses of strdup()
- to ast_strdup() (issue #9983, eliel)
-
-2007-06-14 22:56 +0000 [r69435] Jason Parker <jparker@digium.com>
-
- * /, sounds/Makefile: Merged revisions 69434 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r69434 | qwell | 2007-06-14 17:56:09 -0500 (Thu, 14 Jun 2007) | 1
- line Update to latest versions of sound files. ........
-
-2007-06-14 22:09 +0000 [r69394-69405] Kevin P. Fleming <kpfleming@digium.com>
-
- * include/asterisk/utils.h, main/pbx.c, /, main/say.c,
- cdr/cdr_pgsql.c, cdr/cdr_radius.c, channels/chan_iax2.c,
- cdr/cdr_tds.c, cdr/cdr_csv.c, main/cdr.c, channels/chan_phone.c,
- main/manager.c, cdr/cdr_sqlite.c, apps/app_minivm.c,
- main/callerid.c, main/logger.c, main/stdtime/localtime.c,
- cdr/cdr_odbc.c, main/asterisk.c, cdr/cdr_manager.c,
- channels/chan_mgcp.c, apps/app_voicemail.c: Merged revisions
- 69392 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r69392 | kpfleming | 2007-06-14 16:50:40 -0500 (Thu, 14 Jun 2007)
- | 2 lines use ast_localtime() in every place localtime_r() was
- being used ........
-
- * formats/format_ogg_vorbis.c: oops... somebody patched this module
- without compile-testing it... bad :-)
-
-2007-06-14 21:09 +0000 [r69327-69360] Russell Bryant <russell@digium.com>
-
- * /, main/say.c: Merged revisions 69358 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r69358 | russell | 2007-06-14 16:08:23 -0500 (Thu, 14 Jun 2007) |
- 3 lines Fix some problems with saying dates and times for the
- "tw" langauge (issue #9964, ljmid) ........
-
- * CHANGES: update CHANGES for tw support in voicemail
-
- * apps/app_voicemail.c: Add support for the tw language in
- voicemail (issue #9964, ljmid)
-
- * funcs/func_rand.c, main/frame.c, channels/chan_local.c,
- res/res_features.c, apps/app_record.c, funcs/func_strings.c,
- apps/app_test.c, main/devicestate.c, apps/app_alarmreceiver.c,
- apps/app_ices.c, channels/chan_iax2.c, main/config.c,
- res/res_smdi.c, channels/chan_skinny.c, apps/app_zapscan.c,
- apps/app_zapras.c, apps/app_amd.c, channels/chan_alsa.c,
- cdr/cdr_odbc.c, main/db.c, apps/app_dial.c, formats/format_wav.c,
- channels/chan_agent.c, apps/app_disa.c,
- formats/format_ogg_vorbis.c, channels/iax2-provision.c,
- apps/app_talkdetect.c, apps/app_db.c, res/res_monitor.c,
- apps/app_zapbarge.c, channels/chan_misdn.c,
- channels/chan_features.c, apps/app_macro.c, funcs/func_iconv.c,
- formats/format_g726.c, apps/app_chanspy.c, main/asterisk.c,
- apps/app_voicemail.c, channels/chan_vpb.cc, apps/app_meetme.c,
- res/res_musiconhold.c, cdr/cdr_pgsql.c, channels/chan_gtalk.c,
- apps/app_followme.c, codecs/codec_zap.c, cdr/cdr_radius.c,
- res/res_jabber.c, res/res_config_sqlite.c, main/enum.c,
- cdr/cdr_csv.c, main/cdr.c, main/channel.c, main/dial.c,
- channels/chan_phone.c, apps/app_osplookup.c, apps/app_minivm.c,
- res/res_agi.c, apps/app_mp3.c, main/app.c, apps/app_rpt.c,
- main/dns.c, channels/chan_mgcp.c, apps/app_nbscat.c,
- res/res_config_pgsql.c, funcs/func_version.c,
- channels/chan_zap.c, funcs/func_db.c, channels/chan_sip.c,
- apps/app_festival.c, apps/app_waitforsilence.c, res/res_crypto.c,
- res/res_adsi.c, main/acl.c, apps/app_queue.c, cdr/cdr_tds.c,
- channels/chan_jingle.c, apps/app_channelredirect.c,
- apps/app_directed_pickup.c, main/adsistub.c, main/callerid.c,
- main/file.c, channels/chan_h323.c, channels/chan_nbs.c,
- apps/app_stack.c, main/dsp.c: Add a massive set of changes for
- converting to use the ast_debug() macro. (issue #9957, patches
- from mvanbaak, caio1982, critch, and dimas)
-
-2007-06-14 16:41 +0000 [r69308] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Clean up debug messages a little bit for ss7
- linkset debugging
-
-2007-06-14 15:43 +0000 [r69261] Brett Bryant <bbryant@digium.com>
-
- * main/manager.c: Couple of manager ssl options weren't loading
- because of a typo.
-
-2007-06-14 15:25 +0000 [r69260] Jason Parker <jparker@digium.com>
-
- * funcs/func_groupcount.c, /: Merged revisions 69259 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r69259 | qwell | 2007-06-14 10:21:29 -0500 (Thu,
- 14 Jun 2007) | 12 lines Merged revisions 69258 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r69258 | qwell | 2007-06-14 10:15:53 -0500 (Thu, 14 Jun 2007) | 4
- lines Change a quite broken while loop to a for loop, so
- "continue;" works as expected instead of eating 99% CPU... Issue
- 9966, patch by me. ........ ................
-
-2007-06-13 21:20 +0000 [r69223] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 69221 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r69221 | file | 2007-06-13 17:17:28 -0400 (Wed, 13 Jun 2007) | 2
- lines Let's make chan_iax2 media only native transfers actually
- work. (issue #9376 reported by simone cittadini) ........
-
-2007-06-13 20:03 +0000 [r69187] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 69183 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r69183 | russell | 2007-06-13 14:57:38 -0500 (Wed, 13 Jun 2007) |
- 9 lines Move the logic for destroying a call when no response is
- received to a BYE outside of the block that checks for FLAG_FATAL
- to be set. This flag is only set when the packet is transmitted
- with the reliability set to XMIT_CRITICAL when the original
- packet is transmitted. A BYE is always sent with it set to
- XMIT_RELIABLE, meaning this code could never be encountered. This
- resulted in seeing some SIP channels that would never go away
- with the last packet sent being a BYE. (part of issue #9235,
- patch from jcmoore) ........
-
-2007-06-13 20:00 +0000 [r69185] Joshua Colp <jcolp@digium.com>
-
- * /, channels/iax2-parser.c: Merged revisions 69184 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r69184 | file | 2007-06-13 15:58:59 -0400 (Wed, 13 Jun
- 2007) | 2 lines Add TXMEDIA to list so that it is properly
- displayed during iax2 packet output. ........
-
-2007-06-13 19:47 +0000 [r69182] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 69181 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r69181 | mmichelson | 2007-06-13 14:41:13 -0500 (Wed, 13 Jun
- 2007) | 5 lines Contains a patch for fixing an encoding problem
- when using Outlook to view voicemail emails and attachments. This
- fix has also been tested on Thunderbird, Evolution, Pine, and
- Mutt. (Issue 9336, reported by marwick, patched by mutterc)
- ........
-
-2007-06-13 19:10 +0000 [r69147] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 69144 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r69144 | file | 2007-06-13 15:08:24 -0400 (Wed, 13 Jun 2007) | 2
- lines Really ignore NULL frames and check whether the channel
- hungup or not. (issue #9912 reported by junky) ........
-
-2007-06-13 19:05 +0000 [r69137] Jason Parker <jparker@digium.com>
-
- * channels/chan_agent.c: Completely remove callback mode and all
- references to it from chan_agent. Issue 9969, patch by eliel.
-
-2007-06-13 18:23 +0000 [r69129-69130] Joshua Colp <jcolp@digium.com>
-
- * include/asterisk/app.h, funcs/func_groupcount.c, main/app.c,
- main/cli.c: Use read/write lock based lists for group counting.
-
- * /, main/app.c: Merged revisions 69128 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r69128 | file | 2007-06-13 14:16:00 -0400 (Wed,
- 13 Jun 2007) | 10 lines Merged revisions 69127 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r69127 | file | 2007-06-13 14:12:48 -0400 (Wed, 13 Jun 2007) | 2
- lines Return group counting to previous behavior where you could
- only have one group per category. (issue #9711 reported by
- irroot) ........ ................
-
-2007-06-13 17:37 +0000 [r69081-69108] Jason Parker <jparker@digium.com>
-
- * res/res_config_pgsql.c: Continuation of issue 9968 (revision
- 69081). This should be the last one.
-
- * main/pbx.c, channels/chan_sip.c: Fixes for ast_strlen_zero()
- janitor project. Issue 9968, patch by eliel.
-
-2007-06-13 16:59 +0000 [r69017-69072] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 69071 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r69071 | russell | 2007-06-13 11:56:16 -0500 (Wed, 13 Jun 2007) |
- 3 lines Clarify a bit of logic. This doesn't change behavior in
- any way, but it is helpful when following the logic to debug
- problems like 9235. ........
-
- * /, channels/chan_iax2.c: Merged revisions 69069 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r69069 | russell | 2007-06-13 11:29:12 -0500 (Wed, 13 Jun 2007) |
- 3 lines Fix a place where a chan_iax2 pvt struct was accessed
- without the lock held. This issue was reported to me via email by
- Dmitry Mishchenko. Thanks! ........
-
- * res/snmp/agent.c: Simplify some logic and convert spaces to tabs
-
- * res/snmp/agent.c: The variable used for the return value must be
- declared as static. I broke this when applying the patch, sorry!
- (issue #9637, jeffg)
-
- * include/asterisk/logger.h: Put parenthesis around the level
- argument to ast_debug()
-
- * /, cdr/cdr_pgsql.c: Merged revisions 69016 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r69016 | russell | 2007-06-12 14:40:17 -0500 (Tue, 12 Jun 2007) |
- 4 lines Fix a memory leak pointed out by prashant_jois in
- #asterisk-bugs. PQclear() was not called on the result structure
- after doing a PQexec(). Also, fix up some formatting in passing.
- ........
-
-2007-06-12 19:38 +0000 [r69013-69015] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 69014 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r69014 | file | 2007-06-12 15:36:29 -0400 (Tue, 12 Jun 2007) | 2
- lines Change the full frame dropping log message to debug to
- avoid future bug reports. ........
-
- * /, channels/chan_iax2.c: Merged revisions 69012 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r69012 | file | 2007-06-12 15:26:38 -0400 (Tue, 12 Jun 2007) | 2
- lines Schedule the sending of a PING packet a second later than
- previously so that it does not collide with the LAGRQ. ........
-
-2007-06-12 19:19 +0000 [r68970-69011] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /: Merged revisions 69010 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r69010 | russell | 2007-06-12 14:13:41 -0500 (Tue, 12 Jun 2007) |
- 12 lines In ast_channel_make_compatible(), just return if the
- channels' read and write formats already match up. There are code
- paths that call this function on a pair of channels multiple
- times. This made calls fail that were using g729 in some cases.
- The reason is that codec_g729a will unregister itself from the
- list of available translators will all licenses are in use. So,
- the first time the function got called, the right translation
- path was allocated. However, the second time it got called, the
- code would not find a translation path to/from g729 and make the
- call fail, even if the channel actually already had a g729
- translation path allocated. (SPD-32) ........
-
- * main/pbx.c: Convert pbx.c to use ast_debug() for debug logging.
- (issue #9925, dimas)
-
- * include/asterisk/logger.h: Add a new macro, ast_debug(), which
- combines the check of the value of option_debug and the actual
- call to ast_log(). (issue #9925, dimas)
-
- * doc/ast_appdocs.tex: update the dump of application docs
-
- * apps/app_dial.c, apps/app_privacy.c, apps/app_authenticate.c,
- channels/chan_agent.c, apps/app_image.c, apps/app_chanisavail.c,
- apps/app_transfer.c, apps/app_system.c, apps/app_queue.c,
- apps/app_playback.c, apps/app_controlplayback.c,
- apps/app_osplookup.c, apps/app_sendtext.c, apps/app_minivm.c,
- apps/app_url.c, pbx/pbx_config.c, include/asterisk/options.h,
- apps/app_voicemail.c: Completely remove all of the code related
- to jumping to priority n + 101. yay! (issue #9926, caio1982)
-
-2007-06-12 14:26 +0000 [r68900-68923] Joshua Colp <jcolp@digium.com>
-
- * /, main/rtp.c: Merged revisions 68922 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r68922 | file | 2007-06-12 10:23:11 -0400 (Tue,
- 12 Jun 2007) | 10 lines Merged revisions 68921 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r68921 | file | 2007-06-12 10:18:57 -0400 (Tue, 12 Jun 2007) | 2
- lines Bring RTP back to Asterisk at the end of a native bridge no
- matter what. ........ ................
-
- * main/autoservice.c, main/app.c: Even more minor code cleanup!
-
- * main/channel.c: Minor code cleanup.
-
- * channels/chan_agent.c: Remove old stuff from the
- AgentCallbackLogin days and only autocomplete agents in the agent
- logoff CLI command that are logged in. (issue #9952 reported by
- eliel)
-
-2007-06-11 22:31 +0000 [r68855] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * main/frame.c: corrected CLI 'core show codecs' syntax for issue
- 9945, thanks eserra.
-
-2007-06-11 22:21 +0000 [r68854] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_disa.c, UPGRADE.txt: Issue 8971 - Allow DISA input to be
- ended with a '#'.
-
-2007-06-11 22:07 +0000 [r68816-68831] Jason Parker <jparker@digium.com>
-
- * main/manager.c, configs/manager.conf.sample: Change
- displayconnects option in manager.conf to be per-user. Issue
- 9932, patch by eliel
-
- * /, include/asterisk/time.h: Merged revisions 68814 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r68814 | qwell | 2007-06-11 16:20:15 -0500 (Mon, 11 Jun
- 2007) | 2 lines Solaris 10 sometimes (?) needs this include in
- order to have NULL defined. ........
-
-2007-06-11 20:51 +0000 [r68782] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_directory.c: Merged revisions 68781 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r68781 | tilghman | 2007-06-11 15:45:53 -0500 (Mon, 11 Jun 2007)
- | 2 lines Issue 9947 - fn2 was unused / incorrectly used ........
-
-2007-06-11 17:05 +0000 [r68740] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c:
- Merged revisions 68733 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r68733 | crichter | 2007-06-11 18:57:59 +0200
- (Mo, 11 Jun 2007) | 9 lines Merged revisions 68732 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r68732 | crichter | 2007-06-11 18:49:00 +0200 (Mo, 11
- Jun 2007) | 1 line added check for NULL Pointer when calling
- misdn_new. Asterisk does not allow us to create channels anymore
- when stop gracefully is used :). also modified the
- restart_indicator to 0 ........ ................
-
-2007-06-11 14:41 +0000 [r68662-68685] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: Change channel list to read/write list... I'm
- crazy.
-
- * main/channel.c, /: Merged revisions 68683 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r68683 | file | 2007-06-11 10:33:12 -0400 (Mon,
- 11 Jun 2007) | 10 lines Merged revisions 68682 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r68682 | file | 2007-06-11 10:29:58 -0400 (Mon, 11 Jun 2007) | 2
- lines Improve deadlock handling of the channel list. (issue #8376
- reported by one47) ........ ................
-
- * main/manager.c: Add username completion for manager show user CLI
- command. (issue #9929 reported by eliel)
-
- * configs/sip.conf.sample: Update documentation for proper CLI
- commands. (issue #9936 reported by eserra)
-
-2007-06-11 11:40 +0000 [r68661] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
- channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c,
- channels/misdn/isdn_lib.c: Merged revisions 68644 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r68644 | crichter | 2007-06-11 12:29:18 +0200
- (Mo, 11 Jun 2007) | 9 lines Merged revisions 68631 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r68631 | crichter | 2007-06-11 11:18:01 +0200 (Mo, 11
- Jun 2007) | 1 line fixed problem that the dummybc chanels had no
- lock, checking for the lock now. Also fixed the channel restart
- stuff, we can now specify and restart particular channels too.
- ........ ................
-
-2007-06-11 04:28 +0000 [r68596] Tilghman Lesher <tlesher@digium.com>
-
- * /, pbx/pbx_config.c: Merged revisions 68595 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r68595 | tilghman | 2007-06-10 23:21:30 -0500 (Sun, 10 Jun 2007)
- | 2 lines "dialplan save" produced garbage in the config file
- ........
-
-2007-06-09 01:06 +0000 [r68575] Jason Parker <jparker@digium.com>
-
- * channels/chan_misdn.c: Fix compile errors in chan_misdn.c
- Reported by d1mas in #asterisk-bugs on IRC.
-
-2007-06-08 22:23 +0000 [r68473-68528] Russell Bryant <russell@digium.com>
-
- * /, apps/app_dictate.c: Merged revisions 68527 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r68527 | russell | 2007-06-08 17:23:22 -0500
- (Fri, 08 Jun 2007) | 12 lines Merged revisions 68526 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r68526 | russell | 2007-06-08 17:22:36 -0500 (Fri, 08
- Jun 2007) | 4 lines Don't automatically hang up after running
- Dictate so that callers can exit cleanly using '#' (closes issue
- #9577, patch from Thomas Andrews) ........ ................
-
- * doc/asterisk-mib.txt, res/snmp/agent.c: Add support for
- retrieving the number of channels that are currently bridged via
- SNMP. (closes issue #9637, initial patch from jeffg, modified by
- me)
-
- * include/asterisk/app.h, res/res_agi.c, main/app.c,
- apps/app_controlplayback.c, apps/app_voicemail.c: Add an option
- for ControlPlayback to be able to start at an offset from the
- beginning of the file. Also, add a channel variable that
- indicates the location in the file where the Playback was
- stopped. (closes issue #7655, patch from sharkey)
-
- * main/pbx.c: Add channel variable manager event (issue #7291,
- patch from tonyh and jontow)
-
-2007-06-08 16:03 +0000 [r68453] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 68450 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r68450 | kpfleming | 2007-06-08 10:52:47 -0500 (Fri, 08 Jun 2007)
- | 2 lines actually remember the type/subclass of full frames that
- are in process ........
-
-2007-06-08 15:51 +0000 [r68449] Jason Parker <jparker@digium.com>
-
- * res/res_config_sqlite.c: Fix incorrect logic for param count.
- Issue 9918.
-
-2007-06-08 15:32 +0000 [r68448] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c: Minor formatting change to test changes to
- mantis auto-closing issues (closes issue #6000)
-
-2007-06-08 00:18 +0000 [r68374-68405] Joshua Colp <jcolp@digium.com>
-
- * /, main/say.c: Merged revisions 68401 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r68401 | file | 2007-06-07 20:17:04 -0400 (Thu,
- 07 Jun 2007) | 10 lines Merged revisions 68397 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r68397 | file | 2007-06-07 20:15:33 -0400 (Thu, 07 Jun 2007) | 2
- lines Don't call ast_waitstream_full when the control file
- descriptor and audio file descriptor are not set, simply call
- ast_waitstream! (issue #8530 reported by rickead2000) ........
- ................
-
- * main/dnsmgr.c, /: Merged revisions 68370 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r68370 | file | 2007-06-07 20:02:34 -0400 (Thu,
- 07 Jun 2007) | 10 lines Merged revisions 68368 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r68368 | file | 2007-06-07 19:59:04 -0400 (Thu, 07 Jun 2007) | 2
- lines Do a DNS lookup immediately upon calling the dnsmgr
- function, don't wait until a refresh happens. (issue #9097
- reported by plack) ........ ................
-
-2007-06-07 23:17 +0000 [r68339-68359] Russell Bryant <russell@digium.com>
-
- * /, main/say.c: Merged revisions 68354 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r68354 | russell | 2007-06-07 18:14:45 -0500
- (Thu, 07 Jun 2007) | 11 lines Merged revisions 68351 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r68351 | russell | 2007-06-07 18:13:33 -0500 (Thu, 07
- Jun 2007) | 3 lines Fix a problem where saying a character
- wouldn't properly break out when the caller pressed '#' (issue
- #8113, reported by patbaker82, patch from jamesgolovich (hey,
- long time no see!) and patbaker82) ........ ................
-
- * include/asterisk/devicestate.h, channels/chan_sip.c,
- contrib/asterisk-ng-doxygen, main/devicestate.c,
- include/asterisk/manager.h, res/res_config_sqlite.c, main/rtp.c,
- include/asterisk/http.h, include/asterisk/doxyref.h,
- main/manager.c, include/asterisk/event.h, funcs/func_shell.c,
- apps/app_skel.c, channels/chan_h323.c,
- include/asterisk/strings.h, include/asterisk/stringfields.h: Fix
- a bunch of doxygen errors and document more things (issue #9842,
- snuffy)
-
-2007-06-07 23:00 +0000 [r68327] Jason Parker <jparker@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 68326 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r68326 | qwell | 2007-06-07 18:00:01 -0500 (Thu, 07 Jun 2007) | 5
- lines Fix incorrect French syntax of "old messages". Request for
- feedback was sent to asterisk-dev mailing list, with little
- response. Issue 9118, patch by junky. ........
-
-2007-06-07 22:38 +0000 [r68325] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c: Fix a couple of places that got missed in
- the conversion to using the new API call for creating detached
- threads. (issue #9915, reported by elguro, fixed by me)
-
-2007-06-07 22:18 +0000 [r68321] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 68313 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r68313 | kpfleming | 2007-06-07 17:14:35 -0500 (Thu, 07 Jun 2007)
- | 6 lines some improvements to the IAX2 full frame dropping logic
- recently added: - use inaddrcmp(), since we have it - output the
- type of frame and subclass being dropped, and the type/subclass
- that is already being processed (which caused the drop) ........
-
-2007-06-07 21:22 +0000 [r68284-68289] Russell Bryant <russell@digium.com>
-
- * res/res_jabber.c: Doxygenify a lot of the functions in res_jabber
- (issue #9886, snuffy)
-
- * /, channels/chan_agent.c, apps/app_queue.c: Merged revisions
- 68280 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r68280 | russell | 2007-06-07 16:16:07 -0500 (Thu, 07 Jun 2007) |
- 4 lines Fix loading persistent queue members when using realtime
- configuration for queues. Also, remove an unneeded leading slash
- for the astdb family. (issue #9911, patch by atis) ........
-
-2007-06-07 20:25 +0000 [r68220-68251] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_skinny.c: Merged revisions 68249 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r68249 | qwell | 2007-06-07 15:25:18 -0500 (Thu, 07 Jun
- 2007) | 4 lines Fix an issue with newer phones which require
- packets be padded out to the correct length. Issue 9887, patch by
- DEA. ........
-
- * /, apps/app_voicemail.c: Merged revisions 68211 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r68211 | qwell | 2007-06-07 15:06:00 -0500 (Thu,
- 07 Jun 2007) | 12 lines Merged revisions 68204 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r68204 | qwell | 2007-06-07 15:02:50 -0500 (Thu, 07 Jun 2007) | 4
- lines Don't try to save voicemail greetings unless the user
- presses '1' to accept/save. Issue 9904, patch by me. ........
- ................
-
-2007-06-07 19:51 +0000 [r68201] Olle Johansson <oej@edvina.net>
-
- * CREDITS: Adding Philippe to CREDITS for hard work on detecting
- bugs in our jabber/jingle integration
-
-2007-06-07 19:50 +0000 [r68200] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 68198 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r68198 | mmichelson | 2007-06-07 14:47:42 -0500 (Thu, 07 Jun
- 2007) | 5 lines Submitting a fix for Issue 8016. Added a check to
- make sure that greetings get stored properly. (Issue 8016,
- reported by edhorton, patched by alamantia with modification by
- me. Thanks to Jason Parker for the advice on this). ........
-
-2007-06-07 19:49 +0000 [r68195-68199] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_features.c: Merged revisions 68196 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r68196 | oej | 2007-06-07 21:46:10 +0200 (Thu, 07 Jun
- 2007) | 2 lines Disable chan_features by default in menuselect
- ........
-
- * channels/chan_sip.c: - Doxygen updates - Adding docs on flags to
- be able to clean up a bit
-
-2007-06-07 19:31 +0000 [r68193] Russell Bryant <russell@digium.com>
-
- * /, main/strcompat.c: Merged revisions 68192 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r68192 | russell | 2007-06-07 14:30:30 -0500 (Thu, 07 Jun 2007) |
- 3 lines Include stdarg.h for build issues on Solaris (issue
- #9381) ........
-
-2007-06-07 18:41 +0000 [r68138-68158] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /: Merged revisions 68157 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r68157 | file | 2007-06-07 14:39:52 -0400 (Thu, 07 Jun 2007) | 2
- lines Fix logic when doing a name based channel search for a
- structure when you want to start from a specific point in the
- channel list. (issue #9324 reported by slavon) ........
-
- * doc/queues-with-callback-members.tex: AEL in trunk now uses GOSUB
- so we have to update the queues with callback members example.
- (issue #9813 reported by Mike Anikienko)
-
-2007-06-07 15:48 +0000 [r68118] Russell Bryant <russell@digium.com>
-
- * res/res_jabber.c: Minor formatting change ... testing mantis
- stuff to see if we're done (issue #9790) (closes issue #9816)
-
-2007-06-07 14:23 +0000 [r68072] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 68071 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r68071 | file | 2007-06-07 10:21:59 -0400 (Thu,
- 07 Jun 2007) | 10 lines Merged revisions 68070 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r68070 | file | 2007-06-07 10:19:40 -0400 (Thu, 07 Jun 2007) | 2
- lines Allow the 'g' option to work if used with the 'S' option.
- (issue #9888 reported by gasparz) ........ ................
-
-2007-06-07 10:06 +0000 [r67991-68040] Olle Johansson <oej@edvina.net>
-
- * /, res/res_jabber.c: Merged revisions 68030 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r68030 | oej | 2007-06-07 12:00:17 +0200 (Thu, 07 Jun 2007) | 2
- lines Adding a few Todo's to res_jabber so we don't forget.
- ........
-
- * /, res/res_jabber.c: Merged revisions 68028 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r68028 | oej | 2007-06-07 11:55:13 +0200 (Thu, 07 Jun 2007) | 4
- lines Ok, we found out that this is not about if you have any
- *active* clients using TLS, but if you have initialized TLS at
- all during the lifetime of the module. So if you reload to
- disable TLS, it won't help. ........
-
- * /, res/res_jabber.c: Merged revisions 68027 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r68027 | oej | 2007-06-07 11:42:26 +0200 (Thu, 07 Jun 2007) | 8
- lines If you have a jabber client that uses TLS, refuse unload.
- Bad fix, but will prevent crashes while we are trying to find a
- workaround. Iksemel development seems to have stalled and we
- might have to stop using the TCP/TLS connections in that library
- and use our own, which would scale better from a poll/select
- perspective I guess. It would also make it easier to migrate to
- OpenSSL and stop Asterisk from depending on both OpenSSL and
- GnuTLS. ........
-
- * /, include/asterisk/jabber.h, res/res_jabber.c: Merged revisions
- 67993 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r67993 | oej | 2007-06-07 11:00:44 +0200 (Thu, 07 Jun 2007) | 6
- lines Issue #9738 - Make sure we can unload res_jabber. Patch by
- phsultan - thanks! Due to a bug in the iksemel library, this will
- not work if you are using GTLS in the connection. That's being
- investigated. If you figure out a way to handle that without us
- having to patch iksemel, let us know in the bug report. Thanks.
- ........
-
- * res/res_jabber.c: Simplification of res_jabber code (done at
- Inria, Paris with Philippe)
-
- * main/strcompat.c: Reverting part of #67864 to be able to compile
- agi/eagi-test that relies on this without having ast_log and
- other asterisk api functions available - I could not compile on
- OS/X without reverting this.
-
-2007-06-07 00:12 +0000 [r67925-67944] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 67941 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r67941 | file | 2007-06-06 20:10:48 -0400 (Wed,
- 06 Jun 2007) | 10 lines Merged revisions 67938 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r67938 | file | 2007-06-06 20:09:13 -0400 (Wed, 06 Jun 2007) | 2
- lines Only notify the devicestate system of a peer state change
- when the peer is built from the config file. (issue #9900
- reported by arkadia) ........ ................
-
- * /, main/file.c: Merged revisions 67924 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r67924 | file | 2007-06-06 19:38:15 -0400 (Wed, 06 Jun 2007) | 2
- lines Properly handle cases where a stream can't be written to.
- (issue #9757 reported by junky) ........
-
-2007-06-06 23:12 +0000 [r67920] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Allow overlapdialing directions to be
- configurable. Bug #8554
-
-2007-06-06 22:35 +0000 [r67901] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * channels/chan_iax2.c: added CLI 'iax2 unregister <peername>' for
- issue 9812, thanks eliel
-
-2007-06-06 22:27 +0000 [r67875-67895] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c, configs/sip.conf.sample: Remove our little
- joke that was making fun of email disclaimers which nobody else
- seemed to think was very funny. Oh well ... :)
-
- * /, res/res_snmp.c: Merged revisions 67872 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r67872 | russell | 2007-06-06 17:08:02 -0500 (Wed, 06 Jun 2007) |
- 6 lines Disable reload functionality in res_snmp. It is not
- possible to initialize the snmp library more than once without
- completely unloading the module and loading it again. (issue
- #9571, reported by hristo, additional helpful debug information
- from festr, patch from me) ........
-
-2007-06-06 21:20 +0000 [r67864] Tilghman Lesher <tlesher@digium.com>
-
- * main/udptl.c, main/autoservice.c, main/frame.c,
- channels/chan_local.c, apps/app_readfile.c, res/res_features.c,
- main/threadstorage.c, main/say.c, funcs/func_strings.c,
- apps/app_alarmreceiver.c, main/devicestate.c,
- cdr/cdr_adaptive_odbc.c, channels/chan_iax2.c,
- main/indications.c, main/config.c, main/loader.c, main/cli.c,
- res/res_smdi.c, channels/chan_skinny.c, main/strcompat.c,
- main/http.c, apps/app_externalivr.c, cdr/cdr_odbc.c, main/db.c,
- res/res_speech.c, apps/app_milliwatt.c, main/sched.c,
- apps/app_dial.c, main/pbx.c, channels/chan_agent.c,
- channels/iax2-provision.c, channels/iax2-parser.c,
- main/chanvars.c, res/res_monitor.c, main/netsock.c,
- apps/app_speech_utils.c, channels/chan_misdn.c,
- funcs/func_curl.c, main/fixedjitterbuf.c, apps/app_macro.c,
- res/res_indications.c, apps/app_mixmonitor.c, main/asterisk.c,
- res/res_odbc.c, main/dlfcn.c, apps/app_voicemail.c,
- channels/chan_vpb.cc, apps/app_meetme.c, main/utils.c,
- res/res_musiconhold.c, channels/chan_gtalk.c, cdr/cdr_pgsql.c,
- apps/app_followme.c, codecs/codec_zap.c, res/res_jabber.c,
- res/res_config_sqlite.c, main/enum.c, channels/misdn_config.c,
- main/io.c, main/channel.c, main/cdr.c, funcs/func_enum.c,
- main/dial.c, main/manager.c, apps/app_osplookup.c, main/tdd.c,
- funcs/func_odbc.c, cdr/cdr_sqlite.c, res/res_agi.c,
- apps/app_minivm.c, main/app.c, apps/app_directory.c,
- apps/app_rpt.c, channels/chan_mgcp.c, apps/app_adsiprog.c,
- codecs/codec_lpc10.c, res/res_config_pgsql.c,
- channels/chan_zap.c, main/dnsmgr.c, channels/chan_sip.c,
- apps/app_festival.c, main/translate.c, main/jitterbuf.c,
- main/acl.c, apps/app_queue.c, channels/chan_oss.c, main/rtp.c,
- cdr/cdr_tds.c, main/file.c, main/callerid.c, main/event.c,
- funcs/func_devstate.c, funcs/func_callerid.c, main/dsp.c: Issue
- 9869 - replace malloc and memset with ast_calloc, and other
- coding guidelines changes
-
-2007-06-06 21:16 +0000 [r67813-67863] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 67862 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r67862 | russell | 2007-06-06 16:14:46 -0500 (Wed, 06 Jun 2007) |
- 4 lines Fix a crash when doing call pickups with SIP phones. The
- code unlocked the channel when it should not have. (issue #9652,
- reported by corruptor, fixed by me) ........
-
- * res/res_features.c, include/asterisk/features.h: Constify the
- return values of ast_parking_ext() and ast_pickup_ext()
-
- * main/manager.c: Minor formatting change to test closing mantis
- issues through commit tags (closes issue #9828)
-
- * main/manager.c: Minor formatting change to test closing mantis
- issues through commit tags (closes issue #9828)
-
- * apps/app_voicemail.c: Please forgive this flood of tiny changes
- ... this will be cool when it works how we want it to :) (testing
- mantis+svn) (issue #9828)
-
-2007-06-06 19:46 +0000 [r67808] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 67804 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r67804 | mmichelson | 2007-06-06 14:26:55 -0500 (Wed, 06 Jun
- 2007) | 10 lines Fix for Issue 9810. There was a segfault under a
- specific set of circumstances: 1. VoiceMailMain was configured in
- the dialplan with an extension as its argument 2. A message was
- left for this mailbox 3. Tried to call VoiceMailMain but hung up
- before entering password. This was fixed by checking that a
- pointer was non-null prior to trying to dereference it. (Issue
- 9810, reported by xmarksthespot, patched by Corydon76 with
- modifications by me). ........
-
-2007-06-06 19:44 +0000 [r67787-67807] Russell Bryant <russell@digium.com>
-
- * apps/app_voicemail.c: minor formatting change ... testing
- mantis/svn (issue #9828)
-
- * apps/app_voicemail.c: Don't try to check the result of alloca ...
- ... testing mantis/svn stuff ... (issue #9828)
-
- * main/dsp.c: Yet another minor change to test mantis/svn (issue
- #9828)
-
- * main/dsp.c: minor formatting change ... testing mantis/svn (issue
- #9828)
-
- * main/dsp.c: minor formatting change ... testing mantis/svn (issue
- #9828)
-
- * main/app.c: Formatting change ... testing (issue #9828)
-
-2007-06-06 19:02 +0000 [r67784] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Fixing a crash wherein Asterisk would
- segfault when attempting to leave a voicemail when IMAP storage
- was enabled. Though no bug was reported to the bugtracker, there
- was mention of this made as a note on bug 9810 by edhorton.
-
-2007-06-06 19:00 +0000 [r67697-67782] Russell Bryant <russell@digium.com>
-
- * main/app.c: Make another formatting change ... testing mantis/svn
- stuff (issue #9828)
-
- * main/app.c: Another minor formatting change ... testing
- mantis/svn (issue #9828)
-
- * main/app.c: Minor formatting change ... testing mantis/svn (issue
- #9828)
-
- * channels/chan_iax2.c: Make another small tweak ... mantis/svn
- testing (issue #9828)
-
- * res/res_features.c: Another tiny formatting change for testing
- ... (issue #9828)
-
- * main/app.c: More random formatting changes to test Mantis/SVN
- integration (issue #9828)
-
- * main/app.c: Make a completely arbitrary formatting change to test
- out some Mantis/SVN integration stuff. (issue #9828)
-
- * main/channel.c, /, include/asterisk/linkedlists.h: Merged
- revisions 67716 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r67716 | russell | 2007-06-06 11:55:59 -0500
- (Wed, 06 Jun 2007) | 13 lines Merged revisions 67715 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r67715 | russell | 2007-06-06 11:40:51 -0500 (Wed, 06
- Jun 2007) | 5 lines We have some bug reports showing crashes due
- to a double free of a channel. Add a sanity check to
- ast_channel_free() to make sure we don't go on trying to free a
- channel that wasn't found in the channel list. (issue #8850, and
- others...) ........ ................
-
- * res/res_features.c: Change "show parkedcalls" to "parkedcalls
- show" and mark the previous command as deprecated. Also, convert
- the CLI command to the new style. (issue #9861, patch from eliel)
-
-2007-06-06 13:32 +0000 [r67595-67651] Joshua Colp <jcolp@digium.com>
-
- * /, main/rtp.c: Merged revisions 67650 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r67650 | file | 2007-06-06 09:30:25 -0400 (Wed,
- 06 Jun 2007) | 10 lines Merged revisions 67649 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r67649 | file | 2007-06-06 09:28:34 -0400 (Wed, 06 Jun 2007) | 2
- lines Reinvite the RTP back to the Asterisk machine when the
- timeout happens. (issue #9888 reported by gasparz) ........
- ................
-
- * /, main/translate.c: Merged revisions 67631 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r67631 | file | 2007-06-06 09:18:39 -0400 (Wed, 06 Jun 2007) | 2
- lines Fix plc_samples warning when registering a translator.
- (issue #9897 reported by xylome) ........
-
- * /, apps/app_directed_pickup.c: Merged revisions 67626 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r67626 | file | 2007-06-06 09:16:34 -0400 (Wed, 06 Jun 2007) | 2
- lines Include macroexten while searching for a channel to pick up
- in case they are in a macro. (issue #9491 reported by jamesb63)
- ........
-
- * /, res/res_agi.c: Merged revisions 67597 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r67597 | file | 2007-06-06 08:34:06 -0400 (Wed, 06 Jun 2007) | 2
- lines Make the new "agi debug off" CLI command work. (issue #9890
- reported by eliel) ........
-
- * channels/chan_zap.c: When SS7 is enabled add w/SS7 to the end.
- (issue #9893 reported by Mike Anikienko)
-
- * /, main/devicestate.c: Merged revisions 67594 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r67594 | file | 2007-06-06 08:20:27 -0400 (Wed,
- 06 Jun 2007) | 10 lines Merged revisions 67593 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r67593 | file | 2007-06-06 08:18:36 -0400 (Wed, 06 Jun 2007) | 2
- lines Revert channel name splitting fix for Zap. The moral of the
- story is don't use - in your user/peer names. (issue #9668
- reported by stevedavies) ........ ................
-
-2007-06-05 23:02 +0000 [r67560] Russell Bryant <russell@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 67558 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r67558 | russell | 2007-06-05 18:01:44 -0500 (Tue, 05 Jun 2007) |
- 5 lines Fix some crashes related to the use of the "meetme" CLI
- command. The code for this command was not locking the conference
- list at all. (issue #9351, reported by and patch submitted by
- Junk-Y, committed patch is different and by me) ........
-
-2007-06-05 22:59 +0000 [r67557] Mark Michelson <mmichelson@digium.com>
-
- * main/cli.c: Found a bug where when "core set debug #" is used,
- the verbosity is read as the old value instead of the old debug
- value, leading to an erroneous status message after setting. This
- was purely a cosmetic issue and had no other underlying problems.
-
-2007-06-05 22:04 +0000 [r67529] Steve Murphy <murf@digium.com>
-
- * utils/Makefile, /, pbx/ael/ael.tab.c, pbx/ael/ael.y,
- pbx/pbx_ael.c, pbx/Makefile: Merged revisions 67526 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r67526 | murf | 2007-06-05 15:30:18 -0600 (Tue, 05 Jun
- 2007) | 1 line this fixes bug 9883, wherein macros were not
- allowing the includes construct. fixed and tested, looks OK. Now
- includes can serve as an adjunct to catch. ........
-
-2007-06-05 20:55 +0000 [r67493] Russell Bryant <russell@digium.com>
-
- * /, include/asterisk/linkedlists.h: Merged revisions 67492 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r67492 | russell | 2007-06-05 15:53:28 -0500 (Tue, 05 Jun 2007) |
- 16 lines This bug has been hanging over my head ever since I
- wrote this SLA code. Every time I tried to go debug it by adding
- some debug output, the behavior would change. It turns out I
- wasn't crazy. I had the following piece of code: if (remove)
- AST_LIST_REMOVE_CURRENT(...); Well, AST_LIST_REMOVE_CURRENT was
- not wrapped in braces, so my conditional statement didn't do much
- good at all. It always ran at least all of the macro minus the
- first statement, so I was seeing list entries magically disappear
- when they weren't supposed to. After many hours of debugging, I
- have come to this extremely irritating fix. :) (issues #9581,
- #9497) ........
-
-2007-06-05 20:16 +0000 [r67486] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Merged revisions 67424 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r67424 | mmichelson | 2007-06-05 13:32:50 -0500 (Tue, 05 Jun
- 2007) | 5 lines Fix for bug number 9786, wherein voicemails saved
- to IMAP storage using extensions other than gsm were unable to be
- played over the phone. (Issue 9786, reporter: xmarksthespot,
- Patched by xmarksthe spot with revisions by me, reviewed by
- Russell Bryant). ........
-
-2007-06-05 19:50 +0000 [r67458] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 67457 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r67457 | russell | 2007-06-05 14:48:02 -0500 (Tue, 05 Jun 2007) |
- 2 lines Suppress a bunch of debug output unless option_debug is
- on ........
-
-2007-06-05 18:23 +0000 [r67423] Steve Murphy <murf@digium.com>
-
- * /, pbx/pbx_ael.c: Merged revisions 67420 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r67420 | murf | 2007-06-05 12:17:28 -0600 (Tue, 05 Jun 2007) | 1
- line Added code to automatically add a default case to switches
- that don't have one. In some cases, rather than fall thru, it
- results in a goto with -1 result, which terminates the extension;
- a sort of dialplan seqfault, sort of. This was required to fix
- bug reported in 9881 ........
-
-2007-06-05 18:19 +0000 [r67398-67422] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_skinny.c: Merged revisions 67421 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r67421 | qwell | 2007-06-05 13:18:24 -0500 (Tue, 05 Jun
- 2007) | 4 lines Correctly update date/time on devices throughout
- the life of the device, instead of just at registration. Issue
- 9152, yet another patch by DEA. ........
-
- * main/manager.c: Make sure we default allowmultiplelogin to
- on/yes, per the default stated in the config. Issue 9885, patch
- by eliel.
-
-2007-06-05 17:24 +0000 [r67397] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * channels/misdn/isdn_msg_parser.c: changed #if DEBUG to #ifdef
- DEBUG to fix make failure when configured with --enable-dev-mode
-
-2007-06-05 17:11 +0000 [r67361-67380] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c: Improve the way that the zaptel channel name
- is created by using the Asterisk strings API and by only
- allocating space on the stack
-
- * /: Merged revisions 67360 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r67360 | russell | 2007-06-05 11:56:36 -0500 (Tue, 05 Jun 2007) |
- 5 lines Fix a problem that showed itself by causing Zap channel
- names to be completely bogus on my machine.
- ast_safe_string_alloc() was broken. It called vsnprintf() on a
- va_args list twice without re-initializing it. After the first
- usage, va_end() and va_start() must be called again. ........
-
-2007-06-05 16:21 +0000 [r67345-67350] Christian Richter <christian.richter@beronet.com>
-
- * /, channels/misdn/chan_misdn_config.h: Merged revisions 67334 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r67334 | crichter | 2007-06-05 18:14:07 +0200
- (Di, 05 Jun 2007) | 9 lines Merged revisions 67307 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r67307 | crichter | 2007-06-05 17:42:03 +0200 (Di, 05
- Jun 2007) | 1 line briding is a bool, fixed copy and paste issue.
- ........ ................
-
- * channels/chan_misdn.c, /: Merged revisions 67329 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r67329 | crichter | 2007-06-05 18:11:57 +0200
- (Di, 05 Jun 2007) | 9 lines Merged revisions 67306 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r67306 | crichter | 2007-06-05 17:39:43 +0200 (Di, 05
- Jun 2007) | 1 line simplified the EVENT_SETUP handling in the
- cb_events function a lot. Commented the different possibilities a
- bit and made functions of shared code. When the dialed extension
- does not exist in the extensions.conf we'll jump into the 'i'
- extension if this does exist, else we disconnect the call with
- the cause:1 = No Route to Destination. ........ ................
-
-2007-06-05 15:54 +0000 [r67310] Russell Bryant <russell@digium.com>
-
- * /, include/asterisk/module.h, main/asterisk.c, main/loader.c:
- Merged revisions 67308 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r67308 | russell | 2007-06-05 10:51:53 -0500 (Tue, 05 Jun 2007) |
- 5 lines When shutting down "gracefully", go through and run the
- unload() callbacks for all of the modules. "stop now" is
- considered a non-graceful shutdown and will not go through this
- process. (issue #9804, reported by chrisost, patch by me)
- ........
-
-2007-06-05 15:24 +0000 [r67305] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 67304 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r67304 | file | 2007-06-05 12:22:30 -0300 (Tue, 05 Jun 2007) | 2
- lines Only muck with the thread structure if an idle one was
- found/created. ........
-
-2007-06-05 14:59 +0000 [r67272-67273] Russell Bryant <russell@digium.com>
-
- * doc/CODING-GUIDELINES: add a note about inline comments
-
- * channels/chan_iax2.c: Doxygenify the comments for new members of
- the iax2_thread struct
-
-2007-06-05 14:45 +0000 [r67271] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 67270 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r67270 | kpfleming | 2007-06-05 09:35:52 -0500 (Tue, 05 Jun 2007)
- | 3 lines ensure that a burst of full frames (AST_FRAME_DTMF
- being the prime example) will not be processed out of order...
- this is a brute force fix, but seems to be the safest fix for now
- (thanks to the Digium PQ department for finding this bug)
- ........
-
-2007-06-05 11:48 +0000 [r67240] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /, channels/misdn/chan_misdn_config.h,
- channels/misdn_config.c: Merged revisions 67210 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r67210 | crichter | 2007-06-05 12:25:32 +0200
- (Di, 05 Jun 2007) | 9 lines Merged revisions 67209 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r67209 | crichter | 2007-06-05 12:05:45 +0200 (Di, 05
- Jun 2007) | 1 line added possibility to deactivate bridging per
- port ........ ................
-
-2007-06-04 23:45 +0000 [r67164] Tilghman Lesher <tlesher@digium.com>
-
- * /, funcs/func_math.c: Merged revisions 67162 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r67162 | tilghman | 2007-06-04 18:43:01 -0500
- (Mon, 04 Jun 2007) | 10 lines Merged revisions 67161 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r67161 | tilghman | 2007-06-04 18:41:49 -0500 (Mon, 04
- Jun 2007) | 2 lines According to MATH, 0+1181000386 = 1181000448.
- Oops. ........ ................
-
-2007-06-04 23:32 +0000 [r67160] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 67158 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r67158 | russell | 2007-06-04 18:31:40 -0500 (Mon, 04 Jun 2007) |
- 5 lines Fix up a bunch of places where the iax2 pvt structure can
- disappear and the code did not account for it and crashes.
- (issues #9642, #9569, #9666, probably others ... based on the
- work by stevedavies and mihai, with additional changes from me)
- ........
-
-2007-06-04 23:29 +0000 [r67122-67157] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_skinny.c: Merged revisions 67156 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r67156 | qwell | 2007-06-04 18:26:28 -0500 (Mon, 04 Jun
- 2007) | 6 lines Fix for skinny keepalives. If there is no traffic
- from the phone for (keep_alive * 1100) ms (arbitrarily adding 10%
- for network issues, etc), unregister the device. Issue 8394,
- patch by DEA. ........
-
- * /, channels/chan_mgcp.c: Merged revisions 67121 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r67121 | qwell | 2007-06-04 17:36:57 -0500 (Mon, 04 Jun 2007) | 4
- lines Fixes for dtmf/dialing with mgcp (similar to the recent fix
- for chan_skinny) Issue 9855, patch by DEA. ........
-
-2007-06-04 22:29 +0000 [r67120] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 67119 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r67119 | russell | 2007-06-04 17:28:55 -0500 (Mon, 04 Jun 2007) |
- 6 lines Add comments for two functions that get called with the
- appropriate call locked, but perform operations that could result
- in the pvt structure getting destroyed before returning again,
- causing numerous seg faults all over the module. (inspired by
- issues #9642, #9569, and #9666, and the work done by stevedavies
- and mihai) ........
-
-2007-06-04 22:15 +0000 [r67095] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, /: Merged revisions 67073 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r67073 | murf | 2007-06-04 15:59:34 -0600 (Mon, 04 Jun 2007) | 1
- line This typo has been here since 1.4 forked. It has been the
- source of heartburn to many a dialplan/CDR programmer. ........
-
-2007-06-04 21:48 +0000 [r67070-67072] Russell Bryant <russell@digium.com>
-
- * /, main/rtp.c: Merged revisions 67071 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r67071 | russell | 2007-06-04 16:47:36 -0500 (Mon, 04 Jun 2007) |
- 2 lines Add a missing \n. (pointed out by jcmoore on IRC)
- ........
-
- * channels/chan_iax2.c: Remove a leftover unlock and lock of the
- iax2 pvt struct lock that was left over from my attempt at
- putting pvt structs in a hash table. It can cause seg faults, and
- has no reason to stay. (issue #9642, pointed out by stevedavies)
-
-2007-06-04 19:32 +0000 [r67063-67069] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 67068 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r67068 | file | 2007-06-04 15:31:09 -0400 (Mon, 04 Jun 2007) | 2
- lines Better handle SIP devices that say they have SDP content...
- but really don't. (issue #9398 reported by mthomasslo) ........
-
- * apps/app_dial.c, /: Merged revisions 67066 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r67066 | file | 2007-06-04 13:59:14 -0400 (Mon, 04 Jun 2007) | 2
- lines Initialize cidname variable to nothing since it may be used
- without having been touched. (issue #9661 reported by dimas)
- ........
-
- * /, res/res_features.c: Merged revisions 67064 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r67064 | file | 2007-06-04 13:41:59 -0400 (Mon, 04 Jun 2007) | 2
- lines Returning a value that indicates the parking of a call was
- a success when it really wasn't (because the parking slot
- selected was in use) is the wrong thing to do. (issue #9723
- reported by mdu113) ........
-
- * apps/app_directed_pickup.c: Minor clean up. Constify a few
- variables and use ast_strlen_zero in a few places.
-
-2007-06-04 17:12 +0000 [r67062] Tilghman Lesher <tlesher@digium.com>
-
- * contrib/init.d/rc.debian.asterisk,
- contrib/init.d/rc.mandrake.asterisk, /,
- contrib/init.d/rc.redhat.asterisk,
- contrib/init.d/rc.gentoo.asterisk,
- contrib/init.d/rc.mandrake.zaptel,
- contrib/init.d/rc.slackware.asterisk: Merged revisions 67061 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r67061 | tilghman | 2007-06-04 12:11:43 -0500
- (Mon, 04 Jun 2007) | 10 lines Merged revisions 67060 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r67060 | tilghman | 2007-06-04 12:10:30 -0500 (Mon, 04
- Jun 2007) | 2 lines Add revision Id tags (by request of tzafrir)
- ........ ................
-
-2007-06-04 16:03 +0000 [r67024-67029] Russell Bryant <russell@digium.com>
-
- * /, configure, configure.ac: Merged revisions 67026 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r67026 | russell | 2007-06-04 11:02:31 -0500 (Mon, 04
- Jun 2007) | 6 lines Change the configure script to build a test
- program against libcurl to make sure the results from curl-config
- can be used to compile successfully. This is intended to help
- prevent a situation where you are cross compiling, and the
- configure script finds the curl library installed on the host.
- (issue #9865, reported and patched by zandbelt) ........
-
- * main/ast_expr2f.c, pbx/ael/ael_lex.c, main/app.c: Change javadoc
- style code documentation to the same format we use elsewhere.
- (issue #9864, patch from snuffy)
-
-2007-06-04 15:53 +0000 [r67023] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_jabber.c: Merged revisions 67021 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r67021 | tilghman | 2007-06-04 10:50:16 -0500 (Mon, 04 Jun 2007)
- | 2 lines Issue 9739 - Malformed jid causes a crash ........
-
-2007-06-04 15:50 +0000 [r67016-67022] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 67020 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r67020 | russell | 2007-06-04 10:47:40 -0500 (Mon, 04 Jun 2007) |
- 7 lines Resolve a deadlock in chan_iax2. When handling an
- implicit ACK to a frame that was marked as the final transmission
- for a call, don't call iax2_destroy() for that call while the
- global frame queue is still locked. There is a very nice
- explanation of the deadlock in the report. (issue #9663, thorough
- report and patch from stevedavies, additional positive test
- reports from mihai and joff_oconnell) ........
-
- * include/asterisk/stringfields.h: Fix some compiler warnings in
- C++ modules. (issue #9866, reported by osk, patch by Corydon76)
-
- * channels/chan_sip.c, main/netsock.c: Fix a couple of places where
- "tos" was used instead of "cos". (issue #9540, patch by IgorG)
-
-2007-06-04 11:48 +0000 [r66998] Joshua Colp <jcolp@digium.com>
-
- * apps/app_mixmonitor.c: Add support for autocompleting start/stop
- options of the mixmonitor CLI command. (issue #9862 reported by
- eliel)
-
-2007-06-03 06:10 +0000 [r66981] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_jingle.c, channels/chan_phone.c,
- channels/chan_features.c, channels/chan_h323.c,
- channels/chan_gtalk.c, channels/chan_nbs.c, channels/chan_mgcp.c:
- ast_calloc janitor (Inspired by issue 9860)
-
-2007-06-01 23:39 +0000 [r66957-66959] Russell Bryant <russell@digium.com>
-
- * main/pbx.c: remove a bogus comment that came from copy/paste
-
- * include/asterisk/devicestate.h, include/asterisk.h, main/pbx.c,
- include/asterisk/event_defs.h, main/devicestate.c,
- include/asterisk/pbx.h, apps/app_queue.c, main/asterisk.c: Merge
- major changes to the way device state is passed around Asterisk.
- The two places that cared about device states were app_queue and
- the hint code in pbx.c. The changes include converting it to use
- the Asterisk event system, as well as other efficiency
- improvements. * app_queue: This module used to register a
- callback into devicestate.c to monitor device state changes. Now,
- it is just a subscriber to Asterisk events with the type, device
- state. * pbx.c hints: Previously, the device state processing
- thread in devicestate.c would call ast_hint_state_changed() each
- time the state of a device changed. Then, that code would go
- looking for all the hints that monitor that device, and call
- their callbacks. All of this blocked the device state processing
- thread. Now, the hint code is a subscriber of Asterisk events
- with the type, device state. Furthermore, when this code receives
- a device state change event, it queues it up to be processed by
- another thread so that it doesn't block one of the event
- processing threads.
-
- * channels/chan_iax2.c: Remove 80 bytes in the iax2_registry struct
- that weren't being used
-
-2007-06-01 21:49 +0000 [r66920] Tilghman Lesher <tlesher@digium.com>
-
- * /, funcs/func_odbc.c: Merged revisions 66919 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r66919 | tilghman | 2007-06-01 16:45:44 -0500 (Fri, 01 Jun 2007)
- | 2 lines On some drivers, deallocating the statement handle
- isn't enough. We also have to clear the cursor (nice, Oracle)
- ........
-
-2007-06-01 21:33 +0000 [r66910-66918] Mark Michelson <mmichelson@digium.com>
-
- * /: Merged revisions 66916 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- ........
-
- * /, apps/app_voicemail.c: Merged revisions 66897 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r66897 | mmichelson | 2007-06-01 16:09:30 -0500 (Fri, 01 Jun
- 2007) | 3 lines Submitting a fix for voicemail with IMAP storage.
- Attachments with format specified as gsm were duplicated (i.e.
- two attachments) were left. Thank you very much to xmarksthespot
- for submitting the patch that fixed this. (Issues 9787 and 8873,
- Reported by xmarksthespot and jerjer, patched by xmarksthespot)
- ........
-
-2007-06-01 19:42 +0000 [r66880-66882] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_skinny.c: Merged revisions 66881 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r66881 | russell | 2007-06-01 14:41:30 -0500 (Fri, 01
- Jun 2007) | 6 lines Changes to the way DTMF is handled in the
- core broke dialing in chan_skinny. This patch makes chan_skinny
- usable again. I did not end up testing this, but there are
- multiple positive test reports listed in the bug report. (issue
- #9596, reported by pj, testing by pj and mvanbaak, and the fix
- was written by DEA) ........
-
- * /, apps/app_page.c: Merged revisions 66879 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r66879 | russell | 2007-06-01 14:35:13 -0500 (Fri, 01 Jun 2007) |
- 2 lines List app_meetme as a module that app_page depends on.
- ........
-
-2007-06-01 18:36 +0000 [r66878] Jason Parker <jparker@digium.com>
-
- * res/res_config_sqlite.c: Documentation fixes for
- res_config_sqlite. Issue 9854, patch by tzafrir.
-
-2007-06-01 13:48 +0000 [r66856] Russell Bryant <russell@digium.com>
-
- * configs/sip.conf.sample: Add some more information about the SIP
- Disclaimer header.
-
-2007-05-31 23:04 +0000 [r66822] Tilghman Lesher <tlesher@digium.com>
-
- * /, doc/asterisk.8: Merged revisions 66821 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r66821 | tilghman | 2007-05-31 18:03:28 -0500 (Thu, 31 May 2007)
- | 2 lines Issue 9850 - update preferred command line syntax
- ........
-
-2007-05-31 21:23 +0000 [r66772-66818] Russell Bryant <russell@digium.com>
-
- * configs/sip.conf.sample: fix a typo.
-
- * channels/chan_sip.c, configs/sip.conf.sample: To satisfy some
- legal concerns, add an option for chan_sip to include a
- disclaimer along with SIP messages in the header, X-Disclaimer.
- This is off by default. Also, the text of the disclaimer can be
- customized in sip.conf.
-
- * include/asterisk/app.h, /, include/asterisk/speech.h,
- res/res_speech.c: Merged revisions 66775 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r66775 | russell | 2007-05-31 13:41:58 -0500 (Thu, 31 May 2007) |
- 3 lines Change a couple of header files to not use "new", which
- is a reserved keyword in C++. (issue #9830, reported by osk)
- ........
-
- * res/res_features.c, CHANGES, configs/features.conf.sample: Add
- support for configuring named groups of custom call features in
- features.conf. This allows you to create a feature one time, and
- then map it into groups for various different key mappings for
- the same feature, as well as easy access control to groups of
- features. (patch from bbryant)
-
- * res/res_features.c, configs/features.conf.sample: Revert changes
- that snuck in with revision 66724.
-
- * apps/app_minivm.c: - Don't check if the list is empty needlessly
- - Don't free structures before calling load_config(), because
- load_config() already does it - Use the existing functions for
- freeing the minivm structures instead of replicating the code
- (issue #9846, patch from eliel)
-
-2007-05-31 17:16 +0000 [r66771] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_macro.c: Merged revisions 66770 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r66770 | tilghman | 2007-05-31 12:15:09 -0500
- (Thu, 31 May 2007) | 10 lines Merged revisions 66744 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r66744 | tilghman | 2007-05-31 10:58:45 -0500 (Thu, 31
- May 2007) | 2 lines Issue 9818 - Fix for issue 8329 breaks
- pbx_realtime. Issue 8329 will remain unfixed for pbx_realtime,
- but only because we lack core API to do it. ........
- ................
-
-2007-05-31 16:18 +0000 [r66769] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 66768 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r66768 | file | 2007-05-31 12:14:48 -0400 (Thu,
- 31 May 2007) | 10 lines Merged revisions 66764 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r66764 | file | 2007-05-31 12:12:39 -0400 (Thu, 31 May 2007) | 2
- lines It is now possible for this path of execution to have the
- frame pointer be NULL, therefore we need to check for it before
- trying to access it. (issue #9836 reported by barthpbx) ........
- ................
-
-2007-05-31 15:05 +0000 [r66734] Tilghman Lesher <tlesher@digium.com>
-
- * configs/func_odbc.conf.sample, funcs/func_odbc.c: Issue 9799 -
- Multirow results for func_odbc
-
-2007-05-31 14:52 +0000 [r66724] Russell Bryant <russell@digium.com>
-
- * res/res_features.c, apps/app_minivm.c,
- configs/features.conf.sample: Fix a crash on reload by using
- calloc() instead of malloc() to ensure that data is properly
- initialized. (issue #9765, reported by MatsK, patch from eliel)
-
-2007-05-31 10:26 +0000 [r66705] Olle Johansson <oej@edvina.net>
-
- * include/asterisk/app.h, apps/app_osplookup.c,
- include/asterisk/event.h, apps/app_meetme.c, channels/chan_sip.c,
- include/asterisk/event_defs.h, apps/app_skel.c,
- apps/app_minivm.c, res/res_jabber.c: Issue #9842 - Doxygen
- updates by snuffy. Thanks! (Committed from Media Plaza in
- Utrecht, Netherlands - Open Source VoIP conference)
-
-2007-05-30 23:44 +0000 [r66672] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 66671 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r66671 | mmichelson | 2007-05-30 18:26:39 -0500 (Wed, 30 May
- 2007) | 2 lines Fixed seg-faults when recording greetings in
- voicemail with IMAP enabled. (Issue No. 9734, reported by
- xmarksthespot, patched by me) ........
-
-2007-05-30 17:23 +0000 [r66603-66638] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_zap.c, channels/chan_features.c: This concludes my
- tweaking of things.
-
-2007-05-30 05:17 +0000 [r66539-66585] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_channelredirect.c, channels/chan_vpb.cc,
- res/res_config_odbc.c, funcs/func_shell.c, funcs/func_cdr.c,
- apps/app_zapras.c, res/res_indications.c, apps/app_transfer.c,
- apps/app_stack.c, funcs/func_devstate.c, res/res_config_sqlite.c,
- res/res_odbc.c: Issue 9477 - Improve menuselect labels
-
- * /, funcs/func_strings.c: Merged revisions 66538 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r66538 | tilghman | 2007-05-29 16:56:07 -0500
- (Tue, 29 May 2007) | 10 lines Merged revisions 66537 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r66537 | tilghman | 2007-05-29 16:49:35 -0500 (Tue, 29
- May 2007) | 2 lines If the value of a variable passed to FIELDQTY
- is blank, then FIELDQTY should return 0, not 1. ........
- ................
-
- * funcs/func_enum.c: Shorten description to a much more reasonable
- length
-
-2007-05-29 19:53 +0000 [r66502-66505] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: oops. Thanks Terry.
-
- * /, channels/chan_sip.c: Merged revisions 66503 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r66503 | oej | 2007-05-29 21:32:57 +0200 (Tue, 29 May 2007) | 2
- lines Properly handle 408 request timeout - according to the RFC,
- the dialog dies if a request in a dialog gets this response.
- ........
-
- * /, channels/chan_sip.c: Merged revisions 66474 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r66474 | oej | 2007-05-29 21:02:04 +0200 (Tue, 29 May 2007) | 2
- lines Don't issue hangup on hangup on hangup on hangup (for
- jcmoore) ........
-
-2007-05-29 19:00 +0000 [r66471] Doug Bailey <dbailey@digium.com>
-
- * main/dsp.c: Changed the dtmf detection to integer based goertzel
- algorithm.
-
-2007-05-29 16:46 +0000 [r66438] Joshua Colp <jcolp@digium.com>
-
- * /, main/rtp.c: Merged revisions 66437 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r66437 | file | 2007-05-29 12:44:34 -0400 (Tue, 29 May 2007) | 2
- lines Handle cases where a frame may have no data. (issue #9519
- reported by dmb) ........
-
-2007-05-29 16:19 +0000 [r66432-66433] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 66414 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r66414 | oej | 2007-05-29 18:07:44 +0200 (Tue, 29 May 2007) | 2
- lines Don't reset hangupcause if we already have one ........
-
- * /, channels/chan_sip.c: Merged revisions 66404 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r66404 | oej | 2007-05-29 18:02:50 +0200 (Tue, 29 May 2007) | 2
- lines Tracking down hanging channels, killing them one by one.
- Issue #9235 and related ........
-
-2007-05-29 15:44 +0000 [r66399] Joshua Colp <jcolp@digium.com>
-
- * /, doc/datastores.txt: Merged revisions 66398 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r66398 | file | 2007-05-29 11:43:16 -0400 (Tue, 29 May 2007) | 2
- lines Update datastores documentation. (issue #9801 reported by
- mnicholson) ........
-
-2007-05-29 10:02 +0000 [r66367] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 66363 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r66363 | oej | 2007-05-29 11:41:40 +0200 (Tue,
- 29 May 2007) | 10 lines Merged revisions 66349 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r66349 | oej | 2007-05-29 09:53:14 +0200 (Tue, 29 May 2007) | 2
- lines Issue #9802 - Change inuse counter on CANCEL ........
- ................
-
-2007-05-28 23:28 +0000 [r66313-66315] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Don't try to unregister a peer using the sip
- unregister CLI command if they are not registered. (issue #9811
- reported by eliel)
-
- * channels/chan_sip.c: Due to the way stringfields work the value
- of the url pointer will always be non-NULL so we have to use
- ast_strlen_zero to make sure it is not empty. (issue #9821
- reported by pj)
-
-2007-05-28 18:50 +0000 [r66295] Olle Johansson <oej@edvina.net>
-
- * apps/app_voicemail.c: - Don't re-invent existing headers (some
- already existed in chan_sip) - Rename command so taht module name
- comes first
-
-2007-05-28 15:59 +0000 [r66278] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_iconv.c (added): Issue 7021 - Add ICONV function for
- converting between character sets
-
-2007-05-26 19:35 +0000 [r66225] Joshua Colp <jcolp@digium.com>
-
- * apps/app_minivm.c: Unlock the minivmlock when no configuration is
- found. (issue #9814 reported by eliel)
-
-2007-05-26 06:07 +0000 [r66208] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c: Since this code now uses the API call for
- creating a detached thread, there is no reason to keep a thread
- attribute structure on the conference structure. (Pointed out by
- Tony Mountifield on the asterisk-dev list)
-
-2007-05-25 15:08 +0000 [r66175-66178] Kevin P. Fleming <kpfleming@digium.com>
-
- * /: block change that is already here
-
- * channels/chan_jingle.c, configure, configure.ac: more minor fixes
-
-2007-05-25 14:49 +0000 [r66161] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/say.c: Merged revisions 66159 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r66159 | tilghman | 2007-05-25 09:41:27 -0500
- (Fri, 25 May 2007) | 10 lines Merged revisions 66127 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r66127 | tilghman | 2007-05-25 08:46:35 -0500 (Fri, 25
- May 2007) | 2 lines Issue 9791 - Fix pronunciation of seconds in
- Dutch ........ ................
-
-2007-05-25 14:37 +0000 [r66158] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_jingle.c, /, configure, configure.ac,
- channels/chan_gtalk.c, makeopts.in, res/res_jabber.c: Merged
- revisions 66157 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r66157 | kpfleming | 2007-05-25 10:28:46 -0400 (Fri, 25 May 2007)
- | 3 lines handle the GNUTLS library properly in the configure
- script and build system don't build in OSP support unless we have
- found and are allowed to use SSL support ........
-
-2007-05-25 13:26 +0000 [r66109-66126] Joshua Colp <jcolp@digium.com>
-
- * main/slinfactory.c: Minor tweak... drop translation path if one
- exists when we get an already signed linear frame in. Chances are
- the stream has then switched to signed linear and we no longer
- need the path.
-
- * /, main/slinfactory.c: Merged revisions 66074 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r66074 | file | 2007-05-24 18:16:58 -0400 (Thu, 24 May 2007) | 2
- lines Fix slinfactory logic when dealing with frames coming in
- that may already be in the signed linear format. ........
-
-2007-05-24 22:25 +0000 [r66072-66077] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /: Merged revisions 66076 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r66076 | russell | 2007-05-24 17:23:59 -0500 (Thu, 24 May 2007) |
- 1 line if the string field init fails, clean up the stuff that
- was allocated already ........
-
- * main/channel.c, /: Merged revisions 66070 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r66070 | russell | 2007-05-24 17:07:39 -0500 (Thu, 24 May 2007) |
- 2 lines Check the result of ast_string_field_init() in
- ast_channel_alloc() ........
-
-2007-05-24 22:07 +0000 [r66071] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/aescrypt.c, build_tools/menuselect-deps.in, configure,
- include/asterisk/autoconfig.h.in, include/asterisk/aes_internal.h
- (added), configure.ac, main/aestab.c, include/asterisk/aes.h,
- main/aeskey.c, pbx/pbx_dundi.c, channels/chan_iax2.c,
- makeopts.in: use the OpenSSL AES implementation if it's available
- (unless configured not to)
-
-2007-05-24 20:55 +0000 [r66031] Jason Parker <jparker@digium.com>
-
- * /, configure, configure.ac: Merged revisions 66029-66030 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r66029 | qwell | 2007-05-24 15:53:18 -0500 (Thu, 24 May 2007) | 2
- lines Following moving strip to AC_PATH_TOOL, we need to do
- something similar for ar. ........ r66030 | qwell | 2007-05-24
- 15:54:16 -0500 (Thu, 24 May 2007) | 2 lines Rebuild configure
- script for previous ar fix. ........
-
-2007-05-24 20:51 +0000 [r66028] Joshua Colp <jcolp@digium.com>
-
- * CHANGES, apps/app_voicemail.c: Add ListAllVoicemailUsers manager
- command. (issue #8112 reported by Tony Zhao)
-
-2007-05-24 20:44 +0000 [r65982-66027] Russell Bryant <russell@digium.com>
-
- * /, configure, configure.ac: Merged revisions 66026 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r66026 | russell | 2007-05-24 15:42:53 -0500 (Thu, 24
- May 2007) | 3 lines Checking for the strip application needs to
- be done with AC_PATH_TOOL instead of AC_PATH_PROG to properly
- handle cross compilation environments. ........
-
- * doc/CODING-GUIDELINES: add a note about using the intenal API for
- creating detached threads
-
- * Makefile, /: Merged revisions 65978 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r65978 | russell | 2007-05-24 14:05:08 -0500 (Thu, 24 May 2007) |
- 3 lines Clear CFLAGS before running make for menuselect. (issue
- #9784, reported by ovi, patch by me) ........
-
-2007-05-24 19:05 +0000 [r65979] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, channels/chan_gtalk.c: Merged revisions 65965-65967 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r65965 | kpfleming | 2007-05-24 14:24:55 -0400 (Thu, 24 May 2007)
- | 2 lines don't use uninitialized variables ........ r65966 |
- kpfleming | 2007-05-24 14:25:21 -0400 (Thu, 24 May 2007) | 2
- lines don't reference GnuTLS headers and functions unless the
- configure script found it ........ r65967 | kpfleming |
- 2007-05-24 14:28:48 -0400 (Thu, 24 May 2007) | 2 lines oops, use
- #ifdef instead of #if ........
-
-2007-05-24 18:30 +0000 [r65964-65968] Russell Bryant <russell@digium.com>
-
- * main/pbx.c, include/asterisk/utils.h, channels/chan_zap.c,
- channels/chan_sip.c, apps/app_meetme.c, main/utils.c,
- channels/chan_iax2.c, main/cdr.c, main/manager.c,
- pbx/pbx_spool.c, channels/chan_skinny.c, main/http.c,
- channels/chan_h323.c, pbx/pbx_dundi.c, apps/app_rpt.c,
- apps/app_mixmonitor.c, main/asterisk.c, channels/chan_mgcp.c: Add
- a new API call for creating detached threads. Then, go replace
- all of the places in the code where the same block of code for
- creating detached threads was replicated. (patch from bbryant)
-
- * main/rtp.c: Make this build on *my* machine again, and hopefully
- not break others.
-
-2007-05-24 15:35 +0000 [r65906] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * /, funcs/func_math.c: Merged revisions 65866 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r65866 | dhubbard | 2007-05-24 10:08:56 -0500 (Thu, 24 May 2007)
- | 1 line merged qwell's func_math patch for issue 9507 ........
-
-2007-05-24 15:30 +0000 [r65905] Joshua Colp <jcolp@digium.com>
-
- * main/manager.c, /: Merged revisions 65902 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r65902 | file | 2007-05-24 11:27:23 -0400 (Thu, 24 May 2007) | 2
- lines Add the ability to blacklist certain commands from being
- executed using the Command AMI action. (issue #9240 reported by
- junky) ........
-
-2007-05-24 15:29 +0000 [r65904] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_gtalk.c: Merged revisions 65901 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r65901 | oej | 2007-05-24 17:26:10 +0200 (Thu, 24 May
- 2007) | 2 lines Issue 7672 - fix by zandbelt - Asterisk core dump
- since the GnuTLS interface did not support multithreading
- correctly. ........
-
-2007-05-24 15:28 +0000 [r65903] Jason Parker <jparker@digium.com>
-
- * /, codecs/codec_speex.c, main/translate.c, codecs/codec_ilbc.c,
- .cleancount, include/asterisk/translate.h: Merged revisions 65877
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r65877 | qwell | 2007-05-24 11:14:02 -0400 (Thu, 24 May 2007) | 4
- lines Fix handling of zero-length frames when a codec is capable
- of native PLC. Issue 9183, patch by Mihai. ........
-
-2007-05-24 15:23 +0000 [r65894-65898] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_gtalk.c: Merged revisions 65892 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r65892 | oej | 2007-05-24 17:20:54 +0200 (Thu, 24 May
- 2007) | 2 lines Issue 8193 - NAT issues with gtalk/STUN. Patch by
- phsultan. Thanks! ........
-
- * /, channels/chan_gtalk.c: Merged revisions 65857 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r65857 | oej | 2007-05-24 17:05:10 +0200 (Thu, 24 May
- 2007) | 2 lines Issue 7686, fix by phsultan, NAT issues when
- calling from gtalk to SIP over nat. ........
-
-2007-05-24 15:10 +0000 [r65869] Joshua Colp <jcolp@digium.com>
-
- * /, main/rtp.c: Merged revisions 65863 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r65863 | file | 2007-05-24 11:08:17 -0400 (Thu, 24 May 2007) | 2
- lines I like it when the RTP stack compiles myself... ........
-
-2007-05-24 15:04 +0000 [r65855] Russell Bryant <russell@digium.com>
-
- * /, apps/app_festival.c: Merged revisions 65853 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r65853 | russell | 2007-05-24 10:04:14 -0500 (Thu, 24 May 2007) |
- 4 lines Ensure that frames are fully initialized. This will
- probably fix getting weird timestamp log messages in logs when
- using the Festival app. (issue #9781, patch by me) ........
-
-2007-05-24 14:52 +0000 [r65844] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_gtalk.c: Merged revisions 65841 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r65841 | oej | 2007-05-24 16:48:55 +0200 (Thu, 24 May
- 2007) | 2 lines Issue #8536 - Caller ID not set in CDR for jingle
- ........
-
-2007-05-24 14:50 +0000 [r65843] Russell Bryant <russell@digium.com>
-
- * /, main/rtp.c: Merged revisions 65842 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r65842 | russell | 2007-05-24 09:49:05 -0500 (Thu, 24 May 2007) |
- 5 lines Fix the calculation of the RTT for RTCP. The previous
- code would result in oscillating and incorrect data.
- Additionally, the RTT would sometimes report negative values due
- to incorrect calculations. (issue #9601, patch from davetroy)
- ........
-
-2007-05-24 14:43 +0000 [r65840] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 65839 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r65839 | file | 2007-05-24 10:42:12 -0400 (Thu,
- 24 May 2007) | 10 lines Merged revisions 65837 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r65837 | file | 2007-05-24 10:40:38 -0400 (Thu, 24 May 2007) | 2
- lines Allow RFC2833 to be negotiated when an INVITE comes in
- without SDP and is not matched to a user or peer. (issue #9546
- reported by mcrawford) ........ ................
-
-2007-05-24 14:41 +0000 [r65838] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c, res/res_jabber.c: Issue #8409 and
- accidentally a fix to chan_sip that wasn't supposed to be there
- but is still ok... Sorry. Lack of Tea, really.
-
-2007-05-24 11:38 +0000 [r65814] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_sip.c: Yes Virginia, there is a reason why we have
- stringfields in the sip_pvt structure...
-
-2007-05-24 09:51 +0000 [r65769] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /: Merged revisions 65768 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r65768 | crichter | 2007-05-24 11:37:32 +0200
- (Do, 24 Mai 2007) | 9 lines Merged revisions 65767 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r65767 | crichter | 2007-05-24 11:19:58 +0200 (Do, 24
- Mai 2007) | 1 line we should only activate the generator in
- chan_misdn, when asterisk hask not yet taken the call
- (WAITING4DIGS state). Alerting audio will be generated fomr
- asterisk for example. ........ ................
-
-2007-05-24 03:28 +0000 [r65749] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: - Remove debug variable that was only used
- in one place - convert string handling to the ast_str API -
- Convert strdup() to ast_strdup() and check the result - Minor
- formatting changes
-
-2007-05-24 03:27 +0000 [r65748] Tilghman Lesher <tlesher@digium.com>
-
- * cdr/cdr_adaptive_odbc.c: Oops, should have released this when we
- were done with it.
-
-2007-05-24 02:23 +0000 [r65731] Mark Spencer <markster@digium.com>
-
- * channels/chan_sip.c: Add SendURL support
-
-2007-05-23 21:01 +0000 [r65678-65688] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 65685 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r65685 | kpfleming | 2007-05-23 16:59:19 -0400 (Wed, 23 May 2007)
- | 2 lines start the delayed PBX when receive voice or video full
- frames as well, and comment this delayed-PBX activity ........
-
- * /, channels/chan_sip.c: Merged revisions 65683 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r65683 | kpfleming | 2007-05-23 16:51:56 -0400
- (Wed, 23 May 2007) | 10 lines Merged revisions 65682 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r65682 | kpfleming | 2007-05-23 16:46:22 -0400 (Wed, 23
- May 2007) | 2 lines ensure that variables are set on a newly
- created channel before we start a PBX on it ........
- ................
-
- * /, channels/chan_iax2.c: Merged revisions 65679-65680 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r65679 | kpfleming | 2007-05-23 16:30:24 -0400 (Wed, 23 May 2007)
- | 2 lines don't start a PBX on a new incoming IAX2 channel until
- we have some sort of response to our ACCEPT (ACK or anything
- else) ........ r65680 | kpfleming | 2007-05-23 16:35:50 -0400
- (Wed, 23 May 2007) | 2 lines clear the 'delay PBX' flag when we
- are ready to start the PBX ........
-
- * /, channels/chan_iax2.c: Merged revisions 65677 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r65677 | kpfleming | 2007-05-23 16:07:59 -0400
- (Wed, 23 May 2007) | 10 lines Merged revisions 65676 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r65676 | kpfleming | 2007-05-23 16:06:13 -0400 (Wed, 23
- May 2007) | 2 lines if we are going to set variables on a newly
- created channel, it should be done *before* we start the PBX on
- it ........ ................
-
-2007-05-23 17:17 +0000 [r65659] Russell Bryant <russell@digium.com>
-
- * apps/app_voicemail.c: Don't check for MWI event subscribers
- before creating the MWI event in voicemail. MWI events get
- cached, so go ahead and always generate them so the cache gets
- populated.
-
-2007-05-23 15:37 +0000 [r65640] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Make sure we get the cause code in the REL
-
-2007-05-23 13:10 +0000 [r65591] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 65589 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r65589 | russell | 2007-05-23 08:07:13 -0500
- (Wed, 23 May 2007) | 11 lines Merged revisions 65588 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r65588 | russell | 2007-05-23 08:06:17 -0500 (Wed, 23
- May 2007) | 3 lines Revert revision 62417 as someone reported
- problems with it to Mark. This was related to issue #9588.
- ........ ................
-
-2007-05-23 13:07 +0000 [r65590] Joshua Colp <jcolp@digium.com>
-
- * res/res_musiconhold.c: Fix compiling of res_musiconhold under dev
- mode.
-
-2007-05-23 02:55 +0000 [r65573] Russell Bryant <russell@digium.com>
-
- * main/devicestate.c: Fix a couple minor spelling mistakes.
-
-2007-05-22 20:26 +0000 [r65542] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, build_tools/make_version: Merged revisions 65541 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r65541 | kpfleming | 2007-05-22 16:25:41 -0400 (Tue, 22
- May 2007) | 2 lines when building a version string for a
- developer branch, include the base branch in the version string
- ........
-
-2007-05-22 18:52 +0000 [r65502-65505] Russell Bryant <russell@digium.com>
-
- * main/channel.c, configs/musiconhold.conf.sample,
- include/asterisk/channel.h, res/res_musiconhold.c, CHANGES: Add a
- new feature for Music on Hold. If you set the "digit" option for
- a class in musiconhold.conf, a caller on hold may press this
- digit to switch to listening to that music class. This involved
- adding a new callback for generators, which allow generators to
- get notified of DTMF from the channel they are running on. Then,
- a callback was implemented for the music on hold generators.
- (patch from bbryant)
-
- * channels/chan_zap.c, /, apps/app_voicemail.c: Merged revisions
- 65501 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r65501 | russell | 2007-05-22 13:40:38 -0500 (Tue, 22 May 2007) |
- 3 lines List res_smdi as a dependency for app_voicemail and
- chan_zap (Thanks to mnicholson for pointing it out) ........
-
-2007-05-22 15:25 +0000 [r65455] BJ Weschke <bweschke@btwtech.com>
-
- * /, apps/app_followme.c: Merged revisions 65408 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r65408 | bweschke | 2007-05-22 10:02:56 -0400 (Tue, 22 May 2007)
- | 3 lines Fix a problem with flag recognition. ........
-
-2007-05-22 15:08 +0000 [r65451-65454] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_agent.c: Use ast_strlen_zero where possible. (issue
- #9774 reported by eliel)
-
- * main/cdr.c: Make my compiler happy! Yay!
-
-2007-05-22 12:58 +0000 [r65376] Joshua Colp <jcolp@digium.com>
-
- * res/res_features.c: Don't overwrite a pointer to the first
- channel... that is bad. (issue #9770 reported by tfbu)
-
-2007-05-22 12:52 +0000 [r65375] Russell Bryant <russell@digium.com>
-
- * apps/app_queue.c: Fix a couple of spots in the handling of device
- states that could lead to a double free. (issue #9772, reported
- by Mike Anikienko, fix by me)
-
-2007-05-22 08:21 +0000 [r65343] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /: Merged revisions 65342 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r65342 | crichter | 2007-05-22 10:12:20 +0200
- (Di, 22 Mai 2007) | 9 lines Merged revisions 65328 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r65328 | crichter | 2007-05-22 09:46:39 +0200 (Di, 22
- Mai 2007) | 1 line we stop the tones only when we're in the
- pre-call phase, otherwise e.g. when in CONNECTED state we should
- not stop tones when we receive an Information Message ........
- ................
-
-2007-05-22 02:41 +0000 [r65313] Tilghman Lesher <tlesher@digium.com>
-
- * cdr/cdr_adaptive_odbc.c: Fix for 64-bit platform
-
-2007-05-21 06:56 +0000 [r65298] Russell Bryant <russell@digium.com>
-
- * apps/app_queue.c: I know we have talked about rewriting app_queue
- for Asterisk 1.6, but once I saw this, I couldn't help myself
- from changing it. Previously, for *every* device state change,
- app_queue would spawn a thread to handle it. Now, the device
- state callback just puts the state change in a queue and it gets
- handled by a single state change processing thread.
-
-2007-05-21 02:05 +0000 [r65283] Tilghman Lesher <tlesher@digium.com>
-
- * cdr/cdr_adaptive_odbc.c: Comment a few more things, and remove an
- unnecessary db connection check
-
-2007-05-20 18:01 +0000 [r65233-65253] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_agi.c: Merged revisions 65250 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r65250 | file | 2007-05-20 13:59:58 -0400 (Sun, 20 May 2007) | 2
- lines res_agi needs to export two symbols (ast_agi_register and
- ast_agi_unregister) for usage by others. (issue #9755 reported by
- mnicholson) ........
-
- * res/res_crypto.c, res/res_musiconhold.c: Music on hold and crypto
- no longer need their symbols globally exported. They register the
- function pointers upon loading with their respective stubs.
-
- * main/adsistub.c, main/cryptostub.c: Clean up adsistub file a bit
- (just spacing) and change over the crypto sub to use this
- build_stub macro strategy.
-
- * main/Makefile, main/adsistub.c, res/res_adsi.c: Add the adsistub
- file to the Asterisk makefile, fix a stub definition, and no
- longer make the symbols from res_adsi global since they don't
- need to be.
-
-2007-05-18 22:35 +0000 [r65202-65203] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, /: Merged revisions 65201 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r65201 | murf | 2007-05-18 16:26:51 -0600 (Fri, 18 May 2007) | 1
- line Ugh. The svnmerge didn't catch the shift from cdr.c to
- main/cdr.c, and neither did I. This is the remainder of the 9717
- patch, the fix for the run-away FAIL status for a call ........
-
- * apps/app_dial.c, /, include/asterisk/cdr.h: Merged revisions
- 65200 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r65200 | murf | 2007-05-18 16:06:27 -0600 (Fri,
- 18 May 2007) | 9 lines Merged revisions 65172 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r65172 | murf | 2007-05-18 14:56:20 -0600 (Fri, 18 May 2007) | 1
- line This update will fix the situation that occurs as described
- by 9717, where when several targets are specified for a dial, if
- any one them reports FAIL, the whole call gets FAIL, even though
- others were ringing OK. I rearranged the priorities, so that a
- new disposition, NULL, is at the lowest level, and the
- disposition get init'd to NULL. Then, next up is FAIL, and next
- up is BUSY, then NOANSWER, then ANSWERED. All the related set
- routines will only do so if the disposition value to be set to is
- greater than what's already there. This gives the intended
- effect. So, if all the targets are busy, you'd get BUSY for the
- call disposition. If all get BUSY, but one, and that one rings is
- not answered, you get NOANSWER. If by some freak of nature, the
- NULL value doesn't get overridden, then the disp2str routine will
- report NOANSWER as before. ........ ................
-
-2007-05-18 20:21 +0000 [r65169] Tilghman Lesher <tlesher@digium.com>
-
- * cdr/cdr_adaptive_odbc.c (added),
- configs/cdr_adaptive_odbc.conf.sample (added): Merge
- cdr_adaptive_odbc from developer branch
-
-2007-05-18 18:18 +0000 [r65077-65124] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Related to issue #9235 btw. Merged
- revisions 65123 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r65123 | oej | 2007-05-18 20:16:09 +0200 (Fri,
- 18 May 2007) | 10 lines Merged revisions 65122 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r65122 | oej | 2007-05-18 20:10:46 +0200 (Fri, 18 May 2007) | 2
- lines Not getting an ACK to a 200 OK in the initial invite is
- critical to the call. ........ ................
-
- * /, channels/chan_sip.c: Merged revisions 65076 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r65076 | oej | 2007-05-18 17:18:13 +0200 (Fri,
- 18 May 2007) | 13 lines Merged revisions 65075 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r65075 | oej | 2007-05-18 17:12:09 +0200 (Fri, 18 May 2007) | 5
- lines Issue 9235 - part of the problem, maybe not all. Please
- retry with this patch (and no other patch) if you have problems
- with hanging SIP channels. Thank you. A special Thank You to
- WeBRainstorm that gave me access to his system. ........
- ................
-
-2007-05-18 12:43 +0000 [r65006-65040] Christian Richter <christian.richter@beronet.com>
-
- * /, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c: Merged
- revisions 65039 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r65039 | crichter | 2007-05-18 14:40:46 +0200
- (Fr, 18 Mai 2007) | 9 lines Merged revisions 65007 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r65007 | crichter | 2007-05-18 13:23:11 +0200 (Fr, 18
- Mai 2007) | 1 line fixed a warning regarding Keypad encoding.
- encode the IE sending_complete at the right position. ........
- ................
-
- * channels/chan_misdn.c, /: Merged revisions 64904 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r64904 | crichter | 2007-05-18 10:58:51 +0200
- (Fr, 18 Mai 2007) | 9 lines Merged revisions 64902 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r64902 | crichter | 2007-05-18 10:24:08 +0200 (Fr, 18
- Mai 2007) | 1 line we *need* to send a PROCEEDING when
- sending_complete is set, even if need_more_infos is requested.
- ........ ................
-
-2007-05-18 10:41 +0000 [r64973-64975] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 64974 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r64974 | oej | 2007-05-18 12:37:44 +0200 (Fri, 18 May 2007) | 2
- lines Issue 9487 - stop media flows at hangup of call ........
-
- * channels/chan_sip.c: Makeup, darling.
-
-2007-05-18 10:03 +0000 [r64951-64963] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /: Merged revisions 64515 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r64515 | crichter | 2007-05-16 10:44:51 +0200
- (Mi, 16 Mai 2007) | 9 lines Merged revisions 64513 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r64513 | crichter | 2007-05-16 10:23:42 +0200 (Mi, 16
- Mai 2007) | 1 line in the case immediate=yes, we directly jump
- into the dialplan, where people can use PlayTones to indicate a
- Dialtone, so we don't need to to that by ourself. also we should
- not do a dialtone_indicate for incoming calls on a TE port in
- overlapdialmode. ........ ................
-
- * channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c,
- channels/misdn/isdn_lib.c: Merged revisions 63534 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r63534 | crichter | 2007-05-09 15:17:10 +0200
- (Mi, 09 Mai 2007) | 17 lines Merged revisions 62945,63402,63519
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r62945 | crichter | 2007-05-03 17:39:21 +0200 (Do, 03 Mai 2007) |
- 1 line when we're in state WAITING4DIGS, we use the asterisk
- tone-generator which prods us, so we can't just return -1 in
- misdn_write in this case. Added a MISDN_KEYPAD channel variable,
- and fixed the sending of keypad. this enables us to modify the
- call forward parameters in the switch. ........ r63402 | crichter
- | 2007-05-08 17:07:37 +0200 (Di, 08 Mai 2007) | 1 line added
- application misdn_check_l2l1 which tries to pull up the L1/L2 on
- all ports that have the layers down in a group. It waits then for
- a timeout. This helps for scenarios where multiple PMP BRIs are
- grouped together, or where a provider has a faulty PTP
- Implementation, that looses the L2 after a while. ........ r63519
- | crichter | 2007-05-09 13:26:16 +0200 (Mi, 09 Mai 2007) | 1 line
- release_chan frees ch, so we should never touch ch after
- release_chan, this may cause segfaults. ........ ................
-
- * channels/misdn/isdn_lib_intern.h, channels/misdn/isdn_lib.h,
- channels/chan_misdn.c, /, channels/misdn/ie.c,
- channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c:
- Merged revisions 62912 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r62912 | crichter | 2007-05-03 16:36:32 +0200
- (Do, 03 Mai 2007) | 17 lines Merged revisions 61357,61770,62885
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61357 | crichter | 2007-04-11 14:05:57 +0200 (Mi, 11 Apr 2007) |
- 1 line some fixes for PMP Hold/Retrieve, it should work now, when
- briding=no ........ r61770 | crichter | 2007-04-24 15:50:05 +0200
- (Di, 24 Apr 2007) | 1 line added lock for sending messages to
- avoid double sending. shuffled some empty_chans after the
- cb_event calls, this avoids that a release_complete from a quite
- different call releases a fresh created setup by accident.
- ........ r62885 | crichter | 2007-05-03 15:59:00 +0200 (Do, 03
- Mai 2007) | 1 line fixed the problem that misdn_write did not
- return -1 when called with 0 samples in a frame this resultet in
- a deadlock in some circumstances, when the call ended because of
- a busy extension. added encoding of keypad. ........
- ................
-
- * channels/chan_misdn.c, /, channels/misdn/chan_misdn_config.h,
- channels/misdn/isdn_lib.c, channels/misdn_config.c: Merged
- revisions 59774 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r59774 | crichter | 2007-04-03 09:20:27 +0200
- (Di, 03 Apr 2007) | 17 lines Merged revisions 59623-59624,59639
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59623 | crichter | 2007-04-02 09:12:24 +0200 (Mo, 02 Apr 2007) |
- 1 line we can now make 30 channels on a PRI (before we forgot
- chan 31..) ........ r59624 | crichter | 2007-04-02 09:25:54 +0200
- (Mo, 02 Apr 2007) | 1 line don't be verbose if no need ........
- r59639 | crichter | 2007-04-02 14:08:12 +0200 (Mo, 02 Apr 2007) |
- 1 line added option which allows us to accept incoming SETUP
- Messages without automatically sending Proceeding or Setup
- Acknowledge, this is useful with some broken switches and if you
- want to Release incoming calls without previously having
- acknowledged them. The new option is
- noautorespond_on_setup=yes|no default is no, so we don't break
- the existing behaviour ........ ................
-
- * channels/chan_misdn.c, /: Merged revisions 59254 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r59254 | crichter | 2007-03-27 17:00:10 +0200
- (Di, 27 Mär 2007) | 9 lines Merged revisions 59252 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r59252 | crichter | 2007-03-27 15:56:15 +0200 (Di, 27
- Mär 2007) | 1 line fixed #9355 ........ ................
-
- * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
- channels/misdn/chan_misdn_config.h, channels/misdn/isdn_lib.c,
- channels/misdn_config.c: Merged revisions 59064 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r59064 | crichter | 2007-03-20 14:16:06 +0100
- (Di, 20 Mär 2007) | 21 lines Merged revisions
- 58849-58850,59062-59063 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r58849 | crichter | 2007-03-13 12:58:16 +0100 (Di, 13 Mär 2007) |
- 1 line added method standard_dec for dialing out on groups, to
- avoid conflicts, which caused issues with some ISDN providers
- ........ r58850 | crichter | 2007-03-13 13:58:32 +0100 (Di, 13
- Mär 2007) | 1 line fixed the crypt_keys stuff ........ r59062 |
- crichter | 2007-03-20 10:18:06 +0100 (Di, 20 Mär 2007) | 1 line
- avoid sending a disconnect when we already received one. ........
- r59063 | crichter | 2007-03-20 10:23:22 +0100 (Di, 20 Mär 2007) |
- 1 line modified a loglevel ........ ................
-
- * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
- channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c,
- channels/misdn/isdn_lib.c: Merged revisions 58825-58826 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r58825 | crichter | 2007-03-12 13:43:24 +0100
- (Mo, 12 Mär 2007) | 1 line added UU transceiving and corect
- handling for rdnis ................ r58826 | crichter |
- 2007-03-12 14:08:06 +0100 (Mo, 12 Mär 2007) | 21 lines Merged
- revisions 57034,57523,57753,58558 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r57034 | crichter | 2007-02-28 17:09:27 +0100 (Mi, 28 Feb 2007) |
- 1 line fixed bugs.digium.com bugs: #9157 and bugs.beronet.com
- bugs: #302, #303, #304 ........ r57523 | crichter | 2007-03-02
- 19:32:51 +0100 (Fr, 02 Mar 2007) | 1 line fixed typo ........
- r57753 | crichter | 2007-03-04 11:39:50 +0100 (So, 04 Mar 2007) |
- 1 line fixed another place where the out_cause was hardcoded to
- 16 ........ r58558 | crichter | 2007-03-09 15:43:58 +0100 (Fr, 09
- Mar 2007) | 1 line we can free channel 31 as well, since we can
- occupy it ........ ................
-
-2007-05-18 09:10 +0000 [r64903-64921] Olle Johansson <oej@edvina.net>
-
- * include/asterisk/adsi.h, main/adsistub.c (added), res/res_adsi.c,
- apps/app_voicemail.c: Issue #5930 - Remove dependencies on
- res_adsi.so - clwade A big THANK YOU to clwade for this patch.
- Minor modifications by me.
-
- * channels/chan_sip.c: Another fix for the support for recordings
- controlled by INFO-packets We still lack a setting to
- enable/disable this per peer
-
-2007-05-18 02:55 +0000 [r64869-64870] Russell Bryant <russell@digium.com>
-
- * CHANGES: Add ENUMQUERY and ENUMRESULT to the CHANGES file.
-
- * /, apps/app_queue.c: Merged revisions 64868 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r64868 | russell | 2007-05-17 21:48:51 -0500 (Thu, 17 May 2007) |
- 5 lines Fix a small bug I noticed while working on something
- else. app_queue did not unregister its device state monitoring
- callback in unload_module(). So, this would make Asterisk crash
- on the first device state change after you unload the module.
- ........
-
-2007-05-17 21:20 +0000 [r64821] Tilghman Lesher <tlesher@digium.com>
-
- * /, include/asterisk/linkedlists.h: Merged revisions 64820 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r64820 | tilghman | 2007-05-17 16:19:34 -0500
- (Thu, 17 May 2007) | 10 lines Merged revisions 64819 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r64819 | tilghman | 2007-05-17 16:14:36 -0500 (Thu, 17
- May 2007) | 2 lines How is it that we never caught that this is
- returning the opposite of our documentation, until now? ........
- ................
-
-2007-05-17 17:12 +0000 [r64786] Russell Bryant <russell@digium.com>
-
- * main/manager.c, configs/manager.conf.sample: Add an option that
- lets you only allow one connection at a time for each manager
- user. (issue #8664, reported and original patch by ssokol, patch
- updated by bkruse, and further updated by me)
-
-2007-05-17 16:54 +0000 [r64762] Jason Parker <jparker@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 64761 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r64761 | qwell | 2007-05-17 11:53:27 -0500 (Thu,
- 17 May 2007) | 12 lines Merged revisions 64758 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r64758 | qwell | 2007-05-17 11:52:38 -0500 (Thu, 17 May 2007) | 4
- lines If we have a negative current message, we shouldn't go back
- even further... Issue 9727. ........ ................
-
-2007-05-17 16:53 +0000 [r64757-64760] Russell Bryant <russell@digium.com>
-
- * /, contrib/scripts/astxs (removed): Merged revisions 64759 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r64759 | russell | 2007-05-17 11:52:53 -0500 (Thu, 17 May 2007) |
- 3 lines Remove script that is no longer functional since the
- build system was redone. (issue #9340, reported by junky)
- ........
-
- * apps/app_dial.c, /: Merged revisions 64756 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r64756 | russell | 2007-05-17 11:47:29 -0500 (Thu, 17 May 2007) |
- 3 lines Increase the size of a buffer to support longer dial
- strings for channels. (issue #9291, reported and fix suggested by
- meni) ........
-
-2007-05-17 16:11 +0000 [r64721-64755] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 64754 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r64754 | file | 2007-05-17 12:10:12 -0400 (Thu, 17 May 2007) | 2
- lines Even more direct RTP setup fixes! Don't allow a codec that
- isn't supported to creep into the SDP of either side. (issue
- #9446 reported by marcelbarbulescu) ........
-
- * /, apps/app_voicemail.c: Merged revisions 64720 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r64720 | file | 2007-05-17 09:48:44 -0400 (Thu, 17 May 2007) | 2
- lines Fix authuser support. (issue #9740 reported by
- xmarksthespot) ........
-
-2007-05-17 06:14 +0000 [r64657-64687] Russell Bryant <russell@digium.com>
-
- * README, /: Merged revisions 64686 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r64686 | russell | 2007-05-17 01:13:53 -0500 (Thu, 17 May 2007) |
- 3 lines Update the main README to reflect the new build process
- for 1.4 and above. (issue #9725, patch by eliel) ........
-
- * main/app.c: Ignore this ... playing with jira (AST-1)
-
-2007-05-16 11:01 +0000 [r64494-64611] Olle Johansson <oej@edvina.net>
-
- * /: Blocking patch
-
- * /, channels/chan_sip.c: Below patches with some re-structuring
- for trunk --- Merged revisions 64602 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r64602 | oej | 2007-05-16 12:38:18 +0200 (Wed, 16 May 2007) | 2
- lines Issue #9681 - Handle www-auth on BYE ........
-
- * /, channels/chan_sip.c: Merged revisions 64578 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r64578 | oej | 2007-05-16 12:05:47 +0200 (Wed, 16 May 2007) | 2
- lines Final part of issue #9483 - fixing transfer() of sip calls
- in the dial plan (twilson) ........
-
- * /: Blocking patch that was already committed to trunk
-
- * /, channels/chan_sip.c: Merged revisions 64543 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r64543 | oej | 2007-05-16 11:12:34 +0200 (Wed,
- 16 May 2007) | 10 lines Merged revisions 64535 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r64535 | oej | 2007-05-16 11:08:22 +0200 (Wed, 16 May 2007) | 2
- lines Support SIP uri's starting with SIP: and sip: (reported by
- Tony Mountfield on the mailing list. Thanks!) ........
- ................
-
- * /, channels/chan_sip.c: Merged revisions 64516 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r64516 | oej | 2007-05-16 10:46:18 +0200 (Wed,
- 16 May 2007) | 17 lines Merged following patch with a lot of
- changes for 1.4 ------ Merged revisions 64514 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r64514 | oej | 2007-05-16 10:25:56 +0200 (Wed, 16 May 2007) | 6
- lines Issue #9726 - rlister - Better logging for ACL denials
- While at it, also added better logging and handling of peers that
- are not supposed to register. My patch, stole the issue report
- from Russell. My apologies, Russell :-) ........ ................
-
- * channels/chan_sip.c: Issue #9304 - Update help text to match
- functionality. Patch by kshumard with changes by oej
-
- * channels/chan_sip.c, configs/sip.conf.sample: Issue #6789 -
- Marquis - Add option to support regexten removal when host
- becomes unreachable
-
- * main/event.c: This file really needs more documentation... When
- we implement new API's - please include a small general overview
- in Doxygen
-
- * main/dial.c: Small doxygen updates
-
-2007-05-15 23:05 +0000 [r64469-64480] Russell Bryant <russell@digium.com>
-
- * funcs/func_enum.c, include/asterisk/enum.h, main/enum.c: Add two
- new dialplan functions: ENUMQUERY and ENUMRESULT. These functions
- allow you to initiate an ENUM query using ENUMQUERY, and then
- access the details of all of the results using ENUMRESULT.
- Previously, if you wanted to access multiple results, Asterisk
- would have to do a new DNS lookup every time. (patch by bbryant)
-
- * pbx/pbx_dundi.c: Make sure that DUNDIRESULT is given an ID.
-
-2007-05-15 20:45 +0000 [r64455] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c, configs/zapata.conf.sample: XXX-XXX-XXX
- appears to be the standard ANSI pointcode format
-
-2007-05-15 19:57 +0000 [r64427] Russell Bryant <russell@digium.com>
-
- * /, res/res_features.c: Merged revisions 64426 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r64426 | russell | 2007-05-15 14:52:18 -0500 (Tue, 15 May 2007) |
- 3 lines Properly fix a problem that occurs when you set
- PARKINGEXTEN to an exten where a call is already parked. (issue
- #9723, patch by me) ........
-
-2007-05-14 23:43 +0000 [r64399] Kevin P. Fleming <kpfleming@digium.com>
-
- * /: this does not belong here
-
-2007-05-14 22:25 +0000 [r64384] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Only print the SS7 UP once. Not every time
- we get the test messages on the line.
-
-2007-05-14 21:51 +0000 [r64355] Jason Parker <jparker@digium.com>
-
- * main/Makefile: With libmmime.a as a .PHONY target, asterisk gets
- rebuilt every time, but without proper ASTCFLAGS. This caused a
- problem with the buildinfo.o file not being able to find
- asterisk/build.h This was affecting DESTDIR, but I *think* that
- if asterisk had never been installed before, it would've failed
- also.
-
-2007-05-14 21:17 +0000 [r64354] Russell Bryant <russell@digium.com>
-
- * /, res/res_features.c: Merged revisions 64353 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r64353 | russell | 2007-05-14 16:16:39 -0500 (Mon, 14 May 2007) |
- 4 lines When someone requests a specific parking space using the
- PARKINGEXTEN variable, ensure that no other caller is already
- there. (issue #9723, reported by mdu113, patch by me) ........
-
-2007-05-14 19:35 +0000 [r64323-64325] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 64324 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r64324 | oej | 2007-05-14 21:26:50 +0200 (Mon, 14 May 2007) | 2
- lines Change -2 to XMIT_ERROR to clarify a bit more ........
-
- * /: Blocking patch already committed to trunk
-
-2007-05-14 19:21 +0000 [r64322] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_alsa.c: Merged revisions 64306 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r64306 | russell | 2007-05-14 14:13:00 -0500 (Mon, 14 May 2007) |
- 3 lines Properly handle AST_CONTROL_PROGRESS by just ignoring it.
- An unknown indication will trigger an error and cause sounds to
- stop, which in this case, is ringing. ........
-
-2007-05-14 18:49 +0000 [r64274-64279] Joshua Colp <jcolp@digium.com>
-
- * /, codecs/codec_speex.c: Merged revisions 64278 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r64278 | file | 2007-05-14 14:48:33 -0400 (Mon, 14 May 2007) | 2
- lines Properly set datalen field when doing PLC in codec_speex.
- (issue #9722 reported by mihai) ........
-
- * /, main/devicestate.c: Merged revisions 64276 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r64276 | file | 2007-05-14 14:36:34 -0400 (Mon,
- 14 May 2007) | 10 lines Merged revisions 64275 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r64275 | file | 2007-05-14 14:34:06 -0400 (Mon, 14 May 2007) | 2
- lines Only perform stripping of - strings from the channel name
- for Zap channels. Anywhere else we might remove a legitimate part
- of a device name. (issue #9668 reported by stevedavies) ........
- ................
-
- * channels/chan_sip.c: If no port is specified in the outboundproxy
- setting then use the standard SIP port. (issue #9665 reported by
- tootai)
-
-2007-05-14 18:14 +0000 [r64243-64273] Jason Parker <jparker@digium.com>
-
- * configs/queues.conf.sample: oops - silly typo there
-
- * configs/queues.conf.sample, apps/app_queue.c: Don't allow
- rounding seconds to weird values that may cause "unexpected"
- results. Issue 9514.
-
- * apps/app_queue.c: Add 'c' option to app_queue which allows for
- continuing in the dialplan if the callee hangs up. Issue 9284,
- patch by lyl, modified a little bit by me (I felt 'continue' was
- better than 'keepalive')
-
-2007-05-14 17:25 +0000 [r64242] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /: Merged revisions 64240 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r64240 | file | 2007-05-14 13:23:51 -0400 (Mon, 14 May 2007) | 2
- lines Fix scenario where if a phone that simply called Echo() put
- itself on hold it could never get off hold. ........
-
-2007-05-14 16:08 +0000 [r64225-64226] Russell Bryant <russell@digium.com>
-
- * configure: Regenerate configure script after last change to
- acinclude.m4
-
- * acinclude.m4: Remove an extra space from the macro that checks
- for C defines. (issue #9715, tzafrir)
-
-2007-05-14 14:13 +0000 [r64208] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, main/pbx.c, channels/chan_local.c, /: Merged
- revisions 64193 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r64193 | murf | 2007-05-14 07:58:42 -0600 (Mon, 14 May 2007) | 1
- line As per 9570, worrisome CDR warnings have been removed, that
- are either not helpful, or not relevant. ........
-
-2007-05-14 10:40 +0000 [r64142-64158] Olle Johansson <oej@edvina.net>
-
- * main/channel.c, /: Merged revisions 64157 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r64157 | oej | 2007-05-14 12:39:12 +0200 (Mon, 14 May 2007) | 2
- lines Add hangupcause when we lack codecs for transcoding
- ........
-
- * channels/chan_sip.c: Improve handling network errors on
- transmission to hosts that don't reply or are unreachable With
- this code, the call will fail as soon as we get a network error.
- This may happen on first xmit or a later one, so the retransmit
- code handles this too.
-
-2007-05-12 22:28 +0000 [r64087-64115] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 64114 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r64114 | file | 2007-05-12 18:27:04 -0400 (Sat, 12 May 2007) | 2
- lines This concludes my final adventure with bitmasks and the
- onhold flag. Would anyone care for some peanuts? ........
-
- * /, channels/chan_sip.c: Merged revisions 64086 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r64086 | file | 2007-05-12 17:10:45 -0400 (Sat, 12 May 2007) | 2
- lines Tweak hold flags some more. They can be of three states
- when active: active, inactive, one direction. ........
-
-2007-05-12 19:38 +0000 [r64072] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_enum.c: Issue 9716 - doc/enum.txt no longer exists in
- trunk
-
-2007-05-12 16:33 +0000 [r64045] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 64044 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r64044 | file | 2007-05-12 12:32:15 -0400 (Sat, 12 May 2007) | 2
- lines Ensure the onhold flag is set no matter what when being put
- on hold. ........
-
-2007-05-11 22:52 +0000 [r63967-64030] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c, configs/skinny.conf.sample: Add/fix
- support for Redial, Speeddial, and Messages buttons. Combined
- effort by DEA and mvanbaak.
-
- * main/asterisk.c: oops.. Fix the logic of the last commit.
-
- * Makefile, main/asterisk.c: Better fallback method for
- autosystemname. Issue 9713, patch by Juggie with minor mods by
- me.
-
- * main/manager.c, /: Merged revisions 63982 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r63982 | qwell | 2007-05-11 15:16:17 -0500 (Fri, 11 May 2007) | 7
- lines Hide manager password from "manager show user foo". I
- realize that there are other ways to get this, but we really
- don't need to just show it in plain text so easily. Issue 9273,
- patch by junky ........
-
- * Makefile, main/asterisk.c: Add autosystemname setting to
- asterisk.conf When enabled, it will set the systemname to be the
- hostname of the system Issue 9713, patch by Juggie - slightly
- modified by me, to "failover" to localhost
-
-2007-05-11 18:31 +0000 [r63946] Russell Bryant <russell@digium.com>
-
- * doc/qos.tex: Fix some syntax errors.
-
-2007-05-11 16:37 +0000 [r63906] Tilghman Lesher <tlesher@digium.com>
-
- * Makefile, /, contrib/scripts/safe_asterisk: Merged revisions
- 63905 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r63905 | tilghman | 2007-05-11 11:35:51 -0500
- (Fri, 11 May 2007) | 10 lines Merged revisions 63903 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r63903 | tilghman | 2007-05-11 11:31:03 -0500 (Fri, 11
- May 2007) | 2 lines Issue 9121 - fixups for safe_asterisk script
- ........ ................
-
-2007-05-11 16:21 +0000 [r63901-63902] Russell Bryant <russell@digium.com>
-
- * main/manager.c, /: Merged revisions 63886 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r63886 | russell | 2007-05-11 11:05:43 -0500 (Fri, 11 May 2007) |
- 6 lines When MD5 authentication is not possible because there is
- no challenge present, either because the Challenge action was
- never issued, or some other reason, give a proper error message
- and return an error instead of claiming that the user wasn't
- found. (reported by jsmith on IRC) ........
-
- * res/res_agi.c: Add gender support for AGI SAY NUMBER. (issue
- #9537, patch by chappell)
-
-2007-05-11 15:48 +0000 [r63873] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_features.c: Merged revisions 63872 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r63872 | file | 2007-05-11 11:43:14 -0400 (Fri, 11 May 2007) | 2
- lines Make the PARKINGEXTEN feature of parking actually work.
- (issue #9708 reported by mdu113) ........
-
-2007-05-10 23:16 +0000 [r63832] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 63830 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r63830 | qwell | 2007-05-10 18:15:37 -0500 (Thu,
- 10 May 2007) | 12 lines Merged revisions 63828 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r63828 | qwell | 2007-05-10 18:14:55 -0500 (Thu, 10 May 2007) | 4
- lines Fix an issue with trying to kill a thread before it gets
- created. Issue 9709, patch by nic_bellamy. ........
- ................
-
-2007-05-10 22:25 +0000 [r63805] Russell Bryant <russell@digium.com>
-
- * main/manager.c, /: Merged revisions 63804 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r63804 | russell | 2007-05-10 17:23:42 -0500 (Thu, 10 May 2007) |
- 4 lines Strip terminal escape sequences from CLI command output
- that is going to be sent out over the manager interface. (issue
- #9659, reported by pari, fixed by me) ........
-
-2007-05-10 21:25 +0000 [r63786] Doug Bailey <dbailey@digium.com>
-
- * main/callerid.c: Added check for negative offset in cid spill to
- prevent infinite loops
-
-2007-05-10 20:51 +0000 [r63730-63751] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 63749 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r63749 | oej | 2007-05-10 22:46:41 +0200 (Thu,
- 10 May 2007) | 12 lines Merged revisions 63748 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r63748 | oej | 2007-05-10 22:38:54 +0200 (Thu, 10 May 2007) | 4
- lines Do not allocate SIP pvt's for PEERs we can not reach. This
- was seen as a lot of dialogs being created then immediately
- destroyed at reload/restart of the SIP channel. ........
- ................
-
- * apps/app_minivm.c: Fixing reload. Thanks to Mats Karlsson!
-
-2007-05-09 19:24 +0000 [r63699] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /: Merged revisions 63698 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r63698 | file | 2007-05-09 15:22:39 -0400 (Wed, 09 May 2007) | 2
- lines Use the DTMF frame on the channel when returning a DTMF
- frame from AST_FRAME_NULL or AST_FRAME_VOICE. ........
-
-2007-05-09 19:21 +0000 [r63697] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /: Merged revisions 63612 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r63612 | russell | 2007-05-09 11:55:27 -0500 (Wed, 09 May 2007) |
- 5 lines Modify ast_senddigit_begin() to use the same assumptions
- used elsewhere in the code in that if a channel does not have a
- send_digit_begin() callback, it only cares about DTMF END events.
- (pointed out by Michael Neuhauser on the asterisk-dev list)
- ........
-
-2007-05-09 17:35 +0000 [r63655] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Merged revisions 63654 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r63654 | mattf | 2007-05-09 12:25:21 -0500 (Wed,
- 09 May 2007) | 10 lines Merged revisions 63653 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r63653 | mattf | 2007-05-09 12:20:20 -0500 (Wed, 09 May 2007) | 2
- lines Make sure we only create a DSP if it's requested on
- SUB_REAL ........ ................
-
-2007-05-09 16:56 +0000 [r63613] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 63611 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r63611 | file | 2007-05-09 12:54:56 -0400 (Wed,
- 09 May 2007) | 10 lines Merged revisions 63610 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r63610 | file | 2007-05-09 12:51:03 -0400 (Wed, 09 May 2007) | 2
- lines Properly handle hints that point to multiple devices in
- chan_sip. Why chan_sip is even doing this I have no idea but I
- would rather not go into a rant. (issue #9536 reported by
- rlister) ........ ................
-
-2007-05-09 16:44 +0000 [r63609] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /: Merged revisions 63608 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r63608 | russell | 2007-05-09 11:43:50 -0500 (Wed, 09 May 2007) |
- 5 lines Only call ast_senddigit_begin() in ast_senddigit() if the
- channel has a send_digit_begin() callback. Checking the
- END_DTMF_ONLY flag was the wrong thing to do, because that flag
- indicates that a *bridged* channel only wants DTMF END events
- coming from this channel. ........
-
-2007-05-09 14:52 +0000 [r63567] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_directory.c: Merged revisions 63566 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r63566 | tilghman | 2007-05-09 09:50:33 -0500
- (Wed, 09 May 2007) | 10 lines Merged revisions 63565 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r63565 | tilghman | 2007-05-09 09:48:06 -0500 (Wed, 09
- May 2007) | 2 lines Replicate fix from 51158 (app_voicemail) to
- app_directory (Issue 9224) ........ ................
-
-2007-05-09 13:24 +0000 [r63536] Russell Bryant <russell@digium.com>
-
- * Makefile, /: Merged revisions 63535 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r63535 | russell | 2007-05-09 08:24:03 -0500 (Wed, 09 May 2007) |
- 6 lines I have seen multiple people post questions trying to
- figure out what the message "The configure script must be
- executed before running 'make'" means. So, add another like that
- says to specifically run ./configure. If this isn't obvious
- enough, then they should be using something like AsteriskNOW and
- not installing from source. ........
-
-2007-05-09 13:07 +0000 [r63533] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 63532 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r63532 | oej | 2007-05-09 15:04:14 +0200 (Wed, 09 May 2007) | 2
- lines Don't retransmit 200 OK's on ignore status. (Reported on
- asterisk-users) ........
-
-2007-05-08 22:40 +0000 [r63479] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_macro.c: Merged revisions 63478 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r63478 | tilghman | 2007-05-08 17:38:02 -0500
- (Tue, 08 May 2007) | 10 lines Merged revisions 63477 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r63477 | tilghman | 2007-05-08 17:19:15 -0500 (Tue, 08
- May 2007) | 2 lines Issue 9602 - segfault in app_macro ........
- ................
-
-2007-05-08 16:54 +0000 [r63404-63449] Russell Bryant <russell@digium.com>
-
- * /, res/res_features.c: Merged revisions 63448 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r63448 | russell | 2007-05-08 11:53:09 -0500 (Tue, 08 May 2007) |
- 4 lines I mixed up the use of the find_feature() function, so I
- renamed it find_dynamic_feature, and changed the code to use the
- correct lock when using it. ........
-
- * channels/chan_sip.c, res/res_features.c,
- include/asterisk/features.h: I noted this on the dev list but got
- no response, so I just did it myself. Lock the call features when
- being used in chan_sip.
-
- * /, res/res_features.c: Merged revisions 63445 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r63445 | russell | 2007-05-08 11:30:43 -0500 (Tue, 08 May 2007) |
- 2 lines Use a read/write lock when accessing the built-in
- features. ........
-
- * contrib/scripts/realtime_pgsql.sql (added), /,
- contrib/realtime_pgsql.sql (removed): Merged revisions 63403 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r63403 | russell | 2007-05-08 10:10:37 -0500 (Tue, 08 May 2007) |
- 3 lines Move realtime_pgsql.sql to contrib/scripts to be with the
- rest of the sql examples. (issue #9676, suretec) ........
-
-2007-05-08 06:26 +0000 [r63361] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 63360 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r63360 | tilghman | 2007-05-08 01:22:37 -0500
- (Tue, 08 May 2007) | 10 lines Merged revisions 63359 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r63359 | tilghman | 2007-05-08 01:20:16 -0500 (Tue, 08
- May 2007) | 2 lines Issue 9527 - upon entering a folder, no
- message is selected (curmsg == -1), so deleting causes memory
- corruption (beyond bounds) ........ ................
-
-2007-05-07 22:32 +0000 [r63319-63330] Russell Bryant <russell@digium.com>
-
- * /, contrib/realtime_pgsql.sql (added),
- configs/res_pgsql.conf.sample (added): Merged revisions 63329 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r63329 | russell | 2007-05-07 17:28:50 -0500 (Mon, 07 May 2007) |
- 3 lines Add a sample configuration file and example tables for
- use with res_config_pgsql. (issue #9676, suretec) ........
-
- * apps/app_meetme.c: Make a minor tweak to admin_exec() - don't
- lock the conference list until it is actually necessary.
-
- * apps/app_meetme.c, CHANGES: Add a new application,
- MeetMeChannelAdmin, which is similar to MeetMeAdmin, except it
- lets you operate on a channel by name instead of conference
- member number. It is very useful in combination with the 'X'
- option to ChanSpy. (issue #9671, patch by mnicholson, with some
- small modifications by me)
-
-2007-05-07 21:47 +0000 [r63284-63287] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, include/asterisk/app.h, /, main/app.c: Merged
- revisions 63286 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r63286 | file | 2007-05-07 17:45:01 -0400 (Mon,
- 07 May 2007) | 10 lines Merged revisions 63285 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r63285 | file | 2007-05-07 17:39:52 -0400 (Mon, 07 May 2007) | 2
- lines Properly handle what happens during a masquerade in
- relation to group counting. (issue #9657 reported by ramonpeek)
- ........ ................
-
-2007-05-07 20:07 +0000 [r63228-63255] Olle Johansson <oej@edvina.net>
-
- * /, main/config.c: Merged revisions 63254 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r63254 | oej | 2007-05-07 22:05:15 +0200 (Mon, 07 May 2007) | 2
- lines Don't remove configuration from memory just because one
- section failed. ........
-
- * include/asterisk/module.h, main/loader.c: Constifications
-
- * channels/chan_jingle.c, res/res_jabber.c: Adding external
- referenses for doxygen See
- http://www.asterisk.org/doxygen/trunk/extref.html
-
- * channels/chan_misdn.c: Adding external reference
-
- * channels/chan_misdn.c: Doxyfication... There's a shortage of
- comments in this file...
-
-2007-05-06 20:09 +0000 [r63182] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: Lock iax2 pvt structure when passing off to
- the AMI function, and make sure it exists. (issue #9674 reported
- by arabe)
-
-2007-05-06 13:11 +0000 [r63168] Olle Johansson <oej@edvina.net>
-
- * /, main/file.c: Merged revisions 63152 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r63152 | oej | 2007-05-06 14:28:38 +0200 (Sun, 06 May 2007) | 2
- lines Stop the video stream when you stop playback of all streams
- for a call ........
-
-2007-05-05 08:05 +0000 [r63136] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: - Adding some missing spaces - Correcting
- error messages - Disabling code that doesn't do anything - Making
- sure we always respond to this request, happily
-
-2007-05-04 20:11 +0000 [r63105] Pari Nannapaneni <paripurnachand@digium.com>
-
- * /, configs/manager.conf.sample: Merged revisions 63047 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r63047 | pari | 2007-05-04 11:45:29 -0500 (Fri, 04 May 2007) | 1
- line explanation for httptimeout in manager.conf ........
-
-2007-05-04 20:06 +0000 [r63104] Jason Parker <jparker@digium.com>
-
- * /, res/res_jabber.c: Merged revisions 63099 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r63099 | qwell | 2007-05-04 15:03:49 -0500 (Fri, 04 May 2007) | 4
- lines Fix a crash when checking version attribute in an incoming
- XML caps element. Issue 9667, patch by phsultan. ........
-
-2007-05-04 19:48 +0000 [r63089] Russell Bryant <russell@digium.com>
-
- * main/manager.c: Convert spaces to tabs for indentation.
-
-2007-05-04 18:47 +0000 [r63046-63076] Steve Murphy <murf@digium.com>
-
- * res/res_features.c: According to my testing, it's better if the
- ast_find_call_feature func ran this way instead, as far as the
- snom record button is concerned
-
- * doc/CODING-GUIDELINES, channels/chan_sip.c, res/res_features.c,
- include/asterisk/features.h: a small upgrade to the coding
- standard, and an update to the code that triggered the upgrade.
-
- * channels/chan_sip.c, res/res_features.c, UPGRADE.txt,
- include/asterisk/features.h: Added a small bit of code to support
- the SNOM 360's Record button. Made the find_feature func in
- res_features.c public, so I could use it to find the automon dial
- sequence as configured by the user. When the INFO packet has a
- Record: header with on/off, the sequence is sent as consecutive
- DTMF frames on the phone's channel, triggering the automon
- functionality. The user has to configure the automon in
- features.conf, and set up his dialplan accordingly.
-
-2007-05-04 13:56 +0000 [r63030-63032] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c, channels/chan_iax2.c: Add the new
- ChannelUpdate event to inform manager clients about the PVT ID
- and some other channel driver data that is needed to follow the
- call through the PBX.
-
- * main/manager.c: Add "CoreStatus" - from the moremanager branch.
- This can be extended with more information, ideas and patches are
- welcome, as usual :-)
-
- * include/asterisk.h, main/manager.c, include/asterisk/manager.h,
- include/asterisk/options.h: - Add manager command CoreSettings -
- Add missing option to options.h - Add missing variables to
- asterisk.h - Move manager version to manager.h include file
-
-2007-05-03 16:45 +0000 [r62990] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 62989 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r62989 | file | 2007-05-03 13:44:00 -0300 (Thu,
- 03 May 2007) | 10 lines Merged revisions 62987 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r62987 | file | 2007-05-03 13:42:19 -0300 (Thu, 03 May 2007) | 2
- lines When a peer is seeded or built tell the devicestate core to
- update it's status. This is easier then having chan_sip load
- before pbx_config. (issue #9658 reported by dlynes) ........
- ................
-
-2007-05-03 16:43 +0000 [r62988] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, main/loader.c: Merged revisions 62986 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r62986 | kpfleming | 2007-05-03 11:38:56 -0500 (Thu, 03 May 2007)
- | 2 lines improve loader a bit, by avoiding trying to initialize
- embedded modules twice and avoiding trying to load modules from
- disk when they have been loaded already during the 'preload' pass
- (reported by blitzrage on IRC, patch by me) ........
-
-2007-05-03 15:23 +0000 [r62943] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /: Merged revisions 62942 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r62942 | russell | 2007-05-03 10:23:13 -0500 (Thu, 03 May 2007) |
- 17 lines Fix YADB (Yet Another DTMF Bug) ((C) Russell Bryant,
- 2007, TM, Patent Pending). This set of changes came from a
- debugging session I had with Dwayne Hubbard. When he called into
- his home FXO, ran the Echo application, and pressed a digit, the
- digit would be echoed back and would never end. This is fixed,
- along with a couple other little improvements. * When chan_zap is
- in the middle of playing a digit to a channel, it feeds back null
- frames, not voice frames. So, I have modified ast_read to check
- the timing on emulated DTMF when it receives null frames, in
- addition to where it was doing this on voice frames. * Make a
- tweak to setting the duration on emulated DTMF digits. If there
- was no duration specified, it set it to be the minimum, instead
- of the default. * Instead of timing the emulated digits off of
- the number of samples in audio frames that pass through, just use
- time values. Now there is no code in this section that assumes
- 8kHz audio. ........
-
-2007-05-03 14:44 +0000 [r62911-62914] Steve Murphy <murf@digium.com>
-
- * /: blocking 62913 (1.4) from trunk, as it's already done here
-
- * /, pbx/ael/ael.tab.c, pbx/ael/ael.y,
- pbx/ael/ael-test/ref.ael-test20 (added), pbx/ael/ael.tab.h,
- pbx/ael/ael-test/ael-test20/extensions.ael (added),
- pbx/ael/ael-test/ael-test20 (added): Merged revisions 62883 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r62883 | murf | 2007-05-03 07:54:56 -0600 (Thu, 03 May 2007) | 1
- line These mods fix bug 9623, where an '@' in the eswitch
- contents causes a syntax error. I also updated the regressions.
- ........
-
-2007-05-03 00:25 +0000 [r62824-62843] Kevin P. Fleming <kpfleming@digium.com>
-
- * res/res_config_odbc.c, /: Merged revisions 62842 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r62842 | kpfleming | 2007-05-02 20:23:37 -0400
- (Wed, 02 May 2007) | 10 lines Merged revisions 62841 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r62841 | kpfleming | 2007-05-02 20:23:00 -0400 (Wed, 02
- May 2007) | 2 lines doh... initializing the pointer variable will
- work just a bit better ........ ................
-
- * main/minimime: ignore the archive we build in this directory
-
- * res/res_config_pgsql.c, res/res_config_odbc.c, /: Merged
- revisions 62797,62807 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r62797 | kpfleming | 2007-05-02 19:57:23 -0400
- (Wed, 02 May 2007) | 7 lines improve static Realtime config
- loading from PostgreSQL: don't request sorting on fields that are
- pointless to sort on use ast_build_string() instead of snprintf()
- don't request the list of fieldnames that resulted from the query
- when we both knew what they were before we ran the query _AND_ we
- aren't going to do anything with them anyway (patch by me,
- inspired by blitzrage's bug report about res_config_odbc)
- ................ r62807 | kpfleming | 2007-05-02 20:02:57 -0400
- (Wed, 02 May 2007) | 15 lines Merged revisions 62796 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r62796 | kpfleming | 2007-05-02 19:53:46 -0400 (Wed, 02
- May 2007) | 7 lines increase reliability and efficiency of static
- Realtime config loading via ODBC: don't request fields we aren't
- going to use don't request sorting on fields that are pointless
- to sort on explicitly request the fields we want, because we
- can't expect the database to always return them in the order they
- were created (reported by blitzrage in person (!), patch by me)
- ........ ................
-
-2007-05-02 23:50 +0000 [r62791-62795] Russell Bryant <russell@digium.com>
-
- * CHANGES: Fix some bad grammar.
-
- * apps/app_meetme.c, CHANGES: When a conference is created, the
- UNIQUEID of the channel that caused it to be created will now be
- stored. Then, every channel that joins the conference will have
- the MEETMEUNIQUEID channel variable set with this ID. This can be
- used to relate callers that come and go from long standing
- conferences. (issue #7295, patch by softins)
-
- * CHANGES: Note Hungarian language support in CHANGES
-
- * main/say.c, configs/say.conf.sample: Add Hungarian language
- support to say.c and say.conf. (issue #7077, patch by adomjan)
-
- * main/channel.c, /: Merged revisions 62789 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r62789 | russell | 2007-05-02 17:59:09 -0500 (Wed, 02 May 2007) |
- 20 lines Merge changes from team/russell/inband_dtmf ... Fix some
- issues related to generating inband DTMF. There are two changes
- here: 1) The list of DTMF tones in the senddigit_begin() function
- explicitly specified 100ms of the tone followed by 100ms of
- silence. This really broke things with the way that Asterisk now
- wants complete control over when the digit begins and ends. So,
- regardless of what Asterisk really wanted to do, this was going
- to play out the tone at the length it wanted to. This caused
- various problems like DTMF translation to inband to be extremely
- unreliable. The list of tones has been changed so that the
- correct DTMF tone is played indefinitely until Asterisk tells it
- to stop. 2) ast_write() had to be modified to let a DTMF_END
- frame get processed even when a generator is present. This is how
- the tone will finally get stopped. (issues #8944, #9250, #9348,
- maybe others. Thanks to mdu113 from #8944 for the testing and
- feedback!) ........
-
-2007-05-02 20:57 +0000 [r62741] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, main/pbx.c, /: Merged revisions 62738 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r62738 | murf | 2007-05-02 14:46:07 -0600 (Wed,
- 02 May 2007) | 9 lines Merged revisions 62737 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r62737 | murf | 2007-05-02 14:10:32 -0600 (Wed, 02 May 2007) | 1
- line Some tweaks to satisfy CDR bug 8796, where being in 'h'
- extension louses up the dst field ........ ................
-
-2007-05-02 17:49 +0000 [r62693] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 62692 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r62692 | tilghman | 2007-05-02 12:43:48 -0500
- (Wed, 02 May 2007) | 12 lines Merged revisions 62691 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r62691 | tilghman | 2007-05-02 12:38:16 -0500 (Wed, 02
- May 2007) | 4 lines Issue 9638 - if a text frame is sent with no
- terminating NULL through a bridged IAX connection, the remote end
- will receive garbage characters tacked onto the end. ........
- ................
-
-2007-05-02 17:24 +0000 [r62690] Steve Murphy <murf@digium.com>
-
- * main/channel.c, main/pbx.c, channels/chan_zap.c, /,
- cdr/cdr_radius.c: Merged revisions 62689 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r62689 | murf | 2007-05-02 11:10:50 -0600 (Wed, 02 May 2007) | 1
- line a)In chan_zap, set the clid, src fields in channel_alloc
- call. b)in the channel_alloc func, set the cid_num and name
- fields from the arglist[blush]. c) don't update the channel app &
- app data fields if you are in the 'h' extension. d)the
- load_module func in cdr_radius needs to return DECLINE, SUCCESS.
- ........
-
-2007-05-02 15:46 +0000 [r62671-62673] Russell Bryant <russell@digium.com>
-
- * channels/chan_local.c, CHANGES: Update the device state
- functionality of chan_local such that it will return NOT_INUSE or
- INUSE when Local channels are in use as opposed to just UNKNOWN.
- It will still return INVALID if the extension doesn't exist at
- all. (issue #8048, patch from tim_ringenbach)
-
- * CHANGES: Add the new options for attended transfer to the CHANGES
- file.
-
- * doc/ip-tos.tex (removed), doc/qos.tex (added): For some reason
- when I merged 802.1p support, the new documentation file was not
- properly added. Thanks to IgorG for pointing it out! :)
-
-2007-05-02 12:12 +0000 [r62609-62656] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Add a small message that we're doing
- something. On my systems, there's a long dead period with a
- non-responsive CLI after I issue "load chan_sip.so"
-
- * channels/chan_sip.c: More username body parts to fix... If
- working, this needs to be backported to 1.2, 1.4. But first, some
- serious SIP testing :-)
-
- * channels/chan_sip.c: Handle
- sip:username;parameter=12345@example.com;parameter=1234 URI's
- properly
-
- * /, channels/chan_sip.c: Merged revisions 62624 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r62624 | oej | 2007-05-02 08:15:43 +0200 (Wed, 02 May 2007) | 2
- lines Don't unlock a channel that we already know does not exist
- (propably isue 8228) ........
-
- * CREDITS: Updating CREDITS
-
-2007-05-01 22:24 +0000 [r62549-62593] Russell Bryant <russell@digium.com>
-
- * res/res_features.c, configs/features.conf.sample: In addition to
- making it so attended transfers don't fail unnecessarily, add
- some new options to control what happens when you hangup on an
- attended transfer before the target extension answers the
- transferred channel. You can now have it send the transferee back
- to the transferer. (issue #8413, patch from sergee with very
- minor modifications by me)
-
- * /, res/res_features.c: Merged revisions 62548 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r62548 | russell | 2007-05-01 16:57:10 -0500
- (Tue, 01 May 2007) | 12 lines Merged revisions 62547 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r62547 | russell | 2007-05-01 16:55:19 -0500 (Tue, 01
- May 2007) | 4 lines Remove an unnecessary check that makes it so
- if you hang up after doing an attended transfer before the target
- extension answers the channel, the transfer is not successful.
- (issue #9338, patch by svanlund) ........ ................
-
-2007-05-01 21:41 +0000 [r62546] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 62545 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r62545 | tilghman | 2007-05-01 16:34:43 -0500 (Tue, 01 May 2007)
- | 2 lines Bug 9590 - Memory leaks around find_user() (found by
- rayjay, different fixes by me) ........
-
-2007-05-01 16:27 +0000 [r62415-62498] Russell Bryant <russell@digium.com>
-
- * /, configs/indications.conf.sample: Merged revisions 62497 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r62497 | russell | 2007-05-01 11:26:48 -0500
- (Tue, 01 May 2007) | 11 lines Merged revisions 62496 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r62496 | russell | 2007-05-01 11:26:23 -0500 (Tue, 01
- May 2007) | 3 lines Add indications.conf information for the
- Philippines. (issue #9525, reported and patched by loloski)
- ........ ................
-
- * CHANGES: Add a note to CHANGES about the new support for 802.1p.
- Thanks IgorG!
-
- * CHANGES, apps/app_queue.c, doc/queuelog.tex: This patch adds
- additional information to the EXITWITHKEY and EXITWITHTIMEOUT
- entries in the queue log. (issue #7561, reported and originally
- patched by fkasumovic, patch slightly modified and updated to
- trunk by me)
-
- * include/asterisk/acl.h, main/udptl.c, channels/chan_sip.c,
- include/asterisk/rtp.h, main/acl.c, include/asterisk/netsock.h,
- channels/iax2-provision.c, channels/chan_iax2.c, main/rtp.c,
- main/netsock.c, configs/h323.conf.sample,
- configs/iax.conf.sample, configs/mgcp.conf.sample,
- configs/iaxprov.conf.sample, channels/chan_h323.c,
- pbx/pbx_dundi.c, include/asterisk/udptl.h,
- configs/sip.conf.sample, doc/asterisk.tex, channels/chan_mgcp.c:
- Add support for setting the CoS for VLAN traffic (802.1p) in
- Linux. The file doc/qos.tex has been updated to document the new
- functionality. (issue #9540, patch submitted by IgorG)
-
- * channels/chan_zap.c, /: Merged revisions 62419 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r62419 | russell | 2007-04-30 10:58:28 -0500
- (Mon, 30 Apr 2007) | 12 lines Merged revisions 62417 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r62417 | russell | 2007-04-30 10:57:26 -0500 (Mon, 30
- Apr 2007) | 4 lines This patch fixes an issue where depending on
- the cause code, when the network sends a PRI disconnect, the call
- may not be properly hung up. (issue #9588, reported and patched
- by softins) ........ ................
-
- * channels/chan_sip.c: Don't crash when invalid arguments are
- provided to the CHANNEL() function for a SIP channel. (issue
- #9619, reported by jtodd, original patch by Corydon76, committed
- patch slightly modified by me)
-
- * include/asterisk/http.h, /, main/http.c: Merged revisions 62414
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r62414 | russell | 2007-04-30 10:25:31 -0500 (Mon, 30 Apr 2007) |
- 4 lines When serving dynamic content, include a Cache-Control
- header to instruct the browsers to not store the resulting
- content. (issue #9621, reported by Pari, patch by me) ........
-
-2007-04-30 14:56 +0000 [r62372] Jason Parker <jparker@digium.com>
-
- * configs/iax.conf.sample, /: Merged revisions 62371 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r62371 | qwell | 2007-04-30 09:52:31 -0500 (Mon, 30 Apr
- 2007) | 2 lines Remove unused (and potentially confusing)
- jitterbuffer options from sample config. ........
-
-2007-04-30 14:37 +0000 [r62370] Joshua Colp <jcolp@digium.com>
-
- * /, main/asterisk.c: Merged revisions 62369 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r62369 | file | 2007-04-30 11:36:11 -0300 (Mon,
- 30 Apr 2007) | 10 lines Merged revisions 62368 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r62368 | file | 2007-04-30 11:34:07 -0300 (Mon, 30 Apr 2007) | 2
- lines Update copyright notice. It's now the year 2007! ........
- ................
-
-2007-04-29 05:51 +0000 [r62219-62332] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 62331 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r62331 | russell | 2007-04-29 00:50:37 -0500 (Sun, 29 Apr 2007) |
- 3 lines Fix a bug that made the "language" setting in zapata.conf
- not functional. (issue #9626, reported and fixed by sergee)
- ........
-
- * CHANGES: note MeetMe change in CHANGES
-
- * apps/app_meetme.c: Enable the functionality of the 'o' option to
- "optimize talker" by default.
-
- * channels/iax2.h: Reformat some of iax2.h and convert comments to
- doxygen format
-
- * include/asterisk.h, channels/chan_zap.c, channels/chan_sip.c,
- main/Makefile, res/res_eventtest.c (added),
- configs/voicemail.conf.sample, UPGRADE.txt, CHANGES,
- channels/chan_iax2.c, main/dial.c, include/asterisk/event.h
- (added), include/asterisk/event_defs.h (added), main/event.c
- (added), configs/sip.conf.sample, main/asterisk.c,
- channels/chan_mgcp.c, apps/app_voicemail.c: Merge changes from
- team/russell/events This set of changes introduces a new generic
- event API for use within Asterisk. I am still working on a way
- for events to be shared between servers, but this part is ready
- and can already be used inside of Asterisk. This set of changes
- introduces the first use of the API, as well. I have restructured
- the way that MWI (message waiting indication) is handled. It is
- now event based instead of polling based. For example, if there
- are a bunch of SIP phones subscribed to mailboxes, then chan_sip
- will not have to constantly poll the mailboxes for changes.
- app_voicemail will generate events when changes occur. See
- UPGRADE.txt and CHANGES for some more information on the effects
- of these changes from the user perspective. For developer
- information, see the text in include/asterisk/event.h. As always,
- additional feedback is welcome on the asterisk-dev mailing list.
-
- * doc/ast_appdocs.tex, doc/dundi.tex: Update the DUNDi section of
- the documentation with example usage of DUNDIQUERY and
- DUNDIRESULT. Also, update the automatically generated application
- docs.
-
- * pbx/pbx_dundi.c, CHANGES: Merge changes from
- team/russell/dundi_results This introduces two new dialplan
- functions: DUNDIQUERY and DUNDIRESULT. DUNDIQUERY lets you
- intitiate a DUNDi query from the dialplan. Then, DUNDIRESULT will
- let you find out how many results there are, and access each one
- without having to the query again.
-
- * include/asterisk/lock.h: Remove a message that goes to LOG_ERROR
- that's not really an error.
-
- * configs/queues.conf.sample, CHANGES, apps/app_queue.c: Add a
- min-announce-frequency option to queues.conf which allows you to
- control the minimum amount of time between queue announcements
- for use when the caller's queue position changes frequently.
- (issue #9604, patch by Matthew Roth)
-
- * /, channels/chan_agent.c: Merged revisions 62218 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r62218 | russell | 2007-04-27 16:10:51 -0500 (Fri, 27
- Apr 2007) | 11 lines Fix a weird problem where when a caller
- talking to someone sitting behind an agent channel sent a digit,
- the digit would be played to the agent for forever. This is
- because chan_agent always returned -1 from its send_digit_begin
- and _end callbacks. This non-zero return value indicates to the
- Asterisk core that it would like an inband DTMF generator put on
- the channel. However, this is the wrong thing to do. It should
- *always* return 0, instead. When the digit begin and end
- functions are called on the proxied channel, the underlying
- channel will indicate whether inband DTMF is needed or not, and
- the generator will be put on that one, and not the Agent channel.
- (issue #9615, #9616, reported by jiddings and BigJimmy, and fixed
- by me) ........
-
-2007-04-27 16:18 +0000 [r62175] Jason Parker <jparker@digium.com>
-
- * /, codecs/codec_zap.c: Merged revisions 62174 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r62174 | qwell | 2007-04-27 11:17:46 -0500 (Fri,
- 27 Apr 2007) | 11 lines Merged revisions 62173 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r62173 | qwell | 2007-04-27 11:16:16 -0500 (Fri, 27 Apr 2007) | 3
- lines This transcoder message needn't be a NOTICE. I've seen it
- cause confusion more than a few times. ........ ................
-
-2007-04-27 16:15 +0000 [r62172] Russell Bryant <russell@digium.com>
-
- * main/pbx.c, /: Merged revisions 62171 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r62171 | russell | 2007-04-27 11:14:11 -0500 (Fri, 27 Apr 2007) |
- 6 lines If no variables were passed into
- pbx_substitute_variables_helper_full(), then don't even bother
- creating a temporary bogus channel, since that is only for
- allowing certain functions to operate on the variables as if they
- were on a channel. Most importantly, this fixes a crash. (issue
- #9613, reported by callguy, fixed by me) ........
-
-2007-04-27 14:40 +0000 [r62096-62141] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Issue #9545 Autocomplete for "sip
- unregister" cli command. (eliel) Thanks!
-
- * /, channels/chan_sip.c: Merged revisions 62137 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r62137 | oej | 2007-04-27 16:04:07 +0200 (Fri,
- 27 Apr 2007) | 12 lines Merged revisions 62126 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r62126 | oej | 2007-04-27 15:57:45 +0200 (Fri, 27 Apr 2007) | 4
- lines Issue #7351 - SIP Cancel fails due to the wrong contact
- uri. Reported by PPYY, failed to fix by OEJ final fix by wojtekka
- - THANKS!!!! THis was a hard one to catch. ........
- ................
-
- * /: Blocking patch to 1.4 that was alredy in trunk
-
-2007-04-26 16:35 +0000 [r62039] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 62038 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r62038 | file | 2007-04-26 12:33:52 -0400 (Thu,
- 26 Apr 2007) | 10 lines Merged revisions 62037 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r62037 | file | 2007-04-26 12:30:57 -0400 (Thu, 26 Apr 2007) | 2
- lines Revert previous fix for when the IAX2 channel goes funky
- (that's the technical term). This is causing legit calls to be
- prematurely hung up. (issue #9600 reported by justdave) ........
- ................
-
-2007-04-26 03:24 +0000 [r62006] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /: Merged revisions 62005 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r62005 | file | 2007-04-25 22:19:51 -0500 (Wed, 25 Apr 2007) | 2
- lines Missed an ast_app_group_discard during merge. Thanks
- blitzrage! ........
-
-2007-04-26 01:50 +0000 [r61960-61962] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_monitor.c: Merged revisions 61961 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r61961 | file | 2007-04-25 21:48:55 -0400 (Wed, 25 Apr 2007) | 2
- lines Don't always say that the channel is being paused if it is
- actually being unpaused in the Manager ack message. (reported by
- jsmith in #asterisk-bugs) ........
-
- * /, main/config.c: Merged revisions 61959 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r61959 | file | 2007-04-25 21:27:18 -0400 (Wed,
- 25 Apr 2007) | 10 lines Merged revisions 61958 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61958 | file | 2007-04-25 21:25:03 -0400 (Wed, 25 Apr 2007) | 2
- lines Don't count failed include attempts against the
- configuration include level. (issue #9593 reported by mostyn)
- ........ ................
-
-2007-04-25 22:34 +0000 [r61915] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 61914 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r61914 | kpfleming | 2007-04-25 17:29:53 -0500
- (Wed, 25 Apr 2007) | 10 lines Merged revisions 61913 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r61913 | kpfleming | 2007-04-25 17:24:59 -0500 (Wed, 25
- Apr 2007) | 2 lines handle a very bizarre race condition with
- channels being redirected before a simple switch can be started
- on them (issue #9286) ........ ................
-
-2007-04-25 22:01 +0000 [r61864-61876] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 61870 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r61870 | russell | 2007-04-25 16:59:07 -0500
- (Wed, 25 Apr 2007) | 10 lines Merged revisions 61866 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r61866 | russell | 2007-04-25 16:55:23 -0500 (Wed, 25
- Apr 2007) | 2 lines If the callerid= option is specified, but
- empty, clear any previous data. ........ ................
-
- * /, channels/chan_iax2.c: Merged revisions 61863 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r61863 | russell | 2007-04-25 16:13:15 -0500
- (Wed, 25 Apr 2007) | 10 lines Merged revisions 61862 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r61862 | russell | 2007-04-25 16:06:22 -0500 (Wed, 25
- Apr 2007) | 2 lines Ensure that callerid settings are reset on a
- reload. ........ ................
-
-2007-04-25 19:27 +0000 [r61806] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, include/asterisk/app.h, funcs/func_groupcount.c,
- /, main/app.c, main/cli.c: Merged revisions 61805 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r61805 | file | 2007-04-25 15:21:54 -0400 (Wed,
- 25 Apr 2007) | 10 lines Merged revisions 61804 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61804 | file | 2007-04-25 14:52:50 -0400 (Wed, 25 Apr 2007) | 2
- lines Merge rewritten group counting support. No more storing
- data on the variable list of the channels. That was bad, mmmk?
- (issue #7497 reported by sabbathbh) ........ ................
-
-2007-04-25 16:23 +0000 [r61788-61800] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 61799 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r61799 | russell | 2007-04-25 11:22:07 -0500
- (Wed, 25 Apr 2007) | 11 lines Merged revisions 61798 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r61798 | russell | 2007-04-25 11:20:38 -0500 (Wed, 25
- Apr 2007) | 3 lines Fix a typo where cid_num got copied instead
- of cid_ani. (issue #9587, reported and patched by xrg) ........
- ................
-
- * main/manager.c, /: Merged revisions 61787 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r61787 | russell | 2007-04-24 16:34:53 -0500
- (Tue, 24 Apr 2007) | 12 lines Merged revisions 61786 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r61786 | russell | 2007-04-24 16:33:59 -0500 (Tue, 24
- Apr 2007) | 4 lines Don't crash if a manager connection provides
- a username that exists in manager.conf but does not have a
- password, and also requests MD5 authentication. (ASA-2007-012)
- ........ ................
-
-2007-04-24 19:08 +0000 [r61784] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * channels/chan_zap.c, /: removed #if 0 block from chan_zap
- restart_monitor()
-
-2007-04-24 19:03 +0000 [r61775-61782] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /, include/asterisk/channel.h: Merged revisions
- 61781 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r61781 | russell | 2007-04-24 14:00:06 -0500 (Tue, 24 Apr 2007) |
- 6 lines Improve DTMF handling in ast_read() even more in response
- to a discussion on the asterisk-dev mailing list. I changed the
- enforced minimum length of a digit from 100ms to 80ms.
- Furthermore, I made it now enforce a gap of 45ms in between
- digits. These values are not configurable in a configuration file
- right now, but they can be easily changed near the top of
- main/channel.c. ........
-
- * main/dial.c, /: Merged revisions 61774 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r61774 | russell | 2007-04-24 11:16:41 -0500 (Tue, 24 Apr 2007) |
- 5 lines Add a few more state changes in handle_frame_ownerless()
- so that the SLA code will get notified of these changes even when
- an owner channel is not provided. This isn't from a specific bug
- report, it's just something I noticed while poking around.
- ........
-
-2007-04-24 16:10 +0000 [r61773] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 61772 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r61772 | file | 2007-04-24 12:07:02 -0400 (Tue,
- 24 Apr 2007) | 10 lines Merged revisions 61771 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61771 | file | 2007-04-24 12:05:06 -0400 (Tue, 24 Apr 2007) | 2
- lines Allow RFC2833 to be sent in the response SDP when an INVITE
- comes in without SDP. (issue #9546 reported by mcrawford)
- ........ ................
-
-2007-04-23 18:49 +0000 [r61760-61767] Russell Bryant <russell@digium.com>
-
- * main/manager.c: When building a JSON encoded string in the
- GetConfigJSON manager action, escape the '\' and '"' characters.
- (issue #9475, reported by pari, patch by me)
-
- * main/pbx.c, /: Merged revisions 61765 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r61765 | russell | 2007-04-23 13:17:00 -0500 (Mon, 23 Apr 2007) |
- 5 lines Some dialplan functions, such as CUT(), expect to operate
- on variables on a channel. So, this little hack lets them work in
- places where a channel doesn't exist, such as within DUNDi
- configuration. (issue #9465, reported and patched by Corydon76,
- testing by blitzrage) ........
-
- * main/channel.c, /: Merged revisions 61763 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r61763 | russell | 2007-04-23 12:57:32 -0500 (Mon, 23 Apr 2007) |
- 4 lines Ensure that digits passing through Asterisk have a
- reasonable minimum length. It is currently 100 ms. If someone
- thinks this should be different, feel free to speak up. (related
- to issues #8944, #9250, and #9348) ........
-
- * CHANGES: Add OSP support for IAX2 to the changes file. Also,
- slightly reorganize some of the content.
-
-2007-04-20 21:37 +0000 [r61706-61708] Jason Parker <jparker@digium.com>
-
- * /, main/rtp.c: Merged revisions 61707 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r61707 | qwell | 2007-04-20 16:35:27 -0500 (Fri, 20 Apr 2007) | 8
- lines Avoid invalid seqno cycling detection. Per comment from
- Dave Troy: This adds back in some simple typecasting I had in an
- earlier version which I realize now may be breaking things. Issue
- #9554. ........
-
- * /, main/loader.c: Merged revisions 61705 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r61705 | qwell | 2007-04-20 16:15:29 -0500 (Fri,
- 20 Apr 2007) | 12 lines Merged revisions 61704 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61704 | qwell | 2007-04-20 16:14:27 -0500 (Fri, 20 Apr 2007) | 4
- lines Fix an issue that I noticed while looking over issue 9571.
- The reload timestamp was getting set after reloading the built-in
- stuff, and before the modules. ........ ................
-
-2007-04-20 21:12 +0000 [r61698-61702] Russell Bryant <russell@digium.com>
-
- * channels/iax2-parser.h, funcs/func_channel.c, channels/iax2.h,
- channels/chan_iax2.c, channels/iax2-parser.c: Merge changes from
- team/russell/iax2_osp This set of changes adds OSP support to
- chan_iax2. However, I have modified the patch a bit from what was
- submitted. You now use the CHANNEL() function to get and set the
- OSP token for IAX2. (issue #8531, reported by and original patch
- by homesick, patch updated by me)
-
- * /, main/rtp.c: Merged revisions 61697 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r61697 | russell | 2007-04-20 15:42:02 -0500 (Fri, 20 Apr 2007) |
- 2 lines Remove a stray debug message introduced by a recent
- commit. ........
-
-2007-04-20 19:54 +0000 [r61695] Jason Parker <jparker@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 61694 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r61694 | qwell | 2007-04-20 14:51:49 -0500 (Fri,
- 20 Apr 2007) | 13 lines Merged revisions 61692 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61692 | qwell | 2007-04-20 14:49:54 -0500 (Fri, 20 Apr 2007) | 5
- lines If the '* to hangup' option is not enabled, we don't need
- to disable * as a valid exit key. If it was enabled, this
- statement would've never been checked in the first place. Issue
- #9552 ........ ................
-
-2007-04-20 18:23 +0000 [r61691] Russell Bryant <russell@digium.com>
-
- * main/manager.c, /, include/asterisk/config.h, main/config.c,
- apps/app_voicemail.c: Merged revisions 61690 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r61690 | russell | 2007-04-20 13:19:18 -0500 (Fri, 20 Apr 2007) |
- 4 lines Fix the UpdateConfig manager action to properly treat
- "variables" and "objects" differently (a=b versus a=>b). (issue
- #9568, reported by pari, patch by me) ........
-
-2007-04-20 08:41 +0000 [r61689] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Use the last line in the SDP, even if it
- has no CRLF. Remember Jon Postel :-) This code exists in 1.2 and
- 1.4 but was removed from trunk for some unknown reason.
-
-2007-04-19 04:37 +0000 [r61682-61684] Tilghman Lesher <tlesher@digium.com>
-
- * main/manager.c, /: Merged revisions 61683 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r61683 | tilghman | 2007-04-18 23:36:20 -0500 (Wed, 18 Apr 2007)
- | 2 lines Bug 9557 - simple reason why reading a function always
- returned NULL ........
-
- * funcs/func_groupcount.c, /, funcs/func_timeout.c,
- funcs/func_cdr.c, funcs/func_callerid.c: Merged revisions 61681
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r61681 | tilghman | 2007-04-18 21:45:05 -0500
- (Wed, 18 Apr 2007) | 13 lines Merged revisions 61680 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r61680 | tilghman | 2007-04-18 21:30:18 -0500 (Wed, 18
- Apr 2007) | 5 lines Bug 9557 - Specifying the GetVar AMI action
- without a Channel parameter can cause Asterisk to crash. The
- reason this needs to be fixed in the functions instead of in AMI
- is because Channel can legitimately be NULL, such as when
- retrieving global variables. ........ ................
-
-2007-04-18 22:11 +0000 [r61679] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, sounds/Makefile: Merged revisions 61678 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r61678 | kpfleming | 2007-04-18 17:10:23 -0500 (Wed, 18 Apr 2007)
- | 2 lines allow external build systems to extract the required
- sound file versions ........
-
-2007-04-18 20:48 +0000 [r61671-61677] Olle Johansson <oej@edvina.net>
-
- * /, main/rtp.c: Merged revisions 61676 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r61676 | oej | 2007-04-18 22:46:23 +0200 (Wed, 18 Apr 2007) | 2
- lines Clean upp formatting, add some doxygen stuff while we're in
- cleaning mode... Thanks Kevin! ........
-
- * /, main/rtp.c: Merged revisions 61674 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r61674 | oej | 2007-04-18 22:28:53 +0200 (Wed, 18 Apr 2007) | 2
- lines Issue #9554 - Improve RTCP (Dave Troy) ........
-
- * apps/app_minivm.c (added), configs/extensions_minivm.conf.sample
- (added), configs/minivm.conf.sample (added): Mini-voicemail - an
- embryo for a new voicemail system based on building blocks
- instead of one large monolithic app. Supports multiple templates
- and is designed mostly for voicemail delivery over e-mail.
- There's a todo with a list of ideas in the source code if you
- want to contribute. Feedback is appreciated!
-
-2007-04-16 15:40 +0000 [r61667] Olle Johansson <oej@edvina.net>
-
- * include/asterisk/rtp.h: Doxygen changes
-
-2007-04-14 18:22 +0000 [r61661] Claude Patry <cpatry@gmail.com>
-
- * main/say.c: test my new trunk access ;)
-
-2007-04-13 21:23 +0000 [r61660] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * channels/chan_sip.c: added CLI 'sip unregister <peer>' for issue
- 9326. thanks eliel
-
-2007-04-13 21:22 +0000 [r61659] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, /: Merged revisions 61658 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r61658 | murf | 2007-04-13 15:17:20 -0600 (Fri, 13 Apr 2007) | 1
- line This is a fix to the way CDR merge handles the data that
- results from ForkCDR. ........
-
-2007-04-13 19:18 +0000 [r61649-61657] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 61656 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r61656 | file | 2007-04-13 15:17:08 -0400 (Fri,
- 13 Apr 2007) | 10 lines Merged revisions 61655 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61655 | file | 2007-04-13 15:15:12 -0400 (Fri, 13 Apr 2007) | 2
- lines Add OUTBOUND_GROUP_ONCE variable to app_dial. This behaves
- the same as OUTBOUND_GROUP except it will get unset after use so
- it won't get accidentally inherited. (issue #BE-140) ........
- ................
-
- * /, apps/app_speech_utils.c: Merged revisions 61651 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r61651 | file | 2007-04-13 14:08:02 -0400 (Fri, 13 Apr
- 2007) | 2 lines Do not bother looking for a result if none are
- present. ........
-
- * /, channels/chan_sip.c: Merged revisions 61648 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r61648 | file | 2007-04-13 13:19:53 -0400 (Fri, 13 Apr 2007) | 2
- lines For those very verbose SIP implementations that attach tons
- of info to the Contact header... let's increase our variable
- sizes. (issue #9535 reported by jeffg) ........
-
-2007-04-13 17:15 +0000 [r61647] Russell Bryant <russell@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 61645 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r61645 | russell | 2007-04-13 12:10:19 -0500 (Fri, 13 Apr 2007) |
- 3 lines Eliminate a compiler warning with ODBC_STORAGE enabled so
- that it will build under dev-mode. ........
-
-2007-04-13 17:11 +0000 [r61646] Steve Murphy <murf@digium.com>
-
- * /, channels/chan_oss.c: Merged revisions 61644 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r61644 | murf | 2007-04-13 11:01:02 -0600 (Fri, 13 Apr 2007) | 1
- line A fix for chan_oss that resulted from the CDR changes; it
- helps to use the right info. ........
-
-2007-04-13 16:35 +0000 [r61618-61642] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 61641 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r61641 | file | 2007-04-13 12:32:03 -0400 (Fri, 13 Apr 2007) | 2
- lines Don't assume the callid of a dialog will be set, as in some
- circumstances it may not. (issue #9534 reported by tecnoxarxa)
- ........
-
- * channels/chan_sip.c: Don't treat a host lookup as failed if
- sipregs is not in use when doing a realtime lookup. (issue #9255
- reported by sergee)
-
-2007-04-11 22:19 +0000 [r61575-61599] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * doc/asterisk-conf.tex: clarified 'minmemfree' description in
- doc/asterisk-conf.tex
-
- * main/asterisk.c, doc/asterisk-conf.tex: fixed the '-e' command
- line option for minmemfree. updated doc/asterisk-conf.tex
-
- * main/pbx.c, include/asterisk/options.h, main/asterisk.c: changed
- #if HAVE_SYSINFO to #if defined(HAVE_SYSINFO)
-
- * main/pbx.c, include/asterisk/options.h, main/asterisk.c: added
- HAVE_SYSINFO preprocessor directives for portability and general
- happiness
-
-2007-04-11 20:21 +0000 [r61557] Joshua Colp <jcolp@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac: Add a
- configure script check for sysinfo support.
-
-2007-04-11 19:11 +0000 [r61539] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * main/pbx.c, include/asterisk/options.h, main/asterisk.c: added
- option_minmemfree for use in asterisk.conf to specify the amount
- of minimum free memory prior to accepting calls. added CLI 'core
- show sysinfo' to display system information
-
-2007-04-11 17:07 +0000 [r61522] Joshua Colp <jcolp@digium.com>
-
- * main/logger.c: Output verbose messages to the normal logger as
- well. (issue #9476 reported by gdalgliesh)
-
-2007-04-11 16:06 +0000 [r61478] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 61477 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r61477 | russell | 2007-04-11 11:05:29 -0500
- (Wed, 11 Apr 2007) | 13 lines Merged revisions 61476 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r61476 | russell | 2007-04-11 11:01:25 -0500 (Wed, 11
- Apr 2007) | 5 lines If someone sets the "useragent" option in
- sip.conf to be empty, then don't add the User-Agent header at
- all. It is an optional header, anyway. Also, the bug report says
- that some of Japan's SIP providers don't allow it for some weird
- reason. (issue #9488, reported by makoto, fixed by me) ........
- ................
-
-2007-04-11 15:48 +0000 [r61460] Nadi Sarrar <ns@beronet.com>
-
- * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
- channels/misdn/isdn_lib.c: Merged revisions
- 61342,61372-61373,61443 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r61342 | nadi | 2007-04-11 12:52:28 +0200 (Mi, 11 Apr 2007) | 2
- lines AOCD's are now exported to asterisk channel variables.
- ........ r61372 | nadi | 2007-04-11 15:33:30 +0200 (Mi, 11 Apr
- 2007) | 2 lines Ignore facility messages in case we don't have a
- corresponding channel object. ........ r61373 | nadi | 2007-04-11
- 15:40:26 +0200 (Mi, 11 Apr 2007) | 2 lines Export AOCD variables
- on misdn_hangup. ........ r61443 | nadi | 2007-04-11 17:39:14
- +0200 (Mi, 11 Apr 2007) | 2 lines Don't export AOCD variables on
- misdn_hangup anymore, this was mainly a fix for trunk.. ........
-
-2007-04-11 15:25 +0000 [r61379-61429] Russell Bryant <russell@digium.com>
-
- * funcs/func_devstate.c: Add a minor loop optimization to the
- custom device state callback. Once the correct device is found,
- it should just break out of the loop ...
-
- * /, channels/chan_sip.c: Merged revisions 61427 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r61427 | russell | 2007-04-11 10:09:39 -0500
- (Wed, 11 Apr 2007) | 14 lines Merged revisions 61426 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r61426 | russell | 2007-04-11 10:05:36 -0500 (Wed, 11
- Apr 2007) | 6 lines Fix a bug with switching between host=dynamic
- and using specific hosts for peers. The code would only reset the
- peer's address when it is dynamic if it was a new peer structure.
- Now, it will also reset the address if it was already in the peer
- list, but before the reload, it was not dynamic. (issue #9515,
- reported by caio1982, fixed by me) ........ ................
-
- * /, main/http.c: Merged revisions 61407 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r61407 | russell | 2007-04-11 09:48:01 -0500 (Wed, 11 Apr 2007) |
- 4 lines Add "svgz" to the mimetypes table. (issue #9510, bkruse)
- In passing, constify the elements of the mimetypes table.
- ........
-
- * /, channels/chan_sip.c: Merged revisions 61377 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r61377 | russell | 2007-04-11 09:04:44 -0500
- (Wed, 11 Apr 2007) | 13 lines Merged revisions 61376 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r61376 | russell | 2007-04-11 09:02:54 -0500 (Wed, 11
- Apr 2007) | 5 lines Remove the attempt at reporting configuration
- errors in sip.conf. This can cause a bunch of improper messages
- when using realtime. I give up. As oej tried to convince me when
- I put this in, there is just no easy way to do it. (inspired by a
- message on the -dev list) ........ ................
-
-2007-04-11 14:09 +0000 [r61378] Steve Murphy <murf@digium.com>
-
- * apps/app_voicemail.c: via 8119, a patch to allow voicemail data
- to be stored in RealTime.
-
-2007-04-11 14:01 +0000 [r61375] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Remove duplicate prototype declaration.
- (issue #9517 reported by junky)
-
-2007-04-11 13:41 +0000 [r61374] Steve Murphy <murf@digium.com>
-
- * include/asterisk/config.h, main/config.c: via 8118, a RealTime
- upgrade to make RT a complete storage abstraction. The
- store/destroy mechanisms needed these missing peices.
-
-2007-04-10 23:55 +0000 [r61324] Tilghman Lesher <tlesher@digium.com>
-
- * main/channel.c, main/manager.c, configs/manager.conf.sample,
- include/asterisk/manager.h: Issue 6082 - New DTMF event for
- manager
-
-2007-04-10 22:02 +0000 [r61303] Doug Bailey <dbailey@digium.com>
-
- * channels/chan_zap.c: Added zapata.conf parameter "cid_rxgain" to
- allow the user to adjust the gain bump used during CID
- acquisition.
-
-2007-04-10 20:50 +0000 [r61222-61283] Russell Bryant <russell@digium.com>
-
- * CHANGES: Note the bridge manager action and application in the
- CHANGES file.
-
- * res/res_features.c: Merge changes from team/russell/issue_5841:
- This patch adds a "Bridge" Manager action, as well as a "Bridge"
- dialplan application. The manager action will allow you to steal
- two active channels in the system and bridge them together. Then,
- the one that did not hang up will continue in the dialplan. Using
- the application will bridge the calling channel to an arbitrary
- channel in the system. Whichever channel does not hang up here
- will continue in the dialplan, as well. This patch has been
- touched by a bunch of people over the course of a couple years.
- Please forgive me if I have missed your name in the history of
- things. The most recent patch came from issue #5841, but there is
- also a reference to an earlier version of this patch from issue
- #4297. The people involved in writing and/or reviewing the code
- include at least: twisted, mflorrel, heath1444, davetroy,
- tim_ringenbach, moy, tmancill, serge-v, and me. There are also
- positive test reports from many people.
-
- * main/dial.c, include/asterisk/dial.h: Add an option to the dial
- API for playing music instead of ringing to the caller. I started
- this for use with SLA but ended up deciding not to use it.
- However, there is no reason not to put this part in, anyway.
-
-2007-04-10 16:07 +0000 [r61221] Steve Murphy <murf@digium.com>
-
- * channels/chan_jingle.c: updated ast_channel_alloc() call to
- include the 4 extra args everyone got. Not much info there, as
- the config file evidently does not allow amaflags, or accountcode
- settings; and the pvt's exten doesn't sound like what we need in
- the cdr, either.
-
-2007-04-10 12:47 +0000 [r61184] Nadi Sarrar <ns@beronet.com>
-
- * /, channels/misdn_config.c: Merged revisions 61183 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r61183 | nadi | 2007-04-10 14:43:40 +0200 (Di,
- 10 Apr 2007) | 10 lines Merged revisions 61170 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61170 | nadi | 2007-04-10 14:31:45 +0200 (Di, 10 Apr 2007) | 2
- lines msns config parameter defaults to '*' ........
- ................
-
-2007-04-10 05:41 +0000 [r61152] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, channels/chan_local.c, channels/chan_vpb.cc,
- channels/chan_zap.c, /, channels/chan_sip.c, res/res_features.c,
- channels/chan_agent.c, include/asterisk/channel.h,
- channels/chan_gtalk.c, channels/chan_iax2.c, channels/chan_oss.c,
- main/channel.c, main/cdr.c, channels/chan_phone.c,
- channels/chan_misdn.c, channels/chan_skinny.c,
- channels/chan_features.c, channels/chan_h323.c,
- channels/chan_alsa.c, channels/chan_nbs.c, channels/chan_mgcp.c,
- apps/app_cdr.c, apps/app_voicemail.c: Merged revisions 60989 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r60989 | murf | 2007-04-09 12:32:07 -0600 (Mon, 09 Apr 2007) | 1
- line This is a big improvement over the current CDR fixes. It may
- still need refinement, but this won't have as many folks
- bothered. This also adds the mods from 1.4/r.61136; ........
-
-2007-04-09 22:49 +0000 [r61116] Russell Bryant <russell@digium.com>
-
- * apps/app_dial.c: Remove unused instances of unnamed enums.
-
-2007-04-09 20:01 +0000 [r61073] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 61072 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r61072 | oej | 2007-04-09 21:58:17 +0200 (Mon,
- 09 Apr 2007) | 11 lines Merged revisions 61038 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61038 | oej | 2007-04-09 21:38:59 +0200 (Mon, 09 Apr 2007) | 3
- lines - Don't send ActionID before Response: header. - Don't use
- a blank in an AMI header ........ ................
-
-2007-04-09 19:57 +0000 [r61065-61071] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/minimime/mm_envelope.c, /: Merged revisions 61070 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r61070 | kpfleming | 2007-04-09 14:55:14 -0500 (Mon, 09 Apr 2007)
- | 2 lines fix up some warnings found using --enable-dev-mode
- ........
-
- * /, main/minimime/tests/CVS (removed), main/minimime/Doxyfile
- (removed), main/minimime/tests/messages/CVS (removed): Merged
- revisions 61062 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r61062 | kpfleming | 2007-04-09 14:49:09 -0500 (Mon, 09 Apr 2007)
- | 2 lines remove some more stuff we don't need ........
-
-2007-04-09 19:06 +0000 [r61023] Jason Parker <jparker@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 61022 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r61022 | qwell | 2007-04-09 14:05:48 -0500 (Mon, 09 Apr 2007) | 4
- lines Use the appropriate interface name with COMPLETECALLER.
- Issue 9395. ........
-
-2007-04-09 19:05 +0000 [r60985-61021] Olle Johansson <oej@edvina.net>
-
- * main/manager.c: Add hint to ExtensionStatus AMI event in manager
-
- * channels/chan_sip.c, CHANGES, channels/chan_iax2.c: use
- "ChannelType" in events to indicate which channel driver that
- generates the event. This replaces "ChannelDriver" and "Channel",
- previously used to indicate channel driver. ChannelType is more
- in line with "core show channeltypes"
-
- * res/res_jabber.c: Fix JabberEvents
-
- * /, res/res_jabber.c: Fix missing newline in JabberEvent
-
-2007-04-09 17:23 +0000 [r60937] Jason Parker <jparker@digium.com>
-
- * /, apps/app_directory.c: Merged revisions 60936 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r60936 | qwell | 2007-04-09 12:22:59 -0500 (Mon,
- 09 Apr 2007) | 13 lines Merged revisions 60935 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r60935 | qwell | 2007-04-09 12:22:15 -0500 (Mon, 09 Apr 2007) | 5
- lines Allow matching on names shorter than 3 chars. This also
- fixes the case where somebody wants to match on less then 3
- chars. Issue 9071 ........ ................
-
-2007-04-09 16:30 +0000 [r60917] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * UPGRADE.txt: updated UPGRADE.txt to include format_wav changes
-
-2007-04-09 12:33 +0000 [r60898] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Make RTP session ID and session version
- generation random. (issue #9456 reported by tjardick)
-
-2007-04-09 03:04 +0000 [r60848-60851] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk.h, /, main/asterisk.c: Merged revisions 60850
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r60850 | tilghman | 2007-04-08 22:01:12 -0500
- (Sun, 08 Apr 2007) | 10 lines Merged revisions 60849 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r60849 | tilghman | 2007-04-08 21:49:06 -0500 (Sun, 08
- Apr 2007) | 2 lines Don't check for error when lowering priority
- (according to the manpage, it should never happen anyway). It
- might could happen, though, if another thread messed with the
- priority, so safeguard against that (reported via -dev list).
- ........ ................
-
- * channels/chan_local.c, /: Merged revisions 60847 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r60847 | tilghman | 2007-04-08 21:42:48 -0500
- (Sun, 08 Apr 2007) | 10 lines Merged revisions 60846 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r60846 | tilghman | 2007-04-08 21:37:18 -0500 (Sun, 08
- Apr 2007) | 2 lines Bug 9505 - If the return value for
- local_queue_frame is set, then p->lock is no longer valid.
- ........ ................
-
-2007-04-09 01:06 +0000 [r60763-60799] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 60798 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r60798 | file | 2007-04-08 21:03:14 -0400 (Sun,
- 08 Apr 2007) | 10 lines Merged revisions 60797 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r60797 | file | 2007-04-08 20:59:29 -0400 (Sun, 08 Apr 2007) | 2
- lines When calling a device that then forwards us elsewhere... we
- have to make our channels compatible if it is the only channel
- being dialed. (issue #9445 reported by marcelbarbulescu) ........
- ................
-
- * channels/chan_sip.c: Add counter for sip show registry CLI
- command. (issue #9352 reported by junky)
-
- * /, apps/app_queue.c: Merged revisions 60762 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r60762 | file | 2007-04-08 13:04:44 -0400 (Sun, 08 Apr 2007) | 2
- lines Allow app_queue to use MONITOR_EXEC even if MONITOR_OPTIONS
- is not set. (issue #9495 reported by cduffy) ........
-
-2007-04-08 14:23 +0000 [r60662-60715] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_macro.c: Merged revisions 60713 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r60713 | tilghman | 2007-04-08 09:14:29 -0500
- (Sun, 08 Apr 2007) | 10 lines Merged revisions 60711 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r60711 | tilghman | 2007-04-08 09:00:22 -0500 (Sun, 08
- Apr 2007) | 2 lines Gosub called within a Macro resets the
- arguments improperly and causes general weirdness. (Issue 8329)
- ........ ................
-
- * /, formats/format_wav.c, main/http.c: Merged revisions 60712 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r60712 | tilghman | 2007-04-08 09:12:00 -0500 (Sun, 08 Apr 2007)
- | 2 lines Fix --enable-dev-mode ........
-
- * /, main/file.c: Merged revisions 60661 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r60661 | tilghman | 2007-04-07 20:40:47 -0500
- (Sat, 07 Apr 2007) | 10 lines Merged revisions 60660 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r60660 | tilghman | 2007-04-07 20:39:25 -0500 (Sat, 07
- Apr 2007) | 2 lines Bug 9486 - memory leak when opening a
- filestream ........ ................
-
-2007-04-06 22:29 +0000 [r60641] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * formats/format_wav.c: removed GAIN preprocessor definition,
- removed needsgain from struct wav_desc, removed unnecessary gain
- code from wav_read() and wav_write()
-
-2007-04-06 21:43 +0000 [r60566-60623] Russell Bryant <russell@digium.com>
-
- * main/minimime/Makefile: Filter out -Wundef so that the
- automatically generated C files will compile cleanly
-
- * main/minimime/mytest_files (removed), main/minimime/sys/CVS
- (removed), main/minimime/.cvsignore (removed),
- main/minimime/mm-docs (removed), main/minimime/test (removed):
- Remove a bunch of files that weren't supposed to get added.
-
- * main/minimime/mm-docs/html/mm__envelope_8c.html,
- main/minimime/tests/messages, include/asterisk/autoconfig.h.in,
- main/minimime/mm-docs/html/mm__context_8c.html,
- main/minimime/sys, main/minimime/tests/Makefile,
- main/minimime/tests/CVS/Root, main/minimime/sys/CVS/Entries,
- main/minimime/mm-docs/latex/mm__mimeutil_8c.tex, configure,
- main/strcompat.c, main/http.c, main/minimime/mm_error.c,
- main/minimime/mm-docs/html/globals_func.html,
- main/minimime/mm-docs/html/group__mimeutil.html,
- main/minimime/mm-docs/latex/doxygen.sty,
- main/minimime/mm_param.c, main/minimime/test/CVS, configure.ac,
- main/minimime/.cvsignore, main/minimime/mm_init.c,
- main/minimime/mm-docs/html/mm__queue_8h-source.html,
- main/minimime/mm-docs/html/mm__error_8c.html,
- main/minimime/mm-docs/html/tabs.css, main/minimime/mm_envelope.c,
- main/minimime/mimeparser.h, main/minimime/mimeparser.l,
- main/minimime/mm_context.c,
- main/minimime/mm-docs/html/group__mimepart.html,
- main/minimime/mm-docs/latex/group__envelope.tex,
- main/minimime/tests/messages/CVS,
- main/minimime/mm-docs/html/mm__contenttype_8c.html,
- main/minimime/mm-docs/html/pages.html,
- main/minimime/mm-docs/html/group__error.html,
- main/minimime/mm-docs/latex/group__context.tex,
- main/minimime/mimeparser.y, Makefile.moddir_rules,
- main/minimime/sys/mm_queue.h,
- main/minimime/mm-docs/html/bug.html,
- main/minimime/mm-docs/html/mimeparser_8tab_8h-source.html,
- main/minimime/tests/messages/CVS/Root,
- main/minimime/mm_mimepart.c,
- main/minimime/mm-docs/latex/Makefile,
- main/minimime/mm_internal.h, main/minimime/tests/CVS,
- main/minimime/mm-docs/latex/mm__param_8c.tex,
- main/minimime/tests/parse.c, main/minimime/mm_base64.c,
- main/minimime/mm.h, main/minimime/mm_header.c,
- main/minimime/mm-docs/latex/mm__parse_8c.tex,
- main/minimime/mm-docs/html/mimeparser_8h-source.html,
- main/minimime/mm-docs/html/files.html,
- main/minimime/mm-docs/latex/mm__contenttype_8c.tex,
- main/minimime/mm-docs/html/mm__mem_8h-source.html,
- main/minimime/mm_codecs.c,
- main/minimime/mm-docs/latex/mm__mimepart_8c.tex,
- main/minimime/mytest_files/mytest.c,
- main/minimime/mm-docs/html/mm__mimeutil_8c.html,
- main/minimime/mm-docs/latex/files.tex,
- main/minimime/test/CVS/Entries,
- main/minimime/mm-docs/latex/modules.tex,
- main/minimime/tests/messages/CVS/Repository,
- configs/http.conf.sample, main/minimime/mm_contenttype.c,
- main/minimime/tests/messages/test1.txt,
- main/minimime/mm-docs/html/mm__param_8c.html,
- main/minimime/tests/messages/test3.txt,
- main/minimime/tests/messages/test5.txt,
- main/minimime/tests/messages/test7.txt,
- main/minimime/mm-docs/html/group__contenttype.html,
- main/minimime/mm-docs,
- main/minimime/mytest_files/ast_postdata3.gz, main/minimime
- (added), main/minimime/Make.conf,
- main/minimime/mm-docs/latex/group__contenttype.tex,
- main/minimime/mm_warnings.c, main/minimime/mm_queue.h,
- main/minimime/mm-docs/html/mm__util_8c.html,
- main/minimime/mm-docs/html/doxygen.css, /,
- main/minimime/mm-docs/html/mm__internal_8h.html,
- main/minimime/tests/messages/CVS/Entries, main/minimime/Doxyfile,
- main/minimime/minimime.c, main/minimime/mimeparser.yy.c,
- main/minimime/tests/CVS/Entries.Log, main/minimime/test.sh,
- include/asterisk/compat.h, main/minimime/test/CVS/Repository,
- main/minimime/mm_mimeutil.c, main/minimime/tests,
- main/minimime/mm-docs/latex/group__mimepart.tex,
- main/minimime/tests/CVS/Entries, main/Makefile,
- main/minimime/mm-docs/latex/mm__envelope_8c.tex,
- main/minimime/mm-docs/latex/mm__util_8c.tex,
- main/minimime/mm-docs/latex/pages.tex,
- main/minimime/mm-docs/latex/group__mimeutil.tex,
- main/minimime/mm-docs/latex,
- main/minimime/mm-docs/html/mm_8h-source.html,
- main/minimime/Makefile,
- main/minimime/mm-docs/latex/mm__internal_8h.tex,
- main/minimime/mm-docs/refman.pdf, include/asterisk/manager.h,
- main/minimime/mm-docs/latex/mm__context_8c.tex,
- main/minimime/mm-docs/latex/group__param.tex,
- main/minimime/mm-docs/latex/group__codecs.tex,
- main/minimime/tests/create.c, main/minimime/mm_util.c,
- main/minimime/mm-docs/latex/bug.tex,
- main/minimime/mimeparser.tab.c, main/minimime/mm_util.h,
- main/minimime/mytest_files/ast_postdata,
- main/minimime/mm-docs/html/group__envelope.html,
- main/minimime/mm-docs/html/group__util.html,
- main/minimime/mimeparser.tab.h,
- main/minimime/mm-docs/html/mm__parse_8c.html,
- main/minimime/mm-docs/html,
- main/minimime/mm-docs/latex/group__util.tex,
- main/minimime/mm-docs/html/group__context.html,
- main/minimime/mm-docs/html/mm__internal_8h-source.html,
- main/minimime/mytest_files,
- main/minimime/mm-docs/html/mm__util_8h-source.html,
- main/minimime/sys/CVS,
- main/minimime/mm-docs/html/group__codecs.html, main/manager.c,
- main/minimime/sys/CVS/Repository,
- main/minimime/mm-docs/html/globals.html,
- main/minimime/mm-docs/html/mm__mimepart_8c.html,
- main/minimime/tests/CVS/Repository,
- main/minimime/mm-docs/html/index.html,
- main/minimime/mm-docs/html/modules.html, main/minimime/test,
- main/minimime/mytest_files/ast_postdata2,
- main/minimime/mm-docs/latex/group__error.tex,
- main/minimime/mm-docs/html/mm__header_8c.html,
- main/minimime/strlcpy.c,
- main/minimime/mm-docs/html/group__param.html,
- main/minimime/mm-docs/latex/refman.tex, main/minimime/mm_parse.c,
- main/minimime/mm-docs/latex/mm__header_8c.tex,
- main/minimime/mm-docs/latex/mm__error_8c.tex,
- main/minimime/mm_mem.c,
- main/minimime/mm-docs/html/mm__codecs_8c.html,
- main/minimime/tests/messages/test2.txt,
- main/minimime/tests/messages/test4.txt,
- main/minimime/sys/CVS/Root,
- main/minimime/tests/messages/test6.txt,
- main/minimime/test/CVS/Root, main/minimime/strlcat.c,
- main/minimime/mm_mem.h,
- main/minimime/mm-docs/latex/mm__codecs_8c.tex: Merged revisions
- 60603 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r60603 | russell | 2007-04-06 15:58:43 -0500 (Fri, 06 Apr 2007) |
- 13 lines To be able to achieve the things that we would like to
- achieve with the Asterisk GUI project, we need a fully functional
- HTTP interface with access to the Asterisk manager interface. One
- of the things that was intended to be a part of this system, but
- was never actually implemented, was the ability for the GUI to be
- able to upload files to Asterisk. So, this commit adds this in
- the most minimally invasive way that we could come up with. A lot
- of work on minimime was done by Steve Murphy. He fixed a lot of
- bugs in the parser, and updated it to be thread-safe. The ability
- to check permissions of active manager sessions was added by
- Dwayne Hubbard. Then, hacking this all together and do doing the
- modifications necessary to the HTTP interface was done by me.
- ........
-
- * /, apps/app_meetme.c: Merged revisions 60565 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r60565 | russell | 2007-04-06 14:50:52 -0500 (Fri, 06 Apr 2007) |
- 3 lines When a station picks up a trunk that was on hold, make
- the hints reflect that nobody has the trunk on hold anymore.
- ........
-
-2007-04-06 19:26 +0000 [r60531] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Use the same parameter to the two "Registry"
- AMI events - ChannelDriver
-
-2007-04-06 18:59 +0000 [r60522] Russell Bryant <russell@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 60521 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r60521 | russell | 2007-04-06 13:58:46 -0500 (Fri, 06 Apr 2007) |
- 16 lines Fix a few problems with SLA. (issue #9459, reported by
- francesco_r, fixed by me) * The original behavior was that if one
- station put a call on hold, another one picked it up, and then
- hung up, the code would still consider the call on hold by the
- first station, so the trunk would not be hung up. However, to
- better comply with what most people seem to expect it to behave,
- it will now hang up the trunk. * Fix a problem with "barge=no".
- This was only intended to prevent people from joining calls that
- are in progress. However, it also prevented other people from
- picking up a call that was on hold. This has been fixed. * When
- there are no active stations on a trunk and it is on hold, the
- code now indicates the HOLD and UNHOLD conditions to the trunk
- channel. This allows music on hold to be played to the trunk when
- it is on hold. ........
-
-2007-04-06 18:26 +0000 [r60486-60487] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 60485 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r60485 | mattf | 2007-04-06 13:21:52 -0500 (Fri, 06 Apr 2007) | 2
- lines Make sure we check the faxdetect option before doing fax
- processing ........
-
- * channels/chan_zap.c, /: Merged revisions 60459 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r60459 | mattf | 2007-04-06 12:32:31 -0500 (Fri,
- 06 Apr 2007) | 10 lines Merged revisions 60456 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r60456 | mattf | 2007-04-06 12:03:15 -0500 (Fri, 06 Apr 2007) | 2
- lines There should only be one code path for doing DTMF
- conditionals on channels. This fixes it. ........
- ................
-
-2007-04-06 14:53 +0000 [r60400] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, codecs/codec_zap.c: Merged revisions 60399 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r60399 | kpfleming | 2007-04-06 09:49:51 -0500
- (Fri, 06 Apr 2007) | 10 lines Merged revisions 60398 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r60398 | kpfleming | 2007-04-06 09:41:37 -0500 (Fri, 06
- Apr 2007) | 2 lines remove undocumented 'cardsmode' parameter and
- stop searching for transcoders during reload() ........
- ................
-
-2007-04-06 01:29 +0000 [r60362-60363] Joshua Colp <jcolp@digium.com>
-
- * include/asterisk/speech.h, res/res_speech.c: Major res_speech
- cleanup. It looks much better now!
-
- * /, include/asterisk/speech.h, res/res_speech.c,
- apps/app_speech_utils.c: Merged revisions 60361 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r60361 | file | 2007-04-05 22:14:00 -0300 (Thu, 05 Apr 2007) | 2
- lines Add support for returning different types of results (ie:
- NBest). ........
-
-2007-04-05 23:08 +0000 [r60326] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * /, formats/format_wav.c: Merged revisions 60325 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r60325 | dhubbard | 2007-04-05 17:58:01 -0500 (Thu, 05 Apr 2007)
- | 1 line modified default GAIN for issue 5823, thanks jrwalliker
- ........
-
-2007-04-05 22:40 +0000 [r60324] Steve Murphy <murf@digium.com>
-
- * configs/cdr_custom.conf.sample, /, configs/cdr.conf.sample:
- Merged revisions 60323 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r60323 | murf | 2007-04-05 16:35:11 -0600 (Thu, 05 Apr 2007) | 1
- line Added some clarification to the example configs for CDRs, on
- how to select a backend. Also, made cdr-csv the default if you
- 'make samples', and no other changes. ........
-
-2007-04-05 16:11 +0000 [r60269] Jason Parker <jparker@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 60268 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r60268 | qwell | 2007-04-05 11:10:48 -0500 (Thu,
- 05 Apr 2007) | 13 lines Merged revisions 60267 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r60267 | qwell | 2007-04-05 11:09:41 -0500 (Thu, 05 Apr 2007) | 5
- lines Just because we can't find the voicemail configuration
- file, doesn't mean that the module failed to load. The user could
- be using realtime. Issue #9473 ........ ................
-
-2007-04-05 15:48 +0000 [r60266] Russell Bryant <russell@digium.com>
-
- * /, main/http.c: Merged revisions 60265 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r60265 | russell | 2007-04-05 10:47:17 -0500 (Thu, 05 Apr 2007) |
- 2 lines Add the MIME type for gif by request from Pari ........
-
-2007-04-05 12:57 +0000 [r60215] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 60214 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r60214 | file | 2007-04-05 08:55:02 -0400 (Thu,
- 05 Apr 2007) | 10 lines Merged revisions 60213 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r60213 | file | 2007-04-05 08:52:50 -0400 (Thu, 05 Apr 2007) | 2
- lines Only unlock our pvt and net locks if we are actually going
- to try to lock the owner again. (issue #9472 reported by zoa)
- ........ ................
-
-2007-04-04 23:45 +0000 [r60193] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * main/callerid.c: ast_shrink_phone_number() must ignore
- whitespace, otherwise my CIDCO callerid box gets LINE ERROR
-
-2007-04-04 17:41 +0000 [r60011-60141] Russell Bryant <russell@digium.com>
-
- * main/manager.c, /: Merged revisions 60137 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r60137 | russell | 2007-04-04 12:40:10 -0500
- (Wed, 04 Apr 2007) | 14 lines Merged revisions 60134 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r60134 | russell | 2007-04-04 12:38:47 -0500 (Wed, 04
- Apr 2007) | 6 lines It is valid to redirect channels via the
- manager interface that are not in the UP state. Instead of
- checking for that to prevent to ensure a dead channel doesn't get
- redirected, just use the ast_check_hangup() API call. (issue
- #9457, reported by Callmewind, patch by me) (related to issue
- #8977) ........ ................
-
- * /, channels/chan_sip.c: Merged revisions 60112 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r60112 | russell | 2007-04-04 11:49:45 -0500 (Wed, 04 Apr 2007) |
- 3 lines Add a Content-Length of 0 to the response built by
- transmit_response_with_unsupported(). (issue #9454, reported by
- makoto, fixed by me) ........
-
- * /, channels/chan_sip.c: Merged revisions 60088 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r60088 | russell | 2007-04-04 11:39:04 -0500
- (Wed, 04 Apr 2007) | 12 lines Merged revisions 60083 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r60083 | russell | 2007-04-04 11:37:04 -0500 (Wed, 04
- Apr 2007) | 4 lines Fix the return value of
- handle_common_options() so that it always properly indicates
- whether it handled the option or not. (issue #9455, reported by
- Netview, fixed by me) ........ ................
-
- * /, apps/app_meetme.c: Merged revisions 60069 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r60069 | russell | 2007-04-04 11:26:23 -0500 (Wed, 04 Apr 2007) |
- 4 lines Fix a problem where if a trunk was hung up while it was
- on hold, all of the hints would reflect the line still on hold,
- even though it should reflect that it is back to not in use.
- (issue #9459, reported by francesco_r, fixed by me) ........
-
- * channels/chan_jingle.c, channels/chan_gtalk.c,
- doc/rtp-packetization.txt: Add support for RTP packetization in
- chan_jingle and chan_gtalk. (issue #9416, phsultan)
-
-2007-04-03 19:43 +0000 [r59969] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_speech_utils.c: Merged revisions 59963 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r59963 | file | 2007-04-03 15:40:59 -0400 (Tue, 03 Apr
- 2007) | 2 lines Don't clash when a person both speaks and uses
- DTMF. ........
-
-2007-04-03 19:17 +0000 [r59854-59940] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 59939 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r59939 | russell | 2007-04-03 14:16:53 -0500
- (Tue, 03 Apr 2007) | 12 lines Merged revisions 59938 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r59938 | russell | 2007-04-03 14:15:04 -0500 (Tue, 03
- Apr 2007) | 4 lines Don't attempt to report configuration errors
- in build_user(). oej pointed out that for a "friend" entry, this
- won't work, because all user options are valid for peers, but not
- the other way around. ........ ................
-
- * /, channels/chan_sip.c: Merged revisions 59936 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r59936 | russell | 2007-04-03 13:55:57 -0500
- (Tue, 03 Apr 2007) | 11 lines Merged revisions 59916 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r59916 | russell | 2007-04-03 13:43:54 -0500 (Tue, 03
- Apr 2007) | 3 lines Make chan_sip report when it encounters an
- unknown option. (issue #9440, reported by nightcrawler) ........
- ................
-
- * channels/chan_sip.c: Remove a duplicate function prototype.
- (issue #9444, junky)
-
- * /, main/app.c: Merged revisions 59887 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r59887 | russell | 2007-04-03 13:01:49 -0500
- (Tue, 03 Apr 2007) | 13 lines Merged revisions 59886 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r59886 | russell | 2007-04-03 12:58:19 -0500 (Tue, 03
- Apr 2007) | 5 lines When doing a built-in blind or attended
- transfer, restore the ability to use '#' to terminate the number
- and immediately do the transfer instead of having to dial the
- number and just wait for the feature digit timeout. (issue #8366,
- xueliangliang) ........ ................
-
- * Makefile, /: Merged revisions 59853 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r59853 | russell | 2007-04-03 11:03:35 -0500 (Tue, 03 Apr 2007) |
- 1 line Ensure that menuselect gets executed in dependency check
- mode every time you run make. ........
-
-2007-04-03 11:15 +0000 [r59805] Nadi Sarrar <ns@beronet.com>
-
- * /, channels/misdn/chan_misdn_config.h, channels/misdn_config.c:
- Merged revisions 59804 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r59804 | nadi | 2007-04-03 13:02:46 +0200 (Di,
- 03 Apr 2007) | 15 lines Merged revisions 59788,59803 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r59788 | nadi | 2007-04-03 11:37:00 +0200 (Di, 03 Apr
- 2007) | 2 lines Use the new sysfs way of mISDN 1.2 to check if a
- port is NT or not. ........ r59803 | nadi | 2007-04-03 12:40:58
- +0200 (Di, 03 Apr 2007) | 2 lines ptp is the 5th bit, not the
- 4th. ........ ................
-
-2007-04-02 19:01 +0000 [r59725] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 59724 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r59724 | file | 2007-04-02 14:58:24 -0400 (Mon,
- 02 Apr 2007) | 10 lines Merged revisions 59723 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59723 | file | 2007-04-02 14:55:25 -0400 (Mon, 02 Apr 2007) | 2
- lines Increase the maximum size for a string of mailboxes to
- 1024. (issue #9270 reported by rtucker) ........ ................
-
-2007-04-02 17:40 +0000 [r59693] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: This hashing code is still causing some
- random crashes on my system, and probably others, too. I don't
- really have time to work on it at the moment, so I am just going
- to revert it for now.
-
-2007-04-02 17:38 +0000 [r59692] Steve Murphy <murf@digium.com>
-
- * /, pbx/pbx_ael.c: Merged revisions 59688 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r59688 | murf | 2007-04-02 11:31:32 -0600 (Mon, 02 Apr 2007) | 1
- line continue in for-loop should go to the incrementer, not the
- test. As per 9435, thanks to marcelbarbulescu ........
-
-2007-04-02 16:08 +0000 [r59655] Russell Bryant <russell@digium.com>
-
- * /, main/netsock.c: Merged revisions 59654 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r59654 | russell | 2007-04-02 10:39:07 -0500
- (Mon, 02 Apr 2007) | 14 lines Merged revisions 59608 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r59608 | russell | 2007-04-01 17:35:25 -0500 (Sun, 01
- Apr 2007) | 6 lines Add the SO_REUSEADDR flag to sockets handled
- by netsock. This is needed by the patch that went in for issue
- 7874. chan_iax2 needs to be able to create socket that is
- lisetning on INADDR_ANY, but also be able to bind sockets to
- specific addresses. (Thanks to Stevenson on the asterisk-dev
- mailing list for explaining why this flag was needed.) ........
- ................
-
-2007-03-30 22:54 +0000 [r59574] Jason Parker <jparker@digium.com>
-
- * /, configure, main/Makefile, acinclude.m4: Merged revisions 59573
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r59573 | qwell | 2007-03-30 17:50:31 -0500 (Fri, 30 Mar 2007) | 2
- lines Add linux-uclibc host arch..."thingy". Sorry, I don't know
- what it's called... ........
-
-2007-03-30 20:54 +0000 [r59555] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Update to support multiple CIC groups and
- DPCs per linkset.
-
-2007-03-30 17:57 +0000 [r59453-59523] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, main/channel.c, main/pbx.c, /, res/res_features.c,
- include/asterisk/cdr.h: Merged revisions 59522 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r59522 | murf | 2007-03-30 11:51:17 -0600 (Fri, 30 Mar 2007) | 1
- line several changes via kpflemings review ........
-
- * main/cdr.c, main/channel.c, main/pbx.c, /, res/res_features.c,
- include/asterisk/cdr.h: Merged revisions 59486 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r59486 | murf | 2007-03-30 08:11:59 -0600 (Fri, 30 Mar 2007) | 1
- line These mods fix CDR issues from 8221, 8593, 8680, 8743, and
- perhaps others. Mainly with CDRs generated from transfer
- situations. ........
-
- * /, configs/extensions.conf.sample: Merged revisions 59452 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r59452 | murf | 2007-03-29 18:56:36 -0600 (Thu, 29 Mar 2007) | 1
- line A small clarification to keep bugs from being filed, and
- confusion from rising, if clearglobalvars is set, and globals are
- set in the AEL file. (9419) ........
-
-2007-03-29 23:27 +0000 [r59364-59433] Russell Bryant <russell@digium.com>
-
- * apps/app_voicemail.c: Reduce the ridiculous number of variables
- used in the load_config() function by just having one that can be
- re-used. There is no functional change here (that is intentional,
- anyway!).
-
- * CHANGES, apps/app_voicemail.c: Add the ability for the "voicemail
- show users" CLI command to show users configured in realtime.
-
- * channels/chan_iax2.c: Fix an issue with hashing iax2 pvt
- structures that caused random crashes on systems under heavy load
- such as IAXtel. (possibly related to issue #9403)
-
- * /, res/res_jabber.c: Merged revisions 59363 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r59363 | russell | 2007-03-29 12:43:52 -0500 (Thu, 29 Mar 2007) |
- 6 lines When building a response to a subscription, the "from"
- must be the full Jabber ID. This fixes some problems where jabber
- users are not able to add their Asterisk account to their user
- list, since they are unable to get Asterisk to approve their
- subscription. (issue #8210, reported by caspy, and verified by
- bradtem) ........
-
-2007-03-29 17:42 +0000 [r59362] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 59361 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r59361 | file | 2007-03-29 13:38:55 -0400 (Thu,
- 29 Mar 2007) | 10 lines Merged revisions 59360 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59360 | file | 2007-03-29 13:33:58 -0400 (Thu, 29 Mar 2007) | 2
- lines Keep a global array of variables indicating whether certain
- conference rooms are in use. This ensures that two people going
- into a new dynamic conference when the 'e' option is set don't go
- into the same conference room. (issue #8835 reported by eliel)
- ........ ................
-
-2007-03-29 17:20 +0000 [r59305-59359] Russell Bryant <russell@digium.com>
-
- * /, main/rtp.c: Merged revisions 59358 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r59358 | russell | 2007-03-29 12:17:41 -0500
- (Thu, 29 Mar 2007) | 13 lines Merged revisions 59357 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r59357 | russell | 2007-03-29 12:14:33 -0500 (Thu, 29
- Mar 2007) | 5 lines If an error occurs when reading from an RTP
- socket, and the error code does not indicate that we should try
- again, then return NULL instead of a "null frame". This will
- prevent Asterisk from trying over and over again, and eventually
- causing the system to crash. (issue #8285, john) ........
- ................
-
- * /, channels/chan_iax2.c: Merged revisions 59341 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r59341 | russell | 2007-03-29 11:55:39 -0500 (Thu, 29 Mar 2007) |
- 8 lines When the IAX2 read callback gets called, return NULL
- instead of a "null frame". This will cause Asterisk to hangup the
- call instead of keep trying whatever it was doing. Under normal
- conditions, this function would *never* be called. However, the
- author of this patch says an error will occur that will cause it
- to get called every 100 thousand calls or so. When this does
- happen, it puts the channel in a loop that eventually brings down
- the system. So, hangup up the call is certainly a better
- alternative. (issue #8286, john) ........
-
- * Makefile, /: Merged revisions 59304 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r59304 | russell | 2007-03-29 11:25:41 -0500 (Thu, 29 Mar 2007) |
- 2 lines Export the GTK2 library and include information to sub
- Makefiles. ........
-
-2007-03-29 16:08 +0000 [r59303] Tilghman Lesher <tlesher@digium.com>
-
- * /, cdr/cdr_odbc.c: Merged revisions 59302 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r59302 | tilghman | 2007-03-29 11:07:05 -0500
- (Thu, 29 Mar 2007) | 11 lines Merged revisions 59301 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r59301 | tilghman | 2007-03-29 11:04:46 -0500 (Thu, 29
- Mar 2007) | 3 lines Issue 9415 - No point to getting a diagnostic
- field if we aren't doing anything with the information. (Plus, it
- tends to crash the Postgres ODBC driver.) ........
- ................
-
-2007-03-28 03:40 +0000 [r59290] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_odbc.c: Merged revisions 59289 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r59289 | tilghman | 2007-03-27 22:38:09 -0500 (Tue, 27 Mar 2007)
- | 2 lines Another crash that I thought we had fixed already -
- Issue 9396 ........
-
-2007-03-28 00:09 +0000 [r59286] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * channels/chan_zap.c: added filtering options to 'zap show
- channels' to implement functionality described in issue 6520
-
-2007-03-27 23:38 +0000 [r59282-59285] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 59284 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r59284 | tilghman | 2007-03-27 18:37:31 -0500
- (Tue, 27 Mar 2007) | 10 lines Merged revisions 59283 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r59283 | tilghman | 2007-03-27 18:36:49 -0500 (Tue, 27
- Mar 2007) | 2 lines Oops ........ ................
-
- * /, apps/app_voicemail.c: Merged revisions 59281 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r59281 | tilghman | 2007-03-27 18:32:46 -0500
- (Tue, 27 Mar 2007) | 10 lines Merged revisions 59280 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r59280 | tilghman | 2007-03-27 18:31:20 -0500 (Tue, 27
- Mar 2007) | 2 lines Fix a few remaining bad mmap(2) return values
- ........ ................
-
-2007-03-27 23:22 +0000 [r59274-59279] Russell Bryant <russell@digium.com>
-
- * /, apps/app_directory.c: Merged revisions 59278 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r59278 | russell | 2007-03-27 18:20:22 -0500
- (Tue, 27 Mar 2007) | 11 lines Merged revisions 59277 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r59277 | russell | 2007-03-27 18:19:41 -0500 (Tue, 27
- Mar 2007) | 3 lines Fix the check of the return value from
- mmap(). Thanks to Corydon for catching this one. ........
- ................
-
- * /, apps/app_directory.c: Merged revisions 59275 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r59275 | russell | 2007-03-27 18:16:27 -0500 (Tue, 27 Mar 2007) |
- 3 lines Fix app_directory to actually compile with ODBC_STORAGE,
- and update the code to the latest res_odbc API. ........
-
- * /, apps/Makefile: Merged revisions 59273 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r59273 | russell | 2007-03-27 18:02:12 -0500 (Tue, 27 Mar 2007) |
- 4 lines Fix app_directory when ODBC_STORAGE is being used. The
- Makefile did not properly ensure that this information got copied
- from what was selected for app_voicemail. (issue #9224) ........
-
-2007-03-27 20:11 +0000 [r59272] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_zap.c: Use better english. Renegotiate! Repeat
- after me: renegotiate.
-
-2007-03-27 18:21 +0000 [r59264] Steve Murphy <murf@digium.com>
-
- * /, pbx/pbx_ael.c: Merged revisions 59261 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r59261 | murf | 2007-03-27 12:16:32 -0600 (Tue, 27 Mar 2007) | 1
- line via 9373 (duplicate context in AEL crashes asterisk),
- kpfleming pointed on asterisk-dev, that DECLINE in this case the
- proper thing to do. This change now has it doing the proper
- thing. ........
-
-2007-03-27 18:18 +0000 [r59257-59263] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 59262 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r59262 | russell | 2007-03-27 13:17:47 -0500 (Tue, 27 Mar 2007) |
- 3 lines Fix the check that ensures that the CHANNEL function's
- first argument is "rtpqos". Thanks, Corydon. :) ........
-
- * /, channels/chan_iax2.c: Merged revisions 59259 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r59259 | russell | 2007-03-27 13:05:46 -0500
- (Tue, 27 Mar 2007) | 12 lines Merged revisions 59258 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r59258 | russell | 2007-03-27 13:04:02 -0500 (Tue, 27
- Mar 2007) | 4 lines Fix the use of the "sourceaddress" option
- when "bindaddr" is set to 0.0.0.0 instead of having each
- interface explicitly listed. (issue #7874, patch by stevens)
- ........ ................
-
- * /, channels/chan_sip.c, funcs/func_channel.c: Merged revisions
- 59256 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r59256 | russell | 2007-03-27 11:20:53 -0500 (Tue, 27 Mar 2007) |
- 4 lines Convert the RTPQOS function to just be additional
- parameter of the CHANNEL function. This way, it will be possible
- for other RTP based channel drivers to expose this information in
- the future. ........
-
-2007-03-27 14:09 +0000 [r59233-59253] Steve Murphy <murf@digium.com>
-
- * include/asterisk/config.h: Enhancement via 8118: Realtime API
- extension: add methods store_func and destroy_func, to make
- Realtime a complete database abstraction
-
- * pbx/ael/ael-test/ael-test18/extensions.ael (added),
- pbx/ael/ael-test/ael-test18 (added),
- pbx/ael/ael-test/ref.ael-test18 (added): added the no. 18
- regression test
-
- * pbx/ael/ael-test/ael-test19/extensions.ael (added),
- pbx/ael/ael-test/ref.ael-vtest17, pbx/ael/ael-test/ael-test19
- (added), pbx/ael/ael-test/ref.ael-test7,
- pbx/ael/ael-test/ref.ael-test19 (added),
- pbx/ael/ael-test/ref.ael-vtest13: updated the regressions with
- regards to 9373, the crash on double contexts, and brought other
- regressions up to date
-
- * /, pbx/pbx_ael.c: Merged revisions 59228 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r59228 | murf | 2007-03-26 15:41:32 -0600 (Mon, 26 Mar 2007) | 1
- line fix for 9373 (duplicate context in AEL crashes asterisk). I
- turned a duplicate context from a WARNING to an ERROR. Now you
- get a module load failure, and asterisk just exits. That's better
- than a crash, right\? ........
-
-2007-03-26 21:46 +0000 [r59229-59231] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 59227 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r59227 | tilghman | 2007-03-26 16:37:41 -0500 (Mon, 26 Mar 2007)
- | 2 lines Change this to a single dp function to make oej happy.
- ........
-
-2007-03-26 20:27 +0000 [r59226] Steve Murphy <murf@digium.com>
-
- * /, main/config.c: Merged revisions 59225 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r59225 | murf | 2007-03-26 14:06:12 -0600 (Mon, 26 Mar 2007) | 1
- line Fix for 9257; by eliminating the globals in main/config.c,
- we make it thread-safe, which is a minimum requirement. ........
-
-2007-03-26 19:35 +0000 [r59224] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_speech_utils.c: Merged revisions 59223 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r59223 | file | 2007-03-26 16:34:14 -0300 (Mon, 26 Mar
- 2007) | 2 lines Add ability to specify no timeout. This means as
- soon as the prompt is done playing it moves on to the next
- priority. ........
-
-2007-03-26 18:34 +0000 [r59216-59218] Russell Bryant <russell@digium.com>
-
- * /: Merged revisions 59217 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r59217 | russell | 2007-03-26 13:33:50 -0500 (Mon, 26 Mar 2007) |
- 4 lines Somehow the code for building the email for voicemail got
- out of sync. This change makes a few tweaks to get 1.4 in sync
- with trunk. (issue #9301) ........
-
- * /, apps/app_meetme.c: Merged revisions 59215 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r59215 | russell | 2007-03-26 13:28:29 -0500 (Mon, 26 Mar 2007) |
- 3 lines Fix some codec negotiation problems when CallerID support
- is not enabled in SLA. (issue #9308, reported by twilson)
- ........
-
-2007-03-26 18:14 +0000 [r59214] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_speech_utils.c: Merged revisions 59213 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r59213 | file | 2007-03-26 14:13:06 -0400 (Mon, 26 Mar
- 2007) | 2 lines Make SpeechBackground obey the digit timeout
- value. ........
-
-2007-03-26 17:57 +0000 [r59211] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Merged revisions 59209 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r59209 | russell | 2007-03-26 12:53:07 -0500 (Mon, 26 Mar 2007) |
- 1 line Rename the new dialplan functions to match the variable
- name ........
-
-2007-03-26 17:56 +0000 [r59210] Steve Murphy <murf@digium.com>
-
- * /, main/ast_expr2f.c, pbx/ael/ael.flex, main/ast_expr2.fl: Merged
- revisions 59206 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r59206 | murf | 2007-03-26 11:38:29 -0600 (Mon, 26 Mar 2007) | 1
- line A fix for the flex input files, DONT_COMPILE, and
- STANDALONE_AEL ........
-
-2007-03-26 17:51 +0000 [r59208] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c, include/asterisk/rtp.h, main/rtp.c:
- Merged revisions 59207 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r59207 | russell | 2007-03-26 12:45:55 -0500 (Mon, 26 Mar 2007) |
- 7 lines The AUDIORTPQOS and VIDEORTPQOS variables are not fully
- functional in some because they get set in sip_hangup. So, there
- are common situations where the variables will not be available
- in the dialplan at all. So, this patch provides an alternate
- method for getting to this information by introducing AUDIORTPQOS
- and VIDEORTPQOS dialplan functions. (issue #9370, patch by
- Corydon76, with some testing by blitzrage) ........
-
-2007-03-26 16:48 +0000 [r59204-59205] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Fix bug in which parameter type we are
- passing. This shouldn't be a problem since both types are the
- same underneath.
-
- * channels/chan_zap.c: Small API related SS7 updates.
-
-2007-03-26 15:59 +0000 [r59203] Nadi Sarrar <ns@beronet.com>
-
- * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, configure,
- include/asterisk/autoconfig.h.in, channels/misdn/Makefile,
- channels/misdn/chan_misdn_config.h, configure.ac,
- channels/misdn/isdn_lib.c, channels/misdn_config.c: Merged
- revisions 59202 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r59202 | nadi | 2007-03-26 17:25:53 +0200 (Mo, 26 Mär 2007) | 4
- lines * mISDN >= 1.2 provides a dsp pipeline for i.e. echo
- cancellation modules, make chan_misdn use it. * add a check for
- linux/mISDNdsp.h to configure.ac and update the autogenerated
- files: 'configure', 'autoconfig.h.in' (the 'configure' script was
- not in sync with the latest configure.ac, so the diff is a bit
- bigger than expected). ........
-
-2007-03-26 15:20 +0000 [r59201] Joshua Colp <jcolp@digium.com>
-
- * /, pbx/ael/ael_lex.c: Merged revisions 59200 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r59200 | file | 2007-03-26 11:16:29 -0400 (Mon, 26 Mar 2007) | 2
- lines Have ast_copy_string magically appear in the aelparse
- binary! DONT_OPTIMIZE should now work once again. ........
-
-2007-03-24 01:42 +0000 [r59191-59196] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 59195 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r59195 | file | 2007-03-23 21:39:44 -0400 (Fri,
- 23 Mar 2007) | 10 lines Merged revisions 59194 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59194 | file | 2007-03-23 21:35:49 -0400 (Fri, 23 Mar 2007) | 2
- lines Only try to handle a response if it has a response code.
- (ASA-2007-011) ........ ................
-
- * doc/modules.txt: Update modules.txt to new loader. (issue #9358
- reported by eliel)
-
-2007-03-23 16:17 +0000 [r59190] Steve Murphy <murf@digium.com>
-
- * /, apps/app_macro.c: Merged revisions 59188 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r59188 | murf | 2007-03-23 10:09:01 -0600 (Fri,
- 23 Mar 2007) | 9 lines Merged revisions 59186 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59186 | murf | 2007-03-23 09:57:26 -0600 (Fri, 23 Mar 2007) | 1
- line Added a few words in the Macro doc strings about the
- behavior of macros with hangups (et al.), as per 9337 ........
- ................
-
-2007-03-22 23:41 +0000 [r59181-59183] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 59182 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r59182 | kpfleming | 2007-03-22 16:40:01 -0700 (Thu, 22 Mar 2007)
- | 2 lines don't allow string input to overrun the buffer to hold
- it (ASA-2007-010) ........
-
- * channels/chan_misdn.c, /: Merged revisions 59180 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r59180 | kpfleming | 2007-03-22 16:34:22 -0700 (Thu, 22
- Mar 2007) | 2 lines remove variables that are no longer used
- (--enable-dev-mode is good, developers should be using it)
- ........
-
-2007-03-22 14:48 +0000 [r59146] Steve Murphy <murf@digium.com>
-
- * utils/Makefile, /: Merged revisions 59145 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r59145 | murf | 2007-03-22 08:40:53 -0600 (Thu, 22 Mar 2007) | 1
- line The stuff in utils was compiling with -O6 even if
- DONT_OPTIMIZE is set in menuconfig. Added the include to fix that
- ........
-
-2007-03-21 18:10 +0000 [r59080-59090] Joshua Colp <jcolp@digium.com>
-
- * /, main/http.c: Merged revisions 59089 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r59089 | file | 2007-03-21 14:08:57 -0400 (Wed, 21 Mar 2007) | 2
- lines Add svg mimetype for pari. ........
-
- * /, res/res_monitor.c: Merged revisions 59087 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r59087 | file | 2007-03-21 14:04:58 -0400 (Wed,
- 21 Mar 2007) | 10 lines Merged revisions 59086 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59086 | file | 2007-03-21 14:03:20 -0400 (Wed, 21 Mar 2007) | 2
- lines Indicate the filename changed when it is changed. (issue
- #9311 reported by jsmith) ........ ................
-
- * channels/chan_sip.c: Minor tweak. Only queue up an unhold control
- frame if we are actually on hold. This would have shown itself
- when a call was initially being setup and the SDP data was being
- parsed in.
-
- * /, channels/chan_sip.c: Merged revisions 59081 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r59081 | file | 2007-03-20 23:25:48 -0400 (Tue, 20 Mar 2007) | 2
- lines Until we can do media level parsing for sendrecv/etc just
- use the first value found. This crept up when a phone was offered
- audio+video and returned an inactive video stream. chan_sip
- thought the phone said to put the person on hold but that was
- totally wrong. (issue #9319 reported by benbrown) ........
-
- * main/db.c: Make the database show command spit out how many
- results it got. (issue #9332 reported by junky)
-
-2007-03-20 21:06 +0000 [r59079] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/logger.c: Merged revisions 59078 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r59078 | tilghman | 2007-03-20 16:04:52 -0500 (Tue, 20 Mar 2007)
- | 2 lines Fix defines for inline stack backtraces (only used by
- developers anyway) ........
-
-2007-03-20 20:44 +0000 [r59077] Joshua Colp <jcolp@digium.com>
-
- * /, channels/iax2-parser.c: Merged revisions 59076 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r59076 | file | 2007-03-20 16:42:46 -0400 (Tue, 20 Mar
- 2007) | 2 lines Copy len variable as well, should fix remaining
- IAX2 DTMF issues. ........
-
-2007-03-20 18:18 +0000 [r59071-59073] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_ael.c, include/asterisk/ael_structs.h: The fix for the
- AEL <<security hole>> (bug 9316) is here...
-
- * /: blocking 59070... it was just a repair, doesn't need to be
- here
-
- * /: blocking 59069... will commit these changes with separate
- patch
-
-2007-03-19 22:32 +0000 [r59051] Joshua Colp <jcolp@digium.com>
-
- * main/loader.c: It is possible for mod to become invalid after we
- unload it (if it's a dynamic module) so move it around a bit.
-
-2007-03-19 22:31 +0000 [r59050] Tilghman Lesher <tlesher@digium.com>
-
- * /, funcs/func_strings.c: Merged revisions 59049 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r59049 | tilghman | 2007-03-19 17:29:56 -0500 (Mon, 19 Mar 2007)
- | 2 lines Oops, this should have been a %d all along ........
-
-2007-03-19 15:43 +0000 [r59041] Tilghman Lesher <tlesher@digium.com>
-
- * configs/sip_notify.conf.sample, /: Merged revisions 59040 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r59040 | tilghman | 2007-03-19 10:42:26 -0500 (Mon, 19 Mar 2007)
- | 2 lines Fix unescaped semicolon (reported via -dev list)
- ........
-
-2007-03-18 20:39 +0000 [r59038] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 59037 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r59037 | oej | 2007-03-18 21:37:06 +0100 (Sun, 18 Mar 2007) | 3
- lines Issue #9313, Asterisk crash on SIP return code 0 (reported
- by qwerty1979) (ASA-2007-011) ........
-
-2007-03-18 16:59 +0000 [r59036] BJ Weschke <bweschke@btwtech.com>
-
- * /, apps/app_followme.c: Merged revisions 59035 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r59035 | bweschke | 2007-03-18 12:36:44 -0400 (Sun, 18 Mar 2007)
- | 3 lines Don't return a non-zero return code if the profile
- doesn't exist, to match what the documentation says it already
- does. (#9307 Reported by kkiely) ........
-
-2007-03-16 16:14 +0000 [r58995] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_page.c: Merged revisions 58992 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r58992 | file | 2007-03-16 12:12:28 -0400 (Fri, 16 Mar 2007) | 2
- lines Wait for the async thread to exit when hanging up all of
- the paged phones under all circumstances. (issue #9181 reported
- by PhilSmith) ........
-
-2007-03-16 01:43 +0000 [r58954-58958] Russell Bryant <russell@digium.com>
-
- * /, configs/sla.conf.sample: Merged revisions 58957 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r58957 | russell | 2007-03-15 20:42:37 -0500 (Thu, 15
- Mar 2007) | 1 line fix a couple SLA documentation references
- ........
-
- * /, build_tools/prep_tarball: Merged revisions 58953 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r58953 | russell | 2007-03-15 20:12:40 -0500 (Thu, 15
- Mar 2007) | 2 lines Add the --pdf option to the usage of rubber
- in prep_tarball ........
-
-2007-03-16 00:04 +0000 [r58949-58950] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c, /, doc/ast_appdocs.tex: Merged revisions 58946 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r58946 | tilghman | 2007-03-15 18:52:48 -0500 (Thu, 15 Mar 2007)
- | 2 lines Refashion dump command to match common syntax and
- update the resulting appdocs TeX file ........
-
- * main/pbx.c: Fix trunk so that it compiles again
-
-2007-03-15 23:56 +0000 [r58942-58948] Russell Bryant <russell@digium.com>
-
- * Makefile, /, build_tools/menuselect-deps.in, configure,
- include/asterisk/autoconfig.h.in, configure.ac, makeopts.in:
- Merged revisions 58947 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r58947 | russell | 2007-03-15 18:53:26 -0500 (Thu, 15 Mar 2007) |
- 3 lines Add configure script checking for GTK2 and some
- additional Makefile targets to support gmenuselect ........
-
- * /, doc/asterisk.tex: Merged revisions 58941 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r58941 | russell | 2007-03-15 18:24:09 -0500 (Thu, 15 Mar 2007) |
- 1 line add a link to the rubber homepage ........
-
-2007-03-15 22:52 +0000 [r58936-58938] Russell Bryant <russell@digium.com>
-
- * Makefile, /, doc/asterisk.tex: Merged revisions 58937 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r58937 | russell | 2007-03-15 17:51:29 -0500 (Thu, 15 Mar 2007) |
- 2 lines Add Asterisk version information to the generated PDF
- ........
-
- * /, build_tools/prep_tarball: Merged revisions 58935 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r58935 | russell | 2007-03-15 17:35:52 -0500 (Thu, 15
- Mar 2007) | 2 lines have prep_tarball attempt to build
- asterisk.pdf ........
-
-2007-03-15 22:33 +0000 [r58934] Tilghman Lesher <tlesher@digium.com>
-
- * /, funcs/func_realtime.c: Merged revisions 58933 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r58933 | tilghman | 2007-03-15 17:32:33 -0500 (Thu, 15
- Mar 2007) | 2 lines Function works fine, but the documentation is
- backwards. ........
-
-2007-03-15 22:29 +0000 [r58932] Russell Bryant <russell@digium.com>
-
- * doc/manager.txt (removed), doc/misdn.txt (removed),
- doc/jitterbuffer.tex (added), /, doc/billing.txt (removed),
- doc/extensions.tex (added), doc/queues-with-callback-members.tex
- (added), doc/localchannel.txt (removed), doc/cdrdriver.txt
- (removed), doc/00README.1st (removed), doc/ajam.tex (added),
- doc/manager.tex (added), doc/misdn.tex (added), doc/freetds.txt
- (removed), doc/odbcstorage.txt (removed), configure,
- doc/model.txt (removed), doc/cygwin.txt (removed), doc/sla.tex,
- doc/billing.tex (added), doc/ael.txt (removed),
- doc/channelvariables.txt (removed), doc/callingpres.txt
- (removed), doc/musiconhold-fpm.txt (removed),
- doc/localchannel.tex (added), doc/enum.txt (removed),
- doc/cdrdriver.tex (added), build_tools/make_buildopts_h,
- doc/security.txt (removed), doc/imapstorage.txt (removed),
- doc/PEERING, main/pbx.c, doc/freetds.tex (added),
- doc/odbcstorage.tex (added), doc/privacy.txt (removed),
- configure.ac, doc/iax.txt (removed), doc/channelvariables.tex
- (added), doc/ael.tex (added), doc/enum.tex (added),
- doc/security.tex (added), doc/math.txt (removed), Makefile,
- doc/imapstorage.tex (added), doc/privacy.tex (added),
- doc/realtime.txt (removed), doc/dundi.txt (removed),
- doc/mysql.txt (removed), apps/app_voicemail.c, doc/cliprompt.txt
- (removed), doc/chaniax.txt (removed), doc/app-sms.txt (removed),
- doc/ast_appdocs.tex (added), doc/realtime.tex (added),
- doc/ices.txt (removed), doc/dundi.tex (added), doc/queuelog.txt
- (removed), doc/extconfig.txt (removed), doc/radius.txt (removed),
- doc/cliprompt.tex (added), doc/chaniax.tex (added),
- doc/hardware.txt (removed), doc/mp3.txt (removed),
- doc/app-sms.tex (added), doc/ices.tex (added), doc/asterisk.tex
- (added), doc/configuration.txt (removed), doc/queuelog.tex
- (added), doc/asterisk-conf.txt (removed), doc/sla.pdf (removed),
- doc/ip-tos.txt (removed), doc/hardware.tex (added), doc/h323.txt
- (removed), doc/mp3.tex (added), doc/configuration.tex (added),
- doc/asterisk-conf.tex (added), doc/jitterbuffer.txt (removed),
- doc/channels.txt (removed), doc/ip-tos.tex (added),
- doc/extensions.txt (removed),
- doc/queues-with-callback-members.txt (removed), doc/apps.txt
- (removed), makeopts.in, doc/ajam.txt (removed): Merged revisions
- 58931 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r58931 | russell | 2007-03-15 17:25:12 -0500 (Thu, 15 Mar 2007) |
- 13 lines Merge changes from svn/asterisk/team/russell/LaTeX_docs.
- * Convert most of the doc directory into a single LaTeX formatted
- document so that we can generate a PDF, HTML, or other formats
- from this information. * Add a CLI command to dump the
- application documentation into LaTeX format which will only be
- include if the configure script is run with --enable-dev-mode. *
- The PDF turned out to be close to 1 MB, so it is not included.
- However, you can simply run "make asterisk.pdf" to generate it
- yourself. We may include it in release tarballs or have
- automatically generated ones on the web site, but that has yet to
- be decided. ........
-
-2007-03-15 18:21 +0000 [r58924] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 58923 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r58923 | file | 2007-03-15 15:13:21 -0300 (Thu, 15 Mar 2007) | 2
- lines Don't assume that the pvt structure will still exist after
- calling schedule_delivery as it may not. (issue #9278 reported by
- fmachado) ........
-
-2007-03-14 19:19 +0000 [r58904-58907] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 58906 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r58906 | russell | 2007-03-14 14:18:08 -0500 (Wed, 14 Mar 2007) |
- 4 lines Some people like to put "limitonpeer" instead of
- "limitonpeers" in their configuration. While we're at it, support
- "limitonpeerz" and "limitonpeerssssss". (inspired by issue #9172)
- ........
-
- * /, doc/sla.tex, doc/sla.pdf: Merged revisions 58902 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r58902 | russell | 2007-03-14 12:04:38 -0500 (Wed, 14
- Mar 2007) | 2 lines Add a more basic example setup to the
- examples section ........
-
-2007-03-14 17:01 +0000 [r58900-58901] Olle Johansson <oej@edvina.net>
-
- * cdr/cdr_radius.c: Correct reference to Radius library THanks
- Philippe - Greetings from Lisboa, Portugal
-
- * /, channels/chan_sip.c: Merged revisions 58848 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r58848 | oej | 2007-03-13 12:49:35 +0100 (Tue,
- 13 Mar 2007) | 10 lines Merged revisions 58847 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r58847 | oej | 2007-03-13 12:45:52 +0100 (Tue, 13 Mar 2007) | 2
- lines Issue #9229 - No port in request URI on register to non
- default SIP ports (neelakantan) ........ ................
-
-2007-03-14 16:40 +0000 [r58895-58898] Russell Bryant <russell@digium.com>
-
- * /, doc/security.txt: Merged revisions 58897 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r58897 | russell | 2007-03-14 11:40:22 -0500
- (Wed, 14 Mar 2007) | 11 lines Merged revisions 58896 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r58896 | russell | 2007-03-14 11:38:48 -0500 (Wed, 14
- Mar 2007) | 3 lines Add a note to the security file that the
- Asterisk CLI and log files may contain sensitive information, and
- that people should keep this in mind. ........ ................
-
- * /, apps/app_meetme.c, configs/sla.conf.sample: Merged revisions
- 58894 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r58894 | russell | 2007-03-14 11:33:01 -0500 (Wed, 14 Mar 2007) |
- 8 lines By default, don't attempt to do any CallerID handling at
- all with SLA because it is known to not work properly in some
- situations. However, add an option to enable it for those that
- would like to use it anyway. The short story behind this is that
- to properly handle CallerID with SLA, we need the ability to
- change the CallerID on an existing call, and we are not ready to
- handle that. ........
-
-2007-03-14 01:56 +0000 [r58881] Tilghman Lesher <tlesher@digium.com>
-
- * /, funcs/func_strings.c: Merged revisions 58880 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r58880 | tilghman | 2007-03-13 20:47:08 -0500 (Tue, 13 Mar 2007)
- | 3 lines Issue 9162 - pbx_substitute_variables_helper assumes
- the buffer is initialized to all zeroes. This fixes a case where
- it wasn't. ........
-
-2007-03-13 23:20 +0000 [r58866-58873] Russell Bryant <russell@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 58872 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r58872 | russell | 2007-03-13 18:19:51 -0500 (Tue, 13 Mar 2007) |
- 4 lines Ensure that the blinky lights show that the trunk stopped
- ringing when the trunk hangs up before a station has answered it.
- (issue #9234, reported by francesco_r) ........
-
- * /, configs/sla.conf.sample: Merged revisions 58870 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r58870 | russell | 2007-03-13 18:11:08 -0500 (Tue, 13
- Mar 2007) | 1 line fix the reference to the SLA documentation
- ........
-
- * cdr/cdr_sqlite3_custom.c (added), build_tools/menuselect-deps.in,
- configure, include/asterisk/autoconfig.h.in,
- configs/cdr_sqlite3_custom.conf (added),
- doc/res_config_sqlite.txt (added), cdr/cdr_sqlite.c,
- configs/extconfig.conf.sample, configure.ac, UPGRADE.txt,
- CHANGES, makeopts.in, res/res_config_sqlite.c (added),
- configs/res_config_sqlite.conf (added): Merge changes from
- team/russell/sqlite: * Add new module, cdr_sqlite3_custom which
- allows logging custom CDRs into a SQLite3 database. (issue #7149,
- alerios) * Add new module, res_config_sqlite, which adds realtime
- database configuration support for SQLite version 2. I decided
- that this was ok since we didn't have any realtime support for
- version 3. If someone ports this to version 3, then version 2
- support can be removed or marked deprecated. (issue #7790,
- rbarun_proformatique) * Mark cdr_sqlite as deprecated in favor of
- cdr_sqlite3_custom. Also, note that there were other modules on
- the bug tracker that did not make the cut because they provided
- some duplicated functionality. Those are: * cdr_sqlite3 (issue
- #6754, moy) * cdr_sqlite3 (issue #8694, bsd)
-
-2007-03-13 10:14 +0000 [r58822-58846] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 58845 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r58845 | oej | 2007-03-13 11:03:03 +0100 (Tue, 13 Mar 2007) | 3
- lines Don't hangup the call on OK or errors on MESSAGE and INFO
- inside of a dialog (like video update requests). ........
-
- * /, channels/chan_sip.c: Merged revisions 58843 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r58843 | oej | 2007-03-13 10:12:16 +0100 (Tue, 13 Mar 2007) | 2
- lines Issue #9251 - Clear From URI from user attributes (tgrman)
- ........
-
- * channels/chan_h323.c: Change URL to OpenH323 (thanks, Tzafrir!)
-
-2007-03-12 01:22 +0000 [r58780-58784] Joshua Colp <jcolp@digium.com>
-
- * /, main/rtp.c: Merged revisions 58783 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r58783 | file | 2007-03-11 21:21:12 -0400 (Sun, 11 Mar 2007) | 2
- lines Allow RFC2833 compensation to compensate for even stupider
- implementations by queueing up the end frame at the start, not
- the actual end. (issue #8963 reported by AndrewZ) ........
-
- * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
- 58779 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r58779 | file | 2007-03-11 20:51:16 -0400 (Sun, 11 Mar 2007) | 2
- lines Add matchexterniplocally setting which only substitutes
- your externip/externhost setting if it matches the localnet
- setting. I know of at least two people who need opposite
- settings, so I made it an option! (issue #8821 reported by
- kokoskarokoska) ........
-
-2007-03-11 21:57 +0000 [r58761] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/asterisk.c: grammatical errors are bad, mmmkay?
-
-2007-03-11 16:43 +0000 [r58742] Jason Parker <jparker@digium.com>
-
- * build_tools/cflags.xml, main/asterisk.c: Add CLI command "marko
- show birthday" to show "birthday information" for Mark Spencers
- upcoming 30th birthday. To enable, run `make menuselect` and
- select the option MARKO_BDAY under Compiler Flags.
-
-2007-03-10 18:15 +0000 [r58639-58706] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 58705 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r58705 | russell | 2007-03-10 12:11:11 -0600 (Sat, 10 Mar 2007) |
- 6 lines Fix a few more places in chan_iax2 where the ast_frame
- used for receiving a frame was not properly initialized. -
- Interpolating a frame when the jitterbuffer is in use -
- decrypting a frame when IAX2 encryption is on - frames in an IAX2
- trunk ........
-
- * /, apps/app_meetme.c: Merged revisions 58669 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r58669 | russell | 2007-03-09 21:58:27 -0600 (Fri, 09 Mar 2007) |
- 2 lines Make the compiler happy and initialize a variable.
- ........
-
- * /, doc/sla.txt (removed), doc/sla.tex (added), doc/sla.pdf
- (added): Merged revisions 58638 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r58638 | russell | 2007-03-09 17:59:10 -0600 (Fri, 09 Mar 2007) |
- 8 lines Merge some updates to the SLA documentation. I plan to
- keep working on this to explain all of the expected behavior with
- call handling, configuration details for specific phones, and
- other things. However, I got tired of doing it in plain text, so
- I switched to using LaTeX. I have included the PDF version. I
- haven't been able to get a nice looking plain text version out of
- it yet, but I'm not terribly concerned since this is supposed to
- be more of the manual, while the plain text sample configuration
- file is the reference. ........
-
-2007-03-09 21:10 +0000 [r58592-58605] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 58604 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r58604 | file | 2007-03-09 16:08:19 -0500 (Fri, 09 Mar 2007) | 2
- lines Fix spelling of unavailable in voicemail documentation.
- (issue #9248 reported by tensai) ........
-
- * /, channels/chan_sip.c: Merged revisions 58584 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r58584 | file | 2007-03-09 15:49:47 -0500 (Fri,
- 09 Mar 2007) | 10 lines Merged revisions 58579 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r58579 | file | 2007-03-09 15:46:43 -0500 (Fri, 09 Mar 2007) | 2
- lines If we are unable to lookup the host in a c line we have to
- abort, otherwise the previous data is gone and we will
- (potentially) have no data when all is said and done. ........
- ................
-
-2007-03-08 23:21 +0000 [r58511-58541] Russell Bryant <russell@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 58512 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r58512 | russell | 2007-03-08 16:15:15 -0600 (Thu, 08 Mar 2007) |
- 5 lines Hang up the channel that put the call on hold in the
- event processing thread to avoid a race condition. Also, if the
- station originated the call that it is putting on hold, don't
- hang up the trunk if it was the only station on the call and it
- is hanging up due to hold and not a normal hangup. ........
-
- * channels/chan_zap.c, /: Merged revisions 58510 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r58510 | russell | 2007-03-08 16:06:54 -0600 (Thu, 08 Mar 2007) |
- 3 lines Add a missing break statement so that handling the above
- event does not incorrectly destroy the channel. (issue #9242,
- andrew) ........
-
-2007-03-08 21:34 +0000 [r58480] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_odbc.c: Merged revisions 58479 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r58479 | tilghman | 2007-03-08 15:33:03 -0600 (Thu, 08 Mar 2007)
- | 2 lines Fix segfault (Issue 9236) ........
-
-2007-03-08 20:56 +0000 [r58475] Russell Bryant <russell@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 58474 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r58474 | russell | 2007-03-08 14:54:56 -0600 (Thu, 08 Mar 2007) |
- 3 lines Refactor hold handling a bit so that it does not require
- keeping the call up when a call is put on hold. ........
-
-2007-03-08 18:05 +0000 [r58390-58437] Joshua Colp <jcolp@digium.com>
-
- * /, main/rtp.c: Merged revisions 58436 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r58436 | file | 2007-03-08 13:01:00 -0500 (Thu, 08 Mar 2007) | 2
- lines Make early SDP seeding even smarter! We have to check
- codecs in the make_compatible function too. (issue #9221 reported
- by marcelbarbulescu) ........
-
- * /, main/dsp.c: Merged revisions 58389 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r58389 | file | 2007-03-08 11:07:10 -0500 (Thu,
- 08 Mar 2007) | 10 lines Merged revisions 58388 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r58388 | file | 2007-03-08 11:04:58 -0500 (Thu, 08 Mar 2007) | 2
- lines Only print out debug message if the definition that makes
- the variables shows up was actually defined. (issue #9233
- reported by serginuez) ........ ................
-
-2007-03-08 13:27 +0000 [r58353-58355] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, main/http.c: Merged revisions 58354 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r58354 | kpfleming | 2007-03-08 08:23:46 -0500 (Thu, 08 Mar 2007)
- | 2 lines this change was not needed; fclose() handles closing
- the file descriptor already ........
-
- * /, apps/app_meetme.c, main/http.c: Merged revisions 58351-58352
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r58351 | kpfleming | 2007-03-08 08:17:17 -0500 (Thu, 08 Mar 2007)
- | 2 lines fix two cases where HTTP session file descriptors would
- not be closed ........ r58352 | kpfleming | 2007-03-08 08:17:42
- -0500 (Thu, 08 Mar 2007) | 2 lines fix a compiler warning, and
- overwriting 'res' value ........
-
-2007-03-08 01:06 +0000 [r58304-58321] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c, /, configure, configure.ac: Merged revisions
- 58320 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r58320 | russell | 2007-03-07 19:01:46 -0600 (Wed, 07 Mar 2007) |
- 6 lines If we receive ZT_EVENT_REMOVED, destroy the specified
- channel. (issue #7256, tzafrir) Also, update the configure script
- to make sure that we don't try to build chan_zap if the installed
- version of zaptel does not include ZT_EVENT_REMOVED. ........
-
- * configs/dundi.conf.sample, pbx/pbx_dundi.c, CHANGES: Add the
- ability to dynamically specify weights for responses to DUNDi
- queries. This can be done using a global variable or a dialplan
- function. Using the SHELL() function will allow you to use an
- external script to determine what the weight in the response
- should be. This can be very useful in load balancing
- applications. (inspired by discussions with blitzrage and jsmith
- in #asterisk-bugs)
-
-2007-03-07 20:05 +0000 [r58286] Joshua Colp <jcolp@digium.com>
-
- * main/loader.c: Make the loader less noisy under valgrind.
-
-2007-03-07 18:20 +0000 [r58244] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 58243 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r58243 | russell | 2007-03-07 12:19:19 -0600
- (Wed, 07 Mar 2007) | 17 lines (This bug was reported to me by
- Kinsey Moore) Merged revisions 58242 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r58242 | russell | 2007-03-07 12:17:07 -0600 (Wed, 07 Mar 2007) |
- 7 lines Fix a problem where the Asterisk channel name could be
- that of the wrong IAX2 user for a call. This is because the first
- step of choosing this name is to look for an IAX2 peer that
- happens to have the same IP/port number that this call is coming
- from and assuming that is it. However, this is not always
- correct. So, I have made it change this name after authentication
- happens since at that point, we have an exact match. ........
- ................
-
-2007-03-07 17:55 +0000 [r58241] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c, main/rtp.c: Merged revisions 58240 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r58240 | file | 2007-03-07 12:52:58 -0500 (Wed, 07 Mar 2007) | 2
- lines Ensure we have (or should have) at least one matching codec
- before attempting early bridge SDP seeding. (issue #9221 reported
- by marcelbarbulescu) ........
-
-2007-03-07 08:08 +0000 [r58224] Olle Johansson <oej@edvina.net>
-
- * apps/app_ices.c: Adding reference to ices home page. Anyone that
- has tested with ices2 ?
-
-2007-03-07 01:07 +0000 [r58123-58208] Russell Bryant <russell@digium.com>
-
- * main/file.c: Add the format of the file that is currently being
- played to the verbose message. (issue #9105, junky)
-
- * main/manager.c, /: Merged revisions 58165 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r58165 | russell | 2007-03-06 18:25:19 -0600
- (Tue, 06 Mar 2007) | 12 lines Merged revisions 58164 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r58164 | russell | 2007-03-06 18:20:13 -0600 (Tue, 06
- Mar 2007) | 4 lines If the channels acquired using the manager
- Redirect action are not up, then don't attempt to do anything
- with them. It could lead to weird behavior, including crashes.
- (issue #8977) ........ ................
-
- * include/asterisk/utils.h: Add some documentation on the arguments
- to the base64 encode/decode functions. (inspired by issue #9215)
-
- * apps/app_queue.c: Send a manager AgentComplete event when the
- agent transfers the call, in addition to where it is already sent
- if either side hangs up. (issue #9219, rgollent) In passing, I
- put this code in a function so it would not be duplicated a third
- time.
-
-2007-03-06 23:19 +0000 [r58122] Steve Murphy <murf@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 58121 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r58121 | murf | 2007-03-06 16:10:14 -0700 (Tue,
- 06 Mar 2007) | 9 lines Merged revisions 58115 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r58115 | murf | 2007-03-06 15:52:52 -0700 (Tue, 06 Mar 2007) | 1
- line Fix for 9220: Eyebeam cannot renew subscriptions for
- presence info. Reason: re-SUBSCRIBE requests don't include Accept
- headers, which the rfc says are optional (to put it tersely), (it
- uses MAY), and luckily, the sip_pvt struct has the format info
- stored, so we simply leave it if the format is set, and the
- accept header null. ........ ................
-
-2007-03-06 23:01 +0000 [r58101-58120] Russell Bryant <russell@digium.com>
-
- * /, configs/voicemail.conf.sample: Merged revisions 58119 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r58119 | russell | 2007-03-06 17:00:57 -0600 (Tue, 06 Mar 2007) |
- 3 lines Clarify the documentation of the dialout and
- sendvoicemail options. (issue #9000, caio1982 and serge-v)
- ........
-
- * codecs/codec_zap.c: Sync codec_zap with the one that is in the
- 1.4 branch so that it can actually build here, too.
-
-2007-03-06 20:45 +0000 [r58054-58055] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 58053 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r58053 | oej | 2007-03-06 21:37:07 +0100 (Tue,
- 06 Mar 2007) | 10 lines Merged revisions 58052 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r58052 | oej | 2007-03-06 21:33:21 +0100 (Tue, 06 Mar 2007) | 2
- lines Change error message to proper message ........
- ................
-
- * apps/app_stack.c: Debug control, debug control.
-
-2007-03-06 18:02 +0000 [r58024-58025] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_skinny.c: Merged revisions 58023 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r58023 | russell | 2007-03-06 12:01:20 -0600 (Tue, 06
- Mar 2007) | 3 lines Return an error of transmit_response is
- called without a session. (issue #9002) ........
-
-2007-03-06 08:51 +0000 [r57979-57993] Luigi Rizzo <rizzo@icir.org>
-
- * main/say.c: move declaration to the beginning of a block
-
- * apps/app_meetme.c: remove duplicate const
-
-2007-03-05 20:13 +0000 [r57871-57943] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_zap.c, CHANGES: Add zap show version CLI command.
- This pulls the version/echo canceller in use directly using the
- ZT_GETVERSION ioctl. (issue #9094 reported by tootai)
-
- * /, channels/chan_iax2.c: Merged revisions 57914 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r57914 | file | 2007-03-05 14:19:07 -0500 (Mon, 05 Mar 2007) | 2
- lines Since chan_iax2 does not support reception of DTMF with
- duration ensure that it is set to 0 on the frame. (issue #8521
- reported by gdhgdh) ........
-
- * /, apps/app_meetme.c: Merged revisions 57872 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r57872 | file | 2007-03-05 13:39:28 -0500 (Mon, 05 Mar 2007) | 2
- lines Don't create a listen channel and record the conference
- unless the option is turned on. (issue #9204 reported by
- francesco_r) ........
-
- * apps/app_meetme.c: I like it when app_meetme builds under dev
- mode, don't you?
-
- * /, apps/app_voicemail.c: Merged revisions 57870 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r57870 | file | 2007-03-05 12:52:03 -0500 (Mon,
- 05 Mar 2007) | 10 lines Merged revisions 57869 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r57869 | file | 2007-03-05 12:49:18 -0500 (Mon, 05 Mar 2007) | 2
- lines Make create_dirpath use our standard for return values. -1
- is failure, 0 is success. (issue #9205 reported by ballares)
- ........ ................
-
-2007-03-05 15:30 +0000 [r57827] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, /: Merged revisions 57826 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r57826 | murf | 2007-03-05 08:20:17 -0700 (Mon,
- 05 Mar 2007) | 9 lines Merged revisions 57825 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r57825 | murf | 2007-03-05 07:53:57 -0700 (Mon, 05 Mar 2007) | 1
- line Fixed a typo introduced via 9156 (either the gotos or their
- doc strings are wrong) ........ ................
-
-2007-03-05 04:21 +0000 [r57769-57799] Joshua Colp <jcolp@digium.com>
-
- * /, main/slinfactory.c: Merged revisions 57798 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r57798 | file | 2007-03-04 23:19:53 -0500 (Sun, 04 Mar 2007) | 2
- lines Don't allow a NULL pointer to reach ast_frdup. (issue #9155
- reported by cmaj) ........
-
- * configs/extensions.conf.sample: Remove no longer present CLI
- commands from sample extensions.conf. (issue #9193 reported by
- junky)
-
- * /, res/res_jabber.c: Merged revisions 57770 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r57770 | file | 2007-03-04 22:35:03 -0500 (Sun, 04 Mar 2007) | 2
- lines Don't reference a potentially NULL pointer. (issue #9199
- reported by klolik) ........
-
- * /, main/rtp.c: Merged revisions 57768 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r57768 | file | 2007-03-04 22:22:17 -0500 (Sun, 04 Mar 2007) | 2
- lines Preserve marker bit when P2P bridging. (issue #9198
- reported by edgreenberg) ........
-
-2007-03-03 16:43 +0000 [r57736] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_stack.c: Convert stack apps to use ast_storage channel
- structure
-
-2007-03-03 15:35 +0000 [r57708] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3,
- pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test6,
- pbx/ael/ael-test/ref.ael-test7, pbx/ael/ael-test/ref.ael-vtest13:
- updated the regression tests
-
-2007-03-03 14:40 +0000 [r57651-57691] Tilghman Lesher <tlesher@digium.com>
-
- * main/channel.c, include/asterisk/channel.h: Expand datastores to
- add the notion of inheritance. This will be needed for the
- conversion of IAX2 variables from the current custom method to
- ast_storage.
-
- * /, apps/app_voicemail.c: Merged revisions 57649 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r57649 | tilghman | 2007-03-03 00:45:00 -0600
- (Sat, 03 Mar 2007) | 10 lines Merged revisions 57648 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r57648 | tilghman | 2007-03-03 00:36:55 -0600 (Sat, 03
- Mar 2007) | 2 lines Memory leak of a list, if call recording was
- abandoned ........ ................
-
-2007-03-03 01:11 +0000 [r57621] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * main/say.c: Merged revisions 57620 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r57620 | dhubbard | 2007-03-02 18:59:24 -0600 (Fri, 02 Mar 2007)
- | 1 line submitted patch for Georgian language, issue 9010,
- submitted by Alexander Shaduri ........
-
-2007-03-03 00:01 +0000 [r57557-57590] Russell Bryant <russell@digium.com>
-
- * configs/sla.conf.sample: Add the missing configuration template
- to the sample config file. Thanks to Lacy Moore on the
- asterisk-users list for pointing out that this was missing!
-
- * /, configure, configure.ac: Merged revisions 57556 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r57556 | russell | 2007-03-02 17:03:01 -0600 (Fri, 02
- Mar 2007) | 3 lines Update the check that is used to determine
- whether zaptel transcoder support is present. The interface has
- changed. ........
-
-2007-03-02 18:05 +0000 [r57478-57519] Joshua Colp <jcolp@digium.com>
-
- * main/pbx.c: Don't try to do recursive locking/unlocking when it
- isn't supported.
-
- * /, channels/chan_sip.c: Merged revisions 57477 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r57477 | file | 2007-03-02 12:06:52 -0500 (Fri,
- 02 Mar 2007) | 10 lines Merged revisions 57475 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r57475 | file | 2007-03-02 12:02:46 -0500 (Fri, 02 Mar 2007) | 2
- lines If a SIP message comes in and goes to a method handler that
- requires additional values that may not be present then send back
- an error. ........ ................
-
-2007-03-02 17:03 +0000 [r57476] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, /: Merged revisions 57473 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r57473 | murf | 2007-03-02 09:55:16 -0700 (Fri,
- 02 Mar 2007) | 9 lines Merged revisions 57458 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r57458 | murf | 2007-03-02 09:39:33 -0700 (Fri, 02 Mar 2007) | 1
- line further refinement in wording of goto documentation, as per
- 9156, goto not proceeding to next instruction ........
- ................
-
-2007-03-02 16:59 +0000 [r57474] Russell Bryant <russell@digium.com>
-
- * apps/app_dumpchan.c, main/cli.c: Add the channel's Language to
- the "show channel" CLI command and the DumpChan application.
- (issue #9187, Junky)
-
-2007-03-02 05:57 +0000 [r57438] Steve Murphy <murf@digium.com>
-
- * /, pbx/pbx_ael.c, utils/ael_main.c: Merged revisions 57426 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r57426 | murf | 2007-03-01 22:21:36 -0700 (Thu, 01 Mar 2007) | 1
- line I almost had comma escapes right, but 9184 points out the
- problem-- the escape is removed by pbx_config, and pbx_ael should
- also, before sending it down into the pbx engine. Also, you have
- to insert it back in, if you are generating extensions.conf code
- from the AEL. ........
-
-2007-03-02 00:22 +0000 [r57365-57397] Russell Bryant <russell@digium.com>
-
- * /, main/file.c: Merged revisions 57396 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r57396 | russell | 2007-03-01 18:20:44 -0600 (Thu, 01 Mar 2007) |
- 4 lines Return the correct digit that interrupted the stream.
- This fixes exiting the Background application when using the m
- option. (issue #9176, mjagdis) ........
-
- * /, apps/app_meetme.c, doc/sla.txt, include/asterisk/channel.h,
- configs/sla.conf.sample: Merged revisions 57364 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r57364 | russell | 2007-03-01 17:42:53 -0600 (Thu, 01 Mar 2007) |
- 16 lines Merge changes from svn/asterisk/team/russell/sla_updates
- * Originally, I put in the documentation that only Zap interfaces
- would be supported on the trunk side. However, after a discussion
- with Qwell, we came up with a way to make IP trunks work as well,
- using some things already in Asterisk. So, here it is, this now
- officially supports IP trunks. * Update the SLA documentation to
- reflect how to setup IP trunks. * Add a section in sla.txt that
- describes how to set up an SLA system with voicemail. * Simplify
- the way DTMF passthrough is handled in MeetMe. * Fix a bug that
- exposed itself when using a Local channel on the trunk side in
- SLA. The station's channel needs to be passed to the dial API
- when dialing the trunk. * Change a WARNING message to DEBUG in
- channel.h. This message is of no use to users. ........
-
-2007-03-01 22:23 +0000 [r57319] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_local.c, /: Merged revisions 57318 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r57318 | file | 2007-03-01 17:21:44 -0500 (Thu,
- 01 Mar 2007) | 10 lines Merged revisions 57317 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r57317 | file | 2007-03-01 17:19:32 -0500 (Thu, 01 Mar 2007) | 2
- lines Don't even attempt to optimize things when a proxy channel
- is involved. It will just explode in weird and unexplaineable
- ways. (issue #9175 reported by clegall_proformatique) ........
- ................
-
-2007-03-01 20:24 +0000 [r57293] Russell Bryant <russell@digium.com>
-
- * main/channel.c: Constify the list of codec preferences.
-
-2007-03-01 03:01 +0000 [r57259] TransNexus OSP Development <support@transnexus.com>
-
- * doc/osp.txt: 1. Corrected a typo for www.etsi.org. Thank Patrick.
-
-2007-03-01 00:08 +0000 [r57241] Joshua Colp <jcolp@digium.com>
-
- * main/pbx.c: Minor code cleanup... nothing to write home about.
-
-2007-02-28 23:02 +0000 [r57204-57209] Russell Bryant <russell@digium.com>
-
- * /, doc/sla.txt, configs/sla.conf.sample: Merged revisions 57207
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r57207 | russell | 2007-02-28 17:01:52 -0600 (Wed, 28 Feb 2007) |
- 2 lines minor tweaks to the sla docs ........
-
- * /, apps/app_meetme.c, configs/sla.conf.sample: Merged revisions
- 57203 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r57203 | russell | 2007-02-28 16:07:05 -0600 (Wed, 28 Feb 2007) |
- 7 lines Merge more changes from
- svn/asterisk/team/russell/sla_updates * Add support for private
- hold. By setting "hold=private" for a trunk, only the station
- that put the call on hold will be able to retrieve it from hold.
- Also, by setting "hold=private" for a station, any call that
- station puts on hold can only be retrieved by that station.
- ........
-
-2007-02-28 20:46 +0000 [r57184] Joshua Colp <jcolp@digium.com>
-
- * main/pbx.c, pbx/pbx_dundi.c, include/asterisk/pbx.h,
- pbx/pbx_config.c, apps/app_while.c: Convert the PBX core to use
- read/write locks. This yields a nifty performance improvement
- when it comes to simultaneous calls going through the dialplan.
- Using murf's test the old mutex based core took an average of
- 57.3 seconds while the rwlock based core took 31.1 seconds.
- That's a nifty 26.2 seconds performance improvement. The other
- good part is that if we do need to switch back then we just have
- to change the lock/unlock API calls. I converted everywhere that
- used to touch the mutex locks directly to use them.
-
-2007-02-28 19:59 +0000 [r57145-57147] Russell Bryant <russell@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 57146 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r57146 | russell | 2007-02-28 13:58:56 -0600 (Wed, 28 Feb 2007) |
- 2 lines Minor formatting change ........
-
- * /, apps/app_meetme.c, configs/sla.conf.sample: Merged revisions
- 57144 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r57144 | russell | 2007-02-28 13:56:20 -0600 (Wed, 28 Feb 2007) |
- 6 lines Merge changes from svn/asterisk/team/russell/sla_updates
- * Add support for the "barge=no" option for trunks. If this
- option is set, then stations will not be able to join in on a
- call that is on progress on this trunk. ........
-
-2007-02-28 19:30 +0000 [r57140] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, /: Merged revisions 57139 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r57139 | murf | 2007-02-28 12:23:05 -0700 (Wed,
- 28 Feb 2007) | 9 lines Merged revisions 57118 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r57118 | murf | 2007-02-28 12:12:41 -0700 (Wed, 28 Feb 2007) | 1
- line a small documentation update, to reflect reality in the goto
- doc strings, as per 9156, Goto does not proceed to next prio if
- jump fails ........ ................
-
-2007-02-28 19:00 +0000 [r57094] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 57093 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r57093 | file | 2007-02-28 13:57:52 -0500 (Wed,
- 28 Feb 2007) | 10 lines Merged revisions 57092 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r57092 | file | 2007-02-28 13:55:45 -0500 (Wed, 28 Feb 2007) | 2
- lines Fix a few more issues with the agent logoff CLI command.
- (issue #9123 reported by arbrandes) ........ ................
-
-2007-02-28 18:21 +0000 [r57090] Russell Bryant <russell@digium.com>
-
- * /, apps/app_meetme.c, configs/sla.conf.sample: Merged revisions
- 57089 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r57089 | russell | 2007-02-28 12:20:05 -0600 (Wed, 28 Feb 2007) |
- 8 lines Merge current set of changes from
- svn/asterisk/team/russell/sla_updates * Add support for station
- ring delays. Ring delays can be set globally for a station or for
- specific trunks on the station. * Fix a few bugs in existing
- code. * Restructure and Reorganize code to improve readability
- and maintainability. * Improve formatting of the "sla show
- (trunks|stations)" CLI commands. ........
-
-2007-02-28 17:56 +0000 [r57054-57056] Joshua Colp <jcolp@digium.com>
-
- * /: Merged revisions 57055 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r57055 | file | 2007-02-28 12:55:03 -0500 (Wed, 28 Feb 2007) | 2
- lines Picky compiler... ........
-
- * /, apps/app_speech_utils.c: Merged revisions 57053 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r57053 | file | 2007-02-28 12:45:50 -0500 (Wed, 28 Feb
- 2007) | 2 lines Better handle timeouts when the individual speaks
- after everything has been played but before the timeout ends.
- ........
-
-2007-02-28 17:22 +0000 [r57050] Steve Murphy <murf@digium.com>
-
- * /, pbx/pbx_ael.c: Merged revisions 57049 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r57049 | murf | 2007-02-28 10:15:27 -0700 (Wed, 28 Feb 2007) | 1
- line I was surprised that I had not yet downgraded missing goto
- targets and macro call defs to a warning, in case they are in
- extensions.conf; I rectified this problem. Also, A goto in a
- macro to a target in a catch block was not being found; I fixed
- this too; the cause was that I needed to treat catch statements
- like an extension in the find_match code. ........
-
-2007-02-27 22:17 +0000 [r57011] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c: Properly hangup the original dialed channel, not
- the new channel that appeared from the forwarding. (issue #9161
- reported by PhilSmith)
-
-2007-02-27 17:38 +0000 [r56976] Russell Bryant <russell@digium.com>
-
- * /: (also issue #9159) Merged revisions 56975 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r56975 | russell | 2007-02-27 11:36:09 -0600 (Tue, 27 Feb 2007) |
- 4 lines Fix voicemail email attachments. I missed the conversion
- of one of the line endings and there was an extra one where it
- should not have been. (issue #9128) ........
-
-2007-02-27 00:11 +0000 [r56926-56952] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_zap.c, configs/zapata.conf.sample: Issue 7789 -
- some telcos want the TON set based on the number, but without the
- NANP prefix removed
-
-2007-02-26 20:43 +0000 [r56889] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_alsa.c: Merged revisions 56888 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r56888 | russell | 2007-02-26 14:42:21 -0600 (Mon, 26 Feb 2007) |
- 4 lines Restore the behavior of Asterisk 1.2 where if a device
- was not specified in alsa.conf, then we just use the system
- default, instead of creating our own default of hw:0,0. (issue
- #9139) ........
-
-2007-02-26 20:09 +0000 [r56860] Joshua Colp <jcolp@digium.com>
-
- * /, pbx/pbx_config.c: Merged revisions 56856 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r56856 | file | 2007-02-26 15:07:18 -0500 (Mon,
- 26 Feb 2007) | 10 lines Merged revisions 56850 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r56850 | file | 2007-02-26 15:05:02 -0500 (Mon, 26 Feb 2007) | 2
- lines Obey the clearglobalvars option in extensions reload (or
- dialplan reload depending on your version). (issue #9146 reported
- by ramonpeek) ........ ................
-
-2007-02-26 20:04 +0000 [r56849] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 56847 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r56847 | russell | 2007-02-26 14:04:13 -0600 (Mon, 26 Feb 2007) |
- 2 lines Fix a crash in my last change to iax2_indicate(). (issue
- #9150) ........
-
-2007-02-26 19:34 +0000 [r56811-56840] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_record.c: Merged revisions 56839 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r56839 | file | 2007-02-26 14:33:48 -0500 (Mon, 26 Feb 2007) | 2
- lines Update app_record documentation to use new CLI command,
- core show file formats. (issue #9151 reported by junky) ........
-
- * main/pbx.c, /: Merged revisions 56805 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r56805 | file | 2007-02-26 12:09:53 -0500 (Mon, 26 Feb 2007) | 2
- lines Use ast_strlen_zero to see if the language and/or context
- argument is not present for Background instead of just checking
- if it is NULL. (issue #9141 reported by mjagdis) ........
-
-2007-02-26 16:54 +0000 [r56786] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 56785 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r56785 | russell | 2007-02-26 10:51:18 -0600 (Mon, 26 Feb 2007) |
- 3 lines Do more complete locking of the chan_iax2_pvt struct in
- the indicate callback. (Problem brought up by Ben Smithurst on
- the asterisk-dev list) ........
-
-2007-02-26 16:38 +0000 [r56784] Joshua Colp <jcolp@digium.com>
-
- * /, main/asterisk.c: Merged revisions 56783 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r56783 | file | 2007-02-26 11:36:08 -0500 (Mon, 26 Feb 2007) | 2
- lines Allow both of the show version files and core show file
- versions CLI commands to work. (issue #9135 reported by mvanbaak)
- ........
-
-2007-02-26 01:05 +0000 [r56731-56742] Russell Bryant <russell@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 56740 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r56740 | russell | 2007-02-25 19:04:40 -0600 (Sun, 25 Feb 2007) |
- 2 lines Move a comment to be in the correct struct. ........
-
- * main/asterisk.c: Remove redundant check to ensure that LOW_MEMORY
- is not defined. (issue #9136, mvanbaak)
-
- * channels/chan_iax2.c: There is no need to look in the iaxs array
- for the pvt struct when we already have a pointer to it.
-
-2007-02-25 14:53 +0000 [r56686] Tilghman Lesher <tlesher@digium.com>
-
- * main/channel.c, /: Merged revisions 56685 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r56685 | tilghman | 2007-02-25 08:46:41 -0600
- (Sun, 25 Feb 2007) | 11 lines Merged revisions 56684 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r56684 | tilghman | 2007-02-25 08:38:03 -0600 (Sun, 25
- Feb 2007) | 3 lines Issue 9130 - If prev is the last item on the
- channel list, then evaluating additional conditions (e.g. name
- prefix) will cause a NULL dereference. ........ ................
-
-2007-02-24 20:29 +0000 [r56623-56665] Olle Johansson <oej@edvina.net>
-
- * include/asterisk/http.h, main/channel.c,
- include/asterisk/doxyref.h, include/asterisk/utils.h,
- include/asterisk/zapata.h, apps/app_meetme.c, res/res_limit.c,
- include/asterisk/config.h, channels/chan_h323.c, pbx/pbx_ael.c,
- apps/app_amd.c, include/asterisk/ael_structs.h,
- include/asterisk/jingle.h, main/config.c, main/rtp.c: Doxygen
- additions, corrections
-
- * include/asterisk/doxyref.h, channels/chan_zap.c, main/manager.c,
- include/asterisk/frame.h: Doxygen updates and corrections
-
- * apps/app_osplookup.c, funcs/func_curl.c, res/res_snmp.c,
- apps/app_festival.c, cdr/cdr_sqlite.c, codecs/codec_speex.c,
- contrib/asterisk-ng-doxygen, include/asterisk/jabber.h,
- res/res_crypto.c, channels/chan_h323.c, cdr/cdr_pgsql.c,
- cdr/cdr_radius.c, apps/app_voicemail.c: Creating new doxygen
- macro "\extref" to create page that lists external libraries and
- URLs to these. Please help me add these references. We might want
- to create a similar macro "\linuxpackage" to list the needed
- Linux packages in popular distributions.
-
- * include/asterisk/jabber.h: Add some external references
-
- * include/asterisk/doxyref.h, include/asterisk/jabber.h: Doxygen
- updates for AJI - The Asterisk Jabber API
-
-2007-02-24 02:23 +0000 [r56574-56594] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c, configs/skinny.conf.sample: Allow a
- Skinny device to monitor a dialplan hint (w00t!). See
- skinny.conf.sample for configuration example. Note: Some devices
- (seen on 12SP+/30VIP) will lock up if they monitor too many
- hints. This seems to be a hardware limitation - there isn't
- anything we can do about it.
-
- * channels/chan_skinny.c: Support devicestate requests. Now you
- should be able to subscribe to a Skinny device/line.
-
- * /, channels/chan_skinny.c: Merged revisions 56569 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r56569 | qwell | 2007-02-23 20:02:53 -0600 (Fri, 23 Feb
- 2007) | 4 lines Make sure to set a speeddials parent on creation.
- Don't crash if hold is pressed when no call is active. Don't
- return in places that we shouldn't.. Update softkey map when call
- is connected ........
-
-2007-02-24 01:56 +0000 [r56564] Joshua Colp <jcolp@digium.com>
-
- * apps/app_meetme.c: Make Meetme build again under dev mode.
-
-2007-02-23 23:25 +0000 [r56487-56506] Russell Bryant <russell@digium.com>
-
- * /, main/asterisk.c: Merged revisions 56505 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r56505 | russell | 2007-02-23 17:24:18 -0600
- (Fri, 23 Feb 2007) | 16 lines Merged revisions 56504 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r56504 | russell | 2007-02-23 17:20:55 -0600 (Fri, 23
- Feb 2007) | 8 lines Fix up a couple more signal handlers to not
- do bad things that could cause various undesirable results. The
- other day, I made Asterisk deadlock by hitting Control-C because
- of a bad signal handler. Now, signal handlers just set a flag and
- write to an alert pipe for the flag to be handled. Then, there is
- another thread that is monitoring for these flags. If being run
- in console mode, it is just the main thread. If Asterisk is in
- the background, a thread is created to do it. ........
- ................
-
- * channels/chan_iax2.c: Make the hashing function calculate
- something that makes more sense. (Thanks to bmd on #asterisk-dev
- for pointing out my pointless math).
-
-2007-02-23 21:57 +0000 [r56458] Joshua Colp <jcolp@digium.com>
-
- * /, main/sched.c: Merged revisions 56457 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r56457 | file | 2007-02-23 16:53:41 -0500 (Fri, 23 Feb 2007) | 2
- lines Change log notice to debug. It is possible for a scheduled
- item to execute and be deleted at close to the same time and
- unavoidable. If this happens this message creeps up. ........
-
-2007-02-23 21:20 +0000 [r56408-56447] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Merge team/russell/iax2_performance. There
- is not a large amount of code here and the changes are not very
- invasive. However, they should significantly improve performance
- of chan_iax2 under load. IAX2 media frames only carry the
- *source* call number. So, when one arrives, the correct session
- that it is a part of has to be matched on IP address, port
- number, and call number, instead of just a call number. Had these
- frames carried the *destination* call number, this would not be
- an issue, because that would be a unique identifier that would
- make it easy to immediately identify the correct session. The way
- that chan_iax2 did this matching was extremely inefficient. It
- starts at the first available call number and traverses each call
- number sequentially, locking and unlocking a mutex for each one,
- to try to match against it. It would do this regardless of
- whether the call number was in use or not. So, for a call with a
- local call number of 25000, every single incoming media frame
- would require a traversal that required 25000 mutex lock and
- unlock operations. (Note that the max call number is about 32k).
- I have introduced a hash table of active IAX2 calls to improve
- this lookup process. The hash is done on the IP address, port
- number, and call number. So, for the previous example, a few
- lock/unlock operations may be done versus 25000 for each frame.
-
- * CHANGES: Note that the entries in the CHANGES file only list
- functionality changes
-
- * CHANGES: Add GetConfigJSON to the CHANGES file.
-
- * /, channels/chan_iax2.c: Merged revisions 56407 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r56407 | russell | 2007-02-23 14:20:00 -0600
- (Fri, 23 Feb 2007) | 12 lines Merged revisions 56406 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r56406 | russell | 2007-02-23 14:17:56 -0600 (Fri, 23
- Feb 2007) | 4 lines Don't destroy mutexes before unregistering
- all of the entry points from the core. Also, fix a potential
- memory leak from not destroying the locks for all of the possible
- call numbers (about 32k of them). ........ ................
-
-2007-02-23 19:00 +0000 [r56373] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, build_tools/make_version_h: Merged revisions 56372 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r56372 | kpfleming | 2007-02-23 12:59:09 -0600 (Fri, 23 Feb 2007)
- | 2 lines build special version strings for AADK/S800i builds
- ........
-
-2007-02-23 18:01 +0000 [r56278-56342] Russell Bryant <russell@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 56341 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r56341 | russell | 2007-02-23 11:58:57 -0600 (Fri, 23 Feb 2007) |
- 8 lines The IMAP storage code uses the same code to build the
- email that is used when voicemail is sent via email using
- something like sendmail. In the patch from bug 8033 to fix
- various IMAP storage problems, the line endings in the email file
- were changed in the code from "\n" to "\r\n". However, this
- breaks sending regular voicemail to email. So, this change
- conditionally sets line endings to "\r\n" only if IMAP_STORAGE is
- enabled. (issue #9128, patch by jarjarbinks, modified by me to
- not break IMAP storage) ........
-
- * main/manager.c: Introduce a new manager action, GetConfigJSON,
- which is intended to improve performance of the GUI. This encodes
- the configuration into the JSON format in a manager header,
- "JSON: ". The encoded information can be directly used as a
- javascript object, so no parsing is needed. For large
- configuration files, this can greatly improve loading times in
- the GUI. Furthermore, the encoding takes up a lot less space when
- being transmitted than the other alternatives. (Inspired by
- discussion with Pari) Here is an example of what you get:
- http://localhost:8088/asterisk/rawman?action=getconfigjson&filename=users.conf
- Response: Success JSON:
- {"general":["hasvoicemail=yes"],"6000":["fullname=russell","secret=1234"]}
-
- * main/dial.c, /, apps/app_meetme.c, doc/sla.txt,
- configs/sla.conf.sample: Merged revisions 56277 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r56277 | russell | 2007-02-22 17:08:36 -0600 (Thu, 22 Feb 2007) |
- 18 lines Merge changes from team/russell/sla_updates. This batch
- of changes to the SLA code does a few different things. * I made
- the SLA code event driven instead of having to act in a lot of
- busy loops while dialing things to wait for state changes. This
- makes the code more efficient and readable at the same time. * I
- have implemented a couple of new features. The first is inbound
- trunk ringing timeouts. This is an option that defines how long
- to let an incoming call on a trunk to ring. * I have also
- implemented ring timeouts for stations. They may be specified for
- the entire station, meaning it is how long to let the station
- ring before giving up. You can also specify a ring timeout for a
- specific trunk on a station. So, you can say that you only want a
- specific station to ring 5 seconds if it is line1 ringing, but
- otherwise, there is no timeout. ........
-
-2007-02-22 18:53 +0000 [r56232] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /, channels/chan_sip.c: Merged revisions 56231
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r56231 | file | 2007-02-22 13:49:39 -0500 (Thu,
- 22 Feb 2007) | 10 lines Merged revisions 56230 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r56230 | file | 2007-02-22 13:44:24 -0500 (Thu, 22 Feb 2007) | 2
- lines Only change the original or clone channel if it's the
- channel behind the proxy channel, not if it's just a regular
- bridged channel. ........ ................
-
-2007-02-22 17:36 +0000 [r56209] Kevin P. Fleming <kpfleming@digium.com>
-
- * include/asterisk/module.h: move the ast_module_info structure
- into the special section as well, otherwise when
- restore_globals() is called it will lose its pointer to the
- ast_module structure that the loader put there
-
-2007-02-22 16:48 +0000 [r56188] Joshua Colp <jcolp@digium.com>
-
- * .cleancount: Since I'm a nice guy... let's increment the clean
- count since last night's module changes require a rebuild of
- everything essentially.
-
-2007-02-22 16:25 +0000 [r56187] Russell Bryant <russell@digium.com>
-
- * apps/app_voicemail.c: Fix some compilation problems in
- app_voicemail. There was a parenthesis missing in a function
- prototype, and "#elifdef" is not a valid preprocessor directive.
- (issue #9122, akohlsmith)
-
-2007-02-22 13:58 +0000 [r56156] TransNexus OSP Development <support@transnexus.com>
-
- * doc/osp.txt: Update OSP documention for v1.6.
-
-2007-02-22 10:46 +0000 [r56126] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 56125 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r56125 | oej | 2007-02-22 11:33:55 +0100 (Thu, 22 Feb 2007) | 2
- lines Move message from verbose to debug ........
-
-2007-02-22 02:48 +0000 [r56095] Steve Murphy <murf@digium.com>
-
- * /, sounds/Makefile: Merged revisions 56094 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r56094 | murf | 2007-02-21 19:39:58 -0700 (Wed, 21 Feb 2007) | 1
- line updated the sound tarball versions in Makefile ........
-
-2007-02-22 02:36 +0000 [r56092] Kevin P. Fleming <kpfleming@digium.com>
-
- * funcs, codecs, apps, include/asterisk/module.h,
- Makefile.moddir_rules, Makefile.rules,
- build_tools/make_linker_eo_script (added), cdr, pbx, res,
- channels, formats, main/loader.c: give embedded modules a helping
- hand by backing up and restoring their global variables when they
- are loaded and unloaded
-
-2007-02-22 01:26 +0000 [r56012-56056] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 56055 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r56055 | russell | 2007-02-21 19:24:10 -0600 (Wed, 21 Feb 2007) |
- 3 lines Restructure a little bit of code to reduce nesting. There
- is no functionality change here. ........
-
- * /, channels/chan_sip.c: Merged revisions 56011 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r56011 | russell | 2007-02-21 18:57:36 -0600
- (Wed, 21 Feb 2007) | 11 lines Merged revisions 56010 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r56010 | russell | 2007-02-21 18:53:25 -0600 (Wed, 21
- Feb 2007) | 3 lines If we receive a frame that is not in any of
- the negotiated formats, then drop it. (potentially issue #8781
- and SPD-12) ........ ................
-
-2007-02-22 00:38 +0000 [r56009] Joshua Colp <jcolp@digium.com>
-
- * /, main/cli.c: Merged revisions 56008 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r56008 | file | 2007-02-21 19:35:55 -0500 (Wed, 21 Feb 2007) | 2
- lines Print out deprecation notice on usage output of CLI
- commands. (issue #8925 reported by blitzrage) ........
-
-2007-02-22 00:05 +0000 [r55958-56005] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c: Make filename on email follow subject
- message number, purely for cosmetic purposes for individuals like
- *cough* jsmith *cough*. (issue #9111 reported by sshah)
-
- * /, apps/app_meetme.c: Merged revisions 55957 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r55957 | file | 2007-02-21 15:35:40 -0500 (Wed,
- 21 Feb 2007) | 10 lines Merged revisions 55956 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r55956 | file | 2007-02-21 15:32:16 -0500 (Wed, 21 Feb 2007) | 2
- lines Change naughty warning message to provide useful
- information. If a write now fails on a channel in meetme it will
- tell you the channel name instead of spitting out the wrong error
- message. ........ ................
-
-2007-02-21 20:30 +0000 [r55955] Jason Parker <jparker@digium.com>
-
- * channels/chan_jingle.c, /, channels/chan_gtalk.c: Merged
- revisions 55954 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r55954 | qwell | 2007-02-21 14:27:08 -0600 (Wed, 21 Feb 2007) | 4
- lines Fix locking issue, and accept "transport-accept" as a valid
- accept message. This should solve issues 8970 and 8503. ........
-
-2007-02-21 20:26 +0000 [r55953] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Clarify in the doxygen docs abou RFC2833
- compensation flag.
-
-2007-02-21 20:23 +0000 [r55952] Russell Bryant <russell@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 55951 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r55951 | russell | 2007-02-21 14:22:33 -0600 (Wed, 21 Feb 2007) |
- 3 lines Simplify the last change to app_meetme, and move the call
- to dispose_conf() up into the block where we know a conf exists.
- ........
-
-2007-02-21 20:18 +0000 [r55915-55950] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 55949 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r55949 | file | 2007-02-21 15:16:34 -0500 (Wed, 21 Feb 2007) | 2
- lines Only dispose of the conference if one was created. ........
-
- * /, apps/app_speech_utils.c: Merged revisions 55947 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r55947 | file | 2007-02-21 15:03:38 -0500 (Wed, 21 Feb
- 2007) | 2 lines Only start playing the next file if we have not
- been quieted. ........
-
- * /, channels/chan_sip.c: Merged revisions 55914 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r55914 | file | 2007-02-21 12:18:19 -0500 (Wed, 21 Feb 2007) | 2
- lines Add a flag that indicates whether a SIP dialog is an
- outgoing call or not. SIP_OUTGOING originally did it but it was
- repurposed to the direction of the last transaction, which can
- cause update_call_counter to falsely decrease the wrong counters.
- (please don't hurt me oej) (issue #8943 reported by mdu113)
- ........
-
-2007-02-21 14:07 +0000 [r55870] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, build_tools/make_version: Merged revisions 55869 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r55869 | kpfleming | 2007-02-21 08:06:47 -0600
- (Wed, 21 Feb 2007) | 10 lines Merged revisions 55868 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r55868 | kpfleming | 2007-02-21 08:03:11 -0600 (Wed, 21
- Feb 2007) | 2 lines use new tag version script ........
- ................
-
-2007-02-21 08:39 +0000 [r55835] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 55834 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r55834 | oej | 2007-02-21 09:32:34 +0100 (Wed, 21 Feb 2007) | 2
- lines Issue #8848 - Turn off lamp more quickly after transfer
- (decrement inuse early on transferer's call leg) ........
-
-2007-02-21 02:04 +0000 [r55805] Jason Parker <jparker@digium.com>
-
- * channels/chan_jingle.c, /, channels/chan_gtalk.c: Merged
- revisions 55799 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r55799 | qwell | 2007-02-20 20:01:36 -0600 (Tue, 20 Feb 2007) | 4
- lines Fix segfault when buddy couldn't be found. Issue 7764,
- patch by sailer ........
-
-2007-02-21 01:05 +0000 [r55763] Joshua Colp <jcolp@digium.com>
-
- * main/dns.c: Return trunk to a state where it compiles under
- Darwin. The byte order stuff is ugly, if anyone wants to clean it
- up... by all means do so.
-
-2007-02-21 01:05 +0000 [r55762] Russell Bryant <russell@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 55758 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r55758 | russell | 2007-02-20 19:03:25 -0600 (Tue, 20 Feb 2007) |
- 4 lines Improve the reference counting to fix bugs where people
- report seeing conferences listed that have no members. (issue
- #9073) ........
-
-2007-02-21 00:14 +0000 [r55671-55748] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 55741 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r55741 | file | 2007-02-20 19:11:20 -0500 (Tue, 20 Feb 2007) | 2
- lines Better handle dropped IMAP connections. (issue #9054
- reported by bsmithurst) ........
-
- * /, channels/chan_sip.c: Merged revisions 55717 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r55717 | file | 2007-02-20 18:57:03 -0500 (Tue, 20 Feb 2007) | 2
- lines Return behavior I removed. I did not remember that you
- could just add a localnet entry to make it work. ........
-
- * main/logger.c: Flush out the file pointer. (issue #9115 reported
- by guthrie)
-
- * /, channels/chan_sip.c: Merged revisions 55688 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r55688 | file | 2007-02-20 18:08:45 -0500 (Tue, 20 Feb 2007) | 2
- lines Don't test our own address against the localnet settings.
- At least one person has had issues as a result of this from #7051
- so I'm reversing it. (issue #8821 reported by kokoskarokoska)
- ........
-
- * /, channels/chan_agent.c: Merged revisions 55670 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r55670 | file | 2007-02-20 17:47:00 -0500 (Tue,
- 20 Feb 2007) | 10 lines Merged revisions 55669 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r55669 | file | 2007-02-20 17:39:14 -0500 (Tue, 20 Feb 2007) | 2
- lines Defer clearing callback information if channels are up
- until they are hung up. This ensures the hangup process goes
- smoothly and no channels get hung in limbo. (issue #8088 reported
- by kebl0155) ........ ................
-
-2007-02-20 20:32 +0000 [r55591-55635] Russell Bryant <russell@digium.com>
-
- * /, main/http.c: Merged revisions 55634 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r55634 | russell | 2007-02-20 14:26:06 -0600 (Tue, 20 Feb 2007) |
- 3 lines Add the Asterisk version information to the Server header
- in HTTP responses. (requested by Pari) ........
-
- * /, include/asterisk/manager.h: Merged revisions 55590 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r55590 | russell | 2007-02-20 13:57:07 -0600 (Tue, 20 Feb 2007) |
- 2 lines Increase the maximum number of manager headers to 128, at
- the request of Pari. ........
-
-2007-02-20 16:56 +0000 [r55556] Jason Parker <jparker@digium.com>
-
- * channels/chan_jingle.c, /, channels/chan_gtalk.c,
- res/res_jabber.c: Merged revisions 55555 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r55555 | qwell | 2007-02-20 10:53:45 -0600 (Tue, 20 Feb 2007) | 4
- lines No need to cast nor free with strdupa (thanks file) 55555!
- ........
-
-2007-02-20 16:42 +0000 [r55554] Russell Bryant <russell@digium.com>
-
- * /, configs/sla.conf.sample: Merged revisions 55553 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r55553 | russell | 2007-02-20 10:41:57 -0600 (Tue, 20
- Feb 2007) | 3 lines Change the formatting of sla.conf.sample to
- make it more readable. (issue #9112, blitzrage) ........
-
-2007-02-20 15:19 +0000 [r55534] Joshua Colp <jcolp@digium.com>
-
- * res/res_jabber.c: I like it when trunk builds, so let's make
- res_jabber compile again!
-
-2007-02-20 07:48 +0000 [r55514] Olle Johansson <oej@edvina.net>
-
- * /, res/res_jabber.c: Merged revisions 55483 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r55483 | oej | 2007-02-19 22:12:55 +0100 (Mon, 19 Feb 2007) | 3
- lines - Not sending arguments to an application is not "out of
- memory" - Making error messages a bit more clear ........
-
-2007-02-19 23:27 +0000 [r55495] Jason Parker <jparker@digium.com>
-
- * .cleancount: We need to bump the cleancount when we make API
- changes...
-
-2007-02-19 18:15 +0000 [r55436] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 55435 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r55435 | tilghman | 2007-02-19 12:11:48 -0600
- (Mon, 19 Feb 2007) | 10 lines Merged revisions 55434 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r55434 | tilghman | 2007-02-19 12:09:09 -0600 (Mon, 19
- Feb 2007) | 2 lines forcename and forcegreetings options should
- check to see if the recording already exists ........
- ................
-
-2007-02-19 16:01 +0000 [r55410-55414] Joshua Colp <jcolp@digium.com>
-
- * CHANGES: Clarify last change for SMDI in CHANGES file.
-
- * configs/voicemail.conf.sample, apps/app_voicemail.c: Allow both
- an external application and SMDI to do voicemail notification at
- the same time. (issue #8625 reported by lters)
-
-2007-02-19 15:24 +0000 [r55409] Doug Bailey <dbailey@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 55397 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r55397 | dbailey | 2007-02-19 08:52:59 -0600 (Mon, 19 Feb 2007) |
- 3 lines Changed iax2 process thread to detached to correct memory
- leak due to left over thread context on thread exit. Modified
- module unload process to avoid deadlocks on pthread cancels
- ........
-
-2007-02-18 22:07 +0000 [r55375] Olle Johansson <oej@edvina.net>
-
- * apps/app_voicemail.c: Formatting changes.
-
-2007-02-18 19:13 +0000 [r55351-55352] Joshua Colp <jcolp@digium.com>
-
- * codecs/gsm/inc/proto.h: Return GSM to a state where it actually
- builds under dev mode.
-
- * channels/chan_h323.c: Update chan_h323 to new set_rtp_peer
- definition.
-
-2007-02-18 15:11 +0000 [r55330] Olle Johansson <oej@edvina.net>
-
- * res/res_features.c: Being picky...
-
-2007-02-18 15:03 +0000 [r55329] Kevin P. Fleming <kpfleming@digium.com>
-
- * Makefile, channels/chan_misdn.c, main/srv.c,
- main/editline/refresh.c, pbx/ael/ael.tab.c,
- channels/misdn/isdn_msg_parser.c, channels/chan_oss.c,
- main/enum.c, apps/app_voicemail.c, main/ast_expr2.c: add -Wundef
- to the --enable-dev-mode flags, so that mistyped macro names in
- #if expressions will be caught convert various #if expressions to
- #ifdef for macros that may not be defined (and where the value is
- not important) Note: two of these changes are in bison generated
- files which is going to be inconvenient when they are regenerated
-
-2007-02-18 15:01 +0000 [r55279-55323] Olle Johansson <oej@edvina.net>
-
- * res/res_features.c: Simplify post_manager_event()
-
- * /, apps/app_record.c: Merged revisions 55278 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r55278 | oej | 2007-02-18 13:35:54 +0100 (Sun,
- 18 Feb 2007) | 10 lines Merged revisions 55277 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r55277 | oej | 2007-02-18 13:32:13 +0100 (Sun, 18 Feb 2007) | 2
- lines Documentation update (#9053, jsmith) ........
- ................
-
-2007-02-17 17:41 +0000 [r55220] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 55219 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r55219 | file | 2007-02-17 12:39:32 -0500 (Sat, 17 Feb 2007) | 2
- lines Add missing membername option to AddQueueMember
- documentation. (issue #9088 reported by seanbright) ........
-
-2007-02-17 17:11 +0000 [r55218] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_skinny.c: Merged revisions 55217 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r55217 | qwell | 2007-02-17 11:10:09 -0600 (Sat, 17 Feb
- 2007) | 4 lines Fix an issue where callerid would not be
- displayed on some phones. Issue 8995, initial patch and research
- done by wedhorn ........
-
-2007-02-17 16:48 +0000 [r55087-55198] Joshua Colp <jcolp@digium.com>
-
- * apps/app_queue.c: We want to skip the queue if the name doesn't
- match the specified one, not if they *do*.
-
- * apps/app_queue.c: Increase "queue show" buffer size from 80 to
- 240. This should be more then enough for most cases. (issue #9089
- reported by mvanbaak)
-
- * apps/app_dial.c, /: Merged revisions 55154 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r55154 | file | 2007-02-16 22:55:30 -0500 (Fri,
- 16 Feb 2007) | 10 lines Merged revisions 55153 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r55153 | file | 2007-02-16 22:53:45 -0500 (Fri, 16 Feb 2007) | 2
- lines Answer the channel before recording privacy information.
- (issue #8926 reported by lmamane) ........ ................
-
- * /, apps/app_queue.c: Merged revisions 55129 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r55129 | file | 2007-02-16 21:59:50 -0500 (Fri, 16 Feb 2007) | 2
- lines Make the 'i' option of Queue actually work. (issue #8986
- reported by utis) ........
-
- * channels/chan_jingle.c: Update chan_jingle to new definition of
- set_rtp_peer.
-
- * /, channels/chan_sip.c: Merged revisions 55086 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r55086 | file | 2007-02-16 20:16:59 -0500 (Fri,
- 16 Feb 2007) | 10 lines Merged revisions 55073 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r55073 | file | 2007-02-16 20:09:50 -0500 (Fri, 16 Feb 2007) | 2
- lines Allow chan_sip to handle attended transfers from a SIP
- phone that is sitting behind chan_agent. Yes folks, all it took
- was one line of code. (issue #8784 reported by pzieba) ........
- ................
-
-2007-02-17 01:11 +0000 [r55004-55077] Russell Bryant <russell@digium.com>
-
- * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
- Merged revisions 55052 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r55052 | russell | 2007-02-16 18:40:34 -0600 (Fri, 16 Feb 2007) |
- 3 lines If the pg_config application is found, but there is
- probably executing it, then consider postgres unavailable. (issue
- #8637) ........
-
- * /, codecs/gsm/Makefile: Merged revisions 55050 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r55050 | russell | 2007-02-16 18:31:42 -0600 (Fri, 16 Feb 2007) |
- 3 lines Filter out yet another architecture that does not work
- with the optimizations in the built-in libgsm. (issue 8637, ovi)
- ........
-
- * /, apps/app_meetme.c, configs/meetme.conf.sample: Merged
- revisions 55006 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r55006 | russell | 2007-02-16 16:49:42 -0600
- (Fri, 16 Feb 2007) | 17 lines Merged revisions 55005 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r55005 | russell | 2007-02-16 16:48:22 -0600 (Fri, 16
- Feb 2007) | 9 lines Revert the change I did in revisions 54955,
- 54969, and 54970, in 1.2, 1.4, and trunk. I decided that once a
- conference is created from meetme.conf, it is acceptable behavior
- that the pin can not be changed until the conference goes away. I
- also added a note in meetme.conf to describe this behavior. We
- still have another issue in 1.4 and trunk where some conferences
- with no users don't go away. That is the real bug that needs to
- be addressed here. ........ ................
-
- * apps/app_dumpchan.c: Print the raw read/write formats in the
- DumpChan application. (issue #9083, junky)
-
-2007-02-16 22:20 +0000 [r55003] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 55002 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r55002 | file | 2007-02-16 17:18:46 -0500 (Fri,
- 16 Feb 2007) | 10 lines Merged revisions 54999 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r54999 | file | 2007-02-16 17:13:45 -0500 (Fri, 16 Feb 2007) | 2
- lines Do not send indications through ast_indicate in chan_agent
- but instead go directly to the technology. This way when
- indications are emulated they happen on the Agent channel and do
- not screw up formats on the channels. (issue #8439 reported by
- punkgode) ........ ................
-
-2007-02-16 21:13 +0000 [r54970] Russell Bryant <russell@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 54969 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r54969 | russell | 2007-02-16 15:12:18 -0600
- (Fri, 16 Feb 2007) | 13 lines Merged revisions 54955 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r54955 | russell | 2007-02-16 14:56:58 -0600 (Fri, 16
- Feb 2007) | 5 lines For conferences that are configured in
- meetme.conf, check the configuration file every time someone
- joins the conference instead of only when the conference is first
- created. This is to ensure that changes to the pin numbers in the
- config file are always honored. (issue #9073) ........
- ................
-
-2007-02-16 18:53 +0000 [r54910-54925] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 54924 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r54924 | file | 2007-02-16 13:51:15 -0500 (Fri, 16 Feb 2007) | 2
- lines Need to check macro extension as well as macro context for
- directed pickup. ........
-
- * res/res_features.c, configs/features.conf.sample: Allow the user
- to specify where to enable the respective features for when a
- parked call is picked up. (ie: transfers and parking)
-
-2007-02-16 18:04 +0000 [r54890-54901] Russell Bryant <russell@digium.com>
-
- * /, pbx/pbx_config.c: Merged revisions 54898 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r54898 | russell | 2007-02-16 12:03:41 -0600 (Fri, 16 Feb 2007) |
- 4 lines Fix setting "autofallthrough" to yes by default. It was
- set to enabled in pbx.c. However, if the option was not present
- in extensions.conf, then pbx_config.c would set it back to
- disabled. ........
-
- * /, res/res_features.c: Merged revisions 54888 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r54888 | russell | 2007-02-16 11:40:38 -0600 (Fri, 16 Feb 2007) |
- 3 lines Clean up a few coding guidelines issues - spaces to tabs,
- use sizeof() to pass the size of a static buffer, add spaces ...
- ........
-
-2007-02-16 17:41 +0000 [r54889] Joshua Colp <jcolp@digium.com>
-
- * res/res_features.c, CHANGES, configs/features.conf.sample: Add
- option to features.conf that enables parking via DTMF on picked
- up parked calls. (issue #9082 reported by francesco_r)
-
-2007-02-16 17:26 +0000 [r54887] Jason Parker <jparker@digium.com>
-
- * /, main/asterisk.c: Merged revisions 54886 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r54886 | qwell | 2007-02-16 11:25:21 -0600 (Fri, 16 Feb 2007) | 4
- lines Clarify a restart message. It's silly, but the reporter had
- a very valid point. Issue 9079 ........
-
-2007-02-16 17:07 +0000 [r54885] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 54884 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r54884 | file | 2007-02-16 12:02:35 -0500 (Fri, 16 Feb 2007) | 2
- lines Allow directed pickup to pick up the real context instead
- of the macro context if a Macro is used. (issue #8984 reported by
- jamesb63) ........
-
-2007-02-16 14:31 +0000 [r54773-54862] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Formatting, whitespace fixes
-
- * apps/app_voicemail.c: More cleanups of app_voicemail
-
- * CREDITS, main/channel.c, channels/chan_sip.c,
- channels/chan_skinny.c, include/asterisk/rtp.h,
- include/asterisk/channel.h, channels/chan_gtalk.c, CHANGES,
- include/asterisk/frame.h, main/rtp.c, channels/chan_mgcp.c:
- Adding Realtime Text support (T.140) to Asterisk T.140/RFC 2793
- is a live communication channel, originally created for IP based
- text phones for hearing impaired. Feels very much like the old
- Unix talk application. This code is developed and disclaimed by
- John Martin of Aupix, UK. Tested for interoperability by myself
- and Omnitor in Sweden, the company that wrote most of the
- specifications. A big thank you to everyone involved in this.
-
- * /, channels/chan_sip.c: Merged revisions 54787 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r54787 | oej | 2007-02-16 13:06:23 +0100 (Fri, 16 Feb 2007) | 2
- lines Issue #7541 - Handle multipart attachments to SIP messages
- - even if boundary is quoted. ........
-
- * res/res_agi.c: Issue #9068 - make sure we quote HTML characters
- correctly too (seanbright)
-
- * /, res/res_agi.c: Merged revisions 54772 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r54772 | oej | 2007-02-16 12:39:55 +0100 (Fri,
- 16 Feb 2007) | 10 lines Merged revisions 54771 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r54771 | oej | 2007-02-16 12:38:03 +0100 (Fri, 16 Feb 2007) | 2
- lines Issue #9069 - If we open with TH we should not close with
- /TD. (seanbright) ........ ................
-
-2007-02-16 01:36 +0000 [r54711-54749] Joshua Colp <jcolp@digium.com>
-
- * main/acl.c: Rely on ast_gethostbyname to handle IP addresses, not
- inet_aton. (issue #9056 reported by pj)
-
- * CHANGES, apps/app_chanspy.c: Add 'o' option to Chanspy which
- causes it to only listen to audio coming from the channel, and
- the 'X' option which allows the user to exit to a valid single
- digit extension. (issue #8137 reported by mnicholson)
-
- * /, apps/app_speech_utils.c: Merged revisions 54714 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r54714 | file | 2007-02-15 19:48:48 -0500 (Thu, 15 Feb
- 2007) | 2 lines Don't let dtmf leak over into the engine and let
- it skew the results... also give DTMF results priority. (issue
- #9014 reported by surftek) ........
-
- * main/manager.c: Properly handle an error result from a manager
- action. This could have left the action list permanently locked
- for reading.
-
-2007-02-15 20:29 +0000 [r54654-54686] Olle Johansson <oej@edvina.net>
-
- * apps/app_voicemail.c: - add some notes, asking for help - insert
- a few ast_strlen_zero - Doxygen additions - A few more spaces
-
- * main/io.c: Make file's new comment doxygenified
-
-2007-02-15 16:24 +0000 [r54624] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 54623 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r54623 | file | 2007-02-15 11:19:39 -0500 (Thu,
- 15 Feb 2007) | 10 lines Merged revisions 54622 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r54622 | file | 2007-02-15 11:14:40 -0500 (Thu, 15 Feb 2007) | 2
- lines Use a separate variable to indicate execution should
- continue instead of the return value. (issue #8842 reported by
- pluto70) ........ ................
-
-2007-02-15 15:53 +0000 [r54574-54599] Olle Johansson <oej@edvina.net>
-
- * CHANGES: ...and don't forget to update CHANGES
-
- * channels/chan_sip.c: Add callgroup and pickupgroup to SIPPEER
- function. (thanks ramon)
-
- * CHANGES: Update CHANGES
-
- * channels/chan_sip.c, configs/extconfig.conf.sample,
- doc/realtime.txt: Issue #7443 - amdtech - Optionally SIP
- registrations in another realtime family.
-
-2007-02-15 02:11 +0000 [r54489-54552] Joshua Colp <jcolp@digium.com>
-
- * main/io.c: Clean up the I/O context handler.
-
- * apps/app_flash.c, apps/app_image.c, apps/app_exec.c: Few more
- code clean ups.
-
- * apps/app_milliwatt.c: Clean up app_milliwatt code.
-
- * apps/app_dial.c, /: Merged revisions 54481 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r54481 | file | 2007-02-14 16:07:23 -0500 (Wed, 14 Feb 2007) | 2
- lines Forward begin DTMF frames as well as end. (issue #9068
- reported by mhardeman) ........
-
-2007-02-14 20:45 +0000 [r54464-54466] Olle Johansson <oej@edvina.net>
-
- * main/asterisk.c: Show version in "core show settings"
-
- * CHANGES: Updates and re-organization to make it easier to digest
- this information
-
- * main/cdr.c, main/manager.c, include/asterisk/config.h,
- include/asterisk/cdr.h, include/asterisk/manager.h,
- main/asterisk.c, main/config.c: New CLI command "Core show
- settings" to list some core settings
-
-2007-02-14 17:14 +0000 [r54404] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 54375 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r54375 | mattf | 2007-02-14 10:56:40 -0600 (Wed,
- 14 Feb 2007) | 10 lines Merged revisions 54373 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r54373 | mattf | 2007-02-14 10:25:49 -0600 (Wed, 14 Feb 2007) | 2
- lines When handling glare on a PRI, move the requested channel
- rather than hang up the old one. Fix for 8957 and 9011. ........
- ................
-
-2007-02-14 17:02 +0000 [r54348-54379] Olle Johansson <oej@edvina.net>
-
- * configs/sip.conf.sample: Make documentation match the source
- code.
-
- * channels/chan_sip.c: Issue #9060 - host= parameter in sip.conf
- stopped working caused by outbound proxy patch.
-
- * channels/chan_sip.c: Add port number to SIPPEER dialplan function
-
-2007-02-14 08:34 +0000 [r54325] Paul Cadach <paul@odt.east.telecom.kz>
-
- * codecs/codec_g722.c: I don't know how it worked earlier, but
- valgrind produces core every time you try to load codec_g722.
- Fixed. ;-)
-
-2007-02-14 01:12 +0000 [r54291] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /: Merged revisions 54290 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r54290 | file | 2007-02-13 20:09:40 -0500 (Tue, 13 Feb 2007) | 2
- lines Add G722 to ast_best_codec. If anyone disagrees with it's
- placement, feel free to change it. (issue #9045 reported by gork)
- ........
-
-2007-02-13 22:02 +0000 [r54067-54261] Russell Bryant <russell@digium.com>
-
- * include/asterisk/devicestate.h, apps/app_meetme.c,
- res/res_features.c, include/asterisk/cli.h, main/devicestate.c,
- CHANGES, apps/app_queue.c, funcs/func_devstate.c (added),
- main/cli.c: This introduces a new dialplan function, DEVSTATE,
- which allows you to do some pretty cool things. First, you can
- get the device state of anything in the dialplan: NoOp(SIP/mypeer
- has state ${DEVSTATE(SIP/mypeer)}) NoOp(The conference room 1234
- has state ${DEVSTATE(MeetMe:1234)}) Most importantly, this allows
- you to create custom device states so you can control phone lamps
- directly from the dialplan.
- Set(DEVSTATE(Custom:mycustomlamp)=BUSY) ... exten =>
- mycustomlamp,hint,Custom:mycustomlamp
-
- * /, channels/chan_sip.c: Merged revisions 54204 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r54204 | russell | 2007-02-13 13:42:00 -0600 (Tue, 13 Feb 2007) |
- 5 lines If we fail to create the SIP socket, then return -1 from
- reload_config() so that load_module() will return
- AST_MODULE_LOAD_DECLINE. Otherwise, the console will just get
- spammed with error messages every time chan_sip tries to send a
- message. ........
-
- * /, channels/chan_sip.c: Merged revisions 54235 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r54235 | russell | 2007-02-13 15:31:22 -0600 (Tue, 13 Feb 2007) |
- 2 lines Remove a couple of leftover debug messages ........
-
- * include/asterisk/devicestate.h, /: Merged revisions 54218 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r54218 | russell | 2007-02-13 14:56:50 -0600 (Tue, 13 Feb 2007) |
- 3 lines Fix the documentation on the return values from device
- state provider registration and deletion. ........
-
- * main/asterisk.c: Use spaces instead of tabs in the help text for
- a CLI command
-
- * main/asterisk.c: Simplify WELCOME_MESSAGE to be a single function
- call instead of one for each line.
-
- * include/asterisk/cli.h, main/asterisk.c, main/cli.c: - Constify
- the format string passed to ast_cli() - Simplify printing out the
- warranty and license
-
- * main/dial.c, /, include/asterisk/dial.h: Merged revisions 54103
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r54103 | russell | 2007-02-12 13:17:08 -0600 (Mon, 12 Feb 2007) |
- 2 lines Change ast_set_state_callback() to
- ast_dial_set_state_callback() ........
-
- * main/dial.c, /, apps/app_meetme.c, apps/app_page.c,
- include/asterisk/dial.h: Merged revisions 54066 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r54066 | russell | 2007-02-12 11:58:43 -0600 (Mon, 12 Feb 2007) |
- 4 lines - Add the ability to register a callback to monitor state
- changes in an asynchronous dial operation. - Rename the various
- references to "status" to "state" in the dial API ........
-
-2007-02-12 15:48 +0000 [r54003-54004] Russell Bryant <russell@digium.com>
-
- * configs/users.conf.sample, /: Merged revisions 54002 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r54002 | russell | 2007-02-12 10:38:39 -0500 (Mon, 12
- Feb 2007) | 2 lines Fix a typo where "vmpassword" should be
- "vmsecret" ........
-
- * main/channel.c: Simplify a small bit of logic.
-
-2007-02-12 02:44 +0000 [r53980] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_realtime.c: Formatting fixes
-
-2007-02-11 20:49 +0000 [r53914-53953] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Be careful with debug messages in trunk,
- they tend to stay around for release....
-
- * channels/chan_sip.c: Small fix in outbound proxy support.
-
- * channels/chan_sip.c, configs/sip.conf.sample: Add support for
- outbound proxy for peers and [general] This replaces the older,
- broken, implementation where a setting in [general] did not do
- anything and the [peer] part was broken.
-
- * main/acl.c: Fix debug handling in acl.c
-
-2007-02-10 09:23 +0000 [r53882-53885] Paul Cadach <paul@odt.east.telecom.kz>
-
- * /, channels/chan_h323.c: Merged revisions 53881 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r53881 | pcadach | 2007-02-10 01:09:49 -0800 (Сбт, 10 Фев 2007) |
- 1 line Fix VLDTMF reception ........
-
- * /, apps/app_echo.c: Merged revisions 53880 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r53880 | pcadach | 2007-02-10 01:08:55 -0800 (Сбт, 10 Фев 2007) |
- 1 line Much simpler than previous one ;-) ........
-
- * main/channel.c, /: Merged revisions 53879 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r53879 | pcadach | 2007-02-10 01:07:11 -0800 (Сбт, 10 Фев 2007) |
- 1 line Provide correct DTMF duration ........
-
-2007-02-10 06:14 +0000 [r53851] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, configure, configure.ac: Merged revisions 53850 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r53850 | kpfleming | 2007-02-10 00:06:08 -0600 (Sat, 10
- Feb 2007) | 3 lines don't display the --with-imap message unless
- --with-imap was specified without a path use '-n' instead of '!
- -z' for tests ........
-
-2007-02-10 00:42 +0000 [r53784-53819] Russell Bryant <russell@digium.com>
-
- * include/asterisk/app.h, include/asterisk/utils.h, main/dial.c, /,
- apps/app_meetme.c, channels/chan_sip.c, doc/sla.txt (added),
- include/asterisk/dial.h, configs/sla.conf.sample: Merged
- revisions 53810 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r53810 | russell | 2007-02-09 18:35:09 -0600 (Fri, 09 Feb 2007) |
- 24 lines Merge team/russell/sla_rewrite This is a completely new
- implementation of the SLA functionality introduced in Asterisk
- 1.4. It is now functional and ready for testing. However, I will
- be adding some additional features over the next week, as well.
- For information on how to set this up, see
- configs/sla.conf.sample and doc/sla.txt. In addition to the
- changes in app_meetme.c for the SLA implementation itself, this
- merge brings in various other changes: chan_sip: - Add the
- ability to indicate HOLD state in NOTIFY messages. - Queue HOLD
- and UNHOLD control frames even if the channel is not bridged to
- another channel. linkedlists.h: - Add support for rwlock based
- linked lists. dial.c: - Add the ability to run ast_dial_start()
- without a reference channel to inherit information from. ........
-
- * channels/chan_jingle.c: add another dependency
-
- * /, apps/app_echo.c: Merged revisions 53783 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r53783 | russell | 2007-02-09 18:15:50 -0600 (Fri, 09 Feb 2007) |
- 4 lines When the Echo() application receives the digit '#', echo
- that back as well. Since we already sent the BEGIN frame for that
- digit, it makes sense to send the END as well. ........
-
-2007-02-09 23:53 +0000 [r53782] Kevin P. Fleming <kpfleming@digium.com>
-
- * build_tools/get_moduleinfo, res/res_config_odbc.c, /,
- build_tools/get_makeopts, funcs/func_odbc.c, res/res_adsi.c,
- channels/chan_gtalk.c, apps/app_adsiprog.c, apps/app_voicemail.c:
- Merged revisions 53779-53781 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r53779 | kpfleming | 2007-02-09 17:51:29 -0600 (Fri, 09 Feb 2007)
- | 2 lines fix awk scripts to work when both MODULEINFO and
- MAKEOPTS are present in a source file ........ r53780 | kpfleming
- | 2007-02-09 17:51:41 -0600 (Fri, 09 Feb 2007) | 2 lines add some
- inter-module dependencies ........ r53781 | kpfleming |
- 2007-02-09 17:52:44 -0600 (Fri, 09 Feb 2007) | 2 lines another
- dependency ........
-
-2007-02-09 19:39 +0000 [r53717-53750] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 53749 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r53749 | file | 2007-02-09 14:33:31 -0500 (Fri, 09 Feb 2007) | 2
- lines Temporarily change musicclass on channel to one specified
- in Dial so that the 'm' option functions properly. (issue #8969
- reported by christianbee) ........
-
- * apps/app_queue.c: Clean up documentation of Queue application.
- (issue #9022 reported by seanbright)
-
-2007-02-09 16:43 +0000 [r53716] Kevin P. Fleming <kpfleming@digium.com>
-
- * doc/imapstorage.txt, /, configure, configure.ac: Merged revisions
- 53715 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r53715 | kpfleming | 2007-02-09 10:42:22 -0600 (Fri, 09 Feb 2007)
- | 2 lines clarify the fact that voicemail IMAP storage cannot be
- built against a distro's binary c-client library package (at
- least not at this time) ........
-
-2007-02-09 01:57 +0000 [r53602-53691] Joshua Colp <jcolp@digium.com>
-
- * res/res_musiconhold.c: I'm crazy so I think I'll change the
- musiconhold classes linked list to read/write as well!
-
- * main/manager.c: It is with pleasure that I announce the return of
- rawman support through the HTTP server. (issue #9013 reported by
- Jynger)
-
- * /, apps/app_speech_utils.c: Merged revisions 53601 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r53601 | file | 2007-02-08 12:54:32 -0500 (Thu, 08 Feb
- 2007) | 2 lines Fix timeout issue when utterance is longer then
- timeout itself. ........
-
-2007-02-08 17:19 +0000 [r53580] Jason Parker <jparker@digium.com>
-
- * channels/chan_sip.c: Rename this instance of "busy limit" to
- "busy level" as well
-
-2007-02-08 16:41 +0000 [r53577] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_sip.c, configs/sip.conf.sample: rename busy-limit
- to busy-level, since it is not a limit actually parse the
- busy-limit option from sip.conf, instead of ignoring it
-
-2007-02-08 13:50 +0000 [r53531-53533] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/loader.c: Merged revisions 53532 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r53532 | tilghman | 2007-02-08 07:47:54 -0600 (Thu, 08 Feb 2007)
- | 2 lines Issue 9007 - Mutex not released on early return
- ........
-
- * /, apps/app_voicemail.c: Merged revisions 53530 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r53530 | tilghman | 2007-02-08 07:40:02 -0600
- (Thu, 08 Feb 2007) | 10 lines Merged revisions 53529 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r53529 | tilghman | 2007-02-08 07:36:10 -0600 (Thu, 08
- Feb 2007) | 2 lines Issue 9003 - If fullname is empty, quote()
- passes back "\"" ........ ................
-
-2007-02-07 23:56 +0000 [r53465-53498] Russell Bryant <russell@digium.com>
-
- * /, main/db1-ast/Makefile: Merged revisions 53497 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r53497 | russell | 2007-02-07 17:52:45 -0600 (Wed, 07
- Feb 2007) | 6 lines When building libdb1.a, put the additional
- flags needed at the beginning of ASTCFLAGS, instead of at the
- end. This way, we ensure that we find the local headers first
- before accidentally trying to use headers that exist in locations
- specified in the ASTCFLAGS passed from the main Makefile. (issue
- #8637, ovi) ........
-
- * /, main/Makefile: Merged revisions 53464 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r53464 | russell | 2007-02-07 14:07:39 -0600 (Wed, 07 Feb 2007) |
- 4 lines The clean target actually needs to run "distclean" on
- editline. This is because we need to make sure that its configure
- script gets executed again, because the CFLAGS we want to pass to
- editline may have changed. ........
-
-2007-02-07 17:57 +0000 [r53435] Joshua Colp <jcolp@digium.com>
-
- * /, main/rtp.c: Merged revisions 53434 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r53434 | file | 2007-02-07 12:53:03 -0500 (Wed, 07 Feb 2007) | 2
- lines We can not reliably do P2P bridging with DTMF passing back
- with compensation if we need to listen for DTMF frames. (issue
- #8962 reported by caio1982) ........
-
-2007-02-07 17:46 +0000 [r53431] Russell Bryant <russell@digium.com>
-
- * /, main/rtp.c: Merged revisions 53429 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r53429 | russell | 2007-02-07 11:39:31 -0600 (Wed, 07 Feb 2007) |
- 7 lines When parsing the NTP timestamp in a sender report
- message, you are supposed to take the low 16 bits of the integer
- part, and the high 16 bits of the fractional part. However, the
- code here was erroneously taking the low 16 bits of the
- fractional part. It then shifted the result 16 bits down, so the
- result was always zero. This fix makes it grab the appropriate
- high 16 bits, instead. (issue #8991, pointed out by
- andre_abrantes) ........
-
-2007-02-07 17:06 +0000 [r53359-53400] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_playback.c: Merged revisions 53399 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r53399 | file | 2007-02-07 12:04:44 -0500 (Wed, 07 Feb 2007) | 2
- lines Directly load say.conf in load_module instead of calling
- the reload function. (issue #8946 reported by junky) ........
-
- * /, channels/chan_iax2.c: Merged revisions 53358 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r53358 | file | 2007-02-07 10:43:39 -0500 (Wed,
- 07 Feb 2007) | 10 lines Merged revisions 53357 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r53357 | file | 2007-02-07 10:38:48 -0500 (Wed, 07 Feb 2007) | 2
- lines Fix a few potential memory leaks with realtime users and
- peers. (issue #8999 reported by bsmithurst) ........
- ................
-
-2007-02-07 15:35 +0000 [r53356] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_macro.c: Merged revisions 53355 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r53355 | tilghman | 2007-02-07 09:33:51 -0600
- (Wed, 07 Feb 2007) | 10 lines Merged revisions 53354 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r53354 | tilghman | 2007-02-07 09:30:02 -0600 (Wed, 07
- Feb 2007) | 2 lines Issue 7440 - Macro called from Macro from the
- h extension exits prematurely ........ ................
-
-2007-02-07 09:51 +0000 [r53334] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged
- revisions 53324 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r53324 | crichter | 2007-02-07 10:22:44 +0100
- (Mi, 07 Feb 2007) | 9 lines Merged revisions 52843 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r52843 | crichter | 2007-01-30 15:38:08 +0100 (Di, 30
- Jan 2007) | 1 line fixed some possible segfaults. also fixed an
- very important bug which occurs on high load (when calls are very
- fast generated) ........ ................
-
-2007-02-07 05:25 +0000 [r53247-53297] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_jabber.c: Merged revisions 53294 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r53294 | tilghman | 2007-02-06 23:24:31 -0600 (Tue, 06 Feb 2007)
- | 2 lines Text fix for jabber reload command (reported by bkruse
- via IRC) ........
-
- * main/manager.c, /: Merged revisions 53246 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r53246 | tilghman | 2007-02-06 01:00:52 -0600
- (Tue, 06 Feb 2007) | 10 lines Merged revisions 53245 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r53245 | tilghman | 2007-02-06 00:58:28 -0600 (Tue, 06
- Feb 2007) | 2 lines Issue 8987 - Status could return two
- responses (mnicholson) ........ ................
-
-2007-02-05 21:55 +0000 [r53200] Olle Johansson <oej@edvina.net>
-
- * main/io.c: Doxygen formatting changes
-
-2007-02-05 17:06 +0000 [r53151-53153] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_playback.c: Merged revisions 53152 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r53152 | file | 2007-02-05 11:06:18 -0600 (Mon, 05 Feb 2007) | 2
- lines Ensure say_cfg is NULL when the module is loaded. (issue
- #8946 reported by junky) ........
-
- * /, apps/app_playback.c: Merged revisions 53150 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r53150 | file | 2007-02-05 10:02:00 -0600 (Mon, 05 Feb 2007) | 2
- lines Unregister Playback CLI commands as well as dialplan
- application. (issue #8946 reported by junky) ........
-
-2007-02-05 00:30 +0000 [r53144] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 53143 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r53143 | oej | 2007-02-05 01:18:34 +0100 (Mon, 05 Feb 2007) | 3
- lines Add some comments on queue system behaviour and how it
- affects the SIP channel ........
-
-2007-02-03 22:06 +0000 [r53140-53142] Tilghman Lesher <tlesher@digium.com>
-
- * UPGRADE.txt: Deprecate SetCallerPres application
-
- * apps/app_setcallerid.c, funcs/func_callerid.c: Add CALLERPRES
- dialplan function and deprecate SetCallerPres application
-
- * funcs/func_odbc.c: Fix compiler warnings
-
-2007-02-03 21:06 +0000 [r53139] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 53138 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r53138 | file | 2007-02-03 15:05:02 -0600 (Sat, 03 Feb 2007) | 2
- lines Make SIPDtmfMode application work with recent capability
- changes, and also fix an RTP stack issue when the auto option was
- used. (issue #8972 reported by mdu113) ........
-
-2007-02-03 20:46 +0000 [r53137] Russell Bryant <russell@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 53136 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r53136 | russell | 2007-02-03 14:44:20 -0600
- (Sat, 03 Feb 2007) | 12 lines Merged revisions 53133 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r53133 | russell | 2007-02-03 14:38:13 -0600 (Sat, 03
- Feb 2007) | 4 lines set the DIALSTATUS variable to contain
- "INVALIDARGS" when the dial application exits early because of
- invalid arguments instead of just leaving it empty. (issue #8975)
- ........ ................
-
-2007-02-03 10:12 +0000 [r53132] Paul Cadach <paul@odt.east.telecom.kz>
-
- * /, channels/h323/ast_h323.cxx: Merged revisions 53131 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r53131 | pcadach | 2007-02-03 02:02:55 -0800 (Сбт, 03 Фев 2007) |
- 1 line Remove quote from H.323 vendor string because due to
- compatibilities with Nortel Meridian CS1000 reported at
- www.voip-info.org ........
-
-2007-02-02 20:05 +0000 [r53126-53127] Olle Johansson <oej@edvina.net>
-
- * doc/queue.txt: Update with info about SIP channels and queues
-
- * doc/queue.txt (added): Adding a template for documentation on
- call queues. Please help us add to this! Thanks /OEJ and BJ
-
-2007-02-02 18:21 +0000 [r53111-53125] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Add onHold value to sip show inuse as well.
-
- * /, main/rtp.c: Merged revisions 53120 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r53120 | file | 2007-02-02 11:15:22 -0600 (Fri, 02 Feb 2007) | 2
- lines Correct a copy/pasted error message line for RTCP. ........
-
- * /, main/config.c: Merged revisions 53118 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r53118 | file | 2007-02-02 10:59:53 -0600 (Fri,
- 02 Feb 2007) | 10 lines Merged revisions 53117 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r53117 | file | 2007-02-02 10:58:09 -0600 (Fri, 02 Feb 2007) | 2
- lines Pass the glob expanded filename to process_text_line so
- that error messages contain the actual filename, not the original
- include one. (issue #8959 reported by tzafrir) ........
- ................
-
- * Makefile, /: Merged revisions 53114 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r53114 | file | 2007-02-02 09:29:35 -0600 (Fri, 02 Feb 2007) | 2
- lines Add systemname to asterisk.conf generation per recent
- discussions about it. (issue #8968 reported by blitzrage)
- ........
-
- * main/devicestate.c: Clean up ast_device_state. It's pretty now!
-
- * main/devicestate.c: Switch the devicestate thread to operate the
- same way as the logging thread. Pops all entries off the list to
- be processed, resets the list back to a clean state, and
- processes each entry. The thread won't have to acquire the list
- lock again until it checks to see if there are more to process.
-
- * main/devicestate.c: Read/write lockify the devicestate stuff a
- bit.
-
-2007-02-02 00:26 +0000 [r53110] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c, configs/sip.conf.sample: Patch based on
- this patch with small changes for trunk... Merged revisions 53109
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r53109 | oej | 2007-02-02 01:24:03 +0100 (Fri, 02 Feb 2007) | 4
- lines Disable the direct p2p RTP call setup in SIP. You can
- enable it in sip.conf, but it is now considered experimental
- until we solve the AST_CONTROL_ANSWER with payload and videocaps
- stuff. ........
-
-2007-02-01 22:26 +0000 [r53098-53105] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 53104 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r53104 | file | 2007-02-01 16:24:32 -0600 (Thu,
- 01 Feb 2007) | 10 lines Merged revisions 53103 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r53103 | file | 2007-02-01 16:21:56 -0600 (Thu, 01 Feb 2007) | 2
- lines Copy noncodeccapability over to the joint variable so that
- telephone-event will get transmitted in the sent INVITE. ........
- ................
-
- * /, channels/chan_sip.c: Merged revisions 53097 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r53097 | file | 2007-02-01 15:54:28 -0600 (Thu,
- 01 Feb 2007) | 10 lines Merged revisions 53095 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r53095 | file | 2007-02-01 15:47:11 -0600 (Thu, 01 Feb 2007) | 2
- lines Don't negotiate RFC2833 when not configured to do so.
- (issue #8799 reported by mdu113) ........ ................
-
-2007-02-01 21:27 +0000 [r53094] Russell Bryant <russell@digium.com>
-
- * /, funcs/func_strings.c: Merged revisions 53093 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r53093 | russell | 2007-02-01 15:24:52 -0600 (Thu, 01 Feb 2007) |
- 2 lines Fix the FIELDQTY function to not crash. (reported by
- blitzrage and Corydon on IRC) ........
-
-2007-02-01 21:17 +0000 [r53092] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 53085 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r53085 | oej | 2007-02-01 22:05:34 +0100 (Thu, 01 Feb 2007) | 4
- lines - Clean INC_COUNT flag when we decrement call counter - If
- it's still set at time of dialog destruction, make sure we
- decrement the device call counter properly before we destroy the
- dialog ........
-
-2007-02-01 21:12 +0000 [r53087-53089] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 53088 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r53088 | file | 2007-02-01 15:11:28 -0600 (Thu,
- 01 Feb 2007) | 10 lines Merged revisions 53084 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r53084 | file | 2007-02-01 15:03:10 -0600 (Thu, 01 Feb 2007) | 2
- lines Return previous behavior of having MOH pick up where it was
- left off. (issue #8672 reported by sinistermidget) ........
- ................
-
-2007-02-01 20:44 +0000 [r53080-53083] Olle Johansson <oej@edvina.net>
-
- * /, apps/app_queue.c: Merged revisions 53081 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r53081 | oej | 2007-02-01 21:38:58 +0100 (Thu, 01 Feb 2007) | 2
- lines Change debug level for state change message that is not
- really informative when debugging app_queue ........
-
- * channels/chan_sip.c, configs/sip.conf.sample: Implementing
- "busy-limit". If you set call limit and busy limit, chan_sip will
- indicate BUSY for a device that has reached the busy limit and
- allow calls up to the call limit, allowing for call transfers
- (that generate a new call). If you only set call limit, chan_sip
- will not indicate BUSY until that limit is filled. This affects
- SIP subscriptions, call queues and manager applications.
-
- * /, channels/chan_sip.c: Merged revisions 53079 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r53079 | oej | 2007-02-01 21:28:54 +0100 (Thu, 01 Feb 2007) | 2
- lines Cleaning up the devicestate callback function ........
-
-2007-02-01 20:14 +0000 [r53076-53078] Tilghman Lesher <tlesher@digium.com>
-
- * /, funcs/func_strings.c: Merged revisions 53075 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r53075 | tilghman | 2007-02-01 14:09:52 -0600
- (Thu, 01 Feb 2007) | 10 lines Merged revisions 53074 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r53074 | tilghman | 2007-02-01 14:07:35 -0600 (Thu, 01
- Feb 2007) | 2 lines Bug 8965 - Allow FIELDQTY to work with both
- variables and dialplan functions ........ ................
-
-2007-02-01 19:34 +0000 [r53073] Joshua Colp <jcolp@digium.com>
-
- * /, main/asterisk.c: Merged revisions 53072 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r53072 | file | 2007-02-01 13:33:33 -0600 (Thu, 01 Feb 2007) | 2
- lines Add missing 'F' letter to getopt so it magically becomes a
- valid option. (issue #8960 reported by tzafrir) ........
-
-2007-02-01 19:27 +0000 [r53071] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c, /, funcs/func_strings.c: Merged revisions 53070 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r53070 | tilghman | 2007-02-01 13:21:20 -0600
- (Thu, 01 Feb 2007) | 10 lines Merged revisions 53069 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r53069 | tilghman | 2007-02-01 13:13:53 -0600 (Thu, 01
- Feb 2007) | 2 lines No wonder FIELDQTY doesn't work with
- functions... the documentation in pbx.c was wrong ........
- ................
-
-2007-02-01 19:04 +0000 [r53067] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Signal HOLD status to phones that subscribe
- for status.
-
-2007-02-01 17:42 +0000 [r53065-53066] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 53064 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r53064 | file | 2007-02-01 11:37:44 -0600 (Thu, 01 Feb 2007) | 2
- lines Fix silly logic. We really want to write UDPTL frames out
- when the call is up. ........
-
- * main/db1-ast/hash/hash.c: Make trunk compile under dev mode.
-
-2007-02-01 16:42 +0000 [r53063] Olle Johansson <oej@edvina.net>
-
- * /, configs/sip.conf.sample: Merged revisions 53062 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r53062 | oej | 2007-02-01 17:35:12 +0100 (Thu, 01 Feb
- 2007) | 2 lines Add explanation of port= in combination with
- defaultip= (thanks jsmith) ........
-
-2007-02-01 14:43 +0000 [r53061] Russell Bryant <russell@digium.com>
-
- * apps/app_rpt.c: Remove duplicate calls to pthread_attr_destroy()
- that I put in yesterday by accident.
-
-2007-02-01 11:16 +0000 [r53058-53059] Paul Cadach <paul@odt.east.telecom.kz>
-
- * /, channels/chan_h323.c: Oops -- Merged revisions 53057 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r53057 | pcadach | 2007-02-01 03:07:41 -0800 (Чтв, 01 Фев 2007) |
- 1 line chan_h323 is very stable, so let it built by default
- ........
-
-2007-02-01 00:38 +0000 [r53054] Olle Johansson <oej@edvina.net>
-
- * res/res_features.c: Formatting changes
-
-2007-02-01 00:24 +0000 [r53051-53053] Joshua Colp <jcolp@digium.com>
-
- * /, main/rtp.c: Merged revisions 53052 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r53052 | file | 2007-01-31 18:24:20 -0600 (Wed, 31 Jan 2007) | 2
- lines When going on hold have the side that was put on hold
- reinvite back to Asterisk. When going off hold have the side that
- was taken off hold reinvited back to the other party. ........
-
- * /, main/rtp.c: Merged revisions 53050 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r53050 | file | 2007-01-31 18:19:48 -0600 (Wed, 31 Jan 2007) | 2
- lines Add more frame types to forward in the RTP bridge loops.
- ........
-
-2007-01-31 21:35 +0000 [r52905-53047] Russell Bryant <russell@digium.com>
-
- * main/pbx.c, channels/chan_zap.c, /, apps/app_meetme.c,
- channels/chan_sip.c, apps/app_queue.c, channels/chan_iax2.c,
- main/cdr.c, main/manager.c, pbx/pbx_spool.c,
- channels/chan_skinny.c, channels/chan_h323.c, main/http.c,
- pbx/pbx_dundi.c, apps/app_rpt.c, channels/chan_mgcp.c: Merged
- revisions 53046 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r53046 | russell | 2007-01-31 15:32:08 -0600
- (Wed, 31 Jan 2007) | 11 lines Merged revisions 53045 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r53045 | russell | 2007-01-31 15:25:11 -0600 (Wed, 31
- Jan 2007) | 3 lines Fix a bunch of places where
- pthread_attr_init() was called, but pthread_attr_destroy() was
- not. ........ ................
-
- * /, apps/app_userevent.c: Merged revisions 53042 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r53042 | russell | 2007-01-31 12:18:25 -0600 (Wed, 31 Jan 2007) |
- 2 lines Remove an extra \r\n from manager user events. (issue
- #8955, mnicholson) ........
-
- * /, main/rtp.c: Merged revisions 53040 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r53040 | russell | 2007-01-31 11:45:05 -0600
- (Wed, 31 Jan 2007) | 11 lines Merged revisions 53039 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r53039 | russell | 2007-01-31 11:41:51 -0600 (Wed, 31
- Jan 2007) | 3 lines Use the proper format string to print
- unsigned values in the rtp debug output. (issue #8954, wmis)
- ........ ................
-
- * /, apps/app_queue.c: Merged revisions 53037 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r53037 | russell | 2007-01-31 11:39:28 -0600 (Wed, 31 Jan 2007) |
- 3 lines Only changed the paused status in an existing queue
- member if the paused column exists. ........
-
- * /, apps/app_queue.c: Merged revisions 53035 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r53035 | russell | 2007-01-31 11:34:22 -0600 (Wed, 31 Jan 2007) |
- 4 lines Instead of always creating a realtime queue member as
- unpaused, read the "paused" column and use that value for the
- paused status of the member. (issue #8949, jmls) ........
-
- * /, contrib/init.d/rc.suse.asterisk: Merged revisions 53001 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r53001 | russell | 2007-01-30 17:38:42 -0600 (Tue, 30 Jan 2007) |
- 2 lines Update init script for SuSE 10. (issue #8363, johnlange)
- ........
-
- * /, doc/cdrdriver.txt: Merged revisions 52999 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r52999 | russell | 2007-01-30 17:30:34 -0600 (Tue, 30 Jan 2007) |
- 2 lines Add documentation for using cdr_pgsql. (issue #8942,
- lters) ........
-
- * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
- codecs/codec_gsm.c: Merged revisions 52997 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r52997 | russell | 2007-01-30 17:23:24 -0600 (Tue, 30 Jan 2007) |
- 5 lines When we are checking for a system installed version of
- libgsm, we need to check for gsm.h as well. Furthermore, when
- checking for this header, it may be located in a gsm/ sub
- directory, so check for that, as well. (issue #8773) ........
-
- * /, channels/chan_sip.c: Merged revisions 52952 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r52952 | russell | 2007-01-30 13:33:12 -0600 (Tue, 30 Jan 2007) |
- 5 lines Only set the DTMF flag on the rtp structure if the DTMF
- mode is actually RFC2833, not just that it is not INFO. This
- makes it get set for inband DTMF as well, which is not valid.
- (issue #8936) ........
-
- * /, main/asterisk.c: Merged revisions 52904 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r52904 | russell | 2007-01-30 11:19:39 -0600
- (Tue, 30 Jan 2007) | 17 lines Merged revisions 52903 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r52903 | russell | 2007-01-30 11:12:04 -0600 (Tue, 30
- Jan 2007) | 9 lines The SIGHUP handler was implemented to allow
- admins to send SIGHUP to a running Asterisk process to reload the
- configuration. However, doing the actual reload in the signal
- handler itself is a very bad thing to do, because the reload
- process includes calling non-reentrant functions such as
- malloc/calloc/etc. If Asterisk is running in the background, then
- the reload will happen immediately. However, if running in
- console mode, the reload doesn't work until something is typed at
- the console. That sort of defeats the purpose, but I don't see an
- easy way to get around it at this point. ........
- ................
-
-2007-01-30 15:39 +0000 [r52858-52860] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Use provided variable for name instead of
- one in the structure since the structure was just allocated and
- will be NULL. (issue #8938 reported by st41ker)
-
-2007-01-30 09:13 +0000 [r52818-52820] Paul Cadach <paul@odt.east.telecom.kz>
-
- * /, res/res_odbc.c: Merged revisions 52808 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r52808 | pcadach | 2007-01-30 00:34:26 -0800 (Втр, 30 Янв 2007) |
- 1 line Don't play with free()'d pointers ........
-
- * /, configure, acinclude.m4: Merged revisions 52807 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r52807 | pcadach | 2007-01-30 00:33:22 -0800 (Втр, 30
- Янв 2007) | 1 line Handle non-standard OpenH323/PWLib library
- names ........
-
-2007-01-30 00:16 +0000 [r52764] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 52763 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r52763 | russell | 2007-01-29 18:15:50 -0600
- (Mon, 29 Jan 2007) | 13 lines Merged revisions 52762 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r52762 | russell | 2007-01-29 18:15:06 -0600 (Mon, 29
- Jan 2007) | 5 lines Fix the extraction of the timestamp from
- video frames. It was using the mapping for a mini-frame instead
- of a video-frame, which caused it to get invalid data. (issue
- #8795, mihai) ........ ................
-
-2007-01-29 23:45 +0000 [r52718] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_mixmonitor.c: Merged revisions 52717 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r52717 | file | 2007-01-29 18:43:40 -0500 (Mon,
- 29 Jan 2007) | 10 lines Merged revisions 52716 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r52716 | file | 2007-01-29 18:39:39 -0500 (Mon, 29 Jan 2007) | 2
- lines Now that filename is part of the structure and since it
- comes before postprocess... we have to add it to our postprocess
- line. (reported on asterisk-dev by Boris Bakchiev) ........
- ................
-
-2007-01-29 22:58 +0000 [r52692-52696] Russell Bryant <russell@digium.com>
-
- * /, main/Makefile: Merged revisions 52695 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r52695 | russell | 2007-01-29 16:58:09 -0600 (Mon, 29 Jan 2007) |
- 2 lines Add a missing quotation mark. This was pointed out by
- jcmoore on #asterisk-dev. ........
-
- * main/manager.c, /: Merged revisions 52688 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r52688 | russell | 2007-01-29 16:55:41 -0600 (Mon, 29 Jan 2007) |
- 3 lines Remove a recursive lock of the manager session. This was
- pointed out by zandbelt in issue #8711. ........
-
-2007-01-29 22:13 +0000 [r52680] Tilghman Lesher <tlesher@digium.com>
-
- * /, pbx/pbx_config.c: Merged revisions 52679 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r52679 | tilghman | 2007-01-29 16:12:12 -0600 (Mon, 29 Jan 2007)
- | 2 lines Argument number correction ........
-
-2007-01-29 21:37 +0000 [r52646-52648] Russell Bryant <russell@digium.com>
-
- * /, main/Makefile: Merged revisions 52647 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r52647 | russell | 2007-01-29 15:36:56 -0600 (Mon, 29 Jan 2007) |
- 3 lines ASTLDFLAGS needs to be passed to the editline configure
- script as LDFLAGS. (issue #8928, zandbelt) ........
-
- * /, main/rtp.c: Merged revisions 52645 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r52645 | russell | 2007-01-29 15:26:27 -0600 (Mon, 29 Jan 2007) |
- 6 lines Fix a problem with packet-to-packet bridging and DTMF
- mode translation. P2P bridging can only be used when the DTMF
- modes don't match if the core is monitoring DTMF in both
- directions. Then, the core will handle the translation.
- Otherwise, this bridging method can not be used. (issue #8936)
- ........
-
-2007-01-29 21:03 +0000 [r52635] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Only use locking for bridge information if intense
- P2P bridging is enabled.
-
-2007-01-29 20:51 +0000 [r52612-52613] Russell Bryant <russell@digium.com>
-
- * main/manager.c, /: The changes for trunk are less extensive, but
- include - changing the actionlock to a rwlock - not locking the
- session before doing the action callback The crash issue in 8711
- should not be an issue here. Merged revisions 52611 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r52611 | russell | 2007-01-29 14:39:20 -0600 (Mon, 29
- Jan 2007) | 10 lines The session lock can not be held while
- calling action callbacks. If so, then when the WaitEvent callback
- gets called, then no event can happen because the session can't
- be locked by another thread. Also, the session needs to be locked
- in the HTTP callback when it reads out the output string. This
- fixes the deadlock reported in both 8711 and 8934. Regarding
- issue 8711, there still may be an issue. If there is a second
- action requested before the processing of the first action is
- finished, there could still be some corruption of the output
- string buffer used to build the result. (issue #8711, #8934)
- ........
-
- * apps/app_voicemail.c: Resolve some warnings when not building
- with IMAP_STORAGE
-
-2007-01-29 20:22 +0000 [r52580-52610] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c: Change vmstates list to use linked list
- macros.
-
- * apps/app_voicemail.c: Code cleanup of IMAP storage support in
- app_voicemail.
-
- * /, apps/app_voicemail.c: Merged revisions 52572 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r52572 | file | 2007-01-29 13:59:41 -0500 (Mon, 29 Jan 2007) | 2
- lines Use ast_calloc instead of malloc. ........
-
-2007-01-29 17:49 +0000 [r52524-52525] Joshua Colp <jcolp@digium.com>
-
- * CHANGES, main/cli.c: Add core show channels count CLI command.
- (issue #8932 reported by mr_mehul_shah)
-
- * /, apps/app_voicemail.c: Merged revisions 52523 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r52523 | file | 2007-01-29 12:33:19 -0500 (Mon, 29 Jan 2007) | 2
- lines Set quota information to 0 when creating a vm_state. (issue
- #8924 reported by neutrino88) ........
-
-2007-01-29 17:03 +0000 [r52522] Russell Bryant <russell@digium.com>
-
- * /, main/jitterbuf.c, include/jitterbuf.h: Merged revisions
- 52494,52506 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r52494 | jdixon | 2007-01-28 22:18:36 -0600 (Sun, 28 Jan 2007) |
- 4 lines Fixed problem with jitterbuf, whereas it would not
- complain about, and would allow itself to be overfilled (per the
- max_jitterbuf parameter). Now it rejects any data over and above
- that size, and complains about it. ........ r52506 | russell |
- 2007-01-29 10:54:27 -0600 (Mon, 29 Jan 2007) | 5 lines Clean up a
- few things in the last commit to the adaptive jitterbuffer code.
- - Specifically indicate to the compiler that the "dropem"
- variable only needs one but. - Change formatting to conform to
- coding guidelines. ........
-
-2007-01-28 05:18 +0000 [r52463] Tilghman Lesher <tlesher@digium.com>
-
- * /, configure, configure.ac: Merged revisions 52462 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r52462 | tilghman | 2007-01-27 23:15:07 -0600 (Sat, 27
- Jan 2007) | 2 lines Suggested change to fix normal usage of
- --with-tds=/usr/local (Sean Bright, via asterisk-dev mailing
- list) ........
-
-2007-01-27 02:15 +0000 [r52332-52417] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 52416 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r52416 | file | 2007-01-26 21:13:41 -0500 (Fri,
- 26 Jan 2007) | 10 lines Merged revisions 52415 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r52415 | file | 2007-01-26 21:09:10 -0500 (Fri, 26 Jan 2007) | 2
- lines Make COMPLETECALLER and COMPLETEAGENT output to queue_log
- follow documentation. (issue #7677 reported by amilcar) ........
- ................
-
- * /, channels/chan_iax2.c: Merged revisions 52370 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r52370 | file | 2007-01-26 19:08:18 -0500 (Fri,
- 26 Jan 2007) | 10 lines Merged revisions 52360 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r52360 | file | 2007-01-26 19:03:23 -0500 (Fri, 26 Jan 2007) | 2
- lines Make the last context entry read in the dominant one.
- (issue #8918 reported by pj) ........ ................
-
- * /, main/file.c: Merged revisions 52335 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r52335 | file | 2007-01-26 18:46:47 -0500 (Fri, 26 Jan 2007) | 2
- lines Fix core show file formats CLI command. ........
-
- * main/file.c, main/image.c: Convert some more stuff to read/write
- lists.
-
-2007-01-25 22:49 +0000 [r52168-52308] Joshua Colp <jcolp@digium.com>
-
- * CHANGES, main/db.c: Add DBDel and DBDelTree manager commands.
- (issue #8516 reported by dprado)
-
- * /, main/jitterbuf.c: Merged revisions 52265 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r52265 | file | 2007-01-25 14:18:33 -0500 (Thu,
- 25 Jan 2007) | 10 lines Merged revisions 52264 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r52264 | file | 2007-01-25 14:15:29 -0500 (Thu, 25 Jan 2007) | 2
- lines Allow dequeueing of frames with negative timestamp by
- moving jitterbuffer frames check to jb_next. (issue #8546
- reported by harmen) ........ ................
-
- * channels/chan_sip.c: Use atomic operation functions for
- use/ringing/hold manipulation.
-
- * /, channels/chan_sip.c: Merged revisions 52210 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r52210 | file | 2007-01-25 12:49:39 -0500 (Thu, 25 Jan 2007) | 2
- lines Drop out variables I accidentally put in. ........
-
- * /, channels/chan_sip.c: Merged revisions 52208 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r52208 | file | 2007-01-25 12:14:53 -0500 (Thu, 25 Jan 2007) | 2
- lines Decrement onHold count if we are hung up on and still on
- hold. (issue #8909 reported by alexh42) ........
-
- * /, apps/app_mixmonitor.c: Merged revisions 52163 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r52163 | file | 2007-01-24 20:51:35 -0500 (Wed,
- 24 Jan 2007) | 10 lines Merged revisions 52162 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r52162 | file | 2007-01-24 20:48:52 -0500 (Wed, 24 Jan 2007) | 2
- lines Add another note about audio files being played back to
- each bridged party. (issue #8718 reported by ppyy) ........
- ................
-
-2007-01-25 01:38 +0000 [r52108-52161] Russell Bryant <russell@digium.com>
-
- * configs/users.conf.sample, /, apps/app_voicemail.c: Merged
- revisions 52160 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r52160 | russell | 2007-01-24 19:37:16 -0600 (Wed, 24 Jan 2007) |
- 2 lines By suggestion from kpfleming last week, change
- "vmpassword" to "vmsecret". ........
-
- * /, include/asterisk/dial.h: Merged revisions 52107 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r52107 | russell | 2007-01-24 15:42:47 -0600 (Wed, 24
- Jan 2007) | 3 lines Fix the formatting of doxygen comments to
- properly indicate that the comment documents the previous entity,
- as opposed to the next one. ........
-
-2007-01-24 20:35 +0000 [r52053-52086] Steve Murphy <murf@digium.com>
-
- * UPGRADE.txt, apps/app_chanisavail.c: As per bug 8859 (Add option
- to revert old ChanIsAvail() with 's' option behavior), this
- update makes the 't' option available, which calls
- ast_parse_device_state instead of ast_device_state. This option
- will not dive into the channel driver to find the status of the
- device (which could be good if sip devicestate isn't returning
- full status, for various reasons).
-
- * utils/Makefile, /, utils/check_expr.c: Merged revisions 52052 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r52052 | murf | 2007-01-24 11:26:22 -0700 (Wed,
- 24 Jan 2007) | 9 lines Merged revisions 52002 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r52002 | murf | 2007-01-24 10:43:50 -0700 (Wed, 24 Jan 2007) | 1
- line updated check_expr via 8322 (refactoring of expression
- checking impl); elfring contributed a nice code reorg, I
- contributed some time to get it working again, better messages
- ........ ................
-
-2007-01-24 18:23 +0000 [r52025-52050] Joshua Colp <jcolp@digium.com>
-
- * main/dial.c (added), /, apps/app_page.c, main/Makefile,
- include/asterisk/dial.h (added): Merged revisions 52049 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r52049 | file | 2007-01-24 13:20:05 -0500 (Wed, 24 Jan 2007) | 2
- lines Merge in dialing API and the app_page that uses it. (issue
- #BE-118) ........
-
- * /, channels/chan_sip.c: Merged revisions 52016 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r52016 | file | 2007-01-24 12:59:55 -0500 (Wed, 24 Jan 2007) | 2
- lines Fix changing channel formats when joint capability changes
- and there are no audio formats... I didn't break it originally!
- (issue #8535 reported by ivoc) ........
-
-2007-01-24 09:42 +0000 [r51905-51933] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 51931 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r51931 | oej | 2007-01-24 10:30:21 +0100 (Wed, 24 Jan 2007) | 3
- lines Show capabilities *and* preference in general settings in
- "sip show settings" (reported by Clona/Telio - Thanks!) ........
-
- * include/asterisk/http.h, main/http.c: Doxygen updates
-
- * funcs/func_rand.c, funcs/func_base64.c, funcs/func_module.c,
- funcs/func_md5.c, funcs/func_db.c, funcs/func_version.c,
- funcs/func_timeout.c, funcs/func_env.c, funcs/func_math.c,
- funcs/func_strings.c, funcs/func_sha1.c, funcs/func_logic.c,
- funcs/func_uri.c, funcs/func_global.c, funcs/func_enum.c,
- funcs/func_groupcount.c, funcs/func_odbc.c, funcs/func_shell.c,
- funcs/func_channel.c, funcs/func_cdr.c, funcs/func_callerid.c:
- Doxygen update
-
- * main/udptl.c: Adding some doxygen for udptl.c
-
-2007-01-24 01:00 +0000 [r51850] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /: Merged revisions 51848 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r51848 | russell | 2007-01-23 18:59:58 -0600
- (Tue, 23 Jan 2007) | 14 lines Merged revisions 51843 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r51843 | russell | 2007-01-23 18:57:28 -0600 (Tue, 23
- Jan 2007) | 6 lines Fix an issue related to synchronization of
- recordings when using Monitor(). The bug is a miscalculation of
- the amount to seek the stream for writing to disk when the number
- of samples coming in and out of a channel do not match up. (issue
- #8298, #8887, report and patch by guillecabeza, patch files
- created and testing done by whoiswes) ........ ................
-
-2007-01-24 00:22 +0000 [r51831] Joshua Colp <jcolp@digium.com>
-
- * main/manager.c: Close file after we do the translation, and map
- memory for both reading/writing. (issue #8886 reported by
- cwegener)
-
-2007-01-24 00:21 +0000 [r51830] Russell Bryant <russell@digium.com>
-
- * /, apps/app_while.c: Merged revisions 51829 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r51829 | russell | 2007-01-23 18:19:55 -0600
- (Tue, 23 Jan 2007) | 12 lines Merged revisions 51828 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r51828 | russell | 2007-01-23 18:17:50 -0600 (Tue, 23
- Jan 2007) | 4 lines Don't set a new value for the END_ variable
- on the channel before using the old value. If you do, it will
- lead to accessing a memory address that has been free()'d. (issue
- #8895, arkadia) ........ ................
-
-2007-01-23 22:59 +0000 [r51801] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_phone.c, channels/chan_zap.c, /,
- channels/chan_sip.c, channels/chan_skinny.c,
- channels/chan_features.c, channels/chan_alsa.c,
- channels/chan_gtalk.c, channels/chan_iax2.c, channels/chan_oss.c:
- Merged revisions 51788 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r51788 | file | 2007-01-23 17:46:31 -0500 (Tue, 23 Jan 2007) | 2
- lines Update channel drivers to use module referencing so that
- unloading them while in use will not result in crashes. (issue
- #8897 reported by junky) ........
-
-2007-01-23 22:09 +0000 [r51751-51787] Russell Bryant <russell@digium.com>
-
- * main/manager.c, /: Merged revisions 51781 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r51781 | russell | 2007-01-23 16:04:01 -0600 (Tue, 23 Jan 2007) |
- 6 lines Fix some bugs in process_message(). The manager session
- lock needs to be held when sending some sort of response, or
- calling one of the manager action callbacks. This resolves an
- issue where people using the GUI would get random crashes when
- they start clicking around a lot. (issue #8711, reported and
- debugged by zandbelt) ........
-
- * main/manager.c, /: Merged revisions 51750 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r51750 | russell | 2007-01-23 15:33:15 -0600 (Tue, 23 Jan 2007) |
- 4 lines When traversing the list of manager actions, the iterator
- needs to be initialized to the list head *after* locking the
- list. Also, lock the actions list in one place it is being
- accessed where it was not being done. ........
-
-2007-01-23 20:36 +0000 [r51684-51717] Steve Murphy <murf@digium.com>
-
- * /, res/res_features.c: Merged revisions 51716 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r51716 | murf | 2007-01-23 13:32:54 -0700 (Tue, 23 Jan 2007) | 1
- line this mod from 8593 (dstchannel in cdr is empty when transfer
- call). ........
-
- * /, main/callerid.c: Merged revisions 51683 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r51683 | murf | 2007-01-23 11:58:27 -0700 (Tue, 23 Jan 2007) | 1
- line via 8748 (callerid.c loses name when returning
- PRIVATE_NUMBER flag), the user suggested this mod, saying it
- would allow 'WITHHELD' to appear in the name field, which would
- be useful ........
-
-2007-01-23 15:36 +0000 [r51659] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Issue #8817 - Registry corruption when
- packet retransmits fail. (tootai, patchy by oej)
-
-2007-01-23 06:56 +0000 [r51623] Paul Cadach <paul@odt.east.telecom.kz>
-
- * /, channels/chan_h323.c, channels/Makefile: Merged revisions
- 51615 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r51615 | pcadach | 2007-01-22 22:51:51 -0800 (Пнд, 22 Янв 2007) |
- 1 line Do not abort Asterisk startup if h323 configuration file
- not found (reported by mithraen) ........
-
-2007-01-23 04:45 +0000 [r51463-51592] Joshua Colp <jcolp@digium.com>
-
- * doc/externalivr.txt, apps/app_externalivr.c, CHANGES: Make 'H'
- command do as advertised and add 'E' and 'V' commands to
- ExternalIVR. (issue #8165 reported by mnicholson)
-
- * configs/iax.conf.sample, CHANGES, channels/chan_iax2.c: Add SRV
- Lookup support on outbound calls to chan_iax2. It's listed in the
- RFC so we might want to support it and please don't hurt me Marko
- ... (issue #7812 reported by drorlb)
-
- * /, channels/chan_sip.c: Merged revisions 51558 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r51558 | file | 2007-01-22 22:00:12 -0500 (Mon, 22 Jan 2007) | 2
- lines Only change audio formats on the channel if we have an
- audio format to change to. (issue #8535 reported by ivoc)
- ........
-
- * /: No more conflicts on properties! svnmerge-block be gone!
-
- * /, res/res_musiconhold.c: Merged revisions 51513 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r51513 | file | 2007-01-22 20:45:04 -0500 (Mon,
- 22 Jan 2007) | 10 lines Merged revisions 51512 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r51512 | file | 2007-01-22 20:41:35 -0500 (Mon, 22 Jan 2007) | 2
- lines Yield before reading from zaptel timing source under
- Solaris so that other threads get a chance to do things. (issue
- #7875 reported by bob) ........ ................
-
- * main/autoservice.c: Might as well go crazy here too and make the
- autoservice list read/write.
-
- * main/pbx.c, main/autoservice.c, main/frame.c, main/say.c,
- main/jitterbuf.c, main/devicestate.c, main/utils.c, main/enum.c,
- main/fskmodem.c, main/config.c, main/cli.c, main/io.c,
- main/channel.c, main/cdr.c, main/abstract_jb.c, main/logger.c,
- main/callerid.c, main/file.c, main/app.c, main/image.c,
- main/alaw.c, main/asterisk.c, main/dsp.c: Cosmetic changes. Make
- main source files better conform to coding guidelines and
- standards. (issue #8679 reported by johann8384)
-
- * main/rtp.c: Change RTP protos list to be read/write. Most of the
- time it's only going to be read so making it use mutex locks was
- a waste.
-
- * main/rtp.c: Make the RTP stack better conform to coding
- guidelines. (issue #8679 reported by johann8384)
-
-2007-01-22 19:42 +0000 [r51413] Steve Murphy <murf@digium.com>
-
- * /, pbx/pbx_ael.c: Merged revisions 51409 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r51409 | murf | 2007-01-22 12:28:51 -0700 (Mon, 22 Jan 2007) | 1
- line This fixes 8836, according to dnatural ........
-
-2007-01-22 19:22 +0000 [r51408] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_mixmonitor.c: Merged revisions 51407 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r51407 | file | 2007-01-22 14:13:44 -0500 (Mon,
- 22 Jan 2007) | 10 lines Merged revisions 51406 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r51406 | file | 2007-01-22 14:08:52 -0500 (Mon, 22 Jan 2007) | 2
- lines Move filestream creation to Mixmonitor loop. This will
- prevent a blank file from being created if no frames ever pass
- through to be recorded. (issue #7589 reported by steve_mcneil)
- ........ ................
-
-2007-01-22 19:00 +0000 [r51405] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Remove (to quote Rizzo) "useless" variable.
-
-2007-01-21 03:25 +0000 [r51353] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c: Fix bug introduced during constification (reported by
- tzanger via IRC)
-
-2007-01-20 18:27 +0000 [r51352] Russell Bryant <russell@digium.com>
-
- * include/asterisk/frame.h: Add a comment that the frame type
- constants are transmitted directly over IAX2.
-
-2007-01-20 06:54 +0000 [r51349-51351] Jason Parker <jparker@digium.com>
-
- * /, configs/say.conf.sample: Merged revisions 51350 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r51350 | qwell | 2007-01-20 00:53:49 -0600 (Sat, 20 Jan
- 2007) | 5 lines Fix Italian numeral support in say.conf for
- "_[2-9]00" case. "2131" would've translated to something along
- the lines of (pardon my..Italian {or lack thereof})
- "duecentocentotrentuno", which makes no sense at all. ........
-
- * /, configs/say.conf.sample: Merged revisions 51348 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r51348 | qwell | 2007-01-20 00:16:06 -0600 (Sat, 20 Jan
- 2007) | 8 lines Fix German language support in say.conf Properly
- support 21, 31, 41, 51, 61, 71, 81, and 91. einundzwanzig has the
- same format as zweiundzwanzig (as do all other "_ZX" spoken
- numerals) Fix support for numbers in the 10,000,000 to 99,999,999
- range. Add support for numbers in the 100,000,000 to 999,999,999
- range. ........
-
-2007-01-20 00:13 +0000 [r51314-51344] Russell Bryant <russell@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 51343 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r51343 | russell | 2007-01-19 18:13:06 -0600 (Fri, 19 Jan 2007) |
- 2 lines Remove an unused instance of an unnamed enum. ........
-
- * /, apps/app_meetme.c: Merged revisions 51341 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r51341 | russell | 2007-01-19 16:19:10 -0600 (Fri, 19 Jan 2007) |
- 2 lines Remove another duplicated definition ........
-
- * /, apps/app_meetme.c: Merged revisions 51339 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r51339 | russell | 2007-01-19 15:20:20 -0600 (Fri, 19 Jan 2007) |
- 2 lines Remove a variable that was declared twice. ........
-
- * /, codecs/gsm/Makefile: Merged revisions 51331 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r51331 | russell | 2007-01-19 13:30:54 -0600 (Fri, 19 Jan 2007) |
- 3 lines Add a couple more processors that need optimizations
- excluded. (issue #8637) ........
-
- * /, channels/chan_gtalk.c: Merged revisions 51328 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r51328 | russell | 2007-01-19 13:08:25 -0600 (Fri, 19
- Jan 2007) | 5 lines Fix VLDTMF support in chan_gtalk.
- AST_FRAME_DTMF and AST_FRAME_DTMF_END are actually the same
- thing. So, a digit would have been interpreted incorrectly here.
- Since the channel driver will always have the begin and end
- callbacks called for a digit, only support the button-down and
- button-up messages. ........
-
- * /, .cleancount: Merged revisions 51326 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r51326 | russell | 2007-01-19 13:02:55 -0600 (Fri, 19 Jan 2007) |
- 2 lines Bump the cleancount since my last commit changed the
- channel structure. ........
-
- * channels/chan_zap.c, channels/chan_local.c, main/frame.c, /,
- channels/chan_sip.c, channels/chan_agent.c,
- include/asterisk/channel.h, channels/chan_gtalk.c,
- channels/chan_iax2.c, channels/chan_oss.c, main/rtp.c,
- main/channel.c, channels/chan_jingle.c, channels/chan_phone.c,
- channels/chan_misdn.c, channels/chan_skinny.c,
- channels/chan_features.c, channels/chan_h323.c,
- channels/chan_alsa.c, channels/chan_mgcp.c: Merged revisions
- 51311 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) |
- 23 lines Merge the changes from the /team/group/vldtmf_fixup
- branch. The main bug being addressed here is a problem introduced
- when two SIP channels using SIP INFO dtmf have their media
- directly bridged. So, when a DTMF END frame comes into Asterisk
- from an incoming INFO message, Asterisk would try to emulate a
- digit of some length by first sending a DTMF BEGIN frame and
- sending a DTMF END later timed off of incoming audio. However,
- since there was no audio coming in, the DTMF_END was never
- generated. This caused DTMF based features to no longer work. To
- fix this, the core now knows when a channel doesn't care about
- DTMF BEGIN frames (such as a SIP channel sending INFO dtmf). If
- this is the case, then Asterisk will not emulate a digit of some
- length, and will instead just pass through the single DTMF END
- event. Channel drivers also now get passed the length of the
- digit to their digit_end callback. This improves SIP INFO support
- even further by enabling us to put the real digit duration in the
- INFO message instead of a hard coded 250ms. Also, for an incoming
- INFO message, the duration is read from the frame and passed into
- the core instead of just getting ignored. (issue #8597, maybe
- others...) ........
-
-2007-01-19 18:00 +0000 [r51308-51312] Luigi Rizzo <rizzo@icir.org>
-
- * include/asterisk/strings.h: As the comment in the diff says:
- AST_INLINE_API() is a macro that takes a block of code as an
- argument. Using preprocessor #directives in the argument is not
- supported by all compilers, and it is a bit of an obfuscation
- anyways, so avoid it. As a workaround, define a macro that
- produces either its argument or nothing, and use that instead of
- #ifdef/#endif within the argument to AST_INLINE_API().
-
- * main/rtp.c: in the interest of portability, avoid using %zd when
- all we need is to print is an integer that fits in 16 bits.
-
- * channels/chan_iax2.c: sizeof() is compatible with format %d so
- don't be too picky on printf formats.
-
- * channels/chan_zap.c: remove variable declaration in the middle of
- a block
-
-2007-01-19 17:19 +0000 [r51303-51305] Russell Bryant <russell@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in: Regenerate configure
- script to reflect recent zaptel changes
-
- * include/asterisk/zapata.h: Include tonezone.h for linux, too
-
- * main/asterisk.c: Merged revisions 51302 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r51302 | russell | 2007-01-19 10:56:17 -0600
- (Fri, 19 Jan 2007) | 12 lines Merged revisions 51300 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r51300 | russell | 2007-01-19 10:44:09 -0600 (Fri, 19
- Jan 2007) | 4 lines Fix a memory leak on command line tab
- completion. The container for the matches was freed, but the
- individual matches themselves were not. (issue #8851, arkadia)
- ........ ................
-
-2007-01-19 16:51 +0000 [r51297-51301] Luigi Rizzo <rizzo@icir.org>
-
- * main/Makefile: forgot to add AST_LIBS += $(BKTR_LIB)
-
- * main/channel.c: include "asterisk/zapata.h" to get the zaptel
- headers. this should be the last one left around...
-
- * channels/chan_zap.c: whoops, fix a cut&paste error...
-
- * channels/chan_zap.c: slight change to the initialization of a
- structure, also using '\0' to make it clear we need a (char)0
-
-2007-01-19 16:30 +0000 [r51296] Russell Bryant <russell@digium.com>
-
- * main/manager.c: Break out of the config processing loop for
- manager.conf once the correct user has been found so that 'cat'
- is non-NULL. This way, users.conf is only checked when necessary.
- (issue #8852, akohlsmith, committed patch a bit different)
-
-2007-01-19 16:28 +0000 [r51285-51295] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_zap.c: include "asterisk/zapata.h" to get the
- zaptel headers.
-
- * codecs/codec_zap.c: include "asterisk/zapata.h" to get the zaptel
- headers
-
- * apps/app_meetme.c: include "asterisk/zapata.h" instead of testing
- for the location of the header files. On passing, add a cast to
- insure -Werror clean compilation on FreeBSD 6.x, where time_t
- does not match %ld
-
- * apps/app_zapbarge.c, apps/app_flash.c, apps/app_zapscan.c,
- apps/app_zapras.c, res/res_musiconhold.c, channels/chan_iax2.c,
- apps/app_rpt.c: include "asterisk/zapata.h" instead of looking
- directly for the zaptel.h and tonezone.h
-
- * configure.ac: another freebsd-specific check for zaptel
- compatibility
-
- * include/asterisk/zapata.h (added): Add a stub file to find the
- zaptel headers in the right place, rather than repeating the
- check on every single file. Changes to the individual files are
- coming. The header file name has been suggested by kevin.
- Approved by: kpfleming
-
- * makeopts.in: forgot to add BKTR_INCLUDE and BKTR_LIB in
- makeopts.in
-
- * configure.ac: add comments that AC_USE_SYSTEM_EXTENSIONS and
- AST_PROG_LD do not work on FreeBSD - presumably they depend on
- some auto* feature that is not installed by default. I am not
- sure on what is a proper fix. In my local copy i simply comment
- them out. The AST_PROG_LD is a long standing isse, there were
- attempts to fix it in the past but probably not enough has been
- copied to acinclude.m4, and i had forgotten about it because i
- commented out this call in configure.ac long ago
-
- * configure.ac: Add check for backtrace support on platforms that
- do not have it natively. Part of it leaked in in a previous
- commit.
-
- * configure.ac: remove a useless (and harmful on some platforms)
- -lnsl from IKSEMEL_LIB. Actually i am not even sure whether
- -lgcrypt -lgpg-error are needed.
-
- * configure.ac: simplify checking for zaptel version and location
- (for linux, this is functionally equivalent to the previous
- method; for FreeBSD, it re-adds inspection in $PREFIX/zaptel.h).
- Please wait to regenerate the "configure" file as i have another
- few pending changes to configure.ac Not applicable to 1.4 until
- acinclude.m4 is also updated.
-
-2007-01-19 00:28 +0000 [r51273-51275] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 51274 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r51274 | dhubbard | 2007-01-18 18:17:32 -0600 (Thu, 18 Jan 2007)
- | 3 lines chan_zap compiles without libpri after committing 7877
- patch ........
-
- * channels/chan_zap.c, /: Merged revisions 51272 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r51272 | dhubbard | 2007-01-18 17:56:49 -0600
- (Thu, 18 Jan 2007) | 11 lines Merged revisions 51271 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r51271 | dhubbard | 2007-01-18 17:47:10 -0600 (Thu, 18
- Jan 2007) | 3 lines issue 7877: chan_zap module reload does not
- use default/initialized values on subsequent loads. Reset
- configuration variables to default values prior to parsing
- configuration file. ........ ................
-
-2007-01-18 22:56 +0000 [r51266] Jason Parker <jparker@digium.com>
-
- * main/pbx.c, /, funcs/func_strings.c, apps/app_voicemail.c: Merged
- revisions 51265 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r51265 | qwell | 2007-01-18 16:50:23 -0600 (Thu, 18 Jan 2007) | 4
- lines Add some more checks for option_debug before
- ast_log(LOG_DEBUG, ...) calls. Issue 8832, patch(es) by tgrman
- ........
-
-2007-01-18 21:57 +0000 [r51263] Russell Bryant <russell@digium.com>
-
- * Makefile, /, configure, main/Makefile, acinclude.m4, makeopts.in:
- Merged revisions 51262 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r51262 | russell | 2007-01-18 15:54:23 -0600 (Thu, 18 Jan 2007) |
- 5 lines Ensure that the locations given to the Asterisk configure
- script for ncurses, curses, termcap, or tinfo are further passed
- along to the editline configure script. This fixes some
- cross-compilation environments. (issue #8637, reported by ovi,
- patch by me) ........
-
-2007-01-18 21:15 +0000 [r51257] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/stdtime/localtime.c: Merged revisions 51256 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r51256 | tilghman | 2007-01-18 15:14:24 -0600
- (Thu, 18 Jan 2007) | 10 lines Merged revisions 51255 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r51255 | tilghman | 2007-01-18 15:11:34 -0600 (Thu, 18
- Jan 2007) | 2 lines If a timezone is not specified, assume
- localtime (instead of gmtime) (Issue #7748) ........
- ................
-
-2007-01-18 19:19 +0000 [r51252] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_speech_utils.c: Merged revisions 51251 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r51251 | file | 2007-01-18 14:17:34 -0500 (Thu, 18 Jan
- 2007) | 2 lines Only start timeout once we reach the end of the
- files to play back. ........
-
-2007-01-18 19:03 +0000 [r51249] Jason Parker <jparker@digium.com>
-
- * main/cli.c: Fix filename completion for "module load" and "load"
- CLI commands. Issue 8846
-
-2007-01-18 18:54 +0000 [r51247] Russell Bryant <russell@digium.com>
-
- * main/manager.c: Fix trunk version of manager support for
- users.conf. Now it actually pays attention to the "hasmanager"
- option. (Thanks to Anthony L. for pointing out that this was
- broken!)
-
-2007-01-18 18:39 +0000 [r51244] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 51243 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r51243 | file | 2007-01-18 13:36:35 -0500 (Thu, 18 Jan 2007) | 2
- lines Copy MOH settings when calling a peer so that if they put
- someone on hold or get put on hold themselves they get the right
- music class. (issue #8840 reported by mdu113) ........
-
-2007-01-18 18:36 +0000 [r51242] Jason Parker <jparker@digium.com>
-
- * main/channel.c, /: Merged revisions 51241 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r51241 | qwell | 2007-01-18 12:28:29 -0600 (Thu, 18 Jan 2007) | 2
- lines Fix an issue with deprecated commands ........
-
-2007-01-18 17:52 +0000 [r51237] Tilghman Lesher <tlesher@digium.com>
-
- * contrib/scripts/vmdb.sql, /: Merged revisions 51236 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r51236 | tilghman | 2007-01-18 11:49:41 -0600
- (Thu, 18 Jan 2007) | 10 lines Merged revisions 51235 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r51235 | tilghman | 2007-01-18 11:42:17 -0600 (Thu, 18
- Jan 2007) | 2 lines Document all the fields, including the
- indication that "uniqueid" should not be renamed. ........
- ................
-
-2007-01-18 17:33 +0000 [r51234] Russell Bryant <russell@digium.com>
-
- * main/manager.c, /: Merged revisions 51233 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r51233 | russell | 2007-01-18 11:18:43 -0600 (Thu, 18 Jan 2007) |
- 3 lines Make the "hasmanager" option in users.conf actually have
- an effect. (issue #8740, LnxPrgr3) ........
-
-2007-01-18 06:59 +0000 [r51221] Paul Cadach <paul@odt.east.telecom.kz>
-
- * channels/chan_h323.c: Update ast_append_ha() usage
-
-2007-01-18 05:24 +0000 [r51212-51215] Joshua Colp <jcolp@digium.com>
-
- * apps/app_page.c, CHANGES: Add 's' option to Page application
- which checks devicestate before dialing. (issue #8673 reported by
- sunder)
-
- * /, apps/app_voicemail.c: Merged revisions 51213 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r51213 | file | 2007-01-17 19:48:55 -0500 (Wed, 17 Jan 2007) | 2
- lines Build the IMAP remote directory string better and properly.
- Fix an issue with encoding the GSM voicemail when attaching to
- the voicemail. (issue #8808 reported by akohlsmith) ........
-
- * /, main/rtp.c: Merged revisions 51211 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r51211 | file | 2007-01-17 19:18:44 -0500 (Wed, 17 Jan 2007) | 2
- lines Pass data as well for hold/unhold/vidupdate frames. (issue
- #8840 reported by mdu113) ........
-
-2007-01-17 23:35 +0000 [r51199-51207] Russell Bryant <russell@digium.com>
-
- * /, funcs/func_odbc.c: Merged revisions 51205 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r51205 | russell | 2007-01-17 17:31:11 -0600 (Wed, 17 Jan 2007) |
- 5 lines Fix some instances where when loading func_odbc, a
- double-free could occur. Also, remove an unneeded error message.
- If the failure condition is actually a memory allocation failure,
- a log message will already be generated automatically. ........
-
- * channels/chan_zap.c, /: Merged revisions 51204 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r51204 | russell | 2007-01-17 16:09:52 -0600 (Wed, 17 Jan 2007) |
- 4 lines Instead of dividing the offset by 2 directly, make it
- more clear that the offset is being scaled by the size of the
- elements in the buffer. (Inspired by a discussing on the
- asterisk-dev list about this code) ........
-
- * /, channels/chan_sip.c: Merged revisions 51198 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r51198 | russell | 2007-01-17 15:18:35 -0600
- (Wed, 17 Jan 2007) | 11 lines Merged revisions 51197 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r51197 | russell | 2007-01-17 15:17:21 -0600 (Wed, 17
- Jan 2007) | 3 lines Move the check for a failure of
- ast_channel_alloc() to before locking the pvt structure again.
- Otherwise, on a failure, this will cause a deadlock. ........
- ................
-
-2007-01-17 20:57 +0000 [r51196] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/utils.c: Merged revisions 51195 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r51195 | tilghman | 2007-01-17 14:56:15 -0600
- (Wed, 17 Jan 2007) | 12 lines Merged revisions 51194 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r51194 | tilghman | 2007-01-17 14:52:21 -0600 (Wed, 17
- Jan 2007) | 4 lines When ast_strip_quoted was called with a
- zero-length string, it would treat a NULL as if it were the
- quoting character (and would thus return the string in memory
- immediately following the passed-in string). ........
- ................
-
-2007-01-17 19:43 +0000 [r51193] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: Don't hold channel lock while sleeping/waiting
- for audio stream to get setup. (issue #8834 reported by phsultan)
-
-2007-01-17 17:37 +0000 [r51189] Jason Parker <jparker@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 51186 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r51186 | qwell | 2007-01-17 11:36:53 -0600 (Wed, 17 Jan 2007) | 2
- lines re-add "password" for realtime voicemail ........
-
-2007-01-17 06:37 +0000 [r51183] Joshua Colp <jcolp@digium.com>
-
- * /, main/rtp.c: Merged revisions 51182 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r51182 | file | 2007-01-17 01:36:41 -0500 (Wed, 17 Jan 2007) | 2
- lines Return the correct result when directly writing out a
- packet so that the core doesn't then decide to handle it the
- regular way again. (issue #8833 reported by rcourtna) ........
-
-2007-01-17 01:30 +0000 [r51177] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 51176 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r51176 | kpfleming | 2007-01-16 19:29:12 -0600 (Tue, 16 Jan 2007)
- | 2 lines a few more coding style cleanups and one bug fix (from
- AnthonyL) ........
-
-2007-01-17 00:50 +0000 [r51173] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 51172 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r51172 | file | 2007-01-16 19:46:29 -0500 (Tue, 16 Jan 2007) | 2
- lines Move rescheduling of lagrq/pings into the scheduler
- callback. ........
-
-2007-01-17 00:22 +0000 [r51166-51171] Jason Parker <jparker@digium.com>
-
- * /, main/rtp.c: Merged revisions 51170 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r51170 | qwell | 2007-01-16 18:20:56 -0600 (Tue, 16 Jan 2007) | 4
- lines Fix issue with dtmf continuation packets when the dtmf
- digit is 0... Issue 8831 ........
-
- * contrib/scripts/vmdb.sql, /, apps/app_voicemail.c: Merged
- revisions 51167 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r51167 | qwell | 2007-01-16 16:50:19 -0600 (Tue, 16 Jan 2007) | 6
- lines Fix an issue with IMAP storage and realtime voicemail. Also
- update the vmdb sql script for IMAP specific options. Issue 8819,
- initial patches by bsmithurst (slightly modified by me) ........
-
- * /, doc/voicemail_odbc_postgresql.txt: Merged revisions 51165 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r51165 | qwell | 2007-01-16 16:07:53 -0600 (Tue, 16 Jan 2007) | 2
- lines change documentation to reflect new procedure in 1.4/trunk
- ........
-
-2007-01-16 21:52 +0000 [r51160-51163] Tilghman Lesher <tlesher@digium.com>
-
- * /, doc/voicemail_odbc_postgresql.txt (added): Merged revisions
- 51162 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r51162 | tilghman | 2007-01-16 15:51:15 -0600
- (Tue, 16 Jan 2007) | 10 lines Merged revisions 51161 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r51161 | tilghman | 2007-01-16 15:50:04 -0600 (Tue, 16
- Jan 2007) | 2 lines Add documentation walkthrough on getting
- Postgres to work with voicemail (from Issue 8513) ........
- ................
-
- * /, apps/app_voicemail.c: Merged revisions 51159 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r51159 | tilghman | 2007-01-16 15:28:39 -0600
- (Tue, 16 Jan 2007) | 10 lines Merged revisions 51158 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r51158 | tilghman | 2007-01-16 15:26:06 -0600 (Tue, 16
- Jan 2007) | 2 lines Postgres driver doesn't like a NULL pointer
- when retrieving the length (Bug 8513) ........ ................
-
-2007-01-16 19:01 +0000 [r51155] Kevin P. Fleming <kpfleming@digium.com>
-
- * apps/app_voicemail.c: remove pointless DEBUG message (watch those
- patch merges, people!)
-
-2007-01-16 17:50 +0000 [r51152] Joshua Colp <jcolp@digium.com>
-
- * res/res_features.c, CHANGES, configs/features.conf.sample: Add
- parkedcalltransfers option for res_features. This basically
- enables/disables DTMF based transfers. If you want to get former
- behavior you will have to make sure it is enabled.
-
-2007-01-16 17:47 +0000 [r51151] Matt O'Gorman <mogorman@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 51150 via svnmerge from
- https://svn.digium.com/svn/asterisk/branches/1.4 ........ r51150
- | mogorman | 2007-01-16 11:46:12 -0600 (Tue, 16 Jan 2007) | 2
- lines minor things i missed before i get jumped on ........
-
-2007-01-16 17:42 +0000 [r51149] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_features.c: Merged revisions 51148 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r51148 | file | 2007-01-16 12:39:50 -0500 (Tue,
- 16 Jan 2007) | 10 lines Merged revisions 51145 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r51145 | file | 2007-01-16 12:36:50 -0500 (Tue, 16 Jan 2007) | 2
- lines Return previous behavior. ParkedCalls will be able to do
- DTMF based transfers again. trunk however will get an option to
- allow this to be set on/off. (issue #8804 reported by nortex)
- ........ ................
-
-2007-01-16 17:39 +0000 [r51147] Jason Parker <jparker@digium.com>
-
- * /, main/file.c: Merged revisions 51146 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r51146 | qwell | 2007-01-16 11:36:53 -0600 (Tue, 16 Jan 2007) | 6
- lines Display more useful output when streaming files. Include
- the channel name to which the file is being played. Issue 8828,
- patch by junky. ........
-
-2007-01-16 17:23 +0000 [r51144] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_phone.c, configs/phone.conf.sample, CHANGES: Add
- support for G729 passthrough with Sigma Designs boards. (issue
- #8829 reported by ywalther)
-
-2007-01-16 08:38 +0000 [r51123] Tilghman Lesher <tlesher@digium.com>
-
- * channels/iax2-parser.h, channels/iax2.h, channels/chan_iax2.c,
- channels/iax2-parser.c: IAX2 remote variables - Bug 7619
-
-2007-01-16 05:56 +0000 [r51090] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 51087 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r51087 | file | 2007-01-16 00:55:23 -0500 (Tue,
- 16 Jan 2007) | 10 lines Merged revisions 51085 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r51085 | file | 2007-01-16 00:53:31 -0500 (Tue, 16 Jan 2007) | 2
- lines Add none as a valid callgroup/pickupgroup option. I
- consider it a bug that it would inherit it all the way down and
- not have any way to reset it to nothing - so that's why it is in
- 1.2. (issue #8296 reported by gkloepfer) ........
- ................
-
-2007-01-16 01:20 +0000 [r51058-51060] Russell Bryant <russell@digium.com>
-
- * configs/osp.conf.sample: Fix a couple of typos in the sample
- osp.conf.
-
- * /, main/config.c: Merged revisions 51057 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r51057 | russell | 2007-01-15 19:15:44 -0600 (Mon, 15 Jan 2007) |
- 3 lines It is possible for the config pointer to be NULL here, so
- it needs to be checked before dereferencing it. ........
-
-2007-01-16 00:29 +0000 [r51031] Matt O'Gorman <mogorman@digium.com>
-
- * configs/users.conf.sample, /, apps/app_voicemail.c: Patch allows
- for changing voicemail password in users.conf from voicemail
- main, written by AnthonyL bug #8436
-
-2007-01-15 23:51 +0000 [r50995] Russell Bryant <russell@digium.com>
-
- * /, Makefile.rules: Merged revisions 50994 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r50994 | russell | 2007-01-15 17:49:48 -0600 (Mon, 15 Jan 2007) |
- 2 lines Filter out a few CFLAGS that are not valid CXXFLAGS.
- ........
-
-2007-01-15 21:12 +0000 [r50958] Matt O'Gorman <mogorman@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 50957 via svnmerge from
- https://svn.digium.com/svn/asterisk/branches/1.4 ................
- r50957 | mogorman | 2007-01-15 15:08:07 -0600 (Mon, 15 Jan 2007)
- | 12 lines Merged revisions 50946 via svnmerge from
- https://svn.digium.com/svn/asterisk/branches/1.2 ........ r50946
- | mogorman | 2007-01-15 14:44:53 -0600 (Mon, 15 Jan 2007) | 4
- lines Solves issue with forwarding voicemails from folders other
- than inbox. patch by anthonyl. ........ ................
-
-2007-01-15 18:24 +0000 [r50922] Jason Parker <jparker@digium.com>
-
- * /: These deprecated functions were removed in trunk on purpose.
- No need to re-add.
-
-2007-01-15 16:40 +0000 [r50896] Joshua Colp <jcolp@digium.com>
-
- * main/manager.c, /: Merged revisions 50895 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r50895 | file | 2007-01-15 11:36:07 -0500 (Mon, 15 Jan 2007) | 2
- lines Move event processing into do_message so that it gets
- executed again when events are tripped. ........
-
-2007-01-15 15:08 +0000 [r50868-50869] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, configure, include/asterisk/autoconfig.h.in, main/Makefile,
- configure.ac, Makefile.rules, acinclude.m4, makeopts.in: Merged
- revisions 50867 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r50867 | kpfleming | 2007-01-15 09:03:06 -0600 (Mon, 15 Jan 2007)
- | 2 lines use the ACX_PTHREAD macro from the Autoconf macro
- archive for setting up compiler pthreads support... should
- improve portability to platforms with unusual pthreads
- requirements ........
-
- * codecs/g722: ignore dependency files in this directory
-
-2007-01-15 02:28 +0000 [r50847] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_oss.c: Feature: allow soundcard to be used in both
- modes (autoanswer and not), selectable by how it is called in the
- dialplan. This allows a speaker system hooked up to the soundcard
- to be used for both ring notification, as well as paging.
-
-2007-01-14 22:00 +0000 [r50821] Joshua Colp <jcolp@digium.com>
-
- * /, main/astmm.c: Merged revisions 50820 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r50820 | file | 2007-01-14 16:59:05 -0500 (Sun, 14 Jan 2007) | 2
- lines Add missing newlines for two memory CLI commands. ........
-
-2007-01-14 05:34 +0000 [r50783-50784] Tilghman Lesher <tlesher@digium.com>
-
- * main/config.c: Bug 8803 - Fix crash in API
-
- * /, main/db1-ast/hash/hsearch.c, main/db1-ast/btree/bt_page.c,
- main/db1-ast/recno/rec_search.c, main/db1-ast/btree/bt_get.c,
- main/db1-ast/hash/hash.c, main/db1-ast/db/db.c,
- main/db1-ast/recno/rec_get.c, main/db1-ast/btree/bt_seq.c,
- main/db1-ast/hash/hash_func.c, main/db1-ast/btree/bt_utils.c,
- main/db1-ast/recno/rec_seq.c, main/db1-ast/btree/bt_overflow.c,
- main/db1-ast/btree/bt_search.c, main/db1-ast/btree/bt_conv.c,
- main/db1-ast/btree/bt_close.c, main/db1-ast/btree/bt_put.c,
- main/db1-ast/recno/rec_utils.c, main/db1-ast/hash/hash_bigkey.c,
- main/db1-ast/recno/rec_open.c, main/db1-ast/recno/rec_delete.c,
- main/db1-ast/hash/hash_buf.c, main/db1-ast/hash/hash_page.c,
- main/db1-ast/recno/rec_close.c, main/db1-ast/recno/rec_put.c,
- main/db1-ast/include/ndbm.h, main/db1-ast/btree/bt_debug.c,
- main/db1-ast/mpool/mpool.c, main/db1-ast/btree/bt_split.c,
- main/db1-ast/btree/bt_open.c, main/db1-ast/btree/bt_delete.c,
- main/db1-ast/hash/hash_log2.c: Merged revisions 50782 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r50782 | tilghman | 2007-01-13 23:13:47 -0600
- (Sat, 13 Jan 2007) | 10 lines Merged revisions 50781 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r50781 | tilghman | 2007-01-13 23:01:16 -0600 (Sat, 13
- Jan 2007) | 2 lines Bug 8814 - db should look for its header
- using a relative path, instead of the system path (Fixes FreeWRT)
- ........ ................
-
-2007-01-13 16:47 +0000 [r50755] Kevin P. Fleming <kpfleming@digium.com>
-
- * Makefile, /, build_tools/make_sample_voicemail (added): Merged
- revisions 50754 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r50754 | kpfleming | 2007-01-13 10:45:37 -0600 (Sat, 13 Jan 2007)
- | 2 lines when building the sample greetings for maibox
- 1234@default during 'make samples', build a greeting for each
- language and file format the user selected to install with
- menuselect (reported by Brian Capouch on asterisk-dev) ........
-
-2007-01-13 06:01 +0000 [r50675-50728] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /: Merged revisions 50727 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r50727 | file | 2007-01-13 01:00:24 -0500 (Sat, 13 Jan 2007) | 2
- lines Only write a frame out to the channel if one exists. There
- are cases where one may not and would therefore cause the channel
- driver to segfault. (issue #8434 reported by slimey) ........
-
- * channels/chan_sip.c: Get rid of unneeded code, fix a spelling
- mistake, and use registry state a bit more. (issue #8402 reported
- by rizzo)
-
- * configs/iax.conf.sample: Clarify what the trunkmaxsize value is
- in (bytes).
-
- * configs/iax.conf.sample, CHANGES, channels/chan_iax2.c: Drop
- trunkrealloc option and just have the maximum size be a
- configurable option. This is per Kevin's comments on -dev and my
- own thoughts after I put the previous option in.
-
- * channels/chan_sip.c: Ensure error variable is set to 0 or else we
- might get false error messages. (issue #8798 reported by tootai,
- fix by anthonyl)
-
- * configs/iax.conf.sample, CHANGES, channels/chan_iax2.c: Merge in
- trunkrealloc option for chan_iax2. (issue #8267 reported by
- marcodmb, branch by anthonyl)
-
- * /, res/res_snmp.c: Merged revisions 50674 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r50674 | file | 2007-01-12 22:04:55 -0500 (Fri, 12 Jan 2007) | 2
- lines Only join the snmp thread on an unload if the thread is
- actually running. (issue #8810 reported by junky) ........
-
-2007-01-12 19:25 +0000 [r50648] Jason Parker <jparker@digium.com>
-
- * /, configs/voicemail.conf.sample: Merged revisions 50647 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r50647 | qwell | 2007-01-12 13:24:40 -0600 (Fri, 12 Jan 2007) | 2
- lines Update documentation to state that you shouldn't use
- realtime static with voicemail.conf ........
-
-2007-01-12 18:13 +0000 [r50603-50629] Joshua Colp <jcolp@digium.com>
-
- * main/manager.c: Exit from session loop upon error (ie: they
- disconnected) and don't do any buffer manipulation in do_message.
- get_input will handle it.
-
- * main/manager.c, /: Merged revisions 50602 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r50602 | file | 2007-01-12 11:42:33 -0500 (Fri, 12 Jan 2007) | 2
- lines We need to check for res being 0 in do_message itself,
- otherwise our headers will get lost. ........
-
-2007-01-12 15:01 +0000 [r50538-50571] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/channel.c, main/pbx.c, include/asterisk/channel.h: make the
- automatic post-answer delay happen only when the answer is
- 'automatic' (not done by the Answer() dialplan application)
-
- * main/pbx.c, /: Merged revisions 50562 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r50562 | kpfleming | 2007-01-12 08:42:24 -0600
- (Fri, 12 Jan 2007) | 10 lines Merged revisions 50561 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r50561 | kpfleming | 2007-01-12 08:34:15 -0600 (Fri, 12
- Jan 2007) | 2 lines minor documentation clarification ........
- ................
-
- * main/channel.c: when a channel gets automatically answered by an
- application, sleep a bit to give the audio path (for VOIP
- channels) time to be setup
-
-2007-01-11 05:54 +0000 [r50378-50469] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 50468 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r50468 | file | 2007-01-11 00:53:09 -0500 (Thu, 11 Jan 2007) | 2
- lines Remove check for channel state as it can definitely be
- something other then ring, and also clean up the code a bit. This
- should solve the parking issues and maybe some attended transfer
- issues people have been seeing. ........
-
- * /, channels/chan_sip.c, include/asterisk/rtp.h, main/rtp.c:
- Merged revisions 50466 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r50466 | file | 2007-01-11 00:19:39 -0500 (Thu, 11 Jan 2007) | 2
- lines Add support to see whether NAT was detected (yay symmetric
- RTP) and also add a check in chan_sip so that if NAT has been
- detected and the reinvite behind nat option has been turned off,
- then just do partial bridge. (issue #8655 reported by mnicholson)
- ........
-
- * /, apps/app_speech_utils.c: Merged revisions 50433 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r50433 | file | 2007-01-10 15:25:44 -0500 (Wed, 10 Jan
- 2007) | 2 lines Merge speech-multi branch which adds support for
- joining multiple sound files together to be played one after
- another in SpeechBackground. ........
-
- * /, main/config.c: Merged revisions 50405 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r50405 | file | 2007-01-10 14:46:29 -0500 (Wed, 10 Jan 2007) | 2
- lines Fix parsing when using something like ldap settings. (done
- by anthonyl) ........
-
- * include/asterisk/strings.h: Return the useless casts that ensure
- this file is C++ clean. (issue #8602 reported by mikma)
-
- * /, channels/chan_sip.c: Merged revisions 50377 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r50377 | file | 2007-01-10 13:32:29 -0500 (Wed, 10 Jan 2007) | 2
- lines Fix chan_sip not working issue. Let's not prematurely
- return 0. (issue #8783 reported by st41ker) ........
-
-2007-01-10 16:47 +0000 [r50347] Jason Parker <jparker@digium.com>
-
- * /, cdr/cdr_manager.c: Merged revisions 50346 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r50346 | qwell | 2007-01-10 10:45:36 -0600 (Wed, 10 Jan 2007) | 4
- lines Reverse some logic in cdr_manager, which made it fail to
- load if the config file existed. Issue 8777 ........
-
-2007-01-10 04:56 +0000 [r50267-50302] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 50298 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r50298 | file | 2007-01-09 23:55:13 -0500 (Tue,
- 09 Jan 2007) | 10 lines Merged revisions 50295 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r50295 | file | 2007-01-09 23:51:06 -0500 (Tue, 09 Jan 2007) | 2
- lines Add another return value to dial_exec_full that indicates
- execution is going to continuing at a new
- extension/context/priority and to just let it slide. (issue #8598
- reported by jon) ........ ................
-
- * channels/chan_zap.c: Allow usedistinctiveringdetection and
- distinctiveringaftercid to be reset during a reload. (issue #8739
- reported by tzafrir)
-
- * main/pbx.c, /: Merged revisions 50266 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r50266 | file | 2007-01-09 22:51:29 -0500 (Tue, 09 Jan 2007) | 2
- lines Ensure data's existence before trying to access it. (issue
- #8774 reported by rcourtna) ........
-
-2007-01-10 02:50 +0000 [r50229-50230] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Covert some spaces to tabs, and put a list
- of defines in an enum.
-
- * Makefile, /: Merged revisions 50228 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r50228 | russell | 2007-01-09 21:17:46 -0500
- (Tue, 09 Jan 2007) | 14 lines Merged revisions 50227 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r50227 | russell | 2007-01-09 21:16:45 -0500 (Tue, 09
- Jan 2007) | 6 lines Make the number that represents the major
- version number a single digit instead of 2. Using two digits
- makes it an octal number when put into version.h, which breaks
- the compilation of any out of tree module that checks the version
- for any version after 1.2.7 (reported by Matteo Brancaleoni on
- the asterisk-dev mailing list, who gave credit to vihai for
- pointing it out) ........ ................
-
-2007-01-09 13:45 +0000 [r50152] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 50151 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r50151 | tilghman | 2007-01-09 07:40:45 -0600
- (Tue, 09 Jan 2007) | 12 lines Merged revisions 50150 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r50150 | tilghman | 2007-01-09 07:30:04 -0600 (Tue, 09
- Jan 2007) | 4 lines The advent of realtime has enabled people to
- use commas in the fullname field. This could cause an issue with
- sending voicemails, when the field is unquoted. (Issue 8595)
- ........ ................
-
-2007-01-09 12:25 +0000 [r50132] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Based on the following patch, changed for
- trunk... Merged revisions 50124 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r50124 | oej | 2007-01-09 12:25:20 +0100 (Tue, 09 Jan 2007) | 3
- lines - handle re-invites properly in sip_hangup() - Add some
- invitestate status changes just to be sure ........
-
-2007-01-08 23:40 +0000 [r50099] Jason Parker <jparker@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 50098 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r50098 | qwell | 2007-01-08 17:39:12 -0600 (Mon, 08 Jan 2007) | 4
- lines Fix an issue with voicemail and users.conf, where it
- wouldn't ever parse a password, since it was using "secret"
- instead of "password" Issue 8761, reported by and patch
- suggestion from ssokol. ........
-
-2007-01-08 21:40 +0000 [r50075] Joshua Colp <jcolp@digium.com>
-
- * codecs/codec_zap.c: Move channel acquisition to when the
- translation path is setup, and clean up.
-
-2007-01-08 21:17 +0000 [r50074] Matt O'Gorman <mogorman@digium.com>
-
- * /, apps/app_senddtmf.c: Merged revisions 50073 via svnmerge from
- https://svn.digium.com/svn/asterisk/branches/1.4 ........ r50073
- | mogorman | 2007-01-08 15:11:16 -0600 (Mon, 08 Jan 2007) | 1
- line we can't unlock a channel if we cant find it. - AnthonyL bug
- #8741 ........
-
-2007-01-08 20:10 +0000 [r50033-50056] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Make callback declaration match one used in trunk.
-
- * include/asterisk/lock.h: Change trylock output for what already
- has the lock from an error to a warning.
-
- * /, main/rtp.c: Merged revisions 50032 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r50032 | file | 2007-01-08 13:21:31 -0500 (Mon, 08 Jan 2007) | 2
- lines Disable the more intense packet2packet bridging until the
- bugs can be worked out. ........
-
-2007-01-08 14:31 +0000 [r49931-50007] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 50006 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r50006 | oej | 2007-01-08 15:26:14 +0100 (Mon, 08 Jan 2007) | 11
- lines Issue #8677 - Handle failure of T.38 re-invite This is not
- a fix, but adding an error message to tell the admin that we have
- a bad configuration. We should not send T.38 re-invites to
- devices that can't handle it (with the current architecture where
- you have to hard-code t.38 support per device). To really fix
- this, we need to figure out a way to tell the incoming call that
- the re-invite failed, so we can signal failure on that end and go
- back to the original call. ........
-
- * /, channels/chan_sip.c: Merged revisions 49983 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r49983 | oej | 2007-01-08 14:28:18 +0100 (Mon, 08 Jan 2007) | 3
- lines Issue #8524, support multiple via header values (tardieu)
- Thanks! ........
-
- * main/frame.c, include/asterisk/frame.h, main/rtp.c: Issue #8663 -
- Add passthrough support for MPEG4 (neutrino88).
-
- * /, channels/chan_sip.c: Merged revisions 49945 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r49945 | oej | 2007-01-08 10:08:10 +0100 (Mon, 08 Jan 2007) | 2
- lines We only need one forward declaration ........
-
- * /, channels/chan_sip.c: Merged revisions 49925 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r49925 | oej | 2007-01-08 09:55:03 +0100 (Mon, 08 Jan 2007) | 2
- lines Issue 8735: Terminate state when extension is unavailable
- for subscription ........
-
-2007-01-08 05:13 +0000 [r49891] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 49890 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r49890 | file | 2007-01-08 00:11:54 -0500 (Mon,
- 08 Jan 2007) | 10 lines Merged revisions 49889 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r49889 | file | 2007-01-08 00:10:07 -0500 (Mon, 08 Jan 2007) | 2
- lines Ensure we use the default refresh value of 60 if the remote
- server does not send one. (issue #8746 reported by maethor)
- ........ ................
-
-2007-01-08 03:56 +0000 [r49870] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, configure, configure.ac: Merged revisions 49866 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r49866 | kpfleming | 2007-01-07 21:53:53 -0600 (Sun, 07
- Jan 2007) | 2 lines since we use AC_PATH_TOOL to find tools, we
- should use the results it provides for us (reported by Brian
- Capouch on the asterisk-dev list) ........
-
-2007-01-07 21:46 +0000 [r49832-49835] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_dictate.c: Merged revisions 49834 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r49834 | tilghman | 2007-01-07 15:44:52 -0600
- (Sun, 07 Jan 2007) | 10 lines Merged revisions 49833 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r49833 | tilghman | 2007-01-07 15:43:10 -0600 (Sun, 07
- Jan 2007) | 2 lines If openstream fails, then we crash (Issue
- 8564) ........ ................
-
- * /, channels/chan_sip.c: Merged revisions 49831 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r49831 | tilghman | 2007-01-07 15:24:04 -0600 (Sun, 07 Jan 2007)
- | 2 lines Second condition was a subset of the first, so hold was
- never decremented, thus hint stayed stuck (Issue 8747) ........
-
-2007-01-07 19:00 +0000 [r49816] Joshua Colp <jcolp@digium.com>
-
- * funcs/func_base64.c, funcs/func_blacklist.c,
- funcs/func_callerid.c: One const, two const. Let's stick with
- everything else - one const. Plus older versions of GCC don't
- support double const either.
-
-2007-01-07 16:21 +0000 [r49784-49801] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_config_odbc.c, include/asterisk/config.h,
- res/res_realtime.c, main/config.c, funcs/func_realtime.c: When
- calling the Realtime app more than once, unset fields which were
- previously set are erroneously still set (Bug 6701). After
- discussion, it was determined this should only be changed in
- trunk.
-
- * funcs/func_shell.c, funcs/func_strings.c, funcs/func_cut.c:
- Modifications to allow the output of SHELL() to be split per line
- (Issue 8676)
-
- * funcs/func_shell.c (added): Add function to execute a shell
- command and return the output (Issue 8676)
-
- * main/channel.c: Reduce duplication of code (Issue 6542)
-
-2007-01-07 07:43 +0000 [r49769] Jason Parker <jparker@digium.com>
-
- * main/indications.c: Fix a segfault when using "countries" that
- don't have a matching zone.
-
-2007-01-06 00:28 +0000 [r49743] Jason Parker <jparker@digium.com>
-
- * main/pbx.c, /, res/res_features.c, pbx/pbx_config.c: Merged
- revisions 49742 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r49742 | qwell | 2007-01-05 18:24:38 -0600 (Fri, 05 Jan 2007) | 7
- lines Save 1 whopping byte of allocated memory! This looks like
- it may have been a chicken/egg scenario.. You had to call a
- cleanup func, because everything was allocated. Then since you
- had to call a cleanup func, you were forced to allocate - ie;
- strdup(""). ........
-
-2007-01-06 00:13 +0000 [r49727-49741] Kevin P. Fleming <kpfleming@digium.com>
-
- * funcs/func_base64.c, funcs/func_rand.c, funcs/func_md5.c,
- funcs/func_db.c, channels/chan_zap.c, funcs/func_module.c,
- funcs/func_version.c, funcs/func_timeout.c, funcs/func_env.c,
- funcs/func_strings.c, funcs/func_math.c, funcs/func_vmcount.c,
- funcs/func_cut.c, include/asterisk/channel.h, funcs/func_sha1.c,
- funcs/func_logic.c, funcs/func_uri.c, funcs/func_global.c,
- funcs/func_realtime.c, funcs/func_enum.c, funcs/func_curl.c,
- funcs/func_groupcount.c, funcs/func_odbc.c,
- funcs/func_blacklist.c, funcs/func_cdr.c, funcs/func_channel.c,
- funcs/func_callerid.c: finish const-ifying ast_func_read()
-
- * main/manager.c: probably shouldn't leave the mmap'ed file hanging
- around in memory
-
- * /, configure, acinclude.m4: Merged revisions 49714-49715 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r49714 | kpfleming | 2007-01-05 17:49:52 -0600 (Fri, 05 Jan 2007)
- | 2 lines proper fix for r49712 ........ r49715 | kpfleming |
- 2007-01-05 17:51:31 -0600 (Fri, 05 Jan 2007) | 2 lines one more
- time... ........
-
- * main/manager.c, include/asterisk/config.h, main/config.c: a
- little more const-ification
-
-2007-01-05 23:51 +0000 [r49716] Joshua Colp <jcolp@digium.com>
-
- * codecs/codec_zap.c: It is possible for framein to get called and
- no channel be available, so do a check before we increment the
- count.
-
-2007-01-05 23:41 +0000 [r49711-49713] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, configure, acinclude.m4: Merged revisions 49712 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r49712 | kpfleming | 2007-01-05 17:40:29 -0600 (Fri, 05
- Jan 2007) | 2 lines if --with-foo=<path> is specific for a
- configure option, ensure that it is used for header file checking
- as well ........
-
- * main/pbx.c, /, channels/chan_sip.c, channels/chan_agent.c,
- pbx/pbx_dundi.c, include/asterisk/pbx.h, apps/app_queue.c,
- channels/chan_iax2.c, main/db.c, apps/app_speech_utils.c,
- include/asterisk/astdb.h, apps/app_voicemail.c: const-ify some
- more APIs, and fix rev 49710 from branch-1.4 in a better way here
-
-2007-01-05 23:31 +0000 [r49709] Matt O'Gorman <mogorman@digium.com>
-
- * codecs/codec_zap.c: no need to spam everyone with show transcoder
- messages
-
-2007-01-05 23:17 +0000 [r49706] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c, /, codecs/codec_zap.c: Merged revisions
- 49705 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r49705 | qwell | 2007-01-05 17:16:16 -0600 (Fri, 05 Jan 2007) | 4
- lines Make codec_zap and chan_zap also depend on zaptel. This
- fixes an issue (8727) with zaptel being in a different directory,
- using --with-zaptel. ........
-
-2007-01-05 22:53 +0000 [r49678-49681] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/manager.c, /: Merged revisions 49680 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r49680 | kpfleming | 2007-01-05 16:52:37 -0600 (Fri, 05 Jan 2007)
- | 2 lines don't 'consume' the params list before we try to use it
- again ........
-
- * main/manager.c: use mmap() to read in the results of the manager
- action for an HTTP request, instead of reading it into a buffer
-
- * main/pbx.c, channels/chan_zap.c, /, channels/chan_sip.c,
- apps/app_meetme.c, res/res_features.c, channels/chan_agent.c,
- utils/astman.c, res/res_jabber.c, include/asterisk/manager.h,
- channels/chan_iax2.c, apps/app_queue.c, main/config.c,
- res/res_monitor.c, main/manager.c, include/asterisk/jabber.h,
- apps/app_senddtmf.c, main/db.c: Merged revisions 49676 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r49676 | kpfleming | 2007-01-05 16:16:33 -0600 (Fri, 05 Jan 2007)
- | 2 lines reduce stack consumption for AMI and AMI/HTTP requests
- by nearly 20K in most cases ........
-
-2007-01-05 22:18 +0000 [r49677] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /: Merged revisions 49675 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r49675 | file | 2007-01-05 17:14:47 -0500 (Fri, 05 Jan 2007) | 2
- lines Don't keep repeating the warning over and over when the end
- of the call is reached. (issue #8724 reported by xrg) ........
-
-2007-01-05 17:10 +0000 [r49578-49637] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, channels/chan_sip.c, channels/chan_skinny.c,
- channels/chan_iax2.c: Merged revisions 49636 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r49636 | kpfleming | 2007-01-05 11:09:00 -0600
- (Fri, 05 Jan 2007) | 10 lines Merged revisions 49635 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r49635 | kpfleming | 2007-01-05 10:56:40 -0600 (Fri, 05
- Jan 2007) | 2 lines ensure that threads which are supposed to be
- detached (because we aren't going to wait on them) are created
- properly ........ ................
-
- * main/threadstorage.c: use a rwlock-list for the list of TLS
- objects
-
- * /, channels/chan_iax2.c: Merged revisions 49600 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r49600 | kpfleming | 2007-01-04 18:01:40 -0600 (Thu, 04 Jan 2007)
- | 2 lines revert the dynamic_list insertion change... that was
- not the right thing to do ........
-
- * /, channels/chan_iax2.c: Merged revisions 49581 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r49581 | kpfleming | 2007-01-04 17:50:15 -0600 (Thu, 04 Jan 2007)
- | 3 lines create the IAX2 processing threads as background
- threads so they will use smaller stacks when we create a dynamic
- thread, put it on the dynamic_list right away so we don't lose
- track of it ........
-
- * include/asterisk/strings.h: ensure that the proper
- file/function/line shows up for dynamic string threadstorage
- objects remove pointless casts
-
- * include/asterisk/threadstorage.h: yeah... so... compiling before
- committing seems like it might be a good idea
-
- * build_tools/cflags.xml, include/asterisk.h, /,
- main/threadstorage.c (added), main/Makefile,
- include/asterisk/strings.h, include/asterisk/threadstorage.h,
- main/asterisk.c: Merged revisions 49553 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r49553 | kpfleming | 2007-01-04 16:51:01 -0600 (Thu, 04 Jan 2007)
- | 2 lines add support for tracking thread-local-storage objects
- that exist via 'threadstorage' CLI commands ........
-
-2007-01-04 23:02 +0000 [r49552-49573] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 49568 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r49568 | file | 2007-01-04 18:00:50 -0500 (Thu, 04 Jan 2007) | 2
- lines It's possible for the iax2 pvt to disappear, so if it
- has... don't bother looking for dpentries. ........
-
- * /, main/config.c: Merged revisions 49551 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r49551 | file | 2007-01-04 17:28:29 -0500 (Thu, 04 Jan 2007) | 2
- lines Only free comments and line buffer once we reach the first
- level. (issue #8678 reported by ssokol, fixed by anthonyl)
- ........
-
-2007-01-04 21:59 +0000 [r49538] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/frame.c, /, channels/iax2-parser.c: Merged revisions 49536
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r49536 | kpfleming | 2007-01-04 15:58:42 -0600 (Thu, 04 Jan 2007)
- | 2 lines don't mark these allocations as 'cache' allocations
- when caching has been disabled ........
-
-2007-01-04 21:40 +0000 [r49525] Joshua Colp <jcolp@digium.com>
-
- * main/manager.c: It's pretty difficult to pthread_kill a thread
- that doesn't exist. (issue #8681 reported by bkruse)
-
-2007-01-04 21:06 +0000 [r49524] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, channels/iax2-parser.c: Merged revisions 49523 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r49523 | kpfleming | 2007-01-04 15:06:02 -0600 (Thu, 04
- Jan 2007) | 2 lines if we're going to decrement the frame count
- when we free a frame, we should inrement it when we create one
- :-) ........
-
-2007-01-04 20:27 +0000 [r49491-49507] TransNexus OSP Development <support@transnexus.com>
-
- * doc/osp.txt: 1. Update osp guide.
-
- * configs/osp.conf.sample: 1. Update osp module configuration file.
-
-2007-01-04 18:32 +0000 [r49466] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/iax2-parser.h, /, channels/chan_iax2.c,
- channels/iax2-parser.c: Merged revisions 49465 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r49465 | kpfleming | 2007-01-04 12:31:55 -0600 (Thu, 04 Jan 2007)
- | 2 lines only do IAX2 frame caching for voice and video frames
- ........
-
-2007-01-04 18:28 +0000 [r49464] Matt O'Gorman <mogorman@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 49459 via svnmerge from
- https://svn.digium.com/svn/asterisk/branches/1.4 ................
- r49459 | mogorman | 2007-01-04 12:11:19 -0600 (Thu, 04 Jan 2007)
- | 10 lines Merged revisions 49447 via svnmerge from
- https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49447
- | mogorman | 2007-01-04 11:45:16 -0600 (Thu, 04 Jan 2007) | 2
- lines converted a lot of 256 to PATH_MAX and some white space
- fixes. ........ ................
-
-2007-01-04 18:19 +0000 [r49463] Kevin P. Fleming <kpfleming@digium.com>
-
- * codecs/Makefile, main/frame.c, /, channels/iax2-parser.c: Merged
- revisions 49457,49460-49461 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r49457 | kpfleming | 2007-01-04 12:05:47 -0600 (Thu, 04 Jan 2007)
- | 2 lines make building of codec_gsm against the system GSM
- library actually work ........ r49460 | kpfleming | 2007-01-04
- 12:16:40 -0600 (Thu, 04 Jan 2007) | 2 lines don't define this
- type either if LOW_MEMORY is enabled ........ r49461 | kpfleming
- | 2007-01-04 12:17:01 -0600 (Thu, 04 Jan 2007) | 2 lines don't do
- frame header caching in the core if LOW_MEMORY is defined
- ........
-
-2007-01-04 18:17 +0000 [r49414-49462] Matt O'Gorman <mogorman@digium.com>
-
- * /, channels/iax2-parser.c: Merged revisions 49458 via svnmerge
- from https://svn.digium.com/svn/asterisk/branches/1.4 ........
- r49458 | kpfleming | 2007-01-04 12:06:51 -0600 (Thu, 04 Jan 2007)
- | 2 lines don't do frame caching in LOW_MEMORY mode ........
-
- * /, apps/app_voicemail.c: Merged revisions 49413 via svnmerge from
- https://svn.digium.com/svn/asterisk/branches/1.4 ................
- r49413 | mogorman | 2007-01-04 10:50:56 -0600 (Thu, 04 Jan 2007)
- | 11 lines Merged revisions 49412 via svnmerge from
- https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49412
- | mogorman | 2007-01-04 10:48:43 -0600 (Thu, 04 Jan 2007) | 3
- lines good catch russell sorry i missed that. fix magic number
- with proper sizeof ........ ................
-
-2007-01-03 23:41 +0000 [r49356] Matt O'Gorman <mogorman@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 49355 via svnmerge from
- https://svn.digium.com/svn/asterisk/branches/1.4 ................
- r49355 | mogorman | 2007-01-03 17:32:03 -0600 (Wed, 03 Jan 2007)
- | 14 lines Merged revisions 49354 via svnmerge from
- https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49354
- | mogorman | 2007-01-03 17:22:47 -0600 (Wed, 03 Jan 2007) | 6
- lines When using ODBC_STORAGE VoicemailMain doesn't create the
- subdirectories for a mailbox such as the INBOX directory. this
- patch solves that problem, was written by anthony be-125 ........
- ................
-
-2007-01-03 11:15 +0000 [r49320-49321] Christian Richter <christian.richter@beronet.com>
-
- * doc/misdn.txt, channels/misdn/isdn_lib.h, channels/chan_misdn.c,
- /, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c,
- configs/misdn.conf.sample, channels/misdn/isdn_lib.c,
- channels/misdn_config.c: Merged revisions 49313 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r49313 | crichter | 2007-01-03 10:06:50 +0100
- (Mi, 03 Jan 2007) | 41 lines Merged revisions
- 48319,48321,48467,48552,48576,49135,49303 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) |
- 1 line changed a few debugs to higher debug levels ........
- r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) |
- 1 line added the export and import of the MISDN_ADDRESS_COMPLETE
- Variable to inidcate wether the extension is already completely
- dialed or if there might come additional digits by information
- elements. also added some docs for that. ........ r48467 |
- crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line
- removed FIXUP state. added check for channel allocation conflict
- when we create a setup while the other site creates a setup on
- the same channel, besides the check we resolve this conflict.
- ........ r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18
- Dez 2006) | 1 line when our PTP Partner sends us a SETUP with a
- preselected channel we just accept it, even when we're NT. added
- some checks for segfaults. ........ r48576 | crichter |
- 2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line when we
- reject a channel, because it's in use already, we shouldn't
- process the setup anymore. made the channel allocation a bit
- easier and more understandable, removed a few unused lines
- ........ r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02
- Jan 2007) | 1 line added check for channel ranges in the
- set/empty channel functions. set pmp_l1_check default to no.
- added misdn restart pid cli command. added cleaning of channel
- when we send a RELEASE_COMPLETE. ........ r49303 | crichter |
- 2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines * Added
- check for bridging in misdn_call to avoid setting
- echocancellation when 2 mISDN channels are involved and when
- bridging is set. That lead to a kernel panic before under
- different situations, because we switched about 2 times between
- hardware bridging and echocancelation * readded MISDN_URATE
- variable which got lost before, this should make app_v110 work
- again * fixed typo ........ ................
-
- * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c,
- channels/misdn_config.c: Merged revisions 47989 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r47989 | crichter | 2006-11-24 16:46:13 +0100
- (Fr, 24 Nov 2006) | 9 lines Merged revisions 47968 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r47968 | crichter | 2006-11-23 17:10:23 +0100 (Do, 23
- Nov 2006) | 1 line fixed a litle bug regarding HOLD/RETRIEVE.
- beatufied some logs, changed some loglevels. changed the default
- value of block_on_alarm ........ ................
-
-2007-01-03 03:28 +0000 [r49283] Kevin P. Fleming <kpfleming@digium.com>
-
- * Makefile, /, Makefile.rules: Merged revisions 49282 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r49282 | kpfleming | 2007-01-02 21:21:25 -0600 (Tue, 02
- Jan 2007) | 2 lines various Makefile improvements to get chan_vpb
- (and any other C++ modules) to build properly ........
-
-2007-01-03 01:21 +0000 [r49260] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 49259 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r49259 | file | 2007-01-02 20:19:53 -0500 (Tue, 02 Jan 2007) | 2
- lines Check pvt structure presence before passing to
- send_command. This gets rid of the irritating message about a
- packet without pvt structure. This happens because the scheduled
- item is getting cancelled at almost the exact moment it is
- getting executed. ........
-
-2007-01-02 22:43 +0000 [r49238] Steve Murphy <murf@digium.com>
-
- * /, main/ast_expr2f.c, pbx/ael/ael_lex.c, pbx/ael/ael.flex,
- main/ast_expr2.fl: Merged revisions 49237 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r49237 | murf | 2007-01-02 15:30:53 -0700 (Tue, 02 Jan 2007) | 1
- line This is a slight modification to Josh's edits for #8579;
- both files edited were the produced by flex; so the source files
- need to be changed instead, and the generated files regenerated.
- ........
-
-2007-01-02 20:02 +0000 [r49214-49215] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Removing propably accidentally added debug
- messages sent to verbose channel
-
- * /, channels/chan_sip.c: Merged revisions 49212 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r49212 | oej | 2007-01-02 20:58:45 +0100 (Tue, 02 Jan 2007) | 2
- lines Small cleanup of add_t38sdp - it's always enabled at that
- point in the code ........
-
-2007-01-02 17:04 +0000 [r49187] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_math.c: Tweak description text to match new
- functionality (Issue 7959)
-
-2007-01-02 14:01 +0000 [r49166] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 49165 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r49165 | kpfleming | 2007-01-02 07:59:44 -0600 (Tue, 02 Jan 2007)
- | 2 lines remove comment that is unrelated to this function
- ........
-
-2007-01-02 13:50 +0000 [r49152] Olle Johansson <oej@edvina.net>
-
- * /, configs/features.conf.sample: Update sample config
-
-2007-01-01 23:43 +0000 [r49100-49103] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c, /, build_tools/menuselect-deps.in,
- configure, include/asterisk/autoconfig.h.in, configure.ac,
- codecs/codec_zap.c: Merged revisions 49102 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r49102 | kpfleming | 2007-01-01 17:34:35 -0600 (Mon, 01 Jan 2007)
- | 2 lines check specifically for VLDTMF and transcoding support
- in the system's Zaptel installation, and make only the modules
- that need those features dependent on them (this will allow
- building the other Zaptel-using parts of Asterisk against older
- versions of Zaptel or those on other platforms that haven't
- caught up yet to the Linux version) ........
-
- * Makefile, sounds/Makefile: GNU make already knows what the
- current directory is, there is no need to use 'pwd'
-
- * Makefile, /: Merged revisions 49098-49099 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r49098 | kpfleming | 2007-01-01 16:08:24 -0600 (Mon, 01 Jan 2007)
- | 2 lines revert this change until a better solution can be
- found... 'env -i' was not being used properly, but even when
- changed to do so, this process fails during cross-compilation
- because the menuselect build still sees 'CC' as set to the
- cross-compiler ........ r49099 | kpfleming | 2007-01-01 16:48:03
- -0600 (Mon, 01 Jan 2007) | 2 lines use a simpler (and portable)
- method to ensure that menuselect is built as a host binary
- ........
-
-2007-01-01 20:16 +0000 [r49092-49097] Olle Johansson <oej@edvina.net>
-
- * /: Block cleanup of release branch
-
- * include/asterisk/indications.h: Doxygen documentationification
-
- * main/manager.c: Fix manager too.
-
- * main/frame.c, channels/chan_sip.c, include/asterisk/frame.h: -
- Add error handling to ast_parse_allow_disallow - Use this in
- chan_sip configuration parsing
-
- * include/asterisk/acl.h, channels/chan_sip.c,
- channels/chan_skinny.c, channels/chan_h323.c, main/acl.c,
- channels/chan_iax2.c, channels/chan_mgcp.c: - Implement error
- handling in ast_append_ha - Use this in chan_sip - Document ha
- functions in acl.c
-
-2006-12-31 19:15 +0000 [r49089] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: count is no longer used in the iaxq
- structure really so let's just make this a statically declared
- linked list.
-
-2006-12-31 09:38 +0000 [r49080-49082] Olle Johansson <oej@edvina.net>
-
- * CHANGES: Update CHANGES, make section about SIP. This might be a
- good way to handle other parts of this file too, as it grows.
-
- * configs/sip.conf.sample: Added some docs
-
- * channels/chan_sip.c: Add version number to useragent string -
- Issue #8700, blanchet - THANKS!
-
-2006-12-31 05:20 +0000 [r49075-49076] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_math.c: Add power and right/left shift functions
- (Issue 7959)
-
- * configs/voicemail.conf.sample, UPGRADE.txt, apps/app_voicemail.c:
- 1. Rename 'maxmessage' to 'maxsecs' to differentiate from
- 'maxmsg'. 2. Rename 'minmessage' to 'minsecs' for parity. 3. Make
- 'maxsecs' a per-user option, in addition to global. (Issue #
- 8624)
-
-2006-12-30 18:32 +0000 [r49071-49074] Joshua Colp <jcolp@digium.com>
-
- * /, pbx/pbx_config.c: Merged revisions 49073 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r49073 | file | 2006-12-30 13:31:17 -0500 (Sat, 30 Dec 2006) | 2
- lines IAX has been deprecated for quite some time so we had
- better use IAX2 when creating the dial string for users. (issue
- #8697 reported by ssokol) ........
-
- * main/rtp.c: Clarify why we are reading in a frame in the
- Packet2Packet bridge.
-
-2006-12-30 13:27 +0000 [r49068-49069] Kevin P. Fleming <kpfleming@digium.com>
-
- * sounds/Makefile: now that the 'languageprefix' option defaults to
- 'on', and all channels have a default language of 'en', let's
- install the English sound files into /var/lib/asterisk/sounds/en,
- just like the other languages
-
- * main/channel.c: small formatting fix
-
-2006-12-30 05:49 +0000 [r49064-49067] Joshua Colp <jcolp@digium.com>
-
- * /, main/rtp.c: Merged revisions 49066 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r49066 | file | 2006-12-30 00:46:57 -0500 (Sat, 30 Dec 2006) | 2
- lines If the Packet2Packet bridge is being broken because of a
- masquerade then attempt to read a frame in so the masquerade
- actually happens. Otherwise weirdness will occur. (issue #8696
- reported by kjotte) ........
-
- * funcs/func_odbc.c: Initialize obj pointers to NULL. Gets rid of
- two compiler warnings.
-
- * /, channels/chan_iax2.c: Merged revisions 49063 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r49063 | file | 2006-12-29 22:37:22 -0500 (Fri, 29 Dec 2006) | 2
- lines Initialize the packet queue in load_module instead of just
- declaring the list with the default value. (issue #8695 reported
- by ssokol) ........
-
-2006-12-30 00:51 +0000 [r49062] Steve Murphy <murf@digium.com>
-
- * /, pbx/pbx_ael.c: Merged revisions 49061 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r49061 | murf | 2006-12-29 17:40:37 -0700 (Fri, 29 Dec 2006) | 1
- line A fix for 8661, where the CUT func needed to have comma args
- converted to vertical bars. I hope this change does little harm.
- ........
-
-2006-12-29 13:25 +0000 [r49056] Russell Bryant <russell@digium.com>
-
- * channels/chan_oss.c: Convert various comments to doxygen format.
-
-2006-12-29 11:02 +0000 [r49054] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Removing extra output
-
-2006-12-29 06:26 +0000 [r49053] Russell Bryant <russell@digium.com>
-
- * include/asterisk/smdi.h: Fix a spelling mistake in a comment.
-
-2006-12-29 00:33 +0000 [r49047] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, BUGS: Merged revisions 49046 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r49046 | kpfleming | 2006-12-28 18:32:59 -0600
- (Thu, 28 Dec 2006) | 10 lines Merged revisions 49045 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r49045 | kpfleming | 2006-12-28 18:32:32 -0600 (Thu, 28
- Dec 2006) | 2 lines location of the bug posting guidelines has
- changed ........ ................
-
-2006-12-28 20:13 +0000 [r49030] Tilghman Lesher <tlesher@digium.com>
-
- * configs/func_odbc.conf.sample, funcs/func_odbc.c,
- funcs/func_strings.c: Integrate functionality tested on
- svncommunity users back into trunk
-
-2006-12-28 20:10 +0000 [r49029] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, sounds/Makefile: Merged revisions 49028 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r49028 | kpfleming | 2006-12-28 14:08:59 -0600 (Thu, 28 Dec 2006)
- | 2 lines new versions of sounds ........
-
-2006-12-28 20:05 +0000 [r49026-49027] Joshua Colp <jcolp@digium.com>
-
- * main/http.c: Convert uri_redirects list to read/write locks.
-
- * /, main/http.c: Merged revisions 49024 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r49024 | qwell | 2006-12-28 14:52:46 -0500 (Thu, 28 Dec 2006) | 2
- lines make the uris_lock a rwlock instead of a mutex lock - needs
- to be forward ported to trunk ........
-
-2006-12-28 17:56 +0000 [r49019] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael.tab.c, main/ast_expr2.y, main/ast_expr2f.c,
- pbx/ael/ael_lex.c, include/asterisk/ael_structs.h,
- pbx/ael/ael.tab.h, utils/ael_main.c, main/ast_expr2.fl,
- main/ast_expr2.c: Jason is having problems with the inclusion of
- <err.h>; it appears to be unnecessary for sucessful builds, so I
- either removed or commented out the inclusions from all the AEL
- related code. New outputs from bison/flex are included, etc.
-
-2006-12-27 22:30 +0000 [r49010] Joshua Colp <jcolp@digium.com>
-
- * /, main/ast_expr2f.c, pbx/ael/ael_lex.c: Merged revisions 49009
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r49009 | file | 2006-12-27 17:28:46 -0500 (Wed, 27 Dec 2006) | 2
- lines ast_copy_string is not available when LOW_MEMORY is used
- and things are being built in the utils directory, so we need to
- resort to the old method of strncpy. (issue #8579 reported by
- mottano) ........
-
-2006-12-27 22:14 +0000 [r49007-49008] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/cdr.c, main/channel.c, main/udptl.c, main/pbx.c,
- main/dnsmgr.c, main/frame.c, main/manager.c, /, main/http.c,
- main/logger.c, main/enum.c, main/asterisk.c, main/rtp.c,
- main/term.c: Merged revisions 49006 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r49006 | kpfleming | 2006-12-27 16:06:56 -0600 (Wed, 27 Dec 2006)
- | 2 lines since these variables all have static duration, none of
- them need initializers (they default to zero anyway) ........
-
- * codecs/g722: add file to ignore list
-
-2006-12-27 21:27 +0000 [r49004] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Only include include files once (imported
- from 1.4)
-
-2006-12-27 21:21 +0000 [r48999-49001] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/asterisk.c: apparently we need an explicit message to warn
- people
-
- * main/file.c, UPGRADE.txt, main/asterisk.c, doc/asterisk-conf.txt:
- make the 'languageprefix' option default to on, and deprecate
- turning it off
-
- * /, main/file.c, include/asterisk/options.h, main/asterisk.c:
- Merged revisions 48998 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48998 | kpfleming | 2006-12-27 15:08:30 -0600 (Wed, 27 Dec 2006)
- | 3 lines move extern declaration for this option to a header
- file where it belongs provide an initial value for
- 'languageprefix' option, instead of relying on randomness to
- provide a useful value ........
-
-2006-12-27 20:30 +0000 [r48992-48996] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Only set "rfc2833compensate" option once
-
- * /, channels/chan_sip.c: Only handle T38 options once
-
- * channels/chan_sip.c: -Remove "localmask" setting (deprecated in
- earlier version) - Remove "musiconhold" and "musicclass" settings
- (also deprecated earlier)
-
-2006-12-27 18:34 +0000 [r48989-48990] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 48988 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48988 | kpfleming | 2006-12-27 12:33:22 -0600 (Wed, 27 Dec 2006)
- | 2 lines make the option actually match the documentation
- ........
-
- * include/asterisk/utils.h, include/asterisk/astmm.h, main/frame.c,
- /, main/astmm.c, channels/iax2-parser.c: Merged revisions 48987
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48987 | kpfleming | 2006-12-27 12:29:13 -0600 (Wed, 27 Dec 2006)
- | 2 lines allow 'show memory' and 'show memory summary' to
- distinguish memory allocations that were done for caching
- purposes, so they don't look like memory leaks ........
-
-2006-12-27 18:02 +0000 [r48976-48986] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c, configs/sip.conf.sample: Be politically
- correct
-
- * apps/app_sms.c: From coding guidelines: Comments should explain
- what the code does, not when something was changed or who changed
- it. If you have done a larger contribution, make sure that you
- are added to the CREDITS file.
-
- * /, channels/chan_sip.c, configs/sip.conf.sample: Add support for
- buggy Cisco MWI firmware > 8.0.3 (issue 8575 - flewid)
-
- * /, channels/chan_sip.c: Cleanup of handle_common_options
-
- * /, channels/chan_sip.c: Reset invitestate when sending new invite
-
- * /, channels/chan_sip.c: Issue #8600 - bogus SDP Content Length in
- T.38 re-invite
-
-2006-12-26 05:23 +0000 [r48961-48967] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 48966 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48966 | file | 2006-12-26 00:20:08 -0500 (Tue, 26 Dec 2006) | 2
- lines Get rid of a needless memory allocation and only create a
- conference structure in find_conf_realtime if data was read from
- realtime. (issue #8669 reported by robl) ........
-
- * /, channels/chan_sip.c, include/asterisk/rtp.h, main/rtp.c:
- Merged revisions 48964 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48964 | file | 2006-12-25 23:31:58 -0500 (Mon, 25 Dec 2006) | 2
- lines Add an API call that initializes an RTP structure. We need
- this because chan_sip is cheeky and uses a temporary RTP
- structure for codec purposes, and the API calls that are used
- rely on the lock. (Pointed out on asterisk-dev by Andy Wang)
- ........
-
- * /, configure, configure.ac: Merged revisions 48960 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r48960 | file | 2006-12-25 12:04:48 -0500 (Mon, 25 Dec
- 2006) | 2 lines Clean up autoconf file (gets rid of warnings seen
- when rebuilding configure) and rebuild configure. ........
-
-2006-12-25 06:42 +0000 [r48958-48959] Luigi Rizzo <rizzo@icir.org>
-
- * codecs/g722/g722.h: provide INT16_MIN and INT16_MAX for platforms
- where they are not defined.
-
- * main/channel.c, apps/app_read.c, channels/chan_misdn.c,
- funcs/func_channel.c, include/asterisk/indications.h,
- apps/app_disa.c, main/app.c, res/snmp/agent.c,
- contrib/utils/zones2indications.c, include/asterisk/channel.h,
- res/res_indications.c, main/indications.c: rename the structs
- struct tone_zone_sound and struct tone_zone defined in
- indications.h to ind_tone_zone_sound and ind_tone_zone, to avoid
- conflicts with the structs with the same names defined in
- tonezone.h Hope i haven't missed any instance.
-
-2006-12-25 05:22 +0000 [r48929-48957] Russell Bryant <russell@digium.com>
-
- * /, funcs/func_math.c: Merged revisions 48956 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r48956 | russell | 2006-12-25 00:21:20 -0500
- (Mon, 25 Dec 2006) | 14 lines Merged revisions 48955 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r48955 | russell | 2006-12-25 00:19:48 -0500 (Mon, 25
- Dec 2006) | 6 lines Fix an error introduced by copying and
- pasting the handling of the >= operator for the MATH function. If
- a single equal sign was used as an operator, the function would
- treat it is as if it were the >= operator. Now, it properly
- handles it as an invalid operator. (issue #8665, patch by
- tempest1) ........ ................
-
- * funcs/func_callerid.c: Simplify the if statements used to check
- to see if the argument was "num" or "number". It is not possible
- to ever reach the second part of this conditional statement.
- Thanks to my brother, Brett, for pointing this out. :)
-
- * main/frame.c: Resolve some compiler warnings
-
- * /, channels/chan_oss.c: Merged revisions 48948 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48948 | russell | 2006-12-24 16:19:37 -0500 (Sun, 24 Dec 2006) |
- 3 lines Fix a typo in an error message that indicated that the
- MGCP channel type could not be registered, instead of the correct
- type, OSS. ........
-
- * main/http.c, configs/http.conf.sample: Use spaces as a separator
- for the redirect option to improve readability
-
- * /, channels/chan_iax2.c: Merged revisions 48944 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r48944 | russell | 2006-12-24 02:25:38 -0500
- (Sun, 24 Dec 2006) | 11 lines Merged revisions 48943 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r48943 | russell | 2006-12-24 02:23:07 -0500 (Sun, 24
- Dec 2006) | 3 lines Check for the proper return value on an error
- in a call to mmap(). This was reported by Andy Wang on the
- asterisk-dev list. Thanks! ........ ................
-
- * channels/chan_sip.c: Merged revisions 48940 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r48940 | russell | 2006-12-24 01:49:31 -0500
- (Sun, 24 Dec 2006) | 11 lines Merged revisions 48939 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r48939 | russell | 2006-12-24 01:47:29 -0500 (Sun, 24
- Dec 2006) | 3 lines Remove a couple of misplaced dots in log
- messages. This was reported by Andrea Spadaccini on the
- asterisk-dev mailing list. ........ ................
-
- * main/http.c: Simplify the definition of http_uri_redirect such
- that only one allocation is done for exactly how much memory is
- needed. This was suggested by Luigi on the asterisk-dev mailing
- list. Thanks!
-
- * include/asterisk/http.h, main/http.c, CHANGES,
- configs/http.conf.sample: - Convert the list of URI handlers to
- use the linked list macros. While doing this, implementing
- locking of this list to make it thread-safe. - Add a "redirect"
- option to http.conf that allows redirecting one URI to another. I
- was inspired to do this while playing with the Asterisk GUI. I
- got tired of typing this URL to get to the GUI:
- http://localhost:8088/asterisk/static/config/cfgadvanced.html So,
- now I have the following line in http.conf:
- redirect=/=/asterisk/static/config/cfgadvanced.html Now, I can
- type the following into my browser and go to the GUI:
- http://localhost:8088
-
- * main/manager.c: Remove a debug message. If this is still needed
- for debugging something, it should be made a LOG_DEBUG message.
-
-2006-12-23 19:55 +0000 [r48928] Joshua Colp <jcolp@digium.com>
-
- * include/asterisk/lock.h: We should probably declare the lock...
- and not just the constructor/deconstructor.
-
-2006-12-23 19:51 +0000 [r48927] Russell Bryant <russell@digium.com>
-
- * include/asterisk/lock.h: Use the correct function to destroy an
- rwlock in the destructor for an ast_rwlock_t
-
-2006-12-22 22:34 +0000 [r48871-48907] Jason Parker <jparker@digium.com>
-
- * Makefile, /, main/stdtime/localtime.c: Merged revisions 48906 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48906 | qwell | 2006-12-22 16:33:46 -0600 (Fri, 22 Dec 2006) | 2
- lines Minor fixes for Solaris. ........
-
- * /, channels/chan_skinny.c: Merged revisions 48888 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r48888 | qwell | 2006-12-22 15:40:20 -0600 (Fri, 22 Dec
- 2006) | 2 lines Note to self: Run make before committing...
- ........
-
- * /, channels/chan_skinny.c: Merged revisions 48870 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r48870 | qwell | 2006-12-22 14:43:05 -0600 (Fri, 22 Dec
- 2006) | 2 lines Fix for issue 7774 - patch by alamantia ........
-
-2006-12-22 10:35 +0000 [r48825-48857] Luigi Rizzo <rizzo@icir.org>
-
- * apps/app_sms.c: improve readability of a few macros.
-
- * apps/app_sms.c: make sms_hexdump() thread safe; restructure and
- reduce indentation on some blocks.
-
- * apps/app_sms.c: make isodate thread-safe
-
- * apps/app_sms.c: - use the standard option parsing routines; -
- document existing but undocumented parameters to send a message
- (untested but unchanged; - ad a new option p(N) to set the
- initial message delay to N ms so this can be adapted from the
- dialplan to various countries;
-
-2006-12-21 21:57 +0000 [r48785-48817] Joshua Colp <jcolp@digium.com>
-
- * main/logger.c: Merge non-blocking logger from my branch. This
- should improve things under heavy load with lots of CLI/logging
- output.
-
- * /, redhat/asterisk.spec: Merged revisions 48783 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r48783 | file | 2006-12-21 15:26:29 -0500 (Thu,
- 21 Dec 2006) | 10 lines Merged revisions 48782 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r48782 | file | 2006-12-21 15:25:01 -0500 (Thu, 21 Dec 2006) | 2
- lines Add new silence sound files to the spec for Redhat. (issue
- #8652 reported by alvaro_palma_aste) ........ ................
-
-2006-12-21 20:15 +0000 [r48781] Steve Murphy <murf@digium.com>
-
- * codecs/codec_g722.c: This little mod gets rid of that g722
- compiler warning that breaks builds configured with
- --enable-dev-mode; the previous commit of 48767 was to merge in
- changes for bug 6334, unifying the open mode arguments for saner
- operation.
-
-2006-12-21 19:52 +0000 [r48768] Luigi Rizzo <rizzo@icir.org>
-
- * apps/app_sms.c: put generator functions next to each other.
-
-2006-12-21 19:44 +0000 [r48767] Steve Murphy <murf@digium.com>
-
- * include/asterisk.h, channels/chan_zap.c, apps/app_meetme.c,
- apps/app_festival.c, apps/app_dictate.c, apps/app_record.c,
- res/res_convert.c, channels/chan_iax2.c, res/res_monitor.c,
- cdr/cdr_sqlite.c, res/res_agi.c, main/file.c, main/app.c,
- apps/app_sms.c, apps/app_directory.c, apps/app_chanspy.c,
- apps/app_mixmonitor.c, main/db.c, apps/app_voicemail.c: a quick
- fix to app_sms.c to get rid of cursed compiler warnings so I can
- compile under --enable-dev-mode
-
-2006-12-21 19:36 +0000 [r48736-48766] Luigi Rizzo <rizzo@icir.org>
-
- * main/channel.c: same as in other places, check that
- generator->release is not NULL before calling it. This allows
- generators to set it to NULL when they have nothing to do there.
- Later, the three copies of the code that releases a generator
- should be moved to a function.
-
- * apps/app_sms.c: reduce indentation
-
- * apps/app_sms.c: restructure a block to reduce nesting
-
- * apps/app_sms.c: Add a bit of documentation on this code,
- including pointers to relevant documents and comment on timing
- issues. Initial merge of the code in
- http://bugs.digium.com/view.php?id=8586 by Filippo Grassilli
- (Hyppo) to support the SMS Protocol 2. In this commit i have
- tried to minimize the diffs, so further code cleanup will come in
- subsequent commits.
-
-2006-12-21 15:52 +0000 [r48723] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_config.c: This small update will generate WARNINGS if
- there is garbage in your extensions.conf file (liken extem =>
- instead of exten => !)
-
-2006-12-21 04:05 +0000 [r48680-48709] Joshua Colp <jcolp@digium.com>
-
- * include/asterisk/indications.h, main/indications.c: Really clean
- up indications to use the linkedlists API
-
- * main/pbx.c: Switch list of global variables to read/write locks.
-
- * main/pbx.c: Convert alternate dialplan switch list to use
- read/write locks.
-
-2006-12-21 00:24 +0000 [r48663] Steve Murphy <murf@digium.com>
-
- * configs/iax.conf.sample, main/jitterbuf.c, include/jitterbuf.h,
- CHANGES, channels/chan_iax2.c: As per bug 7978, this version
- introduces the jittertargetextra option in config files
-
-2006-12-21 00:11 +0000 [r48661-48662] Matthew Fredrickson <creslin@digium.com>
-
- * codecs/codec_g722.c: Minor addition giving props to Steve
- Underwood for his hard work. Thanks again Steve!
-
- * codecs/Makefile, codecs/g722/Makefile (added),
- codecs/codec_g722.c (added), codecs/g722/g722_encode.c (added),
- codecs/g722 (added), build_tools/embed_modules.xml,
- codecs/g722/g722_decode.c (added), codecs/g722/g722.h (added),
- codecs/g722_slin_ex.h (added), codecs/slin_g722_ex.h (added): Add
- codec G.722 support.
-
-2006-12-20 04:32 +0000 [r48638-48639] Joshua Colp <jcolp@digium.com>
-
- * apps/app_page.c: Clean up app_page
-
- * /, apps/app_voicemail.c: Merged revisions 48637 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48637 | file | 2006-12-19 21:56:09 -0500 (Tue, 19 Dec 2006) | 2
- lines vms doesn't exist on non-IMAP storage builds. ........
-
-2006-12-20 00:13 +0000 [r48598-48599] Luigi Rizzo <rizzo@icir.org>
-
- * apps/app_sms.c: more formatting cleanup. Move some code into a
- function sms_compose1() in preparation for supporting protocol 2
- as well.
-
- * apps/app_sms.c: formatting and code cleanup. Still a lot of
- copy&pasted code here...
-
-2006-12-19 23:05 +0000 [r48591-48597] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 48596 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48596 | file | 2006-12-19 18:04:30 -0500 (Tue, 19 Dec 2006) | 2
- lines Pass 'vms' pointer to record_and_review so it is then
- passed to the IMAP store file function. (issue #8614 reported by
- punknow) ........
-
- * res/snmp/agent.c: Update res_snmp to use new API declaration of
- pbx_builtin_serialize_variables (issue #8627 reported by
- johann8384)
-
- * /, doc/snmp.txt: Merged revisions 48592 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48592 | file | 2006-12-19 17:00:57 -0500 (Tue, 19 Dec 2006) | 2
- lines find is not the same as bind when it comes to
- documentation. (issue #8626 reported by johann8384) ........
-
- * res/res_limit.c: OpenBSD does not have RLIMIT_AS or RLIMIT_VMEM
- so until someone finds the right rlimit to use then let's not
- support the -v option on OpenBSD. (issue #8543 reported by jtodd)
-
-2006-12-19 21:32 +0000 [r48588-48589] Luigi Rizzo <rizzo@icir.org>
-
- * /: block 48583
-
- * apps/app_sms.c: start documenting this code. On passing, fix the
- bogus datalen on outgoing frames just fixed in 1.4 rev.48583
-
-2006-12-19 21:28 +0000 [r48587] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, channels/Makefile: Merged revisions 48586 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48586 | kpfleming | 2006-12-19 15:28:04 -0600 (Tue, 19 Dec 2006)
- | 2 lines suppress compiler warnings in this module until it can
- be improved ........
-
-2006-12-19 16:36 +0000 [r48580-48581] Luigi Rizzo <rizzo@icir.org>
-
- * apps/app_dial.c: better name for struct dial_localuser.
-
- * main/cli.c: remove now useless extern declarations.
-
-2006-12-19 14:57 +0000 [r48578] Kevin P. Fleming <kpfleming@digium.com>
-
- * res/res_config_odbc.c, /, funcs/func_odbc.c: Merged revisions
- 48577 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48577 | kpfleming | 2006-12-19 08:57:09 -0600 (Tue, 19 Dec 2006)
- | 2 lines use the proper variable type for these unixODBC API
- calls, eliminating warnings on 64-bit platforms that use the
- 'new' 64-bit types for ODBC API calls ........
-
-2006-12-19 09:58 +0000 [r48573-48575] Luigi Rizzo <rizzo@icir.org>
-
- * apps/app_dial.c: introduce a temporary variable for tmp->chan to
- shorten expressions.
-
- * apps/app_dial.c: stop what i think is a memory leak in case Dial
- fails to connect to a channel. Before committing to 1.4 i would
- like some other people to review and test this fix - thanks.
-
- * apps/app_dial.c: move a large block related to privacy handling
- to a separate function.
-
-2006-12-19 03:47 +0000 [r48572] Joshua Colp <jcolp@digium.com>
-
- * Makefile, /: Merged revisions 48571 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48571 | file | 2006-12-18 22:46:12 -0500 (Mon, 18 Dec 2006) | 2
- lines Use env -i to start a fresh environment when going to build
- menuselect. This is more portable then using unset. (issue #8543
- reported by jtodd) ........
-
-2006-12-18 17:44 +0000 [r48568] Luigi Rizzo <rizzo@icir.org>
-
- * include/asterisk/channel.h: unbreak the macro used for
- incrementing the frame counters. I don't know when the bug was
- introduced, but with the typical usage c->fin =
- FRAMECOUNT_INC(c->fin) the frame counters stay to 0.
-
-2006-12-18 17:30 +0000 [r48565-48567] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: Clean up find_idle_thread function and use
- atomic operations for dynamic thread count.
-
- * /, channels/chan_iax2.c: Merged revisions 48564 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48564 | file | 2006-12-18 12:15:49 -0500 (Mon, 18 Dec 2006) | 2
- lines Put thread into proper list if we abort handling due to an
- error, and also hold the lock while putting it back into the
- proper idle list so we don't prematurely get a signal. (issue
- #8604 reported by arkadia) ........
-
-2006-12-18 16:57 +0000 [r48562-48563] Jason Parker <jparker@digium.com>
-
- * configure.ac: ctrl-w != w (nano search) (surprisingly, the fix
- was ever so slightly different in 1.4)
-
-2006-12-18 16:24 +0000 [r48558-48560] Luigi Rizzo <rizzo@icir.org>
-
- * include/asterisk/strings.h: apply the proposed fix for bug 8602
- http://bugs.digium.com/view.php?id=8602 (i am not sure if there
- is still missing cast in front of the alloca() call - being a
- macro this is probably triggered only when actually used). Add
- function ast_str_reset() to reinitialize the string to an empty
- string instead of playing with the internal fields.
-
- * main/cdr.c, main/pbx.c, apps/app_dumpchan.c,
- include/asterisk/cdr.h, include/asterisk/pbx.h, apps/app_queue.c,
- main/cli.c: convert the final clients of ast_build_string to use
- ast_str_*() Now the only module left using it is chan_sip.c
-
- * main/logger.c: debugging shows that we always need more than 128
- bytes for the verbose and logging messages so start with a larger
- buffer from the beginning.
-
-2006-12-18 11:59 +0000 [r48555] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, main/Makefile, codecs/gsm/Makefile, utils/astman.c,
- utils/smsq.c, codecs/ilbc/Makefile, utils/ael_main.c,
- codecs/lpc10/Makefile: Merged revisions 48554 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48554 | kpfleming | 2006-12-18 05:59:24 -0600 (Mon, 18 Dec 2006)
- | 3 lines remove some now-unnecessary explicit includes of
- autoconfig.h clean up per-file dependencies during 'make clean'
- ........
-
-2006-12-18 11:28 +0000 [r48550-48553] Luigi Rizzo <rizzo@icir.org>
-
- * main/manager.c: Replace ast_build_string with ast_str_*(). On
- passing remove presumably duplicate code to generate the message
- for the manager_hooks: in the previous version, the message was
- almost the same as the one sent to regular sessions, with the
- exception of the empty line at the end, and a few (presumably
- unintentional) differences e.g. timestamps, debugging, and
- lowercase headers for "event" and "privilege". now we reuse the
- same message as before.
-
- * funcs/func_realtime.c: replace ast_build_string() with
- ast_str_*(). Unless i am very mistaken, function_realtime_read()
- was broken in that it would always return an empty string
- (because ast_build_string() advanced the pointer to the end of
- the string, and there was no reference to the initial value. This
- commit should fix this problem.
-
- * apps/app_queue.c: replace ast_build_string() with ast_str_*();
- simplify __queues_show()
-
-2006-12-17 18:33 +0000 [r48549] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, build_tools/prep_tarball: Merged revisions 48548 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r48548 | kpfleming | 2006-12-17 12:33:24 -0600 (Sun, 17
- Dec 2006) | 2 lines need an additional argument here to make the
- downloads actually occur ........
-
-2006-12-17 12:47 +0000 [r48543] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_sip.c: define a mask SIP_INSECURE sam as for other
- sets of flags.
-
-2006-12-16 21:38 +0000 [r48522-48529] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
- acinclude.m4: Merged revisions 48528 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48528 | kpfleming | 2006-12-16 15:34:41 -0600 (Sat, 16 Dec 2006)
- | 2 lines use m4 quoting for AC_MSG_NOTICE calls, to keep these
- calls from thinking they have multiple arguments ........
-
- * /, agi, codecs, utils, main/Makefile, apps,
- Makefile.moddir_rules, Makefile.rules, cdr, codecs/ilbc, formats,
- utils/Makefile, agi/Makefile, Makefile, funcs, main/db1-ast,
- codecs/lpc10, build_tools/mkdep (removed), main, codecs/gsm, res,
- pbx, channels: Merged revisions 48525 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48525 | kpfleming | 2006-12-16 15:14:34 -0600 (Sat, 16 Dec 2006)
- | 2 lines simplify dependency tracking system, using the
- compiler's built-in method for generating them, and only doing
- dependency tracking if developer mode is enabled via the
- configure script ........
-
- * funcs/func_curl.c: update to use trunk's version of the
- threadstorage API
-
- * Makefile, include/asterisk.h, /, main/stdtime/localtime.c: Merged
- revisions 48521 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48521 | kpfleming | 2006-12-16 14:12:41 -0600 (Sat, 16 Dec 2006)
- | 2 lines since we really, really have to have autoconfig.h
- included before all other headers (especially system headers),
- the Makefile will now force it to happen (this will fix build
- problems with files like ast_expr2f.c, where we can't control the
- inclusion order in the file itself) ........
-
-2006-12-16 11:23 +0000 [r48515-48520] Luigi Rizzo <rizzo@icir.org>
-
- * main/utils.c: forgot this part...
-
- * main/cli.c: another conversion from ast_build_str to ast_str
-
- * main/translate.c: convert ast_build_str to ast_str_*
-
- * include/asterisk/http.h, main/manager.c, main/http.c,
- include/asterisk/strings.h: replace ast_build_string() with
- ast_str_*() functions. This makes the code easier to follow and
- saves some copies to intermediate buffers.
-
-2006-12-16 04:25 +0000 [r48514] Kevin P. Fleming <kpfleming@digium.com>
-
- * funcs/func_curl.c, /: Merged revisions 48513 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48513 | kpfleming | 2006-12-15 22:25:09 -0600 (Fri, 15 Dec 2006)
- | 2 lines instead of initializing the curl library every time the
- CURL() function is invoked, do it only once per thread (this
- allows multiple calls to CURL() in the dialplan for a channel to
- run much more quickly, and also to re-use connections to the
- server) (thanks to JerJer for frequently complaining about this
- performance problem) ........
-
-2006-12-16 02:42 +0000 [r48508-48512] Luigi Rizzo <rizzo@icir.org>
-
- * res/res_limit.c: prevent a compiler warning
-
- * main/manager.c, main/logger.c, main/utils.c,
- include/asterisk/strings.h, main/cli.c: simplify the
- ast_dynamic_str_*.... routines by renaming them to ast_str ...
- and putting the struct ast_threadstorage pointer into the struct
- ast_str. This makes the code a lot more readable. At this point
- we can use these routines also to replace ast_build_string().
-
- * include/asterisk/utils.h, main/utils.c,
- include/asterisk/strings.h, include/asterisk/threadstorage.h:
- move the dynamic string support in a better place i.e. string.h
- While doing this, add a bit of documentation, and slightly extend
- the functionality as follows: + a max_len of -1 means that we
- take whatever the current size is, and never try to extend the
- buffer; + add support for alloca()-ted dynamic strings, which is
- very useful for all cases where we do an ast_build_string() now.
- Next step is to simplify the interface by using shorter names
- (e.g. ast_str as a prefix) and removing the _thread variant of
- the functions by saving the threadstorage reference into the
- struct ast_str. This can be done by overloading the 'type' field.
- Finally, I will do my best to remove the convoluted interface
- that results from trying to support platforms without va_copy().
-
- * res/res_smdi.c: remove a duplicate include
-
-2006-12-15 19:57 +0000 [r48503-48507] Joshua Colp <jcolp@digium.com>
-
- * /, main/rtp.c: Merged revisions 48506 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48506 | file | 2006-12-15 14:55:28 -0500 (Fri, 15 Dec 2006) | 2
- lines Turn payload_lock into bridge_lock and make it encompass
- all RTP structure contents that may relate to bridge information,
- including who we are bridged to. ........
-
- * /, channels/chan_iax2.c: Merged revisions 48504 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48504 | file | 2006-12-15 14:38:51 -0500 (Fri, 15 Dec 2006) | 2
- lines Hold call structure lock in places where a qualify or peer
- action can destroy it. ........
-
- * /, channels/chan_iax2.c: Merged revisions 48502 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48502 | file | 2006-12-15 14:24:15 -0500 (Fri, 15 Dec 2006) | 2
- lines Lock network retransmission queue in all places that it is
- used. ........
-
-2006-12-15 18:37 +0000 [r48495-48501] Luigi Rizzo <rizzo@icir.org>
-
- * main/manager.c: unbreak the output for http session. Not long ago
- i replaced lseek() with fseek() but forgot that filr FILE's you
- need ftell to give you the current position.
-
- * main/channel.c, include/asterisk/channel.h: remove
- ast_safe_string_alloc() - it is completely equivalent to
- asprintf().
-
- * channels/chan_zap.c: replace ast_safe_string_alloc() with
- asprintf()
-
- * channels/chan_features.c: replace ast_safe_string_alloc() with
- asprintf()
-
- * include/asterisk/threadstorage.h: small documentation
- improvements.
-
-2006-12-15 13:36 +0000 [r48485-48491] Olle Johansson <oej@edvina.net>
-
- * main/tdd.c, include/asterisk/tdd.h: Doxygen changes
-
- * /, channels/chan_sip.c: Issue #8592 - treat 504 as congestion
- (imported from 1.2/1.4)
-
- * /, channels/chan_sip.c: Update to latest IANA specs
-
-2006-12-15 06:34 +0000 [r48479-48480] Joshua Colp <jcolp@digium.com>
-
- * include/asterisk/lock.h: Add support to see what holds the lock
- when doing trylock.
-
- * /, channels/chan_iax2.c: Merged revisions 48478 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48478 | file | 2006-12-15 01:28:05 -0500 (Fri, 15 Dec 2006) | 2
- lines Use a wakeup variable so that we don't wait on IO
- indefinitely if packets need to be retransmitted. ........
-
-2006-12-15 04:03 +0000 [r48476-48477] Luigi Rizzo <rizzo@icir.org>
-
- * main/channel.c, include/asterisk/channel.h: constify
- ast_state2str() and note it is not reentrant.
-
- * main/pbx.c, include/asterisk/channel.h: remove the macro LOAD_OH
- and expand it inline in the only place where it was used.
-
-2006-12-14 17:39 +0000 [r48462-48473] Joshua Colp <jcolp@digium.com>
-
- * /, include/asterisk/rtp.h, main/rtp.c: Merged revisions 48472 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48472 | file | 2006-12-14 12:36:12 -0500 (Thu, 14 Dec 2006) | 2
- lines Payload values on the RTP structure can change AFTER a
- bridge has started. This comes from the packet handling of the
- SIP response when indication that it was answered has been sent.
- Therefore we need to protect this data with a lock when we
- read/write. (issue #8232 reported by tgrman) ........
-
- * /, main/rtp.c: Merged revisions 48461 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48461 | file | 2006-12-13 22:33:30 -0500 (Wed, 13 Dec 2006) | 2
- lines Remove direct RTCP bridging. I've come to the conclusion
- that we should handle this through the core and not just forward
- it on. Should solve a few bugs. ........
-
-2006-12-13 23:08 +0000 [r48458-48459] Luigi Rizzo <rizzo@icir.org>
-
- * main/pbx.c: make sure that showdialplan sends only one 'Response:
- Success ' message even in case of a recursive call.
-
- * main/pbx.c: clean up function manager_show_dialplan_helper()
- reducing indentation and normalizing loops. While doing this,
- remove some unused variables, fix an uninitialized string
- (idaction), and mark some places where the behaviour is not what
- we would expect (e.g. an empty context is reported as an error
- same as a non-existent one). Given that this function is not in
- 1.4, the above can be changed without too many backward
- compatibility concerns. Not applicable to 1.4 or below.
-
-2006-12-13 21:23 +0000 [r48455] Matt O'Gorman <mogorman@digium.com>
-
- * codecs/codec_zap.c: support for deactivating translation paths
- that are no longer available and more descriptive show transcoder
- cli command.
-
-2006-12-13 00:56 +0000 [r48433] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c: revert check for a zaptel transcoder related
- definition that was done in the wrong module.
-
-2006-12-12 23:28 +0000 [r48432] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, build_tools/prep_tarball: Merged revisions 48427 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r48427 | kpfleming | 2006-12-12 17:18:14 -0600 (Tue, 12
- Dec 2006) | 2 lines when making a release, we can always use wget
- and we can't run the configure script to find that out...
- ........
-
-2006-12-12 22:32 +0000 [r48416-48417] Russell Bryant <russell@digium.com>
-
- * include/asterisk/app.h, channels/chan_sip.c,
- include/asterisk/channel.h, include/asterisk/pbx.h: Fix various
- spelling mistakes in comments.
-
- * channels/chan_zap.c: Make chan_zap inform you that your version
- of zaptel is too old instead of just failing to compile. It seems
- like the proper way to do this would be in the configure script.
- However, that wouldn't help existing checkouts unless we forced
- the configure script to be executed after any code was changed.
-
-2006-12-12 19:55 +0000 [r48415] Matt O'Gorman <mogorman@digium.com>
-
- * codecs/codec_zap.c: fixed nubb error on my part, transcoder now
- unlocks and locks correctly, as well as counts in the correct
- direction.
-
-2006-12-12 10:36 +0000 [r48408-48410] Luigi Rizzo <rizzo@icir.org>
-
- * main/manager.c: properly initialize a malloc'ed buffer
-
- * main/manager.c: normalize the scanning of "general" options in
- the config file.
-
- * main/cli.c: Make sure tab-completion works even when we have
- typed a fully matching word (e.g. "sip<TAB>"); this is
- implemented by this one-line change - for (;; dst++, src += n) {
- + for (;src < argindex; dst++, src += n) { However this code is
- not exactly trivial to understand, so i am also adding some
- comments to help figuring out what it does.
-
-2006-12-12 04:14 +0000 [r48402] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 48401 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48401 | file | 2006-12-11 23:13:48 -0500 (Mon, 11 Dec 2006) | 2
- lines Use S_OR in my previous app_voicemail. This is the way it
- should have been done. ........
-
-2006-12-11 23:02 +0000 [r48397-48400] Matt O'Gorman <mogorman@digium.com>
-
- * /, sounds/Makefile: Merged revisions 48399 via svnmerge from
- https://svn.digium.com/svn/asterisk/branches/1.4 ........ r48399
- | mogorman | 2006-12-11 17:02:10 -0600 (Mon, 11 Dec 2006) | 2
- lines new sounds package with 100% more silence ........
-
- * /, apps/app_externalivr.c: Merged revisions 48396 via svnmerge
- from https://svn.digium.com/svn/asterisk/branches/1.4
- ................ r48396 | mogorman | 2006-12-11 16:11:35 -0600
- (Mon, 11 Dec 2006) | 12 lines Merged revisions 48394 via svnmerge
- from https://svn.digium.com/svn/asterisk/branches/1.2 ........
- r48394 | mogorman | 2006-12-11 15:55:43 -0600 (Mon, 11 Dec 2006)
- | 4 lines app_externalivr needs a real silence file, and
- additional changes to add silence files into core instead of
- extra patch provided by bug 8177 with minor additions. ........
- ................
-
-2006-12-11 21:35 +0000 [r48392] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 48391 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48391 | file | 2006-12-11 16:31:23 -0500 (Mon, 11 Dec 2006) | 2
- lines Return non-existant callerid handling to that which it was
- before. In 1.4 and trunk callerid can be allocated but not have
- any contents so we have to use ast_strlen_zero before passing it
- to the relevant functions. (issue #8567 reported by pabelanger)
- ........
-
-2006-12-11 21:04 +0000 [r48390] Matt O'Gorman <mogorman@digium.com>
-
- * codecs/codec_zap.c: add support for dynamic channel creation and
- destruction, and show transcoder to show number of channels in
- use.
-
-2006-12-11 18:11 +0000 [r48389] Luigi Rizzo <rizzo@icir.org>
-
- * main/manager.c: make sure the argument to ast_malloc() is > 0.
- Long explaination: The behaviour of the underlying malloc(0)
- differs depending on the operating system. Some return NULL (SysV
- behaviour); some still allocate a small chunk of memory and
- return a valid pointer (e.g. traditional BSD); some (e.g. FreeBSD
- 6.x) return a non-null pointer that causes a memory fault if
- used, even just for reading. Given the above variety, better
- never call malloc(0).
-
-2006-12-11 17:00 +0000 [r48388] Steve Murphy <murf@digium.com>
-
- * main/app.c: This update fixes the problem reported in bug 8551;
- that ast_app_getdata() behaves differently in trunk for the case
- of a null prompt.
-
-2006-12-11 05:40 +0000 [r48384] Tilghman Lesher <tlesher@digium.com>
-
- * /, funcs/func_strings.c: Merged revisions 48382 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48382 | tilghman | 2006-12-10 23:37:09 -0600 (Sun, 10 Dec 2006)
- | 2 lines STRFTIME() does not actually require an argument (issue
- 8540) ........
-
-2006-12-11 05:38 +0000 [r48378-48383] Joshua Colp <jcolp@digium.com>
-
- * /, main/rtp.c: Merged revisions 48381 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48381 | file | 2006-12-11 00:36:45 -0500 (Mon, 11 Dec 2006) | 2
- lines Merge in my latest RTP changes. Break out RTP and RTCP
- callback functions so they no longer share a common one. ........
-
- * /, apps/app_meetme.c: Merged revisions 48379 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48379 | file | 2006-12-11 00:30:01 -0500 (Mon, 11 Dec 2006) | 2
- lines Use the correct API call to say a device state changed.
- (Yes, I'm a nub.) ........
-
- * /, apps/app_meetme.c: Merged revisions 48377 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48377 | file | 2006-12-10 23:57:38 -0500 (Sun, 10 Dec 2006) | 2
- lines Don't access the conference structure after it has been
- freed. ........
-
-2006-12-11 00:52 +0000 [r48376] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_nbscat.c, /, apps/app_festival.c, apps/app_mp3.c,
- res/res_agi.c, apps/app_zapras.c, apps/app_externalivr.c,
- apps/app_ices.c, res/res_musiconhold.c: Merged revisions 48375
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r48375 | tilghman | 2006-12-10 18:47:21 -0600
- (Sun, 10 Dec 2006) | 13 lines Merged revisions 48374 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r48374 | tilghman | 2006-12-10 18:33:59 -0600 (Sun, 10
- Dec 2006) | 5 lines When doing a fork() and exec(), two problems
- existed (Issue 8086): 1) Ignored signals stayed ignored after the
- exec(). 2) Signals could possibly fire between the fork() and
- exec(), causing Asterisk signal handlers within the child to
- execute, which caused nasty race conditions. ........
- ................
-
-2006-12-10 03:14 +0000 [r48373] Steve Murphy <murf@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 48372 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r48372 | murf | 2006-12-09 20:04:18 -0700 (Sat,
- 09 Dec 2006) | 9 lines Merged revisions 48371 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r48371 | murf | 2006-12-09 19:14:13 -0700 (Sat, 09 Dec 2006) | 1
- line This version applies the patch suggested by stevens in bug
- 7836 (make inbound channel RINGING state consistent with other
- channels). ........ ................
-
-2006-12-09 16:44 +0000 [r48359-48365] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: convert the thread IO state and type to use
- enums.
-
- * /, channels/chan_iax2.c: Merged revisions 48363 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48363 | russell | 2006-12-09 10:59:42 -0500 (Sat, 09 Dec 2006) |
- 8 lines Use locking when accessing the registrations list. This
- list is not actually used very often, so the likelihood of there
- being a problem is pretty small, but still possible. For example,
- if the CLI command to list the registrations was called at the
- same time that a reload was occurring and the registrations list
- was getting destroyed and rebuilt, a crash could occur. In
- passing, go ahead and convert this list to use the linked list
- macros. ........
-
- * channels/chan_iax2.c: chan_iax2 has an extremely large function,
- socket_process(), to handle incoming frames. The function, before
- this commit, was roughly 1400 lines long. So, I am working on
- breaking this up into functions so that the code is easier to
- follow and debug. Also, I will be committing these changes in
- chunks as I do them to ease tracking down any potentially
- introduced problems. Break out roughly 150 lines from
- socket_process() and introduce a new function,
- socket_process_meta() which handles the parsing of an incoming
- meta frame. Also, restructure some of this code a bit to reduce
- the deep nesting that was in this code.
-
- * channels/chan_iax2.c: - Fix a few spelling mistakes - Use
- sizeof() to pass an array size to a function - Use a single bit
- for a variable in the chan_iax2_pvt stuct since that is all it
- needs. - Add some comments about the iaxs, iaxl, and lastused
- arrays.
-
-2006-12-07 18:21 +0000 [r48358] Russell Bryant <russell@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 48357 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r48357 | russell | 2006-12-07 13:17:28 -0500
- (Thu, 07 Dec 2006) | 11 lines Merged revisions 48356 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r48356 | russell | 2006-12-07 13:14:13 -0500 (Thu, 07
- Dec 2006) | 3 lines Ensure that the file position is not
- incremented beyond the total number of files available for
- playback. (issue #8539, ulogic) ........ ................
-
-2006-12-07 16:42 +0000 [r48351] Luigi Rizzo <rizzo@icir.org>
-
- * include/asterisk/http.h, main/manager.c, main/http.c,
- configs/manager.conf.sample: - Generalize the function
- ssl_setup() so that the certificate info are passed as an
- argument. - Update the code in main/http.c to use the new
- interface (the diff is large but mostly mechanical, due to the
- name change of several variables); - And since now it is trivial,
- implement "AMI over TLS", and document the possible options in
- manager.conf - And since the test client (openssl s_client
- -connect host:port ) does not generate \r\n as a line terminator,
- make get_input() also accept just a \n as a line terminator (Mac
- users: do you also need the \r-only version ?) The option parsing
- in manager.conf is not very efficient, and needs to be cleaned up
- and made similar to what we have in http.conf
-
-2006-12-07 16:03 +0000 [r48350] Steve Murphy <murf@digium.com>
-
- * main/manager.c, /: Merged revisions
- 47986,47995,47997,48001,48003-48004,48008-48014,48016,48018-48019
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r47986 | oej | 2006-11-24 07:00:19 -0700 (Fri,
- 24 Nov 2006) | 6 lines Doxygen update - Document cause codes -
- Document a bit more on channel variables - global, predefined and
- local - Fix some doxygen in channel.h. Adding one comment for two
- definitions does not work. They won't be copied to each.
- ................ r47995 | murf | 2006-11-24 10:40:49 -0700 (Fri,
- 24 Nov 2006) | 1 line This fix inspired by a patch supplied in
- bug 8189, which points out problems with the PLC code
- ................ r47997 | murf | 2006-11-24 11:17:25 -0700 (Fri,
- 24 Nov 2006) | 1 line removed the svnmerge-integrated property
- from trunk; it's confusing svnmerge in newly created branches
- ................ r48001 | rizzo | 2006-11-25 02:02:42 -0700 (Sat,
- 25 Nov 2006) | 5 lines set pointers to NULL after freeing memory
- to avoid multiple free() probably 1.4/1.2 issue as well if
- someone can look into that. ................ r48003 | oej |
- 2006-11-25 02:45:57 -0700 (Sat, 25 Nov 2006) | 9 lines - Adding
- comment on suspicious memory allocation. Seems like it's never
- freed, but I don't have a clear understanding of the frame
- allocation/deallocation, so I just mark this for investigation.
- (Reported by Ed Guy). We're trying to see if a free() hurts... -
- Doxygen comments on p2p rtp bridge stuff. I am a bit worried
- about shortcutting rtcp this way, but will need feedback from
- rtcp gurus. This should work for video calls too, and possibly
- UDPTL. ................ r48004 | oej | 2006-11-25 02:48:30 -0700
- (Sat, 25 Nov 2006) | 2 lines Changing ERROR to lesser level.
- Imported from 1.2/1.4 ................ r48008 | rizzo |
- 2006-11-25 10:37:04 -0700 (Sat, 25 Nov 2006) | 7 lines generalize
- a bit the functions used to create an tcp socket and then run a
- service on it. The code in manager.c does essentially the same
- things, so we will be able to reuse the code in here (probably
- moving it to netsock.c or another appropriate library file).
- ................ r48009 | mattf | 2006-11-25 13:30:04 -0700 (Sat,
- 25 Nov 2006) | 1 line Updates to show linkset command
- ................ r48010 | mattf | 2006-11-25 13:54:38 -0700 (Sat,
- 25 Nov 2006) | 2 lines Add ss7 show linkset command
- ................ r48011 | mattf | 2006-11-25 14:32:33 -0700 (Sat,
- 25 Nov 2006) | 1 line Make sure we don't send a group reset on a
- group larger than 32 CICs ................ r48012 | mattf |
- 2006-11-25 14:35:23 -0700 (Sat, 25 Nov 2006) | 1 line bug fix
- ................ r48013 | mattf | 2006-11-25 14:46:58 -0700 (Sat,
- 25 Nov 2006) | 1 line Make compiler happier ................
- r48014 | mattf | 2006-11-25 14:50:42 -0700 (Sat, 25 Nov 2006) | 1
- line Little fix so we use the right message ................
- r48016 | murf | 2006-11-25 17:15:42 -0700 (Sat, 25 Nov 2006) | 9
- lines Merged revisions 48015 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48015 | murf | 2006-11-25 17:01:34 -0700 (Sat, 25 Nov 2006) | 1
- line A little bit of func_cdr documentation upgrade-- no bug#
- involved, although 8221 may have inspired it. ........
- ................ r48018 | murf | 2006-11-25 17:31:13 -0700 (Sat,
- 25 Nov 2006) | 9 lines Merged revisions 48017 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48017 | murf | 2006-11-25 17:26:16 -0700 (Sat, 25 Nov 2006) | 1
- line might as well also document the raw values of the flag vars
- ........ ................ r48019 | russell | 2006-11-25 23:55:33
- -0700 (Sat, 25 Nov 2006) | 6 lines - Add some comments on thread
- storage with a brief explanation of what it is as well as what
- the motivation is for using it. - Add a comment by the
- declaration of ast_inet_ntoa() noting that this function is not
- reentrant, and the result of a previous call to the function is
- no longer valid after calling it again. ................
-
-2006-12-06 20:46 +0000 [r48332-48338] Luigi Rizzo <rizzo@icir.org>
-
- * main/manager.c: remove duplicated code to start the server
- threads, use the infrastructure exposed in http.c earlier today.
- As a bonus, now we can restart the session on a different port
- just reloading the module. On passing, fix a bug in the handling
- of 'enabled' in the configuration file - previously, a missing
- "enabled=" line in manager.conf meant "whatever the state was
- before" instead of a specific value.
-
- * main/manager.c: Part of the transformations necessary to add TLS
- support, as described in
- http://lists.digium.com/pipermail/asterisk-dev/2006-December/025213.html
- In detail, this commit does the following: b) change the function
- get_input() to use fread() instead of read() to collect the data.
- One can still do the ast_wait_for_input() on the original
- descriptor returned by accept(). c) change the function
- send_string() to work on the FILE *. As a side effect, this
- change now really guarantees that we don't spend more than
- "writetimeout" milliseconds on each line sent. d) modify the
- function action_command() so that it creates a temporary file
- descriptor to be passed to ast_cli_command(), and then read back
- the data from the temp file and write it to the output with
- send_string(). The code is similar to what is done in
- generic_http_callback() to support AMI-over-HTTP.
-
-2006-12-06 16:54 +0000 [r48327] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Handle multiple 487's correctly
-
-2006-12-06 16:19 +0000 [r48325] Russell Bryant <russell@digium.com>
-
- * configs/iax.conf.sample, /: Merged revisions 48323 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r48323 | russell | 2006-12-06 11:15:45 -0500
- (Wed, 06 Dec 2006) | 11 lines Merged revisions 48322 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r48322 | russell | 2006-12-06 11:05:54 -0500 (Wed, 06
- Dec 2006) | 3 lines Fix the name of the rtignoreregexpire option
- in the sample configuration file. (issue #8526, arkadia) ........
- ................
-
-2006-12-06 16:17 +0000 [r48324] Luigi Rizzo <rizzo@icir.org>
-
- * include/asterisk/http.h, main/http.c: Make externally visible
- some generic code useful to create and implement services over
- tcp and/or tcp-tls. This commit is nothing more than moving
- structure definitions (and documentation) from main/http.c to
- include/asterisk/http.h (temporary location until we find a
- better place), and removing the 'static' qualifier from
- server_root() and server_start(). The name change (adding the
- ast_ prefix as a minimum, and then possibly a more meaningful
- name) is postponed to future commits. Does not apply to other
- versions of asterisk.
-
-2006-12-06 12:34 +0000 [r48318] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Don't send Contact in SIP Messages
- (imported from 1.2/1.4). Reported by Gunnar at Omnitor.
-
-2006-12-06 07:39 +0000 [r48299-48307] Russell Bryant <russell@digium.com>
-
- * apps/app_osplookup.c, apps/app_meetme.c, apps/app_queue.c,
- apps/app_voicemail.c: Resolve some pointer signedness compiler
- warnings in app_osplookup, and constify a bunch of usage strings
- for CLI commands.
-
- * channels/chan_local.c, channels/chan_skinny.c,
- channels/chan_agent.c, channels/chan_features.c,
- channels/chan_alsa.c, channels/iax2-provision.c,
- channels/chan_gtalk.c, channels/chan_oss.c, channels/chan_mgcp.c:
- Constify a bunch of usage strings for CLI commands.
-
- * res/res_config_pgsql.c, res/res_limit.c, res/res_agi.c,
- res/res_crypto.c, res/res_realtime.c, res/res_jabber.c,
- res/res_odbc.c: Constify a bunch of usage strings for CLI
- commands.
-
- * main/channel.c, main/udptl.c, main/frame.c, main/translate.c,
- main/file.c, pbx/pbx_dundi.c, main/db.c, main/rtp.c: Staticize
- one, and Constify a bunch of usage strings for CLI commands.
-
- * channels/chan_zap.c, channels/chan_sip.c, channels/chan_iax2.c,
- main/asterisk.c, main/cli.c: Constify a bunch of the usage
- strings for CLI commands.
-
- * channels/chan_iax2.c: Instead of creating an unused instance of
- an unnamed enum, give it a name.
-
- * include/asterisk/cli.h: Make the "usage" member of the
- ast_cli_entry struct const to resolve a compiler warning.
-
-2006-12-05 20:46 +0000 [r48282] Joshua Colp <jcolp@digium.com>
-
- * configure: Regenerate configure for Qwell's last commit.
-
-2006-12-05 20:44 +0000 [r48280] Jason Parker <jparker@digium.com>
-
- * /, configure.ac: Merged revisions 48279 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48279 | qwell | 2006-12-05 14:42:52 -0600 (Tue, 05 Dec 2006) | 4
- lines Fix curl version number testing to be much more friendly to
- non-bash shells. Issue 8508, patch by me. This *SHOULD* be POSIX
- compliant now.. ........
-
-2006-12-05 20:39 +0000 [r48277] Olle Johansson <oej@edvina.net>
-
- * include/asterisk/rtp.h, include/asterisk/channel.h, main/rtp.c:
- Doxygen updates
-
-2006-12-05 20:15 +0000 [r48276] Jason Parker <jparker@digium.com>
-
- * main/tdd.c, include/asterisk/fskmodem.h, main/callerid.c,
- main/fskmodem.c: Expand on r48273 (from issue 8506), to translate
- more of the fskmodem stuff to English. r48273 dealt with the
- comments and such, this deals with the code itself. (This
- couldn't have been easily done if it weren't for 48273 - thanks
- again for that merbanan)
-
-2006-12-05 19:41 +0000 [r48269-48273] Olle Johansson <oej@edvina.net>
-
- * include/asterisk/fskmodem.h, main/fskmodem.c: Issue #8506 -
- translate spanish comments in fskmodem to english (according to
- bug guidelines) Thanks merbanan!
-
- * /: Blocking invitestate patch that is already merged to svn
- trunk.
-
- * /, configs/sip.conf.sample: Adding docs on t.38
-
-2006-12-05 14:33 +0000 [r48266] TransNexus OSP Development <support@transnexus.com>
-
- * apps/app_osplookup.c: 1. Change to remove the compiling warning:
- "app_osplookup.c:2169: warning: initialization discards
- qualifiers from pointer target type"
-
-2006-12-05 11:09 +0000 [r48258-48259] Olle Johansson <oej@edvina.net>
-
- * main/frame.c, include/asterisk/frame.h, main/rtp.c: Well, yes...
-
- * main/frame.c, include/asterisk/frame.h, main/rtp.c: Reserving
- flags for coming code (currently in the "videocaps" branch)
- implementing T.140 support in RTP. T.140/RFC 4351 is TDD over IP
- - text telephony for hearing impaired. It defines a realtime text
- chat, much like the old "talk" application in Unix. T.140 is
- character by character in real time. It's not the same as our
- current MESSAGE format - that is more like IM, but can be
- gatewayed to MESSAGE with a text "codec" if needed. More patches
- will follow, as soon as we've separated this code from the video
- capabilities functions in the videocaps branch. Code by John
- Martin, Aupix (disclaimer on file)
-
-2006-12-05 01:46 +0000 [r48253-48255] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 48254 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48254 | tilghman | 2006-12-04 19:41:02 -0600 (Mon, 04 Dec 2006)
- | 2 lines Oops, forgot to release the odbc handle ........
-
- * /, apps/app_voicemail.c: Merged revisions 48252 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r48252 | tilghman | 2006-12-04 19:34:34 -0600
- (Mon, 04 Dec 2006) | 14 lines Merged revisions 48251 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r48251 | tilghman | 2006-12-04 19:26:08 -0600 (Mon, 04
- Dec 2006) | 6 lines If the recording in the database is too
- large, it will fail to retrieve with an mmap error. Not too sure
- why this doesn't happen when we put it in the database, also, but
- since that doesn't seem to be broken, I'm not going to fix it (at
- least until someone reports it). Solution is to ask for the file
- in smaller chunks. (Bug 8385) ........ ................
-
-2006-12-04 21:49 +0000 [r48249] Jason Parker <jparker@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 48248 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48248 | qwell | 2006-12-04 15:48:41 -0600 (Mon, 04 Dec 2006) | 2
- lines Fix an issue which didn't allow unavail/greet/busy/etc
- messages from being saved into ODBC (and probably IMAP). ........
-
-2006-12-04 17:55 +0000 [r48229-48231] Jason Parker <jparker@digium.com>
-
- * /, configs/voicemail.conf.sample: Merged revisions 48230 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48230 | qwell | 2006-12-04 11:54:46 -0600 (Mon, 04 Dec 2006) | 4
- lines Add documentation to voicemail.conf.sample for ODBC
- storage. Issue 8499 - patch by blitzrage. ........
-
- * /, doc/snmp.txt: Merged revisions 48228 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48228 | qwell | 2006-12-04 11:43:24 -0600 (Mon, 04 Dec 2006) | 4
- lines Attempt to document some of the dependencies that are
- needed for net-snmp Issue 8499 - initial patch by blitzrage.
- ........
-
-2006-12-03 06:35 +0000 [r48224] Russell Bryant <russell@digium.com>
-
- * /, sounds/Makefile: Merged revisions 48223 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48223 | russell | 2006-12-03 01:34:14 -0500 (Sun, 03 Dec 2006) |
- 3 lines When "fetch" is in use, instead of "wget", --continue is
- not a valid option. (issue #8451) ........
-
-2006-12-02 22:03 +0000 [r48200-48220] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Cleaning up handle_response a bit.
- (Imported from 1.4)
-
- * .cleancount: Removing two .h files means we need to update
- cleancount to force make depend again (or ?)
-
- * channels/chan_sip.c: Send CANCEL to call with early media
- (PROGRESS INBAND). This is imported from branch "invitestate" and
- "invitestate-1.4" *** *** *** IF YOU HAVE ISSUES WITH
- BYEs/CANCELs - PLEASE UPDATE AND TEST AGAIN! *** Thank you! ***
- *** /Olle
-
- * channels/chan_sip.c: Invitestate updates
-
- * agi/Makefile: Oops. Something is wrong in the agi directory.
- Asking for autoconfig.h. I have it disabled locally, but no
- reason to commit that change.
-
- * apps/app_sms.c: Doxygenification
-
- * main/coef_out.h (removed), main/tdd.c, main/callerid.c,
- main/fskmodem.c, main/coef_in.h (removed): - Code formatting -
- remove coef_in.h and coef_out.h that was only included as data
- definitions in fskmodem.c If you understand spanish, please help
- us translate the comments in fskmodem.c. Thanks!
-
- * /, channels/chan_sip.c, include/asterisk/rtp.h,
- configs/sip.conf.sample, main/rtp.c: - Disable RTP timeouts
- during T.38 transmission - Encapsulate RTP timers to the RTP
- structure, so we have one set for video and one for audio -
- Document RTP keepalive configuration option - Cleanup and
- document the monitor support function to hangup on RTP timeouts -
- Add RTP keepalive to SIP show settings Imported from 1.4 with
- modifications for trunk.
-
-2006-12-01 23:39 +0000 [r48194] Kevin P. Fleming <kpfleming@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 48193 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r48193 | kpfleming | 2006-12-01 17:37:28 -0600
- (Fri, 01 Dec 2006) | 10 lines Merged revisions 48192 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r48192 | kpfleming | 2006-12-01 17:30:59 -0600 (Fri, 01
- Dec 2006) | 2 lines if Dial() is going to send music-on-hold to
- the calling party, it has to send PROGRESS first to ensure that
- the reverse audio path has been setup first (BE-106) ........
- ................
-
-2006-12-01 23:20 +0000 [r48191] Russell Bryant <russell@digium.com>
-
- * Makefile, /, configure, configure.ac, makeopts.in,
- sounds/Makefile: Merged revisions 48190 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48190 | russell | 2006-12-01 18:16:28 -0500 (Fri, 01 Dec 2006) |
- 12 lines FreeBSD 6.1 does not include wget by default. However,
- it has fetch which will work just fine for our purposes of
- downloading the sounds packages. So, check for both wget and
- fetch and the configure script and use what was found to download
- them. If neither one was found, and sound packages are selected
- that must be downloaded, the install process will print out an
- informative error message indicating the situation. Also, fix a
- couple places where "make" was hard coded into some output
- messages by replacing them with the $(MAKE) variable. (issue
- #8451, initial patch by pabelanger, with additional modifications
- by me) ........
-
-2006-12-01 20:49 +0000 [r48188] Olle Johansson <oej@edvina.net>
-
- * main/channel.c: Formatting fix
-
-2006-12-01 20:26 +0000 [r48187] Jason Parker <jparker@digium.com>
-
- * /, configs/extensions.conf.sample: Merged revisions 48186 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r48186 | qwell | 2006-12-01 14:25:51 -0600 (Fri,
- 01 Dec 2006) | 10 lines Merged revisions 48183 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r48183 | qwell | 2006-12-01 14:19:10 -0600 (Fri, 01 Dec 2006) | 2
- lines Fix a small typo - issue 8848, reported by pabelanger
- ........ ................
-
-2006-12-01 19:41 +0000 [r48180] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/cli.c: Merged revisions 48179 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48179 | tilghman | 2006-12-01 13:38:59 -0600 (Fri, 01 Dec 2006)
- | 2 lines Double-unlock error (reported by blitzrage on IRC)
- ........
-
-2006-12-01 18:16 +0000 [r48175-48178] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c, configs/sip.conf.sample: - Remove T.38
- early media, since T.38 requires two way communication (imported
- from 1.4) - Small fixes to limitonpeer
-
- * include/asterisk/threadstorage.h: Tiny doxygen improvement
-
-2006-11-30 21:22 +0000 [r48169] Joshua Colp <jcolp@digium.com>
-
- * /, include/asterisk/rtp.h, channels/chan_gtalk.c, main/rtp.c:
- Merged revisions 48168 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48168 | file | 2006-11-30 16:18:24 -0500 (Thu, 30 Nov 2006) | 2
- lines Do not do a partial bridge for Google Talk since we need to
- handle STUN. (issue #8448 reported by phsultan) ........
-
-2006-11-30 20:55 +0000 [r48164-48167] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Issue #8319 (imported from 1.2, 1.4) -
- Increment nonce-count properly (noriyuki)
-
- * include/asterisk/devicestate.h, main/pbx.c, main/devicestate.c,
- include/asterisk/channel.h, include/asterisk/pbx.h: Documentation
- updates
-
-2006-11-30 20:29 +0000 [r48153-48163] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 48158 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r48158 | file | 2006-11-30 15:07:55 -0500 (Thu,
- 30 Nov 2006) | 10 lines Merged revisions 48157 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r48157 | file | 2006-11-30 15:06:43 -0500 (Thu, 30 Nov 2006) | 2
- lines Only print out debug message if bridged channel is not
- NULL. (issue #8412 reported by jubilex) ........ ................
-
- * /, res/res_features.c: Merged revisions 48155 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r48155 | file | 2006-11-30 14:05:14 -0500 (Thu,
- 30 Nov 2006) | 10 lines Merged revisions 48154 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r48154 | file | 2006-11-30 14:04:11 -0500 (Thu, 30 Nov 2006) | 2
- lines Do not listen for DTMF on the bridge that comes into
- existence when ParkedCall is executed. This means native bridging
- can now occur for this. (issue #8406 reported by kebl0155)
- ........ ................
-
- * main/cdr.c, /: Merged revisions 48152 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r48152 | file | 2006-11-30 13:47:40 -0500 (Thu,
- 30 Nov 2006) | 10 lines Merged revisions 48151 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r48151 | file | 2006-11-30 13:42:45 -0500 (Thu, 30 Nov 2006) | 2
- lines Print certain CDR messages out at the NOTICE level versus
- WARNING since they can occur when used with the CDR applications
- and are perfectly fine. (issue #8367 reported by dartvader)
- ........ ................
-
-2006-11-30 18:25 +0000 [r48149-48150] Olle Johansson <oej@edvina.net>
-
- * main/devicestate.c: Small update
-
- * agi/Makefile, contrib/asterisk-ng-doxygen, agi/eagi-test.c,
- main/devicestate.c, agi/eagi-sphinx-test.c: Doxygen updates
-
-2006-11-30 18:20 +0000 [r48144-48148] Joshua Colp <jcolp@digium.com>
-
- * /, configs/sip.conf.sample: Merged revisions 48143 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r48143 | file | 2006-11-30 12:57:35 -0500 (Thu,
- 30 Nov 2006) | 10 lines Merged revisions 48142 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r48142 | file | 2006-11-30 12:55:23 -0500 (Thu, 30 Nov 2006) | 2
- lines Document 'port' for SIP peers, came up because of the
- current mailing list thread. (issue #8450 reported by blitzrage)
- ........ ................
-
-2006-11-30 17:15 +0000 [r48130-48139] Olle Johansson <oej@edvina.net>
-
- * include/asterisk/doxyref.h, main/devicestate.c: Adding some
- generic docs on extension and device states - watchers and
- providers
-
- * doc/manager.txt, /: Add information on status events
-
- * /, channels/chan_sip.c: Merging patch from 1.2/1.4. I think this
- was originally spotted by Luigi, but hit me in the back today.
-
-2006-11-30 03:29 +0000 [r48116-48123] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: I am pretty sure that oej only meant to
- change the variable name in the source, not the configuration
- option name so let's turn it back to srvlookup instead of
- global_srvlookup. (issue #8442 reported by jtodd)
-
- * /, apps/app_voicemail.c: Merged revisions 48115 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48115 | file | 2006-11-29 16:05:17 -0500 (Wed, 29 Nov 2006) | 2
- lines Use MAILTMPLEN instead of sizeof in mm_login. (issue #8420
- reported by slimey) ........
-
-2006-11-29 20:57 +0000 [r48111-48114] Olle Johansson <oej@edvina.net>
-
- * /, configs/sip.conf.sample: Clarify some settings for status
- reports in subscriptions, queues and manager
-
- * /, configs/sip.conf.sample: Explain RTP timeouts
-
- * main/rtp.c: Change logging for p2p rtp bridge mode
-
-2006-11-29 17:59 +0000 [r48109-48110] Russell Bryant <russell@digium.com>
-
- * include/asterisk/threadstorage.h: - Fix a few spelling mistakes.
- - Add some more documentation for the
- ast_dynamic_str_............() function to document the behavior
- of the function in the case of a partial write. Also, document
- the return value and note that the function should never need to
- be called directly.
-
- * main/utils.c: Go ahead and make this write unconditional. Making
- it conditional is more work in both the append and non-append
- modes. Also, always truncating the partial write makes the
- behavior of the function more consistent, where in any type of
- write, no partial result is left in the buffer. Thanks for the
- feedback, luigi
-
-2006-11-29 16:53 +0000 [r48108] Joshua Colp <jcolp@digium.com>
-
- * /, main/rtp.c: Merged revisions 48107 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r48107 | file | 2006-11-29 11:50:33 -0500 (Wed,
- 29 Nov 2006) | 10 lines Merged revisions 48106 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r48106 | file | 2006-11-29 11:47:10 -0500 (Wed, 29 Nov 2006) | 2
- lines If the frame was duplicated before writing out then we need
- to free it. (issue #8429 reported by edguy3) ........
- ................
-
-2006-11-29 05:08 +0000 [r48103] Russell Bryant <russell@digium.com>
-
- * main/utils.c: Remove an XXX command suggesting that this
- truncation should not be conditional, and also add a more verbose
- comment explaining why it is only needed in the case of appending
- to the string for any curious readers that come along in the
- future.
-
-2006-11-29 04:28 +0000 [r48100-48102] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 48101 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48101 | file | 2006-11-28 23:26:53 -0500 (Tue, 28 Nov 2006) | 2
- lines Don't crash if the mailstream was not created. ........
-
- * sounds/Makefile: Use the proper version of extra sounds. (issue
- #8441 reported by jtodd)
-
-2006-11-28 23:13 +0000 [r48098-48099] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Add a comment to note near some code that
- performs a very expensive operation that occurs for every
- incoming media frame.
-
- * codecs/codec_zap.c: resolve a couple of compiler warnings
-
-2006-11-28 18:28 +0000 [r48096] Jason Parker <jparker@digium.com>
-
- * Makefile, /: Merged revisions 48095 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48095 | qwell | 2006-11-28 12:26:53 -0600 (Tue, 28 Nov 2006) | 2
- lines Export several more variables in top level Makefile.
- Inspired by issue 8438. ........
-
-2006-11-28 17:08 +0000 [r48090] Luigi Rizzo <rizzo@icir.org>
-
- * main/manager.c: don't use outputstr in the struct mansession,
- it's just an extra allocation on a path where we have way too
- many already. Unfortunately the AMI-over-HTTP requires multiple
- copies, because we need to generate a header, then the raw output
- to an intermediate buffer, then convert it to html/xml, and
- finally copy everything into a malloc'ed buffer because that's
- what the generic_http_callback interface expects.
-
-2006-11-28 16:59 +0000 [r48089] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_phone.c, /: Merged revisions 48088 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r48088 | file | 2006-11-28 11:57:16 -0500 (Tue,
- 28 Nov 2006) | 10 lines Merged revisions 48087 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r48087 | file | 2006-11-28 11:56:01 -0500 (Tue, 28 Nov 2006) | 2
- lines According to the research I have done we never needed to
- include compiler.h in the first place so let's not! (issue #8430
- reported by edguy3) ........ ................
-
-2006-11-28 15:53 +0000 [r48062-48086] Luigi Rizzo <rizzo@icir.org>
-
- * main/manager.c: initialize the dynamic string in a sane way.
-
- * main/utils.c: some simplifications to
- ast_dynamic_str_thread_build_va_couldnt_we_choose_a_shorter_name()
- I am unsure whether the truncation of the string in case of a
- failed attempt should be done unconditionally. See the XXX mark.
- Russel, ideas ?
-
- * main/manager.c: do not return 500 Internal error if the AMI
- command provides no output.
-
- * main/manager.c: mosty comment and documentation cleanup on
- waitevent.
-
- * main/manager.c: Move the code to purge stale sessions to a
- function, to simplify the body of the main loop of the accepting
- thread. Rename purge_unused() to purge_events() so one knows what
- the function does.
-
- * main/manager.c: Various simplifications of the code: + use a
- wrapper around ast_carefulwrite(), used in two places, to make
- life easier when we decide to use a different interface to the
- socket. + put an ast_verbose() message on astman_append on a case
- that should never happen now that we use a temporary file for
- AMI-over-HTTP sessions + document and slightly simplify
- process_events() by removing unnecessary parentheses. + in
- get_input(), use ast_wait_for_input() instead of poll(). We may
- want to move to a completely non-blocking
-
- * main/manager.c: More informative message on invalid commands.
-
- * main/manager.c: another normalization of AMI vs HTTP
- identification. Should really define a macro IS_AMI(s) so it is
- clear what we want to do.
-
- * main/manager.c: always use managerid to determine whether this is
- an AMI or HTTP session, and document it.
-
- * main/http.c: In the previous commit i forgot to set the
- poll_timeout to -1, causing the http threads to do busy waiting
- around the socket... Fix the mistake, sorry for the
- inconvenience!
-
- * main/http.c: document the support for running a server on TCP/TLS
- and opening an SSL socket. We are almost ready to make this code
- available to other modules.
-
- * main/http.c, configs/http.conf.sample: add a new http.conf
- option, sslbindaddr. Because https is more secure than http, it
- usually makes sense to keep this service more open than the one
- on the unencrypted port.
-
- * main/http.c: in the helper thread, separate the FILE * creation
- from the actual function doing work on the socket. This is
- another generalization to provide a generic mechanism to open
- TCP/TLS socket with a thread managing the accpet and children
- threads managing the individual sessions.
-
- * main/http.c: staticize a global variable and remove an unused
- field structure.
-
-2006-11-27 18:10 +0000 [r48056] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 48054 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r48054 | file | 2006-11-27 13:06:50 -0500 (Mon,
- 27 Nov 2006) | 10 lines Merged revisions 48053 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r48053 | file | 2006-11-27 13:03:57 -0500 (Mon, 27 Nov 2006) | 2
- lines Use the proper function to get the new message count
- instead of always using the filesystem. (issue #8421 reported by
- slimey) ........ ................
-
-2006-11-27 17:31 +0000 [r48050] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 48049 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r48049 | tilghman | 2006-11-27 11:20:37 -0600
- (Mon, 27 Nov 2006) | 10 lines Merged revisions 48045 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r48045 | tilghman | 2006-11-27 11:15:54 -0600 (Mon, 27
- Nov 2006) | 2 lines Random MOH wasn't really random (bug 8381)
- ........ ................
-
-2006-11-27 15:48 +0000 [r48039-48040] Joshua Colp <jcolp@digium.com>
-
- * pbx/pbx_spool.c: More fixes for referencing a structure after it
- has been freed. (issue #8425 reported by arkadia)
-
- * pbx/pbx_spool.c, /: Merged revisions 48038 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r48038 | file | 2006-11-27 10:32:19 -0500 (Mon,
- 27 Nov 2006) | 10 lines Merged revisions 48037 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r48037 | file | 2006-11-27 10:30:37 -0500 (Mon, 27 Nov 2006) | 2
- lines Do not reference the freed outgoing structure in the debug
- message. (issue #8425 reported by arkadia) ........
- ................
-
-2006-11-27 14:47 +0000 [r48034] Luigi Rizzo <rizzo@icir.org>
-
- * funcs/func_cdr.c: remove an extra comma in an initializer
- Detected by: AST_DEVMODE=yes
-
-2006-11-27 06:59 +0000 [r48032-48033] Olle Johansson <oej@edvina.net>
-
- * include/asterisk/doxyref.h, include/asterisk/threadstorage.h:
- Doxygen updates
-
- * /, channels/chan_sip.c: Change error message (imported from 1.4)
-
-2006-11-26 06:55 +0000 [r48019] Russell Bryant <russell@digium.com>
-
- * include/asterisk/utils.h, include/asterisk/threadstorage.h: - Add
- some comments on thread storage with a brief explanation of what
- it is as well as what the motivation is for using it. - Add a
- comment by the declaration of ast_inet_ntoa() noting that this
- function is not reentrant, and the result of a previous call to
- the function is no longer valid after calling it again.
-
-2006-11-26 00:31 +0000 [r48016-48018] Steve Murphy <murf@digium.com>
-
- * /, funcs/func_cdr.c: Merged revisions 48017 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48017 | murf | 2006-11-25 17:26:16 -0700 (Sat, 25 Nov 2006) | 1
- line might as well also document the raw values of the flag vars
- ........
-
- * /, funcs/func_cdr.c: Merged revisions 48015 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r48015 | murf | 2006-11-25 17:01:34 -0700 (Sat, 25 Nov 2006) | 1
- line A little bit of func_cdr documentation upgrade-- no bug#
- involved, although 8221 may have inspired it. ........
-
-2006-11-25 21:50 +0000 [r48009-48014] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Little fix so we use the right message
-
- * channels/chan_zap.c: Make compiler happier
-
- * channels/chan_zap.c: bug fix
-
- * channels/chan_zap.c: Make sure we don't send a group reset on a
- group larger than 32 CICs
-
- * channels/chan_zap.c: Add ss7 show linkset command
-
- * channels/chan_zap.c: Updates to show linkset command
-
-2006-11-25 17:37 +0000 [r48008] Luigi Rizzo <rizzo@icir.org>
-
- * main/http.c: generalize a bit the functions used to create an tcp
- socket and then run a service on it. The code in manager.c does
- essentially the same things, so we will be able to reuse the code
- in here (probably moving it to netsock.c or another appropriate
- library file).
-
-2006-11-25 09:48 +0000 [r48003-48004] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Changing ERROR to lesser level. Imported
- from 1.2/1.4
-
- * main/rtp.c: - Adding comment on suspicious memory allocation.
- Seems like it's never freed, but I don't have a clear
- understanding of the frame allocation/deallocation, so I just
- mark this for investigation. (Reported by Ed Guy). We're trying
- to see if a free() hurts... - Doxygen comments on p2p rtp bridge
- stuff. I am a bit worried about shortcutting rtcp this way, but
- will need feedback from rtcp gurus. This should work for video
- calls too, and possibly UDPTL.
-
-2006-11-25 09:02 +0000 [r48001] Luigi Rizzo <rizzo@icir.org>
-
- * main/channel.c: set pointers to NULL after freeing memory to
- avoid multiple free() probably 1.4/1.2 issue as well if someone
- can look into that.
-
-2006-11-24 18:17 +0000 [r47995-47997] Steve Murphy <murf@digium.com>
-
- * /: removed the svnmerge-integrated property from trunk; it's
- confusing svnmerge in newly created branches
-
- * /, main/translate.c: This fix inspired by a patch supplied in bug
- 8189, which points out problems with the PLC code
-
-2006-11-24 14:00 +0000 [r47986] Olle Johansson <oej@edvina.net>
-
- * include/asterisk/doxyref.h, main/pbx.c,
- include/asterisk/causes.h, include/asterisk/channel.h: Doxygen
- update - Document cause codes - Document a bit more on channel
- variables - global, predefined and local - Fix some doxygen in
- channel.h. Adding one comment for two definitions does not work.
- They won't be copied to each.
-
-2006-11-23 11:04 +0000 [r47957-47960] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Remove unused memory allocation
-
- * doc/asterisk-conf.txt: Document new configuration option.
-
-2006-11-22 21:49 +0000 [r47933-47945] Joshua Colp <jcolp@digium.com>
-
- * /, main/rtp.c: Merged revisions 47944 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47944 | file | 2006-11-22 16:47:43 -0500 (Wed, 22 Nov 2006) | 2
- lines Video will never reach Packet2Packet bridging and can do
- more harm then good. ........
-
- * CHANGES: Clarify a bit more.
-
- * CHANGES: Need to update the CHANGES file as well for the maxfiles
- option.
-
- * main/asterisk.c: Add support to set the maximum number of files
- open when Asterisk loads using the 'maxfiles' configuration
- option. (issue #7499 reported by rkarlsba)
-
-2006-11-22 11:28 +0000 [r47923] Olle Johansson <oej@edvina.net>
-
- * channels/chan_h323.c: Don't over-deprecate... :-)
-
-2006-11-22 05:49 +0000 [r47912] Mark Spencer <markster@digium.com>
-
- * main/manager.c: Restore some sense of security to manager
-
-2006-11-21 17:34 +0000 [r47898] Joshua Colp <jcolp@digium.com>
-
- * /, main/rtp.c: Merged revisions 47897 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47897 | file | 2006-11-21 12:32:27 -0500 (Tue, 21 Nov 2006) | 2
- lines If we have the non standard G726-32 setting turned on we
- want to return G726-32 to the SDP, not our AAL2 string. (issue
- #8330 reported by voipgate) ........
-
-2006-11-21 15:25 +0000 [r47893] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Treat 101 as 100, not 183 session
- progress
-
-2006-11-21 11:53 +0000 [r47880-47881] Luigi Rizzo <rizzo@icir.org>
-
- * apps/app_dial.c: better fix for the previous bug. In general this
- code needs a deep revision, because the body of do_forward()
- deletes/overwrites the output channel without freeing the resouce
- in some cases, and without notifying the caller. Also, on FreeBSD
- with MALLOC_OPTIONS set i am seeing various panics (duplicate
- freee etc.)
-
- * apps/app_dial.c: do not ast_hangup() on a NULL channel. In the
- original code this would happen in the case of o->forwards >=
- AST_MAX_FORWARDS Likely an 1.2/1.4 isse as well - please someone
- have a look, while I am hunting a few more similar panics now.
-
-2006-11-20 20:04 +0000 [r47866] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 47864-47865 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47864 | tilghman | 2006-11-20 14:01:58 -0600 (Mon, 20 Nov 2006)
- | 2 lines Oops, merge missed release of odbc object ........
- ........
-
-2006-11-20 19:52 +0000 [r47851-47861] Joshua Colp <jcolp@digium.com>
-
- * main/frame.c, /: Merged revisions 47860 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r47860 | file | 2006-11-20 14:51:36 -0500 (Mon,
- 20 Nov 2006) | 10 lines Merged revisions 47859 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r47859 | file | 2006-11-20 14:50:21 -0500 (Mon, 20 Nov 2006) | 2
- lines Don't forget to byte swap if we are exiting the smoother
- feed early. (issue #8287 reported by arturs) ........
- ................
-
- * main/rtp.c: Use RTP/RTCP fds on the RTP structure, don't bother
- storing them.
-
- * /, main/rtp.c: Merged revisions 47852 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47852 | file | 2006-11-20 10:58:50 -0500 (Mon, 20 Nov 2006) | 2
- lines Only remove/destroy the RTCP I/O item if it exists.
- ........
-
- * apps/app_dial.c, /, apps/app_directed_pickup.c,
- include/asterisk/channel.h, .cleancount: Merged revisions 47850
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47850 | file | 2006-11-20 10:51:37 -0500 (Mon, 20 Nov 2006) | 2
- lines Use a separate variable in the channel structure to store
- the context that the channel was dialed from. (issue #8382
- reported by jiddings) ........
-
-2006-11-20 14:08 +0000 [r47847] Steve Murphy <murf@digium.com>
-
- * /: Erased the svnmerge-integrated prop from trunk. Please, in
- your svnmerge-ings, don't let these props leak into the trunk or
- branches.
-
-2006-11-20 11:46 +0000 [r47844-47846] Olle Johansson <oej@edvina.net>
-
- * /, configs/sip.conf.sample: Update docs for videosupport
-
- * /, channels/chan_sip.c: Properly reset schedule items (rizzo)
-
-2006-11-19 04:22 +0000 [r47835-47836] Steve Murphy <murf@digium.com>
-
- * UPGRADE.txt: Added a few words to explain the change to AEL
- concerning Gosub()
-
- * doc/ael.txt: Added a few words of explanation about macros
-
-2006-11-18 22:14 +0000 [r47822-47834] Luigi Rizzo <rizzo@icir.org>
-
- * main/manager.c: comments-only change: document a bit more when
- manager events are delivered to the clients.
-
- * main/cdr.c, res/res_features.c, res/res_realtime.c:
- ESS-ification. no need to bring this in 1.4, it is just code
- cleanup
-
- * include/asterisk/cli.h, main/cli.c: Move this macro from cli.c to
- cli.h so apps can use it without duplicating the macro or the
- code: /*! * In many cases we need to print singular or plural *
- words depending on a count. This macro helps us e.g. * printf("we
- have %d object%s", n, ESS(n)); */ #define ESS(x) ((x) == 1 ? "" :
- "s")
-
- * /, channels/chan_sip.c: Merged revisions 47823 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47823 | rizzo | 2006-11-18 18:59:35 +0100 (Sat, 18 Nov 2006) | 5
- lines fix bug 7450 - Parsing fails if From header contains angle
- brackets (the bug was only in a corner case where the < was right
- after the opening quote, and the fix is trivial). ........
-
- * channels/chan_oss.c: prevent the sound thread from consuming all
- the available CPU doing busy-wait on the output audio device. As
- it is set now, it tries to push a frame every 10ms, which is
- still too frequent but avoids deep restructuring of the code
- (which i should do, though). Note, this is only for ring tones,
- regular audio coming from the network is still delivered as soon
- as it is available. Eventually this could well end up in the 1.4
- branch, but since i am probably the only user of chan_oss there
- isn't much urgency to do that.
-
-2006-11-17 23:18 +0000 [r47821] Steve Murphy <murf@digium.com>
-
- * include/asterisk/file.h, main/channel.c, res/res_features.c,
- main/file.c, main/app.c, apps/app_directory.c,
- apps/app_followme.c, apps/app_voicemail.c: This update fulfils
- the request of bug 7109, which claimed the language arg to
- ast_stream_and_wait() was redundant. Almost all calls just used
- chan->language, and seeing how chan is the first argument, this
- certainly seems redundant. A change of language could just as
- easily be done by simply changing the channel language before
- calling.
-
-2006-11-17 22:56 +0000 [r47815-47818] Luigi Rizzo <rizzo@icir.org>
-
- * main/cli.c: remove a debugging message
-
- * main/cli.c: convert "help" to new style, fix completion of
- arguments past the first one that i broke earlier today.
-
- * main/cli.c: standardize "module show [like]"
-
-2006-11-17 21:51 +0000 [r47814] Jason Parker <jparker@digium.com>
-
- * configs/voicemail.conf.sample, apps/app_voicemail.c: Add ability
- to notify an external application/script that the voicemail
- password was, while also still changing the password
- "internally". Issue 7371, initial patch by pdunkel, with
- rewrite/config comments by me. Additional modifications (yay
- bitmask) by pdunkel.
-
-2006-11-17 21:50 +0000 [r47813] Luigi Rizzo <rizzo@icir.org>
-
- * main/cli.c: describe a bit the patterns that you can have in the
- commands, and add support for wildcard (spelled as '%'). On
- passing fix a bug in the expansion code which was hidden and
- appeared when implementing the wildcard The fix is just the line
- 'src != argindex', in case someone wants to test this on 1.4 -
- but i would just keep this in trunk.
-
-2006-11-17 20:46 +0000 [r47806] Jason Parker <jparker@digium.com>
-
- * apps/app_queue.c: Add ability to add custom queue log via manager
- interface. Issue 7806, patch by alexrch, with slight
- modifications by me.
-
-2006-11-17 18:26 +0000 [r47801] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Add some sense of link state. Don't make
- calls if link is down. Only reset if we're bringing the link up
- for the first time.
-
-2006-11-17 12:26 +0000 [r47787-47790] Luigi Rizzo <rizzo@icir.org>
-
- * main/cli.c: merge the implemenmtation of "core set debug" and
- "core set verbose". No externally visible changes.
-
- * channels/chan_oss.c: remove an unused function
-
- * channels/chan_oss.c: use the regexp cli support on some of the
- command
-
- * include/asterisk/cli.h, main/cli.c: introduce a bit of regexp
- support in the internal CLI api. Now you can specify a cli
- command as "console autoanswer [on|off]" which means the on|off
- argument is optional, or "console {mute|unmute}" which means the
- mute|unmute argument is mandatory. The blocks in [] or {} do not
- necessarily need to be at the end of the string. Completions for
- the variant parts are generated automatically. This should
- significantly simplify the implementation of the various
- handlers.
-
-2006-11-17 01:05 +0000 [r47784] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Make sure we choose the last channel as the
- dchannel if it's not E1 (for BRI). (#8077) Thanks Tzafrir.
-
-2006-11-16 23:20 +0000 [r47783] Jason Parker <jparker@digium.com>
-
- * apps/app_dial.c, /, apps/app_db.c: Merged revisions 47782 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47782 | qwell | 2006-11-16 17:19:46 -0600 (Thu, 16 Nov 2006) | 2
- lines Fix a couple of typos. Initially pointed out by mrobinson.
- ........
-
-2006-11-16 23:05 +0000 [r47779] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_oss.c: convert two entries to new style
-
-2006-11-16 23:00 +0000 [r47778] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, doc/billing.txt: Merged revisions 47777 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r47777 | kpfleming | 2006-11-16 17:00:10 -0600
- (Thu, 16 Nov 2006) | 12 lines update documentation regarding IAX2
- transfers and CDRs Merged revisions 47776 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r47776 | kpfleming | 2006-11-16 16:57:31 -0600 (Thu, 16 Nov 2006)
- | 2 lines update clearly wrong documentation regarding cdr_custom
- ........ ................
-
-2006-11-16 22:51 +0000 [r47775] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c: Remove the interim variable for range
- modifications, and set it on the structure directly. Also move
- the default checking to where it gets set initially. Fixes
- suggested by file.
-
-2006-11-16 22:44 +0000 [r47772] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_oss.c: convert some handlers to new style.
-
-2006-11-16 22:32 +0000 [r47771] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c, configs/zapata.conf.sample: Add ability to
- modify range for dring matching. Issue #8369, patch by ssuehring,
- modified slightly by me.
-
-2006-11-16 22:03 +0000 [r47769-47770] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_oss.c: fix indentation
-
- * main/cli.c: remove an unused function
-
-2006-11-16 21:13 +0000 [r47763-47765] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 47764 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47764 | file | 2006-11-16 16:11:06 -0500 (Thu, 16 Nov 2006) | 2
- lines Compare technology using the pointers instead of a straight
- comparison based on name. (issue #8228 reported by dean bath)
- ........
-
-2006-11-16 20:10 +0000 [r47759] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, configure, configure.ac: Merged revisions 47758 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r47758 | kpfleming | 2006-11-16 14:09:10 -0600 (Thu, 16
- Nov 2006) | 2 lines check for pre-1.4 versions of Zaptel and
- abort the configure script if found with an appropriate error
- message ........
-
-2006-11-16 19:29 +0000 [r47756] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c, configs/sip.conf.sample: Make it possible
- to enable/disable onhold tracking, in order to make life easier
- for realtime users.
-
-2006-11-16 18:32 +0000 [r47747-47752] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_local.c, /: Merged revisions 47751 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r47751 | file | 2006-11-16 13:29:12 -0500 (Thu,
- 16 Nov 2006) | 10 lines Merged revisions 47750 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r47750 | file | 2006-11-16 13:26:50 -0500 (Thu, 16 Nov 2006) | 2
- lines Because of the way chan_local is written we should be extra
- careful and make sure our callback functions have a tech_pvt.
- (issue #8275 reported by mflorell) ........ ................
-
- * /, apps/app_meetme.c: Merged revisions 47748 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47748 | file | 2006-11-16 12:52:48 -0500 (Thu, 16 Nov 2006) | 2
- lines Don't unreference the SLA object if there is no SLA object
- in the devicestate callback. (issue #8354 reported by loloski)
- ........
-
- * /: Be gone 1.2 props!
-
-2006-11-16 17:15 +0000 [r47734-47746] Olle Johansson <oej@edvina.net>
-
- * /: Merging a fix that was already fixed.
-
- * channels/chan_sip.c: Merging implementation of invite states from
- my "invitestate" branch for 1.2. This is a bit more clean
- platform for the handling of BYE/CANCEL than what we had. It
- might also need to changes in other parts of the code, since we
- know the state of the INVITE transaction. Please observe that
- this is is not dialog states at all, this is INVITE transaction
- states. Hello Michael Proctor, and thank you! :-)
-
- * /: Block upgrade to UPGRADE
-
- * /, channels/chan_sip.c, configs/sip.conf.sample: - CANCEL never
- uses authentication - Add docs on canreinvite
-
-2006-11-16 14:58 +0000 [r47727-47732] Luigi Rizzo <rizzo@icir.org>
-
- * main/cli.c: reduce indentation on a large function.
-
- * main/cli.c: use atomic instructions to update the inuse counters
- for CLI entriesC. The lock is not protecting this field. I wonder
- if the field should be declared 'volatile' as well.
-
- * main/cli.c: make kevin (and 64 bit machines) happy and remove a
- cast from char* to int in handling the return values from
- new-style handlers. On passing, note that
- main/loader.c::ast_load_resource() always return 0 so errors are
- not propagated up. I am not sure this is the intended behaviour.
-
-2006-11-16 08:18 +0000 [r47718] Paul Cadach <paul@odt.east.telecom.kz>
-
- * main/channel.c, /, funcs/func_channel.c,
- include/asterisk/channel.h: Merged revisions 44809 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r44809 | pcadach | 2006-10-10 23:44:54 +0700 (Втр, 10
- Окт 2006) | 1 line CHANNEL() function sometime mix parameter and
- value ........
-
-2006-11-15 22:32 +0000 [r47713] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_local.c, /: Merged revisions 47712 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r47712 | file | 2006-11-15 17:31:17 -0500 (Wed,
- 15 Nov 2006) | 10 lines Merged revisions 47711 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r47711 | file | 2006-11-15 17:29:30 -0500 (Wed, 15 Nov 2006) | 2
- lines Make sure that the pvt structure exists before trying to do
- fixup on Local channels. (issue #7937 reported by mada123, fix by
- alamantia with mods by me) ........ ................
-
-2006-11-15 21:57 +0000 [r47710] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 47709 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47709 | tilghman | 2006-11-15 15:56:55 -0600 (Wed, 15 Nov 2006)
- | 2 lines Fix ODBC_STORAGE for when context is NULL ........
-
-2006-11-15 21:36 +0000 [r47708] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /: Merged revisions 47707 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47707 | file | 2006-11-15 16:33:41 -0500 (Wed, 15 Nov 2006) | 2
- lines We need to ensure timelimit stuff is included as well so
- warnings get played. (issue #8050 reported by KNK) ........
-
-2006-11-15 21:21 +0000 [r47706] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Hunting the initreq change for an ACK
-
-2006-11-15 20:59 +0000 [r47703-47704] TransNexus OSP Development <support@transnexus.com>
-
- * apps/app_osplookup.c: 1. Fix the bug that Asterisk hangs up the
- calls if the OSP AuthRsp messages without destination protocol
- infomation. 2. Fix the bug that Asterisk generats wrong dial
- string (no in
- IAX2/[username[:password]@]peer[:port][/exten[@context]][/options]
- format) for IAX. 3. Add support for oh323 channel driver. 4.
- Re-formate the code.
-
- * include/asterisk/astosp.h: 1. Re-format the code.
-
-2006-11-15 20:51 +0000 [r47702] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, main/file.c: Merged revisions 47701 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47701 | kpfleming | 2006-11-15 14:50:06 -0600 (Wed, 15 Nov 2006)
- | 2 lines don't try to call fclose() if fopen() failed ........
-
-2006-11-15 20:40 +0000 [r47700] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: - Don't reply to ACK - Improve SIP
- history for debugging (Imported from 1.4)
-
-2006-11-15 20:28 +0000 [r47685-47694] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 47693 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r47693 | kpfleming | 2006-11-15 14:27:38 -0600
- (Wed, 15 Nov 2006) | 12 lines Merged revisions 47677 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r47677 | kpfleming | 2006-11-15 11:56:42 -0600 (Wed, 15
- Nov 2006) | 4 lines ensure that message duration is included in
- email notifications for forwarded messages (BE-96, fix by me
- after corydon used his clue-bat on me) ensure that duration in
- the message metadata is updated if prepending is done during
- forwarding (related to BE-96) remove prototype for API call that
- does not exist ........ ................
-
- * /, main/config.c: Merged revisions 47690 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r47690 | kpfleming | 2006-11-15 14:01:22 -0600
- (Wed, 15 Nov 2006) | 20 lines Merged revisions 47686,47688-47689
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r47686 | kpfleming | 2006-11-15 13:42:05 -0600 (Wed, 15 Nov 2006)
- | 2 lines clear the category's variable tail pointer as well when
- variables are detached from it ........ r47688 | kpfleming |
- 2006-11-15 13:47:43 -0600 (Wed, 15 Nov 2006) | 2 lines when
- appending a list of variable to a category, ensure the tail
- pointer points to the last variable in the list ........ r47689 |
- kpfleming | 2006-11-15 13:58:46 -0600 (Wed, 15 Nov 2006) | 2
- lines when re-writing the config file, don't repeat the path if
- it hasn't changed ........ ................
-
- * /, main/config.c: Merged revisions 47684 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r47684 | kpfleming | 2006-11-15 12:43:30 -0600
- (Wed, 15 Nov 2006) | 10 lines Merged revisions 47682 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r47682 | kpfleming | 2006-11-15 12:39:47 -0600 (Wed, 15
- Nov 2006) | 2 lines ouch... don't use printf, use
- ast_log/ast_verbose ........ ................
-
-2006-11-15 17:40 +0000 [r47662-47669] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_oss.c: fix indentation
-
- * main/cli.c: small simplifications and localization of variables.
-
- * main/cli.c: new-style "core show channels"
-
- * main/cli.c: more changes to new style of "module load" and
- "load". Under FreeBSD, the filename_completion used in the above
- commands does not work. Not sure why, but on passing i note that
- the function is part of readline and is not reentrant, so it
- needs to be fixed one way or another.
-
- * main/cli.c: move another deprecated command to the new style
-
- * main/cli.c: move "core set debug" to the new style, and remove
- duplicated code for the deprecated handler. On passing fix a long
- standing bug in find_cli() which would not find the longest match
- - this part (trivial, basically just update matchlen on a match)
- must go in 1.4 and possibly 1.2 as well.
-
-2006-11-15 16:09 +0000 [r47657-47661] Olle Johansson <oej@edvina.net>
-
- * /: Messed up earlier, cleaning up...
-
- * /, channels/chan_sip.c: Send proper SIP error message improperly
- when we can't allocate dialog (out of file handles is one cause)
-
- * channels/chan_sip.c: Update doxygen docs to reflect the code...
-
-2006-11-15 15:02 +0000 [r47652-47654] Luigi Rizzo <rizzo@icir.org>
-
- * include/asterisk/cli.h, main/cli.c: one more step cleaning the
- internal CLI interface: the NEW_CLI macro now supports extra
- arguments (to deprecate other commands). use this to implement
- unload and reload, and remove some unused functions. usual
- completion fixes (as these function accept multiple arguments).
- The summary is still a bit inconsistent.
-
- * include/asterisk/cli.h, main/cli.c: update the internal cli api
- following comments from kevin. This change basically simplifies
- the interface of the new-style handler removing almost all the
- tricks used in the previous implementation to achieve backward
- compatibility (which is still present and guaranteed.)
-
-2006-11-15 04:47 +0000 [r47646] Joshua Colp <jcolp@digium.com>
-
- * /, main/rtp.c: Merged revisions 47645 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47645 | file | 2006-11-14 23:45:24 -0500 (Tue, 14 Nov 2006) | 2
- lines If NAT detection is turned on or already detected then say
- NAT is active when setting the remote RTP peer when doing early
- bridging. (issue #8365 reported by marcelbarbulescu) ........
-
-2006-11-15 00:19 +0000 [r47642] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, main/term.c: Merged revisions 47641 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47641 | kpfleming | 2006-11-14 18:19:05 -0600 (Tue, 14 Nov 2006)
- | 2 lines more formatting cleanup, and avoid running off the end
- of the string ........
-
-2006-11-15 00:15 +0000 [r47640] Joshua Colp <jcolp@digium.com>
-
- * /, main/rtp.c: Merged revisions 47639 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47639 | file | 2006-11-14 19:14:07 -0500 (Tue, 14 Nov 2006) | 2
- lines Turn notice about unknown RTCP packet type into a debug
- message instead. ........
-
-2006-11-15 00:06 +0000 [r47636] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, channels/misdn/isdn_lib.c: Merged revisions 47635 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r47635 | kpfleming | 2006-11-14 18:05:44 -0600 (Tue, 14
- Nov 2006) | 2 lines silence compiler warning on 64-bit platforms
- (this variable is an 'int' anyway, comparing it to 'signed long'
- is not useful) ........
-
-2006-11-14 22:19 +0000 [r47633] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 47632 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r47632 | file | 2006-11-14 17:17:16 -0500 (Tue,
- 14 Nov 2006) | 10 lines Merged revisions 47631 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r47631 | file | 2006-11-14 17:15:10 -0500 (Tue, 14 Nov 2006) | 2
- lines Update copyright information in the ADSI logo blob.
- ........ ................
-
-2006-11-14 22:08 +0000 [r47630] Luigi Rizzo <rizzo@icir.org>
-
- * main/cli.c: add missing casts and remove an unused function.
-
-2006-11-14 22:07 +0000 [r47623-47629] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 47628 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47628 | file | 2006-11-14 17:05:03 -0500 (Tue, 14 Nov 2006) | 2
- lines Only keep the video RTP structure around if 1. Video
- support is enabled and 2. A video codec is enabled on the dialog
- ........
-
- * /, funcs/func_uri.c: Merged revisions 47625 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47625 | file | 2006-11-14 16:30:44 -0500 (Tue, 14 Nov 2006) | 2
- lines Small documentation clarification for URIENCODE. (issue
- #8294 reported by salaud) ........
-
- * apps/app_dial.c: Make local copy of arguments to parse. (issue
- #8362 reported by homesick)
-
-2006-11-14 18:58 +0000 [r47622] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 47621 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47621 | tilghman | 2006-11-14 12:54:40 -0600 (Tue, 14 Nov 2006)
- | 3 lines Conversion of res_odbc API to include ast_ prefix did
- not completely transition app_voicemail when ODBC_STORAGE is used
- (reported on IRC by caio1982, not in bugtracker) ........
-
-2006-11-14 17:00 +0000 [r47619-47620] Luigi Rizzo <rizzo@icir.org>
-
- * main/cli.c: fix completion for "module reload mod_1 mod_2 ... "
- (should do the same for the other similar commands, "reload",
- "module unload" and so on.
-
- * main/cli.c: partly convert to new style set-verbose, with
- completion fixes
-
-2006-11-14 16:48 +0000 [r47618] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_amd.c: Merged revisions 47617 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47617 | file | 2006-11-14 11:45:57 -0500 (Tue, 14 Nov 2006) | 2
- lines Use LOG_DEBUG to print out the indication that app_amd is
- using default settings instead of using LOG_NOTICE. This stops
- needless logging of this information under normal circumstances.
- (issue #8361 reported by Seb7) ........
-
-2006-11-14 16:38 +0000 [r47614-47616] Luigi Rizzo <rizzo@icir.org>
-
- * main/cli.c: replace two deprecated functions with calls to the
- standard ones, with fixes to argc/argv
-
- * main/cli.c: remove duplicated implementation for a deprecated
- function, use the original one instead with appropriate changes
- in argc/argv. This is not always applicable, but in some simple
- cases it is.
-
-2006-11-14 16:15 +0000 [r47610-47611] Olle Johansson <oej@edvina.net>
-
- * include/asterisk/cli.h: need to check quoting in the doxygen
- docs...
-
- * include/asterisk/cli.h: Some improvements to doxygen docs
-
-2006-11-14 16:09 +0000 [r47606-47609] Luigi Rizzo <rizzo@icir.org>
-
- * main/cli.c: new-style for 'core show uptime', include 'complete'
- support and better error checking
-
- * main/cli.c: apply previous fix to old-style cli entries as well.
-
- * main/cli.c: fix "core set debug atleast ", and fix the simple
- case where a command can have multiple completions, the first
- ones coming from keywords in previous CLI entries. There is no
- solution yet for the complex case of N1 completions from a CLI
- entry, followed by N2 from the next one, and so on, because the
- _complete() handlers do not tell us how many matches it
- generates, so we don't know how many to skip when interrogating
- the other handlers. The only solution is to avoid, as much as
- possible, multiple CLI entries with the same prefix.
-
- * include/asterisk/cli.h, main/cli.c: Bring in the improved
- internal API for the CLI. WATCH OUT: this changes the binary
- interface (ABI) for modules, so e.g. users of g729 codecs need a
- rebuilt module (but read below). The new way to write CLI
- handlers is described in detail in cli.h, and there are a few
- converted handlers in cli.c, look for NEW_CLI. After converting a
- couple of commands i am convinced that it is reasonably
- convenient to use, and it makes it easier to fix the pending CLI
- issues. On passing, note a bug with the current 'complete'
- architecture: if a command is a prefix of multiple CLI entries,
- we miss some of the possible options. As an example, "core set
- debug" can continue with "channel" from one CLI entry, and "off"
- or "atleast" from another one. We address this problem in a
- separate commit (when i have figured out a fix, that is). ABI
- issues: I asked Kevin if it was ok to make this change and he
- said yes. While it would have been possible to make the change
- without breaking the module ABI, the code would have been more
- convoluted. I am happy to restore the old ABI (while still being
- able to use the "new style" handlers) if there is demand.
-
-2006-11-14 13:17 +0000 [r47595-47600] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Adding some debug output to trace bug in
- realtime
-
- * channels/chan_sip.c: Adding a new debug line for issue #7351 -
- trying to find where an ACK can overwrite the initreq...
-
- * /, channels/chan_sip.c: Issue #8272 imported from 1.2/1.4 - Let
- the peerpoke system destroy it's own packets, please.
-
- * channels/chan_sip.c: Remove flags not used any more (thanks
- Luigi)
-
-2006-11-13 22:40 +0000 [r47586-47587] Matt O'Gorman <mogorman@digium.com>
-
- * codecs/codec_zap.c: oops no parens
-
- * main/frame.c, codecs/codec_zap.c: fix bytesize to 5.3kb for g723
- codec and add support for multimode of tc400p
-
-2006-11-13 21:32 +0000 [r47585] Joshua Colp <jcolp@digium.com>
-
- * /, cdr/cdr_pgsql.c: Merged revisions 47584 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r47584 | file | 2006-11-13 16:28:57 -0500 (Mon,
- 13 Nov 2006) | 10 lines Merged revisions 47583 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r47583 | file | 2006-11-13 16:26:36 -0500 (Mon, 13 Nov 2006) | 2
- lines Initialize global pointers for connection and result to
- NULL. (issue #8356 reported by james) ........ ................
-
-2006-11-13 20:21 +0000 [r47582] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 47581 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r47581 | tilghman | 2006-11-13 14:20:01 -0600
- (Mon, 13 Nov 2006) | 10 lines Merged revisions 47580 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r47580 | tilghman | 2006-11-13 14:18:30 -0600 (Mon, 13
- Nov 2006) | 2 lines Having more than 255 old messages caused
- corruption in the new/old count ........ ................
-
-2006-11-13 19:20 +0000 [r47579] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Small fix for uncommon scenario.
-
-2006-11-13 19:19 +0000 [r47577-47578] Steve Murphy <murf@digium.com>
-
- * /: Blocking 47576 from merging into trunk. Already done in 47577
-
- * main/config.c: This solves bug 8342, whereby a crash occurs under
- certain circumstances while reading a config file with comments--
- a call to CB_ADD shouldn't happen if withcomments is zero
-
-2006-11-13 19:14 +0000 [r47575] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_h323.c: Make chan_h323 build again and make the CLI
- commands work. (reported on asterisk-dev mailing list by Di-Shi
- Sun)
-
-2006-11-13 18:24 +0000 [r47568] Steve Murphy <murf@digium.com>
-
- * /: blocked 47564 from 1.4 to be merged onto trunk; 47566 already
- did it
-
-2006-11-13 18:23 +0000 [r47567] Joshua Colp <jcolp@digium.com>
-
- * configs/queues.conf.sample, CHANGES, apps/app_queue.c: Add
- 'loose' option to joinempty and leavewhenempty which is almost
- exactly like 'strict' except it does not count paused queue
- members as unavailable. (issue #8263 reported by gnarf)
-
-2006-11-13 18:20 +0000 [r47566] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael-test/ref.ael-test3: Eager people beat me to fixing
- the messed if, but we all forgot to update the regressions. Until
- now.
-
-2006-11-13 17:55 +0000 [r47556-47560] Joshua Colp <jcolp@digium.com>
-
- * apps/app_meetme.c: Don't play the "entering conference number
- <insert number here>" prompts if the 'q' option is used. If
- others believe this should be in 1.2/1.4 then we can put it in,
- but I'm uncomfortable doing so right now as it is a change of
- behavior. (issue #8138 reported by tmancill)
-
- * pbx/pbx_ael.c: Clean up last commit to better conform to
- standards.
-
-2006-11-13 17:36 +0000 [r47554-47555] Steve Murphy <murf@digium.com>
-
- * /: Blocking 47553 from 1.4 to trunk... 47554 is done for it.
-
- * pbx/pbx_ael.c: AEL need not complain about parkedcalls not being
- found... just confuses users
-
-2006-11-13 17:10 +0000 [r47543-47552] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_sms.c: Merged revisions 47551 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r47551 | file | 2006-11-13 12:08:07 -0500 (Mon,
- 13 Nov 2006) | 10 lines Merged revisions 47549 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r47549 | file | 2006-11-13 12:05:32 -0500 (Mon, 13 Nov 2006) | 2
- lines When sending an SMS with a user data header properly set
- the UDH flag in the first byte. (issue #8347 reported by
- hoffmeis) ........ ................
-
- * main/cli.c: Return module show to a working state. (issue #8353
- reported by jserve)
-
-2006-11-13 16:08 +0000 [r47541] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Only produce error message once, don't
- fill the screen with them... (Testing SIPP thanks to JerJer and
- Greg)
-
-2006-11-13 14:29 +0000 [r47536-47539] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_sip.c: merge from astobj2-r47450: use UNLINK to
- remove a packet from its queue, and related code rearrangement.
- Approved by: oej This could be made better if we declared struct
- sip_pvt *dialpg = pkt->owner; at the beginning of the function,
- and use it throughout the function. I'll let the boss decide :)
-
- * channels/chan_sip.c: merge from codename-pineapple and astobj2
- 47499: simplify __sip_ack() removing a strcmp for looking up
- packets. no functional change, only performance, so don't need to
- merging to earlier branches now. Approved By: oej
-
- * main/cli.c: stop looking for new entries when we know we are
- done. there is no functional change, so it is not necessary to
- bother merging this to 1.4 now.
-
- * main/cli.c: free memory when unregistering an entry. needs to be
- merged to 1.4
-
-2006-11-13 05:58 +0000 [r47530] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_odbc.c, configs/res_odbc.conf.sample: Feature: allow the
- sanity SQL to be customized per connection class (Issue 6453)
-
-2006-11-13 05:51 +0000 [r47529] Russell Bryant <russell@digium.com>
-
- * /, configure, acinclude.m4: Merged revisions 47527 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r47527 | russell | 2006-11-13 00:48:18 -0500 (Mon, 13
- Nov 2006) | 5 lines AC_PROG_SED is included in autoconf 2.60, but
- apparently it is not included in 2.59. So, to maintain
- compatability with 2.59 since it is a small change, copy this
- macro into acinclude.m4 and rename it to AST_PROG_SED. (issue
- #8345) ........
-
-2006-11-13 05:48 +0000 [r47524-47528] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_odbc.c: Merged revisions 47526 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r47526 | tilghman | 2006-11-12 23:46:18 -0600
- (Sun, 12 Nov 2006) | 10 lines Merged revisions 47525 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r47525 | tilghman | 2006-11-12 23:45:11 -0600 (Sun, 12
- Nov 2006) | 2 lines If the execute fails a second time, make sure
- that we don't pass back a stale handle ........ ................
-
- * channels/chan_zap.c, /: Merged revisions 47523 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r47523 | tilghman | 2006-11-12 19:12:01 -0600
- (Sun, 12 Nov 2006) | 10 lines Merged revisions 47522 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r47522 | tilghman | 2006-11-12 18:34:44 -0600 (Sun, 12
- Nov 2006) | 2 lines Don't play dialtone if the seizing the
- channel fails (Bug 7754) ........ ................
-
-2006-11-12 20:47 +0000 [r47521] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Part of patch in #7403 to improve tag
- checking in pedantic mode (stephen_dredge)
-
-2006-11-12 19:22 +0000 [r47520] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: The use of an ifdef to check for FreeBSD is
- useless here because the two versions of the format string are
- identical. Also, since each format is only used once, get rid of
- the use of defines all together. (issue #8344, julieng) In
- passing, also clean up the formatting a but to get rid of the
- nesting without the use of braces, as defined in the coding
- guidelines.
-
-2006-11-12 16:15 +0000 [r47508-47514] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Restore auto-framing (DEA). Imported from
- 1.4
-
- * /, channels/chan_sip.c: - Support UDPTL as well as udptl in T38
- sdp.
-
- * /, channels/chan_sip.c: - Don't hangup because of failed
- re-invite. Go back to previous state. - Keep RTP running during
- T.38 session We might improve the code to issue ast_rtp_stop if
- T.38 re-invite not fails later on in the code, but I don't see
- many reasons to.
-
- * /, channels/chan_sip.c: - Add some comments to t.38 code - Remove
- improper blocking of ptime: in SDP
-
-2006-11-12 06:31 +0000 [r47493-47498] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 47497 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r47497 | russell | 2006-11-12 01:23:23 -0500
- (Sun, 12 Nov 2006) | 12 lines Merged revisions 47496 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r47496 | russell | 2006-11-12 01:09:03 -0500 (Sun, 12
- Nov 2006) | 4 lines Only do the check to determine whether the
- channel calling this function is an IAX2 channel when getting the
- IP address using the special argument, CURRENTCHANNEL. (issue
- #8341, jcovert) ........ ................
-
- * Makefile, /: Merged revisions 47494 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47494 | russell | 2006-11-11 10:31:08 -0500 (Sat, 11 Nov 2006) |
- 6 lines Add the target "menuconfig" as an alias for the
- "menuselect" target. This is just a favor to users so that if you
- accidentally type "make menuconfig" instead of "make menuselect",
- it still works. (inspired by a comment on IRC from wangster
- calling me an "especially devious asterisk developer" for having
- it be menuselect instead of menuconfig. :) ) ........
-
- * /, main/term.c: Merged revisions 47492 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47492 | russell | 2006-11-11 10:18:02 -0500 (Sat, 11 Nov 2006) |
- 2 lines Tweak the formatting of this new function to better
- conform to coding guidelines. ........
-
-2006-11-11 02:12 +0000 [r47491] Matt O'Gorman <mogorman@digium.com>
-
- * main/logger.c, include/asterisk/term.h, main/term.c: safe
- terminal output is sweet.
-
-2006-11-10 22:06 +0000 [r47478] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Make sure we don't use 32bits for a value
- that only requires 1 bit. Also, fix a compiler warning for one of
- the SS7 functions.
-
-2006-11-10 21:55 +0000 [r47467-47477] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Add some history and fix some debug
- output for autodestruct.
-
- * /, channels/chan_sip.c: Proper fix for adding debug...
-
- * /, channels/chan_sip.c: Make sure we destroy dialog in case of
- loop
-
- * /, channels/chan_sip.c: Cleanup imported from 1.4
-
-2006-11-10 20:05 +0000 [r47459-47465] Joshua Colp <jcolp@digium.com>
-
- * pbx/pbx_dundi.c: Fine, take this.
-
- * main/cli.c: A trunk that builds is a productive trunk.
-
- * pbx/pbx_dundi.c: Hello compiler working, goodbye compiler
- warning. (fix compiler warning introduced from pbx_dundi
- optimizations)
-
- * /, channels/chan_h323.c: Merged revisions 47457 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47457 | file | 2006-11-10 14:36:25 -0500 (Fri, 10 Nov 2006) | 2
- lines Let's give this a go... ........
-
-2006-11-10 19:35 +0000 [r47456] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Add fix for 7321. Ability to hide
- calleridname from zapata.conf
-
-2006-11-10 19:01 +0000 [r47455] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Issue 8336- fix support for multipart SDP
- (imported from 1.2/1.4). (Alphaque)
-
-2006-11-10 17:22 +0000 [r47445] Luigi Rizzo <rizzo@icir.org>
-
- * build_tools/prep_moduledeps: manual merge from 1.4: grep -m not
- available on bsd, use head -1 which works for all
-
-2006-11-10 17:01 +0000 [r47439] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c, channels/chan_skinny.c,
- channels/chan_h323.c, channels/chan_iax2.c, channels/chan_mgcp.c,
- main/cli.c: Merged revisions 47436 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47436 | tilghman | 2006-11-10 10:51:55 -0600 (Fri, 10 Nov 2006)
- | 2 lines Discussion of these CLI changes resulted in more
- consistency (Bug 8236) ........
-
-2006-11-10 16:55 +0000 [r47438] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_chanspy.c: Merged revisions 47437 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47437 | file | 2006-11-10 11:53:16 -0500 (Fri, 10 Nov 2006) | 2
- lines Only split up extension and context if a value exists.
- (issue #8332 reported by loloski) ........
-
-2006-11-10 16:38 +0000 [r47434-47435] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 47433 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47433 | kpfleming | 2006-11-10 10:36:49 -0600 (Fri, 10 Nov 2006)
- | 2 lines if adding a queue member is LOG_NOTICE, then removing
- them should be LOG_NOTICE, not LOG_DEBUG ........
-
- * /, apps/app_queue.c: Merged revisions 47432 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47432 | kpfleming | 2006-11-10 10:34:04 -0600 (Fri, 10 Nov 2006)
- | 2 lines reflect addition/removal of dynamic queue members in
- queue_log, so that people using dialplan replacement for
- AgentCallbackLogin can still track login/logout (issue #7736,
- reported/patched by whoiswes but this commit was written by me
- and covers all three paths for AQM/RQM) ........
-
-2006-11-10 13:14 +0000 [r47415-47419] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Ripping out bad support for 491 replies
- to INVITE's. Let's try again properly later.
-
- * /, channels/chan_sip.c: Fix badly defined flag. Thanks fenlander!
-
- * channels/chan_sip.c: Small simplification and documentation
- correction.
-
-2006-11-10 04:30 +0000 [r47408-47410] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_dundi.c: Various little bits of code cleanup to reduce
- nesting, remove useless casts, and to remove a duplicated error
- message after a memory allocation error
-
- * include/asterisk/app.h, apps/app_read.c, main/app.c: Add the
- ability to specify multiple prompts to the Read() dialplan
- application, similar to Background() and Playback(). (issue
- #7897, jsmith, with some modifications)
-
-2006-11-10 03:45 +0000 [r47399-47406] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_h323.c: Merged revisions 47405 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47405 | file | 2006-11-09 22:44:36 -0500 (Thu, 09 Nov 2006) | 2
- lines Fix building of chan_h323 by completeing some structure
- definitions. (issue #8327 reported by Mithraen) ........
-
- * main/pbx.c: This should already be called while locked.
-
- * /, apps/app_voicemail.c: Merged revisions 47398 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47398 | file | 2006-11-09 17:32:30 -0500 (Thu, 09 Nov 2006) | 2
- lines Do conversion in a more easier to read and working way for
- \r, \n, and \t. (issue #8324 reported by johnlange) ........
-
-2006-11-09 21:32 +0000 [r47392] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c, /, build_tools/prep_moduledeps,
- apps/app_voicemail.c: Merged revisions 47391 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47391 | russell | 2006-11-09 16:26:27 -0500 (Thu, 09 Nov 2006) |
- 7 lines Work around an issue that caused menuselect to display a
- bogus description for app_voicemail and chan_zap. These modules
- use some preprocessor directives to determine what it will report
- to Asterisk as its description. However, the way we extract this
- information from the source files for menuselect is not smart
- enough to figure this out. (issue #8326, #8328) ........
-
-2006-11-09 17:08 +0000 [r47382] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_phone.c, /: Merged revisions 47380 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r47380 | file | 2006-11-09 11:53:25 -0500 (Thu,
- 09 Nov 2006) | 10 lines Merged revisions 47379 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r47379 | file | 2006-11-09 11:48:05 -0500 (Thu, 09 Nov 2006) | 2
- lines Don't include compiler.h on kernels 2.6.18 and higher as,
- well, it's apparently going to be removed. This should make all
- you FC6 fans happy as your Asterisk will now build without any
- mods. ........ ................
-
-2006-11-09 16:30 +0000 [r47353-47378] Russell Bryant <russell@digium.com>
-
- * /, main/cli.c: Merged revisions 47377 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47377 | russell | 2006-11-09 11:28:15 -0500 (Thu, 09 Nov 2006) |
- 2 lines fix tab completion for "core debug channel" and "core no
- debug channel" ........
-
- * /, main/cli.c: Merged revisions 47375 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47375 | russell | 2006-11-09 11:24:02 -0500 (Thu, 09 Nov 2006) |
- 3 lines Fix "core show channel". Also, fix tab completion for
- both "core show channel" and "core show channels". ........
-
- * /, main/cli.c: Merged revisions 47372 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47372 | russell | 2006-11-09 11:18:33 -0500 (Thu, 09 Nov 2006) |
- 3 lines Fix "core debug channel <whatever>". I guess someone
- needs to go through and audit every CLI command that changed
- number of arguments ... ........
-
- * /, main/cli.c: Merged revisions 47366 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47366 | russell | 2006-11-09 10:49:39 -0500 (Thu, 09 Nov 2006) |
- 3 lines Fix another CLI command, "core show uptime" ... (issue
- #8323, reported by johnlange, fixed by myself) ........
-
- * /, main/asterisk.c: Merged revisions 47352 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47352 | russell | 2006-11-09 01:31:37 -0500 (Thu, 09 Nov 2006) |
- 3 lines fix "core show version" to reflect the new number of
- arguments for this CLI command (issue #8316, kshumard) ........
-
-2006-11-09 00:46 +0000 [r47343-47351] Steve Murphy <murf@digium.com>
-
- * /: Blocking 47344 from automerging into trunk
-
- * /: Blocking 47348 from automerging into trunk
-
- * main/channel.c: This mod via bug 7531
-
- * channels/chan_skinny.c: committed in behalf of bug 8190
-
-2006-11-08 22:35 +0000 [r47341] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: - Add Max-Forwards higher in the packet,
- following recommendations - Fix documentation for
- sip_pvt_lock/unlock - doxygen does not inherit like zapata.conf
- !!! - Change doc for a sip_pvt setting
-
-2006-11-08 21:59 +0000 [r47337-47339] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/frame.c: restore display of G.722 codec
-
- * /, channels/chan_sip.c: Merged revisions 47333 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47333 | kpfleming | 2006-11-08 12:07:16 -0600 (Wed, 08 Nov 2006)
- | 2 lines add simple fix for SDP to report proper sample rate for
- G.722 media sessions ........
-
-2006-11-08 18:26 +0000 [r47335] Joshua Colp <jcolp@digium.com>
-
- * main/pbx.c, CHANGES: Display CID matching information when using
- dialplan show. (issue #8279 reported by caio1982)
-
-2006-11-08 17:06 +0000 [r47325-47332] Russell Bryant <russell@digium.com>
-
- * /, utils/streamplayer.c: Merged revisions 47331 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47331 | russell | 2006-11-08 12:03:09 -0500 (Wed, 08 Nov 2006) |
- 5 lines I occasionally get email from users that are trying to
- figure out what this does, or due to some misunderstanding as to
- what it is supposed to do, can't get it to work. So, I have added
- some text here to hopefully explain what this application does
- and does not do. ........
-
- * /, configure, configure.ac, acinclude.m4: Merged revisions 47327
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47327 | russell | 2006-11-08 11:31:59 -0500 (Wed, 08 Nov 2006) |
- 4 lines Copy the macros from libtool.m4 to our own acinclude.m4
- such that libtool is no longer required to be installed to be
- able to generate the configure script. ........
-
-2006-11-08 15:28 +0000 [r47321] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_sip.c: coding guidelines, coding guidelines, coding
- guidelines
-
-2006-11-08 13:59 +0000 [r47314-47318] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_sip.c: merge from team/rizzo/astobj2 rev.47271
- avoid doing p > 0 when p is a pointer; move a lock closer to the
- place where it is needed Approved By: oej
-
- * channels/chan_sip.c: merge from team/rizzo/astobj2 rev.47246 Same
- as for peers and users, replace ASTOBJ_UNREF(r,
- sip_registry_destroy) with unref_registry(r); Approved By: oej
-
- * channels/chan_sip.c: merge from team/rizzo/astobj2, rev 47243,
- 47244, 47245: Replace ASTOBJ_UNREF(peer, sip_destroy_peer) with
- unref_peer(peer); This places the name of the destructor in one
- place only (where it should be), eliminates the chance of errors
- in case you specify the wrong destructor, and also lets the
- compiler do type checking on the argument, again helping with
- keeping the code clean. Same for users. remove two duplicate
- definitions. Approved By: oej
-
- * channels/chan_sip.c: merge rev.47224 from team/rizzo/astobj2:
- hide dialoglist lock/unlocking in wrapper functions. Approved By:
- oej
-
- * channels/chan_sip.c: silence compiler about uninitialized
- variables. The compiler is wrong, but it has the last word.
-
-2006-11-08 08:01 +0000 [r47313] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Destroy dialog properly at unload (rizzo)
-
-2006-11-08 07:21 +0000 [r47306] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_jingle.c, channels/chan_gtalk.c: fix compilation.
- Overall i think the previous change to ast_channel_alloc() to
- close bug 7506 should have been done by defining an
- ast_set_callerid_noevent() function that does the setting without
- generating the event. Lot less code duplication, and easier to
- handle.
-
-2006-11-08 03:13 +0000 [r47304-47305] Russell Bryant <russell@digium.com>
-
- * configure.ac: add a comment about where AC_PROG_LD comes from
-
- * aclocal.m4 (removed), /: remove aclocal.m4 from the tree since it
- is just an intermediate file created while generating the
- configure script.
-
-2006-11-07 23:14 +0000 [r47295-47300] Luigi Rizzo <rizzo@icir.org>
-
- * main/asterisk.c: fix "core show profile" parsing. Needs to go in
- 1.4 too, but ENOTIME now
-
- * apps/app_queue.c: %ld and time_t don't match, so cast the
- argument to long to ease portability problems
-
-2006-11-07 21:47 +0000 [r47290] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, channels/chan_local.c, channels/chan_vpb.cc,
- channels/chan_zap.c, channels/chan_sip.c, res/res_features.c,
- channels/chan_agent.c, main/utils.c, include/asterisk/channel.h,
- channels/chan_gtalk.c, channels/chan_iax2.c, channels/chan_oss.c,
- main/channel.c, channels/chan_jingle.c, channels/chan_phone.c,
- channels/chan_misdn.c, channels/chan_skinny.c,
- channels/chan_features.c, channels/chan_h323.c,
- channels/chan_alsa.c, channels/chan_nbs.c,
- include/asterisk/stringfields.h, channels/chan_mgcp.c,
- apps/app_voicemail.c: A fair number of changes for the sake of
- bug 7506
-
-2006-11-07 20:16 +0000 [r47285-47288] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_local.c, /: Merged revisions 47287 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r47287 | file | 2006-11-07 15:14:58 -0500 (Tue, 07 Nov
- 2006) | 2 lines This is not the commit you are looking for...
- ........
-
- * channels/chan_local.c, /: Merged revisions 47284 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r47284 | file | 2006-11-07 15:08:52 -0500 (Tue, 07 Nov
- 2006) | 2 lines Make MOH work as it did before in chan_local,
- without this then it can go funky when transfers and MOH are
- involved. (issue #7671 reported by jmls) ........
-
-2006-11-07 18:56 +0000 [r47280] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, configs/musiconhold.conf.sample: Merged revisions 47279 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47279 | kpfleming | 2006-11-07 12:56:21 -0600 (Tue, 07 Nov 2006)
- | 2 lines clean up sample config, and make native file playback
- the more obvious default choice ........
-
-2006-11-07 18:50 +0000 [r47278] Matt O'Gorman <mogorman@digium.com>
-
- * apps/app_voicemail.c: rge overhaul to voicemail imap support.
- Allows support for more imap servers, also a better
- implementation of several parts of the original work. patch
- provided by 8033 with major upgrades. minor differences from 1.4
- patch do to changes in app_voicemail
-
-2006-11-07 17:33 +0000 [r47269] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Break -> continue to make stuff work...
- Thanks, Luigi!
-
-2006-11-07 14:25 +0000 [r47257-47259] Kevin P. Fleming <kpfleming@digium.com>
-
- * /: remove another broken property merge
-
- * /: remove properties that shouldn't be merged to this branch
-
- * /: use editable URL for menuselect, and switch to trunk
-
-2006-11-07 13:26 +0000 [r47251-47252] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: issue #8265 - don't reply to ACK.
- Imported from 1.2, 1.4
-
- * include/asterisk/frame.h: Stealing Tilghman's explanation from
- the -dev list and producing documentation...
-
-2006-11-07 08:34 +0000 [r47242] Luigi Rizzo <rizzo@icir.org>
-
- * main/utils.c: explain why ast_carefulwrite is written the way it
- is, and also that it doesn't really work as claimed.
-
-2006-11-07 01:28 +0000 [r47232-47240] Russell Bryant <russell@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 47239 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r47239 | russell | 2006-11-06 20:25:10 -0500
- (Mon, 06 Nov 2006) | 13 lines Merged revisions 47238 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r47238 | russell | 2006-11-06 20:22:58 -0500 (Mon, 06
- Nov 2006) | 5 lines If random order is enabled for files mode
- music on hold, set a random initial position, instead of always
- starting at the first file, and doing the random operation only
- when switching to the next file. (bug reported by John Lange on
- the asterisk-dev mailing list) ........ ................
-
- * utils/check_expr.c: check for failure after call to calloc()
- (issue #8295)
-
-2006-11-06 17:27 +0000 [r47230] Kevin P. Fleming <kpfleming@digium.com>
-
- * UPGRADE.txt: minor change to test live syncing
-
-2006-11-06 17:05 +0000 [r47229] Joshua Colp <jcolp@digium.com>
-
- * main/manager.c, utils/astman.c, include/asterisk/manager.h: Add
- support for manager hooks, so you could fire off manager events
- over IRC if you were crazy enough. (issue #5161 reported by anthm
- with mods by moi)
-
-2006-11-05 01:04 +0000 [r47210-47213] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_dundi.c: Make pbx_dundi compile again. Sorry. :(
-
- * configs/zapata.conf.sample: List ss7 with the rest of the valid
- signalling types. Group SS7 options together and comment them out
- by default.
-
-2006-11-04 22:16 +0000 [r47209] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Don't lock dialoglist if monitor runs
- __sip_destroy. Hmmm. I did not change pbx_dundi and yet it
- doesn't compile ;-)
-
-2006-11-04 22:08 +0000 [r47206-47207] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_dundi.c: use the AST_MODULE_LOAD_* return codes from
- load_module()
-
- * pbx/pbx_dundi.c: simplify a couple of loops
-
-2006-11-04 21:48 +0000 [r47205] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Move IP address selection for media out of
- add_sdp
-
-2006-11-04 21:44 +0000 [r47204] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_dundi.c: Do some minor cleanup to the section of code
- that sets the EID by getting the mac address for an ethernet
- interface
-
-2006-11-04 21:17 +0000 [r47200-47203] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Make srvlookup global_srvlookup to follow
- the rest of the code
-
- * channels/chan_sip.c: Simplify previous patch
-
- * channels/chan_sip.c, configs/sip.conf.sample: Adding new config
- option "limitpeersonly" to only apply call limits to the peer
- side of a type=friend. This is for trying to support BJ in his
- quest to solve some issues with the queue system and type=friend
- objects. BJ: Please test!
-
- * /, channels/chan_sip.c: Importing patch for Invite/replaces from
- 1.4
-
-2006-11-04 18:12 +0000 [r47197-47198] Russell Bryant <russell@digium.com>
-
- * /, main/cli.c: Merged revisions 47196 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47196 | russell | 2006-11-04 13:10:22 -0500 (Sat, 04 Nov 2006) |
- 2 lines Fix another bug in "core set debug" ... ........
-
- * /, main/asterisk.c, main/cli.c: Merged revisions 47195 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47195 | russell | 2006-11-04 12:59:39 -0500 (Sat, 04 Nov 2006) |
- 2 lines Really fix the "core set debug" and "core set verbose"
- CLI commands. ........
-
-2006-11-04 17:45 +0000 [r47194] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Reverting rev 47093 until we have an
- agreement on how to implement this, if at all.
-
-2006-11-04 17:40 +0000 [r47193] Russell Bryant <russell@digium.com>
-
- * /, main/cli.c: Merged revisions 47192 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47192 | russell | 2006-11-04 12:38:24 -0500 (Sat, 04 Nov 2006) |
- 3 lines fix the "atleast" option to the "core set verbose" and
- "core set debug" CLI commands ........
-
-2006-11-04 11:00 +0000 [r47179-47189] Luigi Rizzo <rizzo@icir.org>
-
- * apps/app_dial.c: move out another large block to a large
- function, and document some possibly missing parts in the privacy
- screening code. Now that it is more streamlined it is easier to
- see differences in handling the various cases. Have not tested
- the code in depth.
-
- * res/res_agi.c: useless cast removal...
-
- * main/logger.c: remove many unnecessary casts
-
- * main/app.c: remove a useless cast
-
- * configs/manager.conf.sample: document the "debug" parameter, and
- the change manager list -> manager show
-
- * apps/app_dial.c: fix indentation of a block, and do minor
- simplifications at the end of another one.
-
- * apps/app_dial.c: complete previous commit.
-
- * apps/app_dial.c: move another block into a function. On passing,
- avoid two null-pointer string dereference while printing messages
- (which are sometimes not fatal in some platforms, but still
- wrong). These two lines at least should be merged to 1.4 once i
- am done with all the changes here.
-
- * apps/app_dial.c: move a large block into a separate function.
- Mark with XXX a possible bug in previous code which used the
- wrong source in case of a forwarded call. the function
- do_forward() needs to be split further, as the initial part is
- replicated in another places (with some minor differences, most
- likely forgotten when updating after the copy).
-
-2006-11-03 23:27 +0000 [r47178] Steve Murphy <murf@digium.com>
-
- * channels/chan_sip.c: This fix introduced via bug 8233
-
-2006-11-03 23:24 +0000 [r47160-47177] Luigi Rizzo <rizzo@icir.org>
-
- * apps/app_dial.c: another small set of simplifications
-
- * apps/app_dial.c: change HANDLE_CAUSE into a function.
-
- * apps/app_dial.c: remove redundant checks
-
- * apps/app_dial.c: start integrating the simplifications proposed
- in bug 0005860, as usual a bit at a time to ease locating new
- bugs or fixes worth merging into other branches. In this commit,
- introduce a macro, S_REPLACE, that replaces a string possibly
- freeing the previous value. In one of these places (see the
- comment marked XXX) the previous code might leak memory - if so,
- this ought to be merged in 1.4 The macro might be worth putting
- in one of the global headers (e.g. include/asterisk/strings.h) as
- the construct is used in a million places in the asterisk code.
-
-2006-11-03 19:15 +0000 [r47146] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c: One has to create the path and filename in
- order to copy a file there. (issue #8278 reported by davebath)
-
-2006-11-03 18:53 +0000 [r47072-47132] Luigi Rizzo <rizzo@icir.org>
-
- * main/manager.c, include/asterisk/manager.h: add a new
- cli/manager.conf option "debug" to enable/disable debugging code
- in the manager. At the moment the debugging code is very
- lightweight, if the option is enabled manager messages also carry
- a sequence number and the info where they have been generated
- e.g. SequenceNumber: 10 File: chan_sip.c Line: 11927 Func:
- handle_response_register It is not worthwhile having this as a
- compile time option right now, because the extra work involved at
- runtime is just checking one variable.
-
- * channels/chan_zap.c: remove old/useless usecnt stuff
-
- * channels/chan_vpb.cc: remove old/useless usecnt stuff. I think
- this module doesn't compile, anyways, because it has not been
- updated to the new module interface.
-
- * main/cli.c: Fix "core show channels" and "core show modules". Not
- sure it applies like this to 1.4 because of deprecate versions of
- the same command(s).
-
- * res/res_jabber.c: move variable declarations to the beginning of
- a block.
-
- * /: block other changes of mine already applied to trunk.
-
- * /: block more changes of mine already applied to trunk
-
- * /: blocking 47107
-
- * /: blocking 47108
-
- * channels/chan_sip.c: Save the 'From' header received in a
- REGISTER message so we can show it e.g. in the Manager interface.
- This information is available as a callerid (or something like
- that) during a call, but not when a device is registered but
- silent. It may be useful to have it available e.g. when
- developing a user interface/operator panel, to map numbers to
- names. experimental, so not committed to 1.4
-
- * channels/chan_jingle.c, channels/chan_gtalk.c: remove useless
- usecnt stuff
-
- * channels/chan_phone.c: remove useless usecnt stuff
-
- * channels/chan_alsa.c: remove useless usecnt stuff
-
- * channels/chan_agent.c: remove useless usecnt stuff
-
- * channels/chan_features.c: remove useless usecnt handling
-
- * channels/chan_skinny.c: remove useless usecnt handling code
-
-2006-11-02 23:55 +0000 [r47052-47054] Tilghman Lesher <tlesher@digium.com>
-
- * main/udptl.c, /, channels/chan_skinny.c, res/res_agi.c,
- channels/chan_h323.c, res/res_jabber.c, main/rtp.c: Merged
- revisions 47053 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47053 | tilghman | 2006-11-02 17:49:13 -0600 (Thu, 02 Nov 2006)
- | 2 lines More changes making the CLI more consistent with
- "category verb arguments" (continuation of issue 8236) ........
-
- * main/pbx.c, channels/chan_local.c, main/frame.c,
- channels/chan_sip.c, /, res/res_features.c, res/res_crypto.c,
- channels/chan_agent.c, res/res_musiconhold.c, apps/app_queue.c,
- channels/chan_iax2.c, main/config.c, main/cli.c, main/channel.c,
- main/manager.c, channels/chan_skinny.c, res/res_agi.c,
- channels/chan_features.c, main/logger.c, main/file.c,
- main/http.c, res/res_indications.c, main/image.c, res/res_odbc.c,
- main/asterisk.c, channels/chan_mgcp.c, apps/app_voicemail.c:
- Merged revisions 47051 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r47051 | tilghman | 2006-11-02 17:00:20 -0600 (Thu, 02 Nov 2006)
- | 2 lines Reverse change of "show" to "list" and make several
- other commands more consistent with "category verb arguments"
- ........
-
-2006-11-02 21:40 +0000 [r47037] Joshua Colp <jcolp@digium.com>
-
- * main/pbx.c, include/asterisk/pbx.h: Let's make
- application/function/hint lists read/write lists... just for
- kicks
-
-2006-11-02 21:34 +0000 [r47035] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Updates to do unblock correctly
-
-2006-11-02 20:24 +0000 [r46999-47021] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Move check for codec translators to an
- earlier place in the call, so we can fail gracefully (imported
- from 1.4)
-
- * /, channels/chan_sip.c: Disable code for not implemented
- functionality (T38 over RTP/TCP)
-
-2006-11-02 18:34 +0000 [r46991-46994] Russell Bryant <russell@digium.com>
-
- * include/asterisk/astobj.h: Sure enough, some of the uses of
- astobj are doing recursive locking. This doesn't work with
- rwlocks, so, this is reverted for now.
-
- * include/asterisk/astobj.h: astobj was already set up to use read
- and write locks. Now that we have wrappers for them, use them
- here.
-
-2006-11-02 18:01 +0000 [r46967-46972] Joshua Colp <jcolp@digium.com>
-
- * main/translate.c: Convert translation core linked list over to
- read/write based one, since it spends most of it's time only
- reading.
-
- * include/asterisk/linkedlists.h: Add AST_RWLIST_* set of macros
- which implement linked lists using read/write locks, the actual
- list manipulation is still done via the old macros.
-
-2006-11-02 17:51 +0000 [r46966] Russell Bryant <russell@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 46965 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r46965 | russell | 2006-11-02 12:49:54 -0500
- (Thu, 02 Nov 2006) | 11 lines Merged revisions 46964 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r46964 | russell | 2006-11-02 12:47:56 -0500 (Thu, 02
- Nov 2006) | 3 lines ignore files in a music on hold directory
- that begin with '.' (issue #8249, cboie) ........
- ................
-
-2006-11-02 16:51 +0000 [r46940] Joshua Colp <jcolp@digium.com>
-
- * include/asterisk/lock.h: Set the AST_RWLOCK_INIT_VALUE to the
- PTHREAD_RWLOCK_INIT_VALUE if it is available, that way outside
- stuff can determine whether to use a constructor or deconstructor
- for initialization instead of using the init value.
-
-2006-11-02 16:50 +0000 [r46939] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Changes to show blocked/unblocked states, as
- well as in service, out of service state
-
-2006-11-02 16:45 +0000 [r46938] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 46937 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r46937 | kpfleming | 2006-11-02 10:45:32 -0600 (Thu, 02 Nov 2006)
- | 2 lines don't send INVITE when we have determined that we can't
- offer any audio formats due to lack of trancoding support (or
- incorrect configuration) ........
-
-2006-11-02 16:28 +0000 [r46931-46935] Joshua Colp <jcolp@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac,
- include/asterisk/lock.h: I'm crazy so I will add this... pthread
- rwlock wrappers, along with autoconf stuff that detects the
- presence of the initializer and the ability to set the kind of
- lock (in our case we rather like writer preferred locks so writer
- starvation doesn't occur... but on something like Darwin we don't
- get that)
-
- * /, channels/chan_sip.c: Merged revisions 46930 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r46930 | file | 2006-11-02 11:06:39 -0500 (Thu,
- 02 Nov 2006) | 10 lines Merged revisions 46920 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r46920 | file | 2006-11-02 11:02:27 -0500 (Thu, 02 Nov 2006) | 2
- lines Repeat after me oej: I will at least make sure my code
- compiles before I commit it. ........ ................
-
-2006-11-02 16:03 +0000 [r46926] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Add simple down event support
-
-2006-11-02 15:47 +0000 [r46906] Nadi Sarrar <ns@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/misdn_config.c:
- find_free_chan_in_stack: cleanup buggy usage
-
-2006-11-02 15:31 +0000 [r46902] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Don't overwrite pkt->flags (imported from
- 1.2/1.4)
-
-2006-11-02 14:15 +0000 [r46846-46886] Russell Bryant <russell@digium.com>
-
- * main/callerid.c: various whitespace changes to reduce indentation
- and to better conform to formatting guidelines
-
- * main/callerid.c: Change the buffer used in callerid_feed() and
- callerid_feed_jp() to be allocated on the stack using alloca()
- instead of using malloc() since they are only used locally to
- these functions.
-
- * /, main/say.c: Merged revisions 46857 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r46857 | russell | 2006-11-01 18:01:48 -0500 (Wed, 01 Nov 2006) |
- 2 lines fix saying one hundred and two hundred in hebrew (issue
- #7810, eldadran) ........
-
- * CHANGES: Add a couple of things to the CHANGES file
-
- * Makefile, /, configure, codecs/gsm/Makefile, configure.ac,
- build_tools/strip_nonapi, makeopts.in: Merged revisions 46847 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r46847 | russell | 2006-11-01 17:51:21 -0500 (Wed, 01 Nov 2006) |
- 3 lines Fixes for cross-compilation on mips (issue #8058,
- ywalther, with some modifications) ........
-
- * aclocal.m4, /, build_tools/menuselect-deps.in, configure,
- build_tools/embed_modules.xml, configure.ac: Merged revisions
- 46845 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r46845 | russell | 2006-11-01 17:32:12 -0500 (Wed, 01 Nov 2006) |
- 5 lines Add a check in the configure script to determine whether
- ld is GNU ld or not. This is needed because module embedding only
- works for gnu ld. GNU ld is now listed as a dependency for all of
- the module embedding options in menuselect. (issue #8143)
- ........
-
-2006-11-01 20:38 +0000 [r46823] Matt O'Gorman <mogorman@digium.com>
-
- * /, channels/chan_gtalk.c: Merged revisions 46822 via svnmerge
- from https://svn.digium.com/svn/asterisk/branches/1.4 ........
- r46822 | mogorman | 2006-11-01 14:35:41 -0600 (Wed, 01 Nov 2006)
- | 2 lines bind address support from bug 8164 ........
-
-2006-11-01 19:48 +0000 [r46801] Steve Murphy <murf@digium.com>
-
- * res/res_config_odbc.c: a fix for bug 8251; the var_val needs to
- accept longer strings or mass confusion and a lot of lost time is
- the result
-
-2006-11-01 18:41 +0000 [r46782] Joshua Colp <jcolp@digium.com>
-
- * /, main/Makefile: Merged revisions 46780 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r46780 | file | 2006-11-01 13:39:47 -0500 (Wed, 01 Nov 2006) | 2
- lines Force poll() emulation for Darwin to always be on. It's too
- broken to consider being used. This resolves the console issue
- OSX users have been seeing. I would have liked to autoconf this
- but I haven't been able to come up with a test case that works.
- Que sera. ........
-
-2006-11-01 18:40 +0000 [r46779-46781] Russell Bryant <russell@digium.com>
-
- * doc/channelvariables.txt, pbx/pbx_dundi.c: Add the ability to
- pass options to the Dial application when using the DUNDi switch
- in the dialplan by setting the DUNDIDIALARGS channel variable.
- (issue #8084, patch by bluecrow76, with small modifications and
- documentation updates)
-
- * /, res/res_monitor.c: Merged revisions 46778 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r46778 | russell | 2006-11-01 13:26:35 -0500
- (Wed, 01 Nov 2006) | 17 lines Merged revisions 46776 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r46776 | russell | 2006-11-01 13:24:17 -0500 (Wed, 01
- Nov 2006) | 9 lines soxmix and Asterisk expect different file
- extensions for certain formats. This was already handled for the
- wav49 format. However, it was not handled for ulaw and alaw. I
- fixed this in such a way that using the alternate extensions for
- ulaw and alaw will only happen if we know we're calling soxmix,
- and not a custom script defined using the MONITOR_EXEC variable.
- The wav49 processing was left alone so that external scripts will
- see no behavior change. (issue #7550, reported by mnicholson,
- proposed patch by junky, committed fix is a bit different)
- ........ ................
-
-2006-11-01 18:26 +0000 [r46777] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 46775 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r46775 | file | 2006-11-01 13:21:34 -0500 (Wed, 01 Nov 2006) | 2
- lines It's another round of chan_iax2 fixes! Should hopefully fix
- the deadlock issues people have been reporting. IAXtel now has
- qualify turned on for 800 peers and it is handling it fine.
- ........
-
-2006-11-01 18:16 +0000 [r46759-46774] Steve Murphy <murf@digium.com>
-
- * CHANGES: OOps. forgot to add this to CHANGES
-
- * main/say.c, apps/app_voicemail.c: This introduces Brazilian
- Portuguese via 7663
-
- * main/config.c: Cleanups suggested by Russell.
-
-2006-11-01 17:09 +0000 [r46758] Luigi Rizzo <rizzo@icir.org>
-
- * res/res_features.c: move variable declaration in the middle of a
- block
-
-2006-11-01 16:51 +0000 [r46745] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 46744 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r46744 | russell | 2006-11-01 11:39:09 -0500 (Wed, 01 Nov 2006) |
- 2 lines Prevent an infinite loop when config processing gets to a
- jitterbuffer option ........
-
-2006-11-01 00:07 +0000 [r46732] Matt O'Gorman <mogorman@digium.com>
-
- * res/res_features.c: change default return extension after parking
- timeout. 6953 with minor changes.
-
-2006-10-31 22:19 +0000 [r46719] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, main/translate.c, include/asterisk/translate.h: Merged
- revisions 46714 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r46714 | kpfleming | 2006-10-31 15:47:48 -0600 (Tue, 31 Oct 2006)
- | 2 lines add an API so that translators can activate/deactivate
- themselves when needed ........
-
-2006-10-31 22:07 +0000 [r46717-46718] Jason Parker <jparker@digium.com>
-
- * main/translate.c: Fix "core show translation" output. Issue
- #8243, patch by Damin.
-
-2006-10-31 18:10 +0000 [r46683-46696] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_iax2.c: remove old/useless usecount handling
-
- * channels/chan_sip.c: remove old/useless usecount stuff.
-
- * channels/chan_oss.c: remove old/useless usecount management code.
-
-2006-10-31 15:22 +0000 [r46661] Russell Bryant <russell@digium.com>
-
- * main/manager.c: Fix the new send text manager command. There is
- no way this could have worked. - Check the channel name string
- length to be zero, not non-zero - Check the message string length
- to be zero, not non-zero - unlock the channel *after* calling
- sendtext
-
-2006-10-31 13:56 +0000 [r46582-46650] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Set #define for TIMER T1 value
-
- * channels/chan_sip.c: Cleaning up code
-
- * funcs/func_enum.c, /, include/asterisk/enum.h, main/enum.c: Issue
- #80898 - Restoring func_enum (otmar)
-
- * main/manager.c: Add manager sendtext action. (Issue 6131, ZX81 -
- thanks!)
-
- * /, channels/chan_sip.c, configs/sip.conf.sample: Fix rport
- handling. ...where did the 1.2 properties come from, really?
- they're back.
-
- * /, channels/chan_sip.c: - If peer that register fails ACL, fail
- him - Remove the 1.2 props I've set by mistake earlier
-
- * /: Block patch that only applies to 1.4
-
- * main/loader.c: Take two, using find_resource on Kevin's
- suggestion. Might need better locking support, giving up if we
- can't get the lock. Right now, using existing locking in
- find_resource
-
-2006-10-31 06:37 +0000 [r46556-46565] Russell Bryant <russell@digium.com>
-
- * apps/app_cdr.c: add author doxygen tag (issue #8241, kshumard)
-
- * /, contrib/init.d/rc.redhat.asterisk: Merged revisions 46563 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r46563 | russell | 2006-10-31 01:30:53 -0500 (Tue, 31 Oct 2006) |
- 3 lines Start Asterisk later in the boot process to ensure it
- starts after stuff like MySQL (issue #8253, Alric) ........
-
- * /, main/utils.c: Merged revisions 46561 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r46561 | russell | 2006-10-31 01:19:56 -0500
- (Tue, 31 Oct 2006) | 11 lines Merged revisions 46560 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r46560 | russell | 2006-10-31 01:18:36 -0500 (Tue, 31
- Oct 2006) | 3 lines When handling the case where the hostname is
- just an IPV4 numeric address, be sure to set the address type.
- (issue #8247, alexr) ........ ................
-
- * /, res/res_agi.c: Merged revisions 46558 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r46558 | russell | 2006-10-31 01:14:13 -0500
- (Tue, 31 Oct 2006) | 11 lines Merged revisions 46557 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r46557 | russell | 2006-10-31 01:13:09 -0500 (Tue, 31
- Oct 2006) | 3 lines fix some copy/paste bugs in the checking of
- arguments for the "control stream file" AGI command (issue #8255,
- mnicholson) ........ ................
-
- * /, main/translate.c: Merged revisions 46554 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r46554 | russell | 2006-10-31 00:55:07 -0500 (Tue, 31 Oct 2006) |
- 5 lines Add a small tweak to the code that checks to see whether
- destination formats are translatable based on the source format.
- If we have already determined that there is no translation path
- in one direction, don't bother checking the other direction.
- ........
-
-2006-10-30 23:11 +0000 [r46541] Steve Murphy <murf@digium.com>
-
- * apps/app_dial.c, utils/astman.c: These changes submitted by moy
- via bug 6992, to add a Dial 'End' event to asterisk. I include
- some changes to astman to cover other events that have been
- added.
-
-2006-10-30 22:27 +0000 [r46529] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, main/translate.c: Merged revisions 46526 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r46526 | kpfleming | 2006-10-30 16:19:55 -0600 (Mon, 30 Oct 2006)
- | 3 lines when unregistering a translator, don't rebuild the
- translation matrix unless needed when filtering formats out of an
- offer, ensure we check for translation ability in both directions
- ........
-
-2006-10-30 21:56 +0000 [r46513-46514] Olle Johansson <oej@edvina.net>
-
- * funcs/func_module.c: show, list, view, display... whatever.
-
- * funcs/func_module.c (added), include/asterisk/module.h,
- main/loader.c: Adding dialplan function IFMODULE, so you can
- create dialplans that handle various PBX installations and checks
- if a module is loaded before using it. example
- IFMODULE(chan_sip3.so) issue #6671 in the bug tracker, finally
- gone. Thanks to mithraen for keeping it updated.
-
-2006-10-30 21:46 +0000 [r46512] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, include/asterisk/linkedlists.h: Merged revisions 46511 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r46511 | kpfleming | 2006-10-30 15:46:07 -0600 (Mon, 30 Oct 2006)
- | 2 lines ensure that items removed from a list are always
- unlinked from the list (next pointer set to NULL) ........
-
-2006-10-30 21:22 +0000 [r46508-46509] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Update sip list to eventlist format.
-
- * main/pbx.c, main/manager.c, include/asterisk/manager.h: Issue
- #3930 - Add manager command for listing dialplan (coded april
- 2005, in bugtracker since)
-
-2006-10-30 21:11 +0000 [r46507] Joshua Colp <jcolp@digium.com>
-
- * /, configure, configure.ac: Merged revisions 46506 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r46506 | file | 2006-10-30 16:09:13 -0500 (Mon, 30 Oct
- 2006) | 2 lines Don't explicitly link in crypt as it is not used
- on some platforms. ........
-
-2006-10-30 19:56 +0000 [r46476-46489] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c, configs/sip.conf.sample: Change name of
- "contact" setting to "callback" which better reflects what it is
- to the person that configures asterisk. That we use it internally
- in the contact header is a totally different story. Still not
- convinced this is a good option.
-
- * channels/chan_sip.c: Globals need the "global_" prefix in
- chan_sip, and need to be reset to default value at reload.
-
-2006-10-30 18:17 +0000 [r46475] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 46474 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r46474 | file | 2006-10-30 13:13:07 -0500 (Mon, 30 Oct 2006) | 2
- lines We need to lock the pvt structure during retransmission as
- another worker thread may be doing something as well. ........
-
-2006-10-30 18:04 +0000 [r46466] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Make sure we give the linkset number, not
- the offset in the linksets array
-
-2006-10-30 18:02 +0000 [r46461] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Small conversion to ast_channel_unlock
-
-2006-10-30 17:32 +0000 [r46459] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Specify which linkset we're getting the
- messages from in the message
-
-2006-10-30 16:59 +0000 [r46439] Olle Johansson <oej@edvina.net>
-
- * main/rtp.c: In debug mode, recognize that someone is sending
- zrtp, even though we can't do anything with it yet. Ideally a
- first step would be a passthrough mode.
-
-2006-10-30 16:50 +0000 [r46436] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Don't make errors when we don't need them
-
-2006-10-30 16:33 +0000 [r46379-46434] Olle Johansson <oej@edvina.net>
-
- * include/asterisk/file.h, include/asterisk/doxyref.h, /,
- channels/chan_sip.c, main/ast_expr2f.c,
- include/asterisk/module.h, formats/format_ogg_vorbis.c,
- main/app.c, include/asterisk/channel.h, include/asterisk/lock.h,
- include/asterisk/frame.h, main/asterisk.c, apps/app_voicemail.c:
- Issue 8246 Doxygen updates (kshumard) THANK YOU!
-
- * /: The RTCP patch started in trunk, so don't start all over again
- :-)
-
- * main/asterisk.c: Small formatting changes
-
- * main/rtp.c: Bind RTCP to the same IP as RTP. I currently don't
- see this as a bug that needs to be fixed in 1.4/1.2 too, but feel
- free to backport if you see it that way. RTCP now binds to ALL IP
- addresses on the host, RTP to a specific address.
-
- * /, channels/chan_sip.c: Issue #7869 - Stop retransmission of 302
- redirects.
-
- * /, channels/chan_sip.c: Issue #7608 - Notifications sent with
- wrong content-type (imported from 1.2, 1.4)
-
- * /: Block patch from other branch
-
- * channels/chan_sip.c: Issues related to issue #7828 - segfault
- with MWI subscriptions and realtime.
-
- * /, channels/chan_sip.c: - Fix the OUTGOING stuff (merge from 1.4)
- - Make sure we UNREF authpeer when not needed
-
- * apps/app_voicemail.c: Spelling fix.
-
- * channels/chan_sip.c: Documentation update again
-
- * channels/chan_sip.c: Documentation update (I guess)
-
- * channels/chan_sip.c: Documentation correction
-
- * channels/chan_sip.c: maxtime is not needed any more now that we
- actually set the T1 timer based on the qualify result.
-
- * /, channels/chan_sip.c: Only accept message once
-
- * channels/chan_sip.c: Adding documentation inspired by a virtual
- drink with an anonymous man in New Jersey
-
- * channels/chan_sip.c: Don't duplicate function if not needed... -
- removing transmit_reinvite_with_t38_sdp in favour of adding an
- argument to transmit_reinvite_with_sdp
-
- * /, channels/chan_sip.c: Merge from 1.4 : Don't send 183
- reliably...
-
- * channels/chan_sip.c: - Don't lock the dialoglist during the whole
- destruction of a single SIP dialog. Only lock when needed - when
- we remove the dialog from the dialog list If this doesn't lead to
- severe problems, it might help with some locking issues in
- 1.4/1.2. - Remove the term "interface" as a synonym for a SIP
- dialog. Sorry, Mark, but no one understands it... ;-)
-
-2006-10-28 16:39 +0000 [r46378] Joshua Colp <jcolp@digium.com>
-
- * utils/Makefile, /: Merged revisions 46377 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r46377 | file | 2006-10-28 12:37:44 -0400 (Sat, 28 Oct 2006) | 2
- lines Don't build muted on OpenBSD, it is not supported. ........
-
-2006-10-27 19:28 +0000 [r46372] BJ Weschke <bweschke@btwtech.com>
-
- * apps/app_queue.c: Let's make sure we hold the mutex lock before
- we go looking at values in the queue structure that could
- potentially be changing while we're running.
-
-2006-10-27 19:04 +0000 [r46371] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 46370 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r46370 | russell | 2006-10-27 14:03:32 -0500 (Fri, 27 Oct 2006) |
- 4 lines move the copy of the default settings to the global
- settings back out of process_zap, so that they aren't overwritten
- when process_zap is called multiple times ........
-
-2006-10-27 18:59 +0000 [r46369] BJ Weschke <bweschke@btwtech.com>
-
- * configs/queues.conf.sample, CHANGES, apps/app_queue.c: * Added
- option to run macro when a queue member is connected to a caller,
- see queues.conf.sample for details. * Added QUEUE_VARIABLES
- function to set queue variables added setqueuevar and
- setqueueentryvar options for each queue, see queues.conf.sample
- for details. (#8216, jmls reported and submitted)
-
-2006-10-27 18:31 +0000 [r46368] Olle Johansson <oej@edvina.net>
-
- * /, contrib/asterisk-ng-doxygen: raise the pressure on Christian
- :-)
-
-2006-10-27 17:46 +0000 [r46366] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: First pass at implementation to be able to
- block and unblock zap channels for use.
-
-2006-10-27 17:45 +0000 [r46365] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Put this patch on hold pending further
- testing...
-
-2006-10-27 17:42 +0000 [r46359-46364] Russell Bryant <russell@digium.com>
-
- * /, res/res_agi.c, apps/app_externalivr.c, res/res_musiconhold.c,
- main/asterisk.c: Merged revisions 46363 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r46363 | russell | 2006-10-27 12:39:31 -0500 (Fri, 27 Oct 2006) |
- 5 lines We should always be using _exit() after a fork() or
- vfork() instead of exit(). This is because exit() does some extra
- cleanup which in some implementations of vfork(), for example,
- can actually modify the state of the parent process, causing very
- weird bugs or crashes. (issue #7971, Nick Gavrikov) ........
-
- * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c: Add
- the ability to customize some of the prompts used within the
- voicemail application by configuring them in voicemail.conf
- (issue #7415, patch by fkasumovic, with some fixes and
- documentation updates by myself)
-
- * channels/chan_zap.c, /: Merged revisions 46358 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r46358 | russell | 2006-10-27 10:32:40 -0500 (Fri, 27 Oct 2006) |
- 5 lines Instead of iterating all of the options once to look for
- jitterbuffer options, and then again for everything else, move
- the processing of jitterbuffer options into the main loop so that
- there are no erroneous messages about ignoring unknown options.
- (issue #8226) ........
-
-2006-10-27 11:18 +0000 [r46354] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
- channels/misdn/chan_misdn_config.h,
- channels/misdn/isdn_msg_parser.c, configs/misdn.conf.sample,
- channels/misdn/isdn_lib.c, channels/misdn_config.c: Merged
- revisions 46351-46353 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r46351 | crichter | 2006-10-27 11:49:20 +0200
- (Fr, 27 Okt 2006) | 9 lines Merged revisions 46176 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r46176 | crichter | 2006-10-25 10:41:59 +0200 (Mi, 25
- Okt 2006) | 1 line added nttimeout option to configure wether we
- disconnect calls on NT timeouts or not during an overlapdial
- session ........ ................ r46352 | crichter | 2006-10-27
- 11:58:44 +0200 (Fr, 27 Okt 2006) | 1 line fixed not compile
- issue, which was just introduced ................ r46353 |
- crichter | 2006-10-27 12:03:23 +0200 (Fr, 27 Okt 2006) | 9 lines
- Merged revisions 46350 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r46350 | crichter | 2006-10-27 11:24:01 +0200 (Fr, 27 Okt 2006) |
- 1 line fixed a bug which caused chan_misdn to try to allocate 2
- times the same channel on high load, which then caused
- instability of mISDN. removed a useless function from isdn_lib.c
- ........ ................
-
-2006-10-26 20:27 +0000 [r46348] Jason Parker <jparker@digium.com>
-
- * /, apps/app_page.c: Merged revisions 46347 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r46347 | qwell | 2006-10-26 15:25:44 -0500 (Thu, 26 Oct 2006) | 2
- lines Fix small formatting issue, that causes misaligned line
- ........
-
-2006-10-26 20:22 +0000 [r46346] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Show if the channel is ready for video or
- T.38 udptl
-
-2006-10-26 18:04 +0000 [r46341] Jason Parker <jparker@digium.com>
-
- * contrib/scripts/astgenkey.8: oops - somebody forgot to change
- this - long ago, probably.
-
-2006-10-26 17:52 +0000 [r46330-46339] Russell Bryant <russell@digium.com>
-
- * main/pbx.c, apps/app_osplookup.c, main/manager.c,
- apps/app_meetme.c, apps/app_festival.c, main/say.c,
- apps/app_alarmreceiver.c, apps/app_sms.c, apps/app_rpt.c,
- main/rtp.c, apps/app_voicemail.c: fix various spelling mistakes
- in comments (issue #8237, jmls)
-
- * /, main/translate.c: Merged revisions 46329 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r46329 | russell | 2006-10-26 11:31:05 -0500 (Thu, 26 Oct 2006) |
- 11 lines - If the source has no audio or no video portion, do not
- call powerof() to get the format index. - Don't run through the
- audio and video loops if there is no audio or video portion of
- the source If 0 is passed to powerof, it will return -1. This
- value of -1 was then being used as an array index in these loops,
- which caused a crash on some systems. Other than this issue, this
- code works as we expected it to. If a format is not in the
- source, and we have to translation path to it, it is not offered
- in the list of acceptable destination formats. (fixes issue
- #8231) ........
-
-2006-10-26 12:47 +0000 [r46308-46319] Luigi Rizzo <rizzo@icir.org>
-
- * main/manager.c: fix a problem that i recently introduced when the
- manager receives long commands.
-
- * configs/sip.conf.sample: document the match_auth_username option
-
-2006-10-26 04:19 +0000 [r46299] Russell Bryant <russell@digium.com>
-
- * /, doc/backtrace.txt: Merged revisions 46298 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r46298 | russell | 2006-10-25 23:18:00 -0500 (Wed, 25 Oct 2006) |
- 2 lines update backtrace documentation to reflect changes in 1.4
- (issue #8230, kshumard) ........
-
-2006-10-26 01:38 +0000 [r46288] Mark Spencer <markster@digium.com>
-
- * main/manager.c, main/config.c: Fix comment preservation code
- (thanks murf!)
-
-2006-10-25 20:21 +0000 [r46259-46277] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Old todo: Don't add Contact headers on
- BYE and CANCEL.
-
- * channels/chan_sip.c: First stab at transaction direction fix,
- this for trunk for testing
-
- * /, channels/chan_sip.c: Ugly code to try to remove issue
- discovered by Luigi as well as attack bug #7608
-
-2006-10-25 19:24 +0000 [r46256] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Send CPG when we get a CONTROL_PROGRESS
- frame and make sure that it sends ACM (not CPG) when we get
- CONTROL_PROCEEDING.
-
-
-2006-10-25 19:14 +0000 [r46251] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c, configs/zapata.conf.sample: Update changes
- to do US style point code parsing/formatting (xxx.xxx.xxx)
-
-2006-10-25 19:10 +0000 [r46250] Russell Bryant <russell@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 46249 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r46249 | russell | 2006-10-25 14:08:18 -0500 (Wed, 25 Oct 2006) |
- 2 lines update warning message to include "agi" option (issue
- #8225, jmls) ........
-
-2006-10-25 17:12 +0000 [r46238] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, sounds/sounds.xml, sounds/Makefile: Merged revisions 46237 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r46237 | kpfleming | 2006-10-25 12:08:58 -0500 (Wed, 25 Oct 2006)
- | 2 lines add support for prebuilt G.722 prompts and music on
- hold files ........
-
-2006-10-25 16:01 +0000 [r46215-46224] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merge from 1.4
-
- * /: Block change to 1.4 to block change to 1.2... This is
- confusing, but I think I got it right.
-
-2006-10-25 14:55 +0000 [r46201-46203] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, channels/chan_sip.c, main/translate.c,
- include/asterisk/translate.h: Merged revisions
- 46082-46083,46152-46153 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r46082 | kpfleming | 2006-10-23 22:45:42 -0500 (Mon, 23 Oct 2006)
- | 2 lines add an API call to allow channel drivers to determine
- which media formats are compatible (passthrough or transcode)
- with the format an existing channel is already using ........
- r46083 | kpfleming | 2006-10-23 22:53:32 -0500 (Mon, 23 Oct 2006)
- | 2 lines ensure that the translation matrix is properly
- lock-protected every place it is used ........ r46152 | kpfleming
- | 2006-10-24 18:45:19 -0500 (Tue, 24 Oct 2006) | 2 lines if
- multiple translators are registered for the same source/dest
- combination, ensure that the lowest-cost one is always inserted
- earlier in the list ........ r46153 | kpfleming | 2006-10-24
- 19:10:54 -0500 (Tue, 24 Oct 2006) | 2 lines code zone experiment:
- don't offer formats in the outbound INVITE that aren't either
- passthrough or translatable ........
-
- * channels/chan_iax2.c: restore bugfix that was reverted by
- trunk_mtu patch
-
- * channels/chan_sip.c, /, apps/app_record.c, apps/app_softhangup.c,
- res/res_adsi.c, main/utils.c, pbx/dundi-parser.c,
- apps/app_ices.c, apps/app_getcpeid.c, apps/app_queue.c,
- channels/chan_iax2.c, main/cli.c, main/cdr.c,
- channels/chan_phone.c, pbx/pbx_spool.c, channels/chan_features.c,
- channels/chan_h323.c, pbx/pbx_ael.c, channels/chan_alsa.c,
- pbx/pbx_realtime.c, apps/app_sms.c, channels/chan_nbs.c,
- main/image.c, main/db.c, channels/chan_mgcp.c, cdr/cdr_custom.c,
- apps/app_parkandannounce.c, apps/app_voicemail.c: Merged
- revisions 46200 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r46200 | kpfleming | 2006-10-25 09:32:08 -0500 (Wed, 25 Oct 2006)
- | 2 lines apparently developers are still not aware that they
- should be use ast_copy_string instead of strncpy... fix up many
- more users, and fix some bugs in the process ........
-
-2006-10-25 14:26 +0000 [r46199] Olle Johansson <oej@edvina.net>
-
- * CREDITS, configs/iax.conf.sample, channels/chan_iax2.c: Ok,
- second attempt...
-
-2006-10-25 14:18 +0000 [r46198] Luigi Rizzo <rizzo@icir.org>
-
- * CHANGES: document a couple of recently introduced feature also
- including the version number where the feature appeared.
-
-2006-10-25 14:14 +0000 [r46183-46197] Olle Johansson <oej@edvina.net>
-
- * CREDITS, configs/iax.conf.sample, channels/chan_iax2.c: On the
- other hand, don't use 1.4 patches for trunk... Sorry.
-
- * CREDITS, configs/iax.conf.sample, channels/chan_iax2.c: Add
- ability to adapt the IAX trunk packets to the MTU size, to avoid
- bad audio when the number of channels fill the MTU on a given
- link. In the future, this needs to be configurable per peer with
- trunking enabled.
-
- * channels/chan_sip.c: Adding comments in the source is more
- persistent than just adding them to the commit message :-)
-
- * channels/chan_sip.c: Always add doxygen comments to new
- functions, more lines than one are appreciated really. (Read the
- coding guidelines). I've worked hard to make chan_sip a better
- place to code in, let's keep it that way and don't add more stuff
- without comments. Thank you.
-
-2006-10-25 00:32 +0000 [r46155] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/frame.c, /, main/translate.c, formats/format_pcm.c,
- channels/chan_h323.c, channels/chan_iax2.c,
- include/asterisk/frame.h, main/rtp.c: Merged revisions 46154 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r46154 | kpfleming | 2006-10-24 19:26:17 -0500 (Tue, 24 Oct 2006)
- | 2 lines add passthrough and file format support for G.722 16KHz
- audio (issue #5084, original patch by andrew, updated by
- mithraen) ........
-
-2006-10-24 20:22 +0000 [r46141] Mark Spencer <markster@digium.com>
-
- * res/res_agi.c: Fix FastAGI to not wait for the non-existant pid
-
-2006-10-24 19:33 +0000 [r46131] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 46130 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r46130 | file | 2006-10-24 15:29:56 -0400 (Tue, 24 Oct 2006) | 2
- lines We need to initialize our scheduler pthread condition...
- yes. ........
-
-2006-10-24 17:14 +0000 [r46104-46120] Luigi Rizzo <rizzo@icir.org>
-
- * main/manager.c: i really think it is safe to commit this version,
- that simplifies the manager queue handling as described in the
- comment, and will make a lot easier to make further work on this
- code.
-
- * channels/chan_sip.c: correct fix for the bug i previously
- introduced - the strings are meant to be always initialized,
- independently from their content.
-
-2006-10-24 05:24 +0000 [r46094] Russell Bryant <russell@digium.com>
-
- * Makefile, /: Merged revisions 46093 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r46093 | russell | 2006-10-24 01:23:33 -0400 (Tue, 24 Oct 2006) |
- 3 lines Restore the ability to remove the firmware directory
- without causing the installation to fail (issue #8111) ........
-
-2006-10-24 03:15 +0000 [r46081] Kevin P. Fleming <kpfleming@digium.com>
-
- * doc/imapstorage.txt, /: Merged revisions 46080 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r46080 | kpfleming | 2006-10-23 22:13:08 -0500 (Mon, 23 Oct 2006)
- | 2 lines simplify and correct voicemail IMAP storage build
- instructions ........
-
-2006-10-24 03:09 +0000 [r46079] Tilghman Lesher <tlesher@digium.com>
-
- * main/channel.c, /: Merged revisions 46078 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r46078 | tilghman | 2006-10-23 22:01:00 -0500 (Mon, 23 Oct 2006)
- | 3 lines Pass through a frame if we don't know what it is,
- rather than trying to pass a NULL, which will segfault a channel
- driver (Bug 8149) ........
-
-2006-10-24 01:28 +0000 [r46055-46068] Russell Bryant <russell@digium.com>
-
- * utils/muted.c, /, utils/ael_main.c: Merged revisions 46067 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r46067 | russell | 2006-10-23 21:27:42 -0400 (Mon, 23 Oct 2006) |
- 7 lines In muted.c, check the return value of strdup. In
- ael_main.c, check the return value of calloc. (issue #8157) In
- passing fix a few minor bugs in ael_main.c. The last argument to
- strncpy() was a hard-coded 100, where it should have been 99. I
- changed this to use sizeof() - 1. ........
-
- * /, apps/app_meetme.c: Merged revisions 46065 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r46065 | russell | 2006-10-23 21:04:14 -0400 (Mon, 23 Oct 2006) |
- 2 lines Fix the descriptions of some of the MeetMeAdmin options
- (issue #8098, mflorell) ........
-
- * channels/chan_sip.c: Fix a seg fault on a registration. Line
- 7706, in parse_register_contact, explicitly passes NULL as the
- "pass" argument to this function.
-
-2006-10-23 21:46 +0000 [r46003-46045] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_sip.c: Unlike ast_strdup(), ast_strdupa() does not
- take a NULL pointer as argument, so fix the places where this
- might happen. This is also a fix that ought to go into 1.4 [The
- difference between the two functions is a bit confusing, and in
- asterisk i believe all string handling functions should be able
- to handl a NULL string as argument, but changing the API in trunk
- and not in 1.4 would make backporting harder.]
-
- * channels/chan_sip.c: remove a useless check for ocseq = 0. As
- discussed on the mailing lists, 0 is a legal value for Cseq, so
- there is no point to treat it specially.
-
- * channels/chan_sip.c: get_header() always returns a non-NULL
- value, so checking for NULL is certainly wrong and usually
- disables the checks that we want to make instead. This commit
- fixes a number of the above bugs where the result of get_header()
- is immediately checked for NULL. This is certainly a candidate
- for merging into 1.4
-
- * channels/chan_sip.c: put another duplicated block of code in a
- function.
-
- * channels/chan_sip.c: reformat a statement and comment a
- potentially wrong assignement (altering state on an unvalidated
- message).
-
- * channels/chan_sip.c: Remove unnecessary casts from const char *
- to char *, if necessary by slightly rearranging the code.
-
- * channels/chan_sip.c: another use for parse_uri(). On passing,
- remove a wrong comment (that probably I wrote myself!) and
- introduce a temporary variable to avoid a misleading cast.
-
-2006-10-23 17:08 +0000 [r46000] Russell Bryant <russell@digium.com>
-
- * /, res/res_jabber.c: Merged revisions 45999 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r45999 | russell | 2006-10-23 13:07:45 -0400 (Mon, 23 Oct 2006) |
- 2 lines don't crash when an incoming message has no "from" (issue
- #8205, jmls) ........
-
-2006-10-23 16:54 +0000 [r45945-45989] Luigi Rizzo <rizzo@icir.org>
-
- * main/utils.c: use autodetected support for gethostbyname_r
-
- * channels/chan_sip.c: + make sure parse_uri never returns NULL
- pointers - this simplifies its usage. + add another client for
- parse_uri, in handling Contact: strings (on passing, document the
- content of the "fullcontact" field); + in register_verify(), mark
- with XXX what i believe is another misinterpretation on the URI
- format when '@' is missing. No code changed here, so no fixes
- applied.
-
- * channels/chan_sip.c: After reading better the SIP RFC on sip URI
- (19.1.1) fix parse_uri() to interpret a missing userinfo section
- as a domain-only URI, and comment a wrong interpretation of the
- above in check_user_full(). The function has been patched to
- preserve the existing behaviour (in what admittedly is a corner
- case, but could be received under attacks). Hopefully the From:
- based matching will go away soon!
-
- * channels/chan_sip.c: in function get_also_info(), move argument
- stripping before splitting around the @, otherwise the
- refer_to_domain might contain arguments as well, causing
- failures. I think this is a true bug that ought to be fixed in
- 1.4 as well.
-
- * channels/chan_sip.c: start putting the URI parsing code in one
- place, introducing the function parse_uri() that splits a URI in
- its components. Right now use it only in one place, because the
- custom parsing that is done here and there sometimes has bugs
- that i want to figure out first.
-
- * channels/chan_sip.c: put common code in function terminate_uri()
- so we need to fix it only in one place.
-
- * channels/chan_sip.c: More cleanup of check_user_full with no
- functional change apart from a small (but disabled by default)
- new option. In detail: + introduce a new value for enum
- check_auth_result, AUTH_DONT_KNOW, used (read below) when a
- function does not have a conclusive response. Possibly this is
- the same as AUTH_NOT_FOUND, but need to check further. + move the
- large blocks (checking in the users list and in the peers list,
- respectively) from check_user_full() to separate functions. They
- return AUTH_DONT_KNOW in case they don't find a match, so the
- caller know that it has to try the next method. There is still
- some duplication of code here, but i have not tried yet to remove
- it. + [new option] a new option in sip.conf, match_auth_username,
- has been introduced, and disabled by default. If set, and the
- incoming request carries authentication info, the username to
- match in the users list is taken from there rather than from the
- From: field. This change is easy to identify, being made of - one
- line to declare the variable match_auth_username - a block of 15
- lines in check_user_full() - one line in sip list settings - two
- lines for parsing the config file. check_user_full() is now a lot
- cleaner - basically a sequence of checks that are applied to the
- request. This will help future work with new matching schemes.
-
-2006-10-23 00:33 +0000 [r45929] Joshua Colp <jcolp@digium.com>
-
- * /, cdr/cdr_odbc.c: Merged revisions 45928 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r45928 | file | 2006-10-22 20:27:39 -0400 (Sun,
- 22 Oct 2006) | 10 lines Merged revisions 45927 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r45927 | file | 2006-10-22 20:25:28 -0400 (Sun, 22 Oct 2006) | 2
- lines Don't leak memory mmmk? ........ ................
-
-2006-10-22 21:57 +0000 [r45917] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /: Merged revisions 45916 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r45916 | crichter | 2006-10-22 23:44:46 +0200
- (Sun, 22 Oct 2006) | 9 lines Merged revisions 45808 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r45808 | crichter | 2006-10-21 14:35:13 +0200 (Sat, 21
- Oct 2006) | 1 line fixed issue, that if chan_misdn is loaded and
- couldn't be initialized it would cause a segfault after 'reload'.
- Reported by Drew/Matt thx. ........ ................
-
-2006-10-22 21:08 +0000 [r45904-45915] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_sip.c: more streamlining of check_user_full
-
- * channels/chan_sip.c: simplify the flow of function
- check_user_full() A large block needs reindentation now, but we
- don't do that because it can be moved to a separate function.
-
- * channels/chan_sip.c: put duplicated code in functions.
-
-2006-10-22 19:34 +0000 [r45893] Russell Bryant <russell@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in: regenerate the
- configure script and autoconfig.h.in to reflect recent changes
- for https support for the built in http server
-
-2006-10-22 19:09 +0000 [r45858-45892] Luigi Rizzo <rizzo@icir.org>
-
- * main/Makefile, configure.ac, main/http.c,
- configs/http.conf.sample: Fix a few issues in the previous
- (disabled) HTTPS code, and support linux as well (using
- fopencookie(), which should be available in glibc). Update
- configure.ac to check for funopen (BSD) and fopencookie(glibc),
- and while we are at it also for gethostbyname_r (the generated
- files need to be updated, or you need to run bootstrap.sh
- yourself). Document the new options in http.conf.sample (names
- are only tentative, better ones are welcome). At this point we
- can safely enable the option. Anyone willing to try this on Sun
- and Apple platforms ?
-
- * main/http.c: Implement https support. The changes are not large.
- Most of the diff comes from putting the global variables
- describing an accept session into a structure, so we can reuse
- the existing code for running multiple accept threads on
- different ports. Once this is done, and if your system has the
- funopen() library function (and ssl, of course), it is just a
- matter of calling the appropriate functions to set up the ssl
- connection on the existing socket, and everything works on the
- secure channel now. At the moment, the code is disabled because i
- have not implemented yet the autoconf code to detect the presence
- of funopen(), and add -lssl to main/Makefile if ssl libraries are
- present. And a bit of documentation on the http.conf arguments,
- too. If you want to manually enable https support, that is very
- simple (step 0 1 2 will be eventually detected by ./configure,
- the rest is something you will have to do anyways). 0. make sure
- your system has funopen(3). FreeBSD does, linux probably does
- too, not sure about other systems. 1. uncomment the following
- line in main/http.c // #define DO_SSL /* comment in/out if you
- want to support ssl */ 2. add -lssl to AST_LIBS in main/Makefile
- 3. add the following options to http.conf sslenable=yes
- sslbindport=4433 ; pick one you like sslcert=/tmp/foo.pem ; path
- to your certificate file. 4. generate a suitable certificate e.g.
- (example from mini_httpd's Makefile: openssl req -new -x509 -days
- 365 -nodes -out /tmp/foo.pem -keyout /tmp/foo.pem and here you
- go: https://localhost:4433/asterisk/manager now works.
-
- * main/http.c: it is useless and possibly wrong to use ast_cli() to
- send the reply back to http clients. Use fprintf/fwrite instead,
- since we are already using a FILE * to read the input. If you
- wonder why, this is because it makes it trivial to implement
- https support (as long as your system has funopen()). And this is
- what i am going to put in with the next few commits...
-
-2006-10-22 04:44 +0000 [r45847] Joshua Colp <jcolp@digium.com>
-
- * Makefile, main/Makefile: Let's have build.h created a bit earlier
- so that func_version can use it and not stop the build on a fresh
- machine that has never had Asterisk installed on it before...
-
-2006-10-21 20:24 +0000 [r45836] Luigi Rizzo <rizzo@icir.org>
-
- * main/http.c: the default port number was erroneously stored in
- host order, and reading from the config file used ntohs instead
- of htons. this ought to be merged to 1.4 as well.
-
-2006-10-21 18:52 +0000 [r45820] Joshua Colp <jcolp@digium.com>
-
- * /, main/loader.c: Merged revisions 45817 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r45817 | file | 2006-10-21 14:48:58 -0400 (Sat, 21 Oct 2006) | 2
- lines Don't use promotion on Darwin because it doesn't seem to
- work quite right in all cases, this should solve the unresolved
- symbol issue people have been seeing. ........
-
-2006-10-21 18:50 +0000 [r45819] Russell Bryant <russell@digium.com>
-
- * /, res/res_monitor.c: Merged revisions 45818 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r45818 | russell | 2006-10-21 14:49:46 -0400 (Sat, 21 Oct 2006) |
- 3 lines Add a couple missing unregistrations of manager actions
- and remove duplicate unregistrations of applications. (issue
- #8194, jmls) ........
-
-2006-10-20 20:59 +0000 [r45786] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_sip.c: introduce sip_pvt_lock() and
- sip_pvt_unlock() wrappers to lock these data structures. This
- improve readability, and also hides the underlying locking
- mechanism so it is a lot easier to add diagnostic code, or move
- the object locks somewhere else, etc. On passing, rename the lock
- field in sip_pvt to pvt_lock, also for ease of readability.
-
-2006-10-20 19:04 +0000 [r45776] Joshua Colp <jcolp@digium.com>
-
- * Makefile, /: Merged revisions 45775 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r45775 | file | 2006-10-20 15:03:03 -0400 (Fri, 20 Oct 2006) | 2
- lines Pass DESTDIR and ASTSBINDIR so that the utilities get
- installed in the proper location (reported on asterisk-dev
- mailing list) ........
-
-2006-10-20 15:54 +0000 [r45764] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: put the constants for whether methods can
- create a dialog or not in an enum
-
-2006-10-20 11:24 +0000 [r45753] Luigi Rizzo <rizzo@icir.org>
-
- * main/manager.c: minor comment changes, code rearrangement and
- field renaming to minimize diffs with future modifications. The
- current implementation is problematic for the following reasons:
- + all insertions are O(N) because the event list does not have a
- tail pointer; + there is only a single lock protecting both
- session and users queues. + the implementation of the queue
- itself is not documented. I think i have figured it out, more or
- less, but am unclear on whether there is proper locking in place
- The rewrite (which i have working locally) uses a tailq so
- insertions are O(1), separate locks for the event and session
- queues, and has a documented implementation so hopefully we can
- figure out if/where bug exist.
-
-2006-10-20 08:14 +0000 [r45742-45743] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Let's repair the SIP attack shield :-)
-
- * main/manager.c: Doxygen corrections
-
-2006-10-19 22:06 +0000 [r45712-45724] Steve Murphy <murf@digium.com>
-
- * funcs/func_version.c (added): This new function, VERSION(),
- created via bug report 8176, may help dialplan programmers in the
- future. In the meantime, they can use the algorithm I outline on
- the bug report notes; If anyone invents something better, I'd
- hope they post it
-
- * utils/astman.c: astman was slightly weirding out over the new
- Dial and Newcallerid events
-
-2006-10-19 17:26 +0000 [r45696] Luigi Rizzo <rizzo@icir.org>
-
- * main/manager.c: more fixes to comments and very minor code
- rearrangement.
-
-2006-10-19 17:25 +0000 [r45693-45695] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_jabber.c: Merged revisions 45694 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r45694 | file | 2006-10-19 13:24:40 -0400 (Thu, 19 Oct 2006) | 2
- lines Let's remember to unregister JabberStatus too (issue #8184
- reported by jmls) ........
-
- * /, apps/app_externalivr.c: Merged revisions 45692 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r45692 | file | 2006-10-19 13:19:47 -0400 (Thu,
- 19 Oct 2006) | 10 lines Merged revisions 45691 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r45691 | file | 2006-10-19 13:16:37 -0400 (Thu, 19 Oct 2006) | 2
- lines Respect language selection when seeing if the file exists
- (issue #8178 reported by mnicholson) ........ ................
-
-2006-10-19 17:07 +0000 [r45690] Luigi Rizzo <rizzo@icir.org>
-
- * main/manager.c: implement proper XML/HTML formatting of multiple
- messages (e.g. the result of waitevent). Also fix some comments.
-
-2006-10-19 16:06 +0000 [r45679] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 45678 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r45678 | file | 2006-10-19 12:03:09 -0400 (Thu, 19 Oct 2006) | 2
- lines If the jitterbuffer is forced on then we can't partially
- bridge (reported by wangster on #asterisk-dev) ........
-
-2006-10-19 10:05 +0000 [r45648-45668] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_sip.c: move a large block out of do_monitor() and
- into a function, to improve readability.
-
- * channels/chan_sip.c: + move the definition of netlock as it was
- not related to the comment just above; + decouple the struct
- definition and variable declaration (iflist);
-
- * main/manager.c: more documentation of data structure and
- functions. Of interest: + ast_get_manager_by_name_locked() is now
- without the ast_ prefix as it is a local function; +
- unuse_eventqent() renamed to unref_event(), and returns the
- pointer to the next entry. + marked with XXX a couple of usages
- of unref_event() because i suspect we are addressing the wrong
- entry.
-
-2006-10-19 07:17 +0000 [r45647] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Cleaning up... Removing duplicate (again)
-
-2006-10-19 02:16 +0000 [r45634] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_sip.c, include/asterisk/threadstorage.h: restore
- freeing of threadstorage objects without custom cleanup functions
- allow custom threadstorage init functions to return failure use a
- custom init function for chan_sip's temp_pvt, to improve
- performance a bit
-
-2006-10-19 01:04 +0000 [r45623-45624] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merge fix to not leak the stringfields of
- a thread speicif sip_pvt. This also includes the fix not to leak
- the actual sip_pvt. Merged revisions 45622 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r45622 | russell | 2006-10-18 20:59:51 -0400 (Wed, 18 Oct 2006) |
- 2 lines Don't leak the actual thread-specific sip_pvt struct
- ........
-
- * main/channel.c, main/frame.c, main/manager.c,
- channels/chan_sip.c, channels/chan_skinny.c, main/logger.c,
- main/utils.c, channels/iax2-parser.c,
- include/asterisk/threadstorage.h, main/cli.c: Extend the thread
- storage API such that a custom initialization function can be
- called for each thread specific object after they are allocated.
- Note that there was already the ability to define a custom
- cleanup function. Also, if the custom cleanup function is used,
- it *MUST* call free on the thread specific object at the end.
- There is no way to have this magically done that I can think of
- because the cleanup function registered with the pthread
- implementation will only call the function back with a pointer to
- the thread specific object, not the parent ast_threadstorage
- object.
-
-2006-10-18 22:40 +0000 [r45611] Luigi Rizzo <rizzo@icir.org>
-
- * main/manager.c: silent warning from a debugging message (which
- will go away soon, anyways)
-
-2006-10-18 22:19 +0000 [r45610] Joshua Colp <jcolp@digium.com>
-
- * apps/app_meetme.c, CHANGES: Just for Nicholson - here's an
- option, C, to Meetme that will allow it to continue in the
- dialplan if the person is kicked out. (issue #7994 reported by
- mnicholson with mods by myself)
-
-2006-10-18 21:41 +0000 [r45597-45599] Luigi Rizzo <rizzo@icir.org>
-
- * main/manager.c: remove trailing whitespace
-
- * main/manager.c: ouch! remember to unlink temporary files once
- done with them.
-
- * main/manager.c: + move output_format variables in the http
- section of the file; + more comments on struct mansession and
- global variables; + small improvements to the session matching
- code so it supports multiple sessions from the same IP
-
-2006-10-18 21:05 +0000 [r45596] Joshua Colp <jcolp@digium.com>
-
- * /, main/asterisk.c: Merged revisions 45595 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r45595 | file | 2006-10-18 17:03:34 -0400 (Wed, 18 Oct 2006) | 2
- lines Don't modify things if we are using vfork as this is very
- bad and may cause unexpected behavior (issue #7970 reported by
- Nick Gavrikov) ........
-
-2006-10-18 17:53 +0000 [r45572-45583] Luigi Rizzo <rizzo@icir.org>
-
- * main/manager.c: another bunch of comments on the data structures.
-
- * main/manager.c: despite the large changes, this commit only moves
- functions around so that functions belonging to the same group
- are close to each other. At the beginning of each group i have
- added a bit of documentation to explain what the group does and
- what is the typical flow - basically, all i have learned by code
- inspection over the past few days should be documented for you to
- read. I have not put many doxygen annotations just because i am
- not sure what are the proper ones. Hopefully some doxygen experts
- will jump in. Next on the plate: try to figure out how "struct
- eventqent" are supposed to work.
-
- * main/manager.c: more comment and formatting fixes, small
- simplifications to functions get_input() and session_do()
-
-2006-10-18 16:45 +0000 [r45571] Matt O'Gorman <mogorman@digium.com>
-
- * main/manager.c: rizzo compile then commit, maybe even run it too
- ^_^
-
-2006-10-18 15:49 +0000 [r45529-45561] Luigi Rizzo <rizzo@icir.org>
-
- * main/manager.c: comment and cleanup the main thread. On passing,
- fix a bug: close the socket if the allocation of a structure for
- the new session fails. (the bugfix is a candidate for 1.4)
-
- * main/manager.c: create a new (internal, for the time being)
- function astman_start_ack() to start manager responses that need
- further lines. This removes a lot of duplicate code from the
- various handlers that at the moment build an ActionID string
- themselves. Once settled, the function should move to manager.h
- so it can be used by other files (chan_agent, chan_iax2,
- chan_sip, chan_zap, res_jabber and app_queue). I am not totally
- clear if there is a preferred position for the ActionID: line in
- a message. Some instances put it at the end, but one would argue
- that it is preferable to have it at the beginning.
-
- * main/manager.c: more indentation cleanup from previous commits,
- and remove the "busy" field from struct mansession as it was not
- used correctly anyways.
-
- * main/manager.c: create proper handlers for "Challenge" and
- "Login" actions, rather than use inline code for them. Things are
- more readable this way, and also error processing is more
- consistent.
-
- * main/manager.c: fix indentation from a commit of a couple of days
- ago
-
- * main/manager.c: another batch of simplifications to
- authenticate() (they are committed a bit at a time so it is
- easier to revert them in case we find a bug at a later time).
-
-2006-10-18 12:15 +0000 [r45528] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Remove duplicate declarations...
-
-2006-10-18 11:59 +0000 [r45463-45518] Luigi Rizzo <rizzo@icir.org>
-
- * main/manager.c, configs/manager.conf.sample: remove unused fields
- and unimplemented options.
-
- * main/manager.c: first pass as simplifying authenticate(),
- avoiding whitespace changes
-
- * main/manager.c: more code simplifications
-
- * main/manager.c: simplify ast_strings_to_mask
-
- * main/manager.c: add a comment to remember that a block of code is
- completely redundant.
-
- * main/manager.c: + move the enum declaration for output formats
- near the head of the file, so it can be used from more places; +
- make the declaration of contenttype[] more robust; + remove the
- wrappers around __xml_translate(), since they were used only in
- one place, and rename to xml_translate(). This allows for a bit
- of simplifications. + document the output produced by the above
- function.
-
- * main/manager.c: merge xml_translate() and html_translate() into
- one function since they do similar things. Add a small form on
- top of the html output so request like
- http://foo:8088/asterisk/manager will suggest you what to do.
- Note: i suspect there is still a bug somewhere in the session
- matching code, as sometimes you have to login twice in order for
- the following commands to be recognised. Apart from this, the cli
- now is basically usable from a web form!
-
- * main/http.c: introduce uri_decode() so that '+' are translated
- into ' ' (e.g. browsers do this when they encode input strings
- from a form).
-
- * main/http.c: various code simplifications to reduce nesting
- depth, minor optimizations to avoid extra calls of strlen(), and
- some variable localization. One feature worth backporting is the
- move of ast_variables_destroy() to a different place in
- handle_uri() to avoid leaking memory in case a uri is not found.
-
-2006-10-18 03:03 +0000 [r45453] Joshua Colp <jcolp@digium.com>
-
- * /, main/rtp.c: Merged revisions 45452 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r45452 | file | 2006-10-17 23:02:08 -0400 (Tue, 17 Oct 2006) | 2
- lines Don't segfault if you're using a channel driver that
- doesn't turn RTCP on ........
-
-2006-10-18 02:46 +0000 [r45440-45442] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /: Merged revisions 45441 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r45441 | russell | 2006-10-17 22:41:36 -0400 (Tue, 17 Oct 2006) |
- 7 lines Don't attempt to access private data members of the
- pthread_mutex_t object, because this does not work on all linux
- systems. Instead, just access the reentrancy field in the
- ast_mutex_info struct when DEBUG_THREADS is enabled. If
- DEBUG_CHANNEL_LOCKS is enabled, the developer probably has
- DEBUG_THREADS on as well. (issue #8139, me) ........
-
- * configs/sip_notify.conf.sample, /: Merged revisions 45439 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r45439 | russell | 2006-10-17 22:19:07 -0400 (Tue, 17 Oct 2006) |
- 2 lines update entry to reboot a snom phone (issue #7850,
- pnlarsson) ........
-
-2006-10-17 23:06 +0000 [r45426] Steve Murphy <murf@digium.com>
-
- * res/res_agi.c: As per bug 6779, this patch is now applied to
- trunk; while I was at it, I corrected a reference to a CLI
- command, to follow the new regime.
-
-2006-10-17 22:32 +0000 [r45409-45411] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, build_tools/prep_tarball (added): Merged revisions 45410 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r45410 | kpfleming | 2006-10-17 17:31:54 -0500 (Tue, 17 Oct 2006)
- | 2 lines add a project-specific script to be used during release
- preparation ........
-
- * main/channel.c, /, channels/chan_sip.c, channels/chan_iax2.c,
- include/asterisk/stringfields.h, main/ast_expr2.c: Merged
- revisions 45408 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r45408 | kpfleming | 2006-10-17 17:24:10 -0500 (Tue, 17 Oct 2006)
- | 3 lines optimize the 'quick response' code a bit more... no
- more malloc() or memset() for each response expand stringfields
- API a bit to allow reusing the stringfield pool on a structure
- when needed, and remove some unnecessary code when the structure
- was being freed ........
-
-2006-10-17 21:09 +0000 [r45379-45398] Joshua Colp <jcolp@digium.com>
-
- * main/manager.c: Warning be gone!
-
- * /, channels/chan_sip.c: Merged revisions 45378 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r45378 | file | 2006-10-17 16:30:34 -0400 (Tue, 17 Oct 2006) | 2
- lines Don't create a "real" pvt structure for requests that
- shouldn't be able to create one. Instead use a temporary pvt and
- fill it with enough information so we can send a reply. ........
-
-2006-10-17 19:57 +0000 [r45365] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c, doc/channelvariables.txt: Issue #5484
- (branch sipdiversion) - Support for Diversion header in redirects
- of calls with 302 redirection. (tinning)
-
-2006-10-17 18:08 +0000 [r45351] Luigi Rizzo <rizzo@icir.org>
-
- * main/manager.c: simplify authority_to_str() using
- ast_build_string()
-
-2006-10-17 17:54 +0000 [r45335] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Issue #7254 - Add support of "423 Interval
- too brief" to outbound SIP registrations. Thanks, tardieu!
-
-2006-10-17 17:51 +0000 [r45334] Luigi Rizzo <rizzo@icir.org>
-
- * main/manager.c: Improve the XML formatting of responses coming
- from web interface. Normal responses are sequences of lines of
- the form "Name: value", with \r\n as line terminators and an
- empty line as a response terminator. Generi CLI commands,
- however, do not have such a clean formatting, and the existing
- code failed to generate valid XML for them. Obviously we can only
- use heuristics here, and we do the following: - accept either \r
- or \n as a line terminator, trimming trailing whitespace; - if a
- line does not have a ":" in it, assume that from this point on we
- have unformatted data, and use "Opaque-data:" as a name; - if a
- line does have a ":" in it, the Name field is not always a legal
- identifier, so replace non-alphanum characters with underscores;
- All the above is to be refined as we improve the formatting of
- responses from the CLI. And, all the above ought to go as a
- comment in the code rather than just in a commit message...
-
-2006-10-17 17:51 +0000 [r45331-45333] Olle Johansson <oej@edvina.net>
-
- * /, configs/sip.conf.sample: Update of docs
-
- * channels/chan_sip.c: - Remove unneeded code that won't be reached
- now that we kill responses to unkonwn dialogs earlier in the
- process. - move debug message.
-
-2006-10-17 17:41 +0000 [r45330] Luigi Rizzo <rizzo@icir.org>
-
- * main/manager.c: open a temporary file to receive the output from
- cli commands invoked through the http interface. It is not
- terribly efficient but better than no output at all. Todo: use a
- configurable /tmp directory instead of a hardwired one.
-
-2006-10-17 17:22 +0000 [r45328] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, LICENSE: Merged revisions 45327 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r45327 | kpfleming | 2006-10-17 12:22:25 -0500
- (Tue, 17 Oct 2006) | 10 lines Merged revisions 45326 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r45326 | kpfleming | 2006-10-17 12:22:01 -0500 (Tue, 17
- Oct 2006) | 2 lines provide licensing language for IAXy firmware
- file ........ ................
-
-2006-10-17 17:19 +0000 [r45325] Luigi Rizzo <rizzo@icir.org>
-
- * main/manager.c: document xml_copy_escape() and add an extra
- function, namely replace non-alphanum chars with underscore. This
- is useful when building field names in xml formatting.
-
-2006-10-17 16:27 +0000 [r45295-45316] Olle Johansson <oej@edvina.net>
-
- * /: ...block this one too... Only applies to 1.4 since the fix for
- trunk was different.
-
- * /: Block patch from 1.4 that does not apply here.
-
- * channels/chan_sip.c: Get rid of the ignore variable that was only
- partially replaced by the flag.
-
-2006-10-16 20:26 +0000 [r45234-45286] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c, configs/sip.conf.sample: In the course of a
- data this has been turned into an option to ignore replies, then
- ignore responses and finally I'm just getting rid of the option
- altogether and making it the default no matter what. C'est la
- vie!
-
- * /: Woof.
-
- * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
- 45280 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r45280 | file | 2006-10-16 16:06:18 -0400 (Mon,
- 16 Oct 2006) | 10 lines Merged revisions 45265 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r45265 | file | 2006-10-16 15:59:54 -0400 (Mon, 16 Oct 2006) | 2
- lines Use responses rather then replies even though they mean the
- same thing. ........ ................
-
- * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
- 45262 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r45262 | file | 2006-10-16 15:37:34 -0400 (Mon,
- 16 Oct 2006) | 10 lines Merged revisions 45260 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r45260 | file | 2006-10-16 15:28:18 -0400 (Mon, 16 Oct 2006) | 2
- lines Add 'ignoreoodreplies' option which will not create a pvt
- structure on a SIP response but instead basically drop it.
- ........ ................
-
- * apps/app_directed_pickup.c: It's new directed pickup! This now
- features a more sane way of finding the channel to pick up (I
- snuck it into the tree on Friday... bet you didn't know I'd
- actually use it eh?). PICKUPMARK now also works in a different
- way, you should prefix it with _ when setting it so it gets
- inherited onto the channel(s) created in app_dial as directed
- pickup will now look for it on the target channel, not the
- originating channel. (BE-85)
-
-2006-10-16 14:03 +0000 [r45224] Olle Johansson <oej@edvina.net>
-
- * CREDITS, /: Update
-
-2006-10-16 14:00 +0000 [r45219] Luigi Rizzo <rizzo@icir.org>
-
- * main/manager.c: + comment some unclear fields of struct
- mansession; + let some commands (Challenge, Login) be processed
- even if already authenticated, as it doesn't harm and prevents
- some incorrect error messages + remove custom code for Logoff -
- the existing handler was ok. Some indentation fixes may be
- necessary
-
-2006-10-16 13:20 +0000 [r45194-45209] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: When adding new functions, please add a
- forward declaration. I *know* it is not required, but it makes
- navigation easier and will help when splitting up this large
- source code file. Thank you!
-
- * /, channels/chan_sip.c: Importing rev 45196 from 1.4 - don't kill
- dialog for a bad response
-
- * channels/chan_sip.c: A B2BUA should *not* issue proxy auth.
-
-2006-10-16 11:29 +0000 [r45151-45185] Luigi Rizzo <rizzo@icir.org>
-
- * main/manager.c: + comment some unclear requirements for
- master_eventq + remove the need for an snprintf in
- astman_get_header() + fix comment for manager list eventq +
- localize one variable and minor code simplifications.
-
- * main/manager.c: protect access to first_action with actionlock.
- Mark with XXX one place (during command execution) where
- navigation should be protected with actionlock, but is not
- because it would block requests for a long time. To solve this
- properly we need to put reference counts in the struct
- manager_action. A suboptimal fix is to copy the record on a
- search and then unlock the list while we work on the copy.
-
- * main/http.c: comment some functions, and more small code
- simplifications
-
- * main/http.c: fix indentation of a large block after changes in
- previous commit (basically whitespace only).
-
- * main/http.c: simplify string parsing routines using ast_skip_*()
- functions.
-
- * main/http.c: don't forget to close a descriptor on a malloc
- failure. On passing, small rearrangement of the code to reduce
- indentation. There is a bit more cleanup planned for this file,
- so a merge to 1.4 will be done when it is all done.
-
- * main/http.c: typo: serer -> server
-
-2006-10-14 04:36 +0000 [r45142] Steve Murphy <murf@digium.com>
-
- * funcs/func_rand.c: update the doc string for both AEL and
- extensions.conf users.
-
-2006-10-13 23:03 +0000 [r45126] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, main/acl.c: Merged revisions 45125 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r45125 | kpfleming | 2006-10-13 18:02:48 -0500 (Fri, 13 Oct 2006)
- | 7 lines
- ------------------------------------------------------------------------
- r45119 | kpfleming | 2006-10-13 17:57:42 -0500 (Fri, 13 Oct 2006)
- | 2 lines don't drop the entire permit/deny list when an attempt
- is made to add an invalid entry (BE-92)
- ------------------------------------------------------------------------
- ........
-
-2006-10-13 21:20 +0000 [r45105-45109] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c: Inherit the context and extension until the
- channel is answered
-
- * /, res/res_speech.c: Merged revisions 45106 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r45106 | file | 2006-10-13 17:06:09 -0400 (Fri, 13 Oct 2006) | 2
- lines Clear the quiet flag too since we are restarting a
- recognition again (reported on -dev by Stephan Edelman) ........
-
- * /, res/res_speech.c: Merged revisions 45104 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r45104 | file | 2006-10-13 17:01:13 -0400 (Fri, 13 Oct 2006) | 2
- lines Check return value from engine in case of failure (ie: out
- of licenses) (reported on -dev mailing list) ........
-
-2006-10-13 19:24 +0000 [r45089] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /: Merged revisions 45088 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r45088 | crichter | 2006-10-13 21:19:46 +0200 (Fr, 13
- Okt 2006) | 1 line avoiding warning, fixing potential bug
- ........
-
-2006-10-13 18:45 +0000 [r45080] Joshua Colp <jcolp@digium.com>
-
- * codecs/lpc10/median.c, codecs/lpc10/encode.c,
- codecs/lpc10/ivfilt.c, /, codecs/lpc10/bsynz.c,
- codecs/lpc10/prepro.c, codecs/lpc10/invert.c,
- codecs/lpc10/placev.c, codecs/lpc10/irc2pc.c,
- codecs/lpc10/decode.c, codecs/lpc10/dcbias.c,
- codecs/lpc10/pitsyn.c, codecs/lpc10/difmag.c,
- codecs/lpc10/voicin.c, codecs/lpc10/synths.c,
- codecs/lpc10/preemp.c, codecs/lpc10/hp100.c,
- codecs/lpc10/lpfilt.c, codecs/lpc10/rcchk.c,
- codecs/lpc10/mload.c, codecs/lpc10/lpcenc.c,
- codecs/lpc10/vparms.c, codecs/lpc10/dyptrk.c,
- codecs/lpc10/lpcini.c, codecs/lpc10/random.c,
- codecs/lpc10/ham84.c, codecs/lpc10/chanwr.c,
- codecs/lpc10/placea.c, codecs/lpc10/tbdm.c,
- codecs/lpc10/analys.c, codecs/lpc10/onset.c,
- codecs/lpc10/energy.c, codecs/lpc10/lpcdec.c,
- codecs/lpc10/deemp.c: Merged revisions 45079 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r45079 | file | 2006-10-13 14:42:49 -0400 (Fri, 13 Oct 2006) | 2
- lines And file said... let the compiler warnings STOP! ........
-
-2006-10-13 18:08 +0000 [r45078] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael-test/ref.ael-vtest17 (added),
- pbx/ael/ael-test/ael-vtest17/extensions.ael (added),
- pbx/ael/ael-test/ael-vtest17 (added),
- pbx/ael/ael-test/ref.ael-test3, pbx/pbx_ael.c: Correction for bug
- 8128 in trunk
-
-2006-10-13 17:06 +0000 [r45052-45067] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_chanspy.c: Merged revisions 45066 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r45066 | file | 2006-10-13 13:05:02 -0400 (Fri,
- 13 Oct 2006) | 10 lines Merged revisions 45060 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r45060 | file | 2006-10-13 13:01:22 -0400 (Fri, 13 Oct 2006) | 2
- lines Turn on volume adjustment if it needs to be on (issue #8136
- reported by mnicholson) ........ ................
-
- * /, apps/app_playback.c: Merged revisions 45051 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r45051 | file | 2006-10-13 12:20:58 -0400 (Fri, 13 Oct 2006) | 2
- lines Move say.conf existence check to do_say function since it
- is called from multiple places (issue #8144 reported by kshumard)
- ........
-
-2006-10-13 16:20 +0000 [r45050] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 45049 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r45049 | kpfleming | 2006-10-13 11:19:35 -0500
- (Fri, 13 Oct 2006) | 10 lines Merged revisions 45048 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r45048 | kpfleming | 2006-10-13 11:18:08 -0500 (Fri, 13
- Oct 2006) | 2 lines when sending a call to a peer, use the proper
- socket if we have multiple bindings (reported on asterisk-dev)
- ........ ................
-
-2006-10-13 16:02 +0000 [r45032-45047] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 45040 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r45040 | file | 2006-10-13 12:01:17 -0400 (Fri, 13 Oct 2006) | 2
- lines Complete merging in RPID screen changes (issue #8101
- reported by hristo, patch by oej in revision 44757) ........
-
- * main/dnsmgr.c, /: Merged revisions 45031 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r45031 | file | 2006-10-13 11:53:22 -0400 (Fri,
- 13 Oct 2006) | 10 lines Merged revisions 45030 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r45030 | file | 2006-10-13 11:49:53 -0400 (Fri, 13 Oct 2006) | 2
- lines Pass the right value to usleep for sleeping, and always add
- the background refresh item back into the scheduler if enabled
- since it is deleted during reload. (issue #8142 reported by
- p_lindheimer) ........ ................
-
-2006-10-13 15:47 +0000 [r45029] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
- main/utils.c: Merged revisions 45027 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r45027 | kpfleming | 2006-10-13 10:41:14 -0500 (Fri, 13 Oct 2006)
- | 2 lines use a configure script test for PMTU discovery control
- instead of just assuming it's available on Linux ........
-
-2006-10-13 15:42 +0000 [r45028] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged
- revisions 45026 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r45026 | crichter | 2006-10-13 16:45:39 +0200
- (Fr, 13 Okt 2006) | 9 lines Merged revisions 45020 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r45020 | crichter | 2006-10-13 15:11:13 +0200 (Fr, 13
- Okt 2006) | 1 line fixed some echocandisable issues when bridged.
- this caused a kernel panic sometimes..also some minor formatting
- fixes ........ ................
-
-2006-10-13 11:18 +0000 [r45009-45010] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_sip.c: Try to avoid the use of 'z' modifier in
- cases where it is not necessary - rather, cast the argument to
- int. In this case, the string is in a UDP packet and as such
- limited to 64k so its length can be safely represented in an int
- without truncation (besides, this is just a debugging message!)
-
- * channels/chan_sip.c: arguments to auth_headers() needed to be
- swapped here. To avoid the same mistake in the future (due to
- slightly confusing variable names), add a comment. On passing,
- remove a redundant initialization.
-
-2006-10-13 08:23 +0000 [r45000] Christian Richter <christian.richter@beronet.com>
-
- * /, channels/misdn/isdn_msg_parser.c: Merged revisions 44994 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r44994 | crichter | 2006-10-13 09:52:41 +0200
- (Fr, 13 Okt 2006) | 9 lines Merged revisions 44993 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r44993 | crichter | 2006-10-13 09:40:07 +0200 (Fr, 13
- Okt 2006) | 1 line fixed issue, that the hangupcause got a wrong
- isdn cause at RELEASE_COMPLETE ........ ................
-
-2006-10-12 20:41 +0000 [r44983] Matt O'Gorman <mogorman@digium.com>
-
- * /, channels/chan_gtalk.c: Merged revisions 44982 via svnmerge
- from https://svn.digium.com/svn/asterisk/branches/1.4 ........
- r44982 | mogorman | 2006-10-12 15:34:49 -0500 (Thu, 12 Oct 2006)
- | 2 lines fix for bug 7764. ........
-
-2006-10-12 19:16 +0000 [r44957-44973] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_sip.c: eliminate compiler warning
-
- * /, channels/chan_sip.c: Merged revisions 44971 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44971 | kpfleming | 2006-10-12 14:14:24 -0500 (Thu, 12 Oct 2006)
- | 2 lines we can only send one 'a=ptime' attribute per media
- session, not one for each format ........
-
- * include/asterisk/utils.h, /, channels/chan_sip.c, main/utils.c,
- main/netsock.c: Merged revisions 44956 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r44956 | kpfleming | 2006-10-12 13:38:51 -0500
- (Thu, 12 Oct 2006) | 10 lines Merged revisions 44955 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r44955 | kpfleming | 2006-10-12 13:31:26 -0500 (Thu, 12
- Oct 2006) | 2 lines ensure that IAX2 and SIP sockets allow UDP
- fragmentation when running on Linux (thanks to Brian Candler on
- the asterisk-dev list for the tip) ........ ................
-
-2006-10-12 16:57 +0000 [r44944-44946] Russell Bryant <russell@digium.com>
-
- * main/manager.c, /: Merged revisions 44945 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44945 | russell | 2006-10-12 12:56:32 -0400 (Thu, 12 Oct 2006) |
- 2 lines fix a silly typo in a comment that I saw while reading
- the commit list ........
-
- * pbx/pbx_dundi.c: put flags in an enum and remove a couple of
- unused defines
-
-2006-10-12 16:11 +0000 [r44943] Joshua Colp <jcolp@digium.com>
-
- * Makefile, /: Merged revisions 44942 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44942 | file | 2006-10-12 12:08:50 -0400 (Thu, 12 Oct 2006) | 2
- lines Pass off AUDIO_LIBS so muted can link on OSX (issue #8135
- reported by ssokol) ........
-
-2006-10-12 15:12 +0000 [r44933] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_sip.c: + move [almost] all instances of
- WWW-Authenticate/Proxy-Authenticate and friends in a function,
- auth_headers(), which is used to simplify the interface of
- do_{proxy|register}_auth(). + use PROXY_AUTH = 407, WWW_AUTH =
- 401 as values for enum sip_auth_type; No functional change, only
- code cleanup.
-
-2006-10-12 13:04 +0000 [r44922] Nadi Sarrar <ns@beronet.com>
-
- * main/manager.c, /: Merged revisions 44921 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44921 | nadi | 2006-10-12 14:55:25 +0200 (Do, 12 Okt 2006) | 2
- lines append_event must be called while holding the session lock
- ........
-
-2006-10-12 10:26 +0000 [r44912] Russell Bryant <russell@digium.com>
-
- * /, res/res_jabber.c: Merged revisions 44911 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44911 | russell | 2006-10-12 06:24:36 -0400 (Thu, 12 Oct 2006) |
- 2 lines change some debug output to use LOG_DEBUG instead of
- verbose output ........
-
-2006-10-11 23:36 +0000 [r44900-44901] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_sip.c: reduce indentation of two large blocks
-
- * channels/chan_sip.c: operator != also works between booleans...
-
-2006-10-11 16:57 +0000 [r44889] Jason Parker <jparker@digium.com>
-
- * /, main/db1-ast/Makefile: Merged revisions 44888 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r44888 | qwell | 2006-10-11 11:57:06 -0500 (Wed, 11 Oct
- 2006) | 3 lines These are already set by the parent Makefile..
- There is no need to have this here (it doesn't actually work
- anyways). ........
-
-2006-10-11 13:45 +0000 [r44876-44877] Russell Bryant <russell@digium.com>
-
- * doc/linkedlists.txt (removed): Remove doc/linkedlists.txt as it
- is no longer needed. The top of the file reads: As of 2004-12-23,
- this documentation is no longer maintained. The doxygen
- documentation generated from linkedlists.h should be referred to
- in its place, as it is more complete and better maintained.
-
- * channels/chan_sip.c: Revert Luigi's accidental commit of his
- local changes when debugging some SIP authentication issues. This
- was committed in revision 44844, where the commit message was
- just "small formatting cleanup", so I am pretty sure he didn't
- mean to commit this part.
-
-2006-10-11 13:21 +0000 [r44844-44875] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_sip.c: remove duplicate prototypes. Have not
- checked if there are more.
-
- * channels/chan_sip.c: simplify and comment
- handle_response_peerpoke()
-
- * channels/chan_sip.c: fix indentation of a function after previous
- commit (whitespace-only change)
-
- * channels/chan_sip.c: handle_response_peerpoke() does not need to
- return anything. (Reindentation in the next commit.)
-
- * channels/chan_sip.c: small formatting cleanup
-
-2006-10-11 08:45 +0000 [r44840-44843] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /: Merged revisions 44563 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r44563 | crichter | 2006-10-06 14:53:41 +0200
- (Fr, 06 Okt 2006) | 9 lines Merged revisions 44460 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r44460 | crichter | 2006-10-05 12:02:38 +0200 (Do, 05
- Okt 2006) | 1 line fixed segfault which happens during
- hold/transfer action ........ ................
-
- * channels/chan_misdn.c, /: Merged revisions 44562 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r44562 | crichter | 2006-10-06 14:52:01 +0200
- (Fr, 06 Okt 2006) | 9 lines Merged revisions 44335 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r44335 | crichter | 2006-10-04 17:26:59 +0200 (Mi, 04
- Okt 2006) | 1 line if INFORMATION Message come with keypad
- instead of called party number, we just use the keypad as called
- party number. ........ ................
-
- * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
- channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample,
- channels/misdn/isdn_lib.c, channels/misdn_config.c: Merged
- revisions 44561 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r44561 | crichter | 2006-10-06 14:50:25 +0200
- (Fr, 06 Okt 2006) | 9 lines Merged revisions 44334 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r44334 | crichter | 2006-10-04 17:13:58 +0200 (Mi, 04
- Okt 2006) | 1 line added the option 'reject_cause' to make it
- possible to set the RELEASE_COMPLETE - cause on the 3. incoming
- PMP channel, which is automatically rejected because chan_misdn
- does not support that kind of callwaiting. Therefore chan_misdn
- supports now 3 incoming channels on a PMP BRI Port.
- misdn_lib_get_free_bc now gets the info if the requested channel
- is incoming or outgoing to make the 3. channel possible ........
- ................
-
- * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
- channels/misdn/isdn_lib.c: Merged revisions 44559 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r44559 | crichter | 2006-10-06 12:44:34 +0200
- (Fr, 06 Okt 2006) | 9 lines Merged revisions 44149 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r44149 | crichter | 2006-10-02 15:28:14 +0200 (Mo, 02
- Okt 2006) | 1 line fixed the hold/retrieve/transfer issues,
- removed a useless bc field, added setting of frame.delivery
- fields, some minor code cleanups ........ ................
-
-2006-10-10 20:52 +0000 [r44831] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_rpt.c: More whitespace fixes
-
-2006-10-10 17:23 +0000 [r44820] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 44819 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44819 | file | 2006-10-10 13:21:44 -0400 (Tue, 10 Oct 2006) | 2
- lines Move some stuff around so that a NOTIFY dialog won't hang
- around until the end of the world under certain circumstances
- ........
-
-2006-10-10 16:46 +0000 [r44810] Tilghman Lesher <tlesher@digium.com>
-
- * /, funcs/func_logic.c: Merged revisions 44808 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44808 | tilghman | 2006-10-10 11:42:19 -0500 (Tue, 10 Oct 2006)
- | 2 lines Lost of a bit of logic when this was simplified between
- 1.2 and 1.4 (Bug 8117) ........
-
-2006-10-10 16:31 +0000 [r44789-44807] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 44806 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44806 | file | 2006-10-10 12:30:00 -0400 (Tue, 10 Oct 2006) | 2
- lines Bail out if we have no refer structure and we get a refer
- response ........
-
- * /, channels/chan_sip.c: Merged revisions 44788 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44788 | file | 2006-10-10 11:23:14 -0400 (Tue, 10 Oct 2006) | 2
- lines Only set DTMF information if an RTP structure exists
- ........
-
-2006-10-10 14:54 +0000 [r44787] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged
- revisions 44786 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r44786 | crichter | 2006-10-10 15:50:26 +0200
- (Di, 10 Okt 2006) | 9 lines Merged revisions 44785 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r44785 | crichter | 2006-10-10 15:34:33 +0200 (Di, 10
- Okt 2006) | 1 line (re)added support of dynamical enabling hdlc
- on bchannels ........ ................
-
-2006-10-10 08:08 +0000 [r44770-44774] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_sip.c: clarify the use of the standard SIP port
- number, 5060, and rename the old DEFAULT_SIP_PORT as
- STANDARD_SIP_PORT to make it clear that this is not something we
- can change, unlike other defaults.
-
- * channels/chan_sip.c: improve formatting of SIP packets when
- dumped to the verbose output stream, so it is easier to find them
- in the log.
-
-2006-10-09 18:23 +0000 [r44768] Joshua Colp <jcolp@digium.com>
-
- * funcs/func_timeout.c: Timeout values are in seconds (issue #7122
- reported by jmls)
-
-2006-10-09 16:15 +0000 [r44765] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_skinny.c: Merged revisions 44764 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r44764 | qwell | 2006-10-09 11:12:35 -0500 (Mon, 09 Oct
- 2006) | 4 lines Fix a problem where phones that go "missing"
- never got unregistered. Issue #8067, reported by pj, patch by
- Anthony LaMantia (with minor whitespace modifications) ........
-
-2006-10-09 15:52 +0000 [r44762-44763] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 44759 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44759 | file | 2006-10-09 11:41:28 -0400 (Mon, 09 Oct 2006) | 2
- lines Properly avoid a collision with iax2_hangup (issue #8115
- reported by vazir) ........
-
-2006-10-09 11:20 +0000 [r44753] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Being pedantic... "media" is easier to
- understand than "data" in the function name... :-)
-
-2006-10-09 09:04 +0000 [r44745-44752] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_sip.c: slightly restructure sipsock_read() removing
- a "goto"
-
- * channels/chan_sip.c: use S_OR in one place
-
- * channels/chan_sip.c: update_call_counter(): indentation fixes and
- small simplifications at the top of the function.
-
- * channels/chan_sip.c: localize some variables and reduce nesting
- depth (indentation will be fixed by a separate commit).
-
- * channels/chan_sip.c: small simplification to initreqprep()
-
- * channels/chan_sip.c: Simplify function parse_request() using a
- single loop instead of two very similar ones.
-
- * channels/chan_sip.c: do not dereference p if we know it is NULL.
-
-2006-10-07 20:42 +0000 [r44697-44731] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Fix some debug output for setsockopt for TOS
-
- * channels/chan_sip.c: - move definition of global_autoframing to
- the same place as other globals - set initial value at
- load/reload - Add questionmarks for someone to fill in for
- doxygen
-
- * channels/chan_sip.c: Add/change doxygen and comments
-
- * configs/sip.conf.sample: Recommend using "sip reload" since it's
- much easier to learn and remember.
-
- * channels/chan_sip.c: Explain usage of DEFAULT_SIP_PORT
-
- * channels/chan_sip.c: Do *NOT* use DEFAULT_SIP_PORT in these
- comparisions, since users may change that, but the protocol
- clearly states that if we DO NOT mention a port it is 5060.
- DEFAULT_SIP_PORT is whatever we default to listen to. I believe
- it's the third time I revert a patch like this.
-
-2006-10-07 14:48 +0000 [r44685-44686] Paul Cadach <paul@odt.east.telecom.kz>
-
- * /, channels/h323/ast_h323.cxx, channels/chan_h323.c,
- channels/h323/ast_h323.h, channels/h323/chan_h323.h: Merged
- revisions 44684 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44684 | pcadach | 2006-10-07 20:39:34 +0600 (Сбт, 07 Окт 2006) |
- 1 line Propagate caller's transfer capability too ........
-
- * include/asterisk/callerid.h, main/callerid.c, CHANGES,
- funcs/func_callerid.c: Extend CALLERID() function for "pres" and
- "ton" values
-
-2006-10-07 12:50 +0000 [r44641-44675] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_sip.c: slightly restructure the code that computes
- the channel's name
-
- * channels/chan_sip.c: put repeated code to set nat mode in a
- function.
-
- * channels/chan_sip.c: put common code in a function to avoid
- repetitions.
-
- * channels/chan_sip.c: remove hardwired usage of 5060, use
- DEFAULT_SIP_PORT instead
-
- * channels/chan_sip.c: improve and document function
- get_in_brackets(), introducing a helper function
- find_closing_quote() of more general use.
-
- * channels/chan_sip.c: when possible, use ast_set2_flags instead of
- ast_set/ast_clr . Also mark XXX some dubious places.
-
-2006-10-06 21:29 +0000 [r44632] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, include/asterisk/linkedlists.h: Merged revisions 44631 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44631 | kpfleming | 2006-10-06 16:28:03 -0500 (Fri, 06 Oct 2006)
- | 2 lines ensure that mutex locks inside list heads are
- initialized properly on platforms that require constructor
- initialization (issue #8029, patch from timrobbins) ........
-
-2006-10-06 21:10 +0000 [r44630] Joshua Colp <jcolp@digium.com>
-
- * /, main/rtp.c: Merged revisions 44628 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44628 | file | 2006-10-06 17:08:54 -0400 (Fri, 06 Oct 2006) | 2
- lines Remove the seqno check for RFC2833, the handler is smart
- enough to not need it. ........
-
-2006-10-06 21:04 +0000 [r44616-44626] Luigi Rizzo <rizzo@icir.org>
-
- * main/manager.c: basically fix indentation of a large function
- after previous simplifications. On passing, use a single exit
- point. (once done with the cleanup i will merge the changes into
- 1.4, if applicable)
-
- * main/manager.c: s cannot be null here, so remove the useless test
- and error-handling block.
-
- * main/manager.c: simplify logic in preparation to reduce
- indentation
-
-2006-10-06 18:47 +0000 [r44606] Joshua Colp <jcolp@digium.com>
-
- * /, main/rtp.c: Merged revisions 44605 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44605 | file | 2006-10-06 14:46:28 -0400 (Fri, 06 Oct 2006) | 2
- lines When the sequence number rolls over then reset the recorded
- sequence number for DTMF (issue #8106 reported by bungalow)
- ........
-
-2006-10-06 17:27 +0000 [r44595] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_rpt.c: Massive cleanup of the rpt code, updating to
- current coding guidelines
-
-2006-10-06 16:56 +0000 [r44582] Joshua Colp <jcolp@digium.com>
-
- * /, main/file.c: Merged revisions 44581 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r44581 | file | 2006-10-06 12:53:48 -0400 (Fri,
- 06 Oct 2006) | 10 lines Merged revisions 44580 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r44580 | file | 2006-10-06 12:52:14 -0400 (Fri, 06 Oct 2006) | 2
- lines Even more frames to treat as though the remote side
- disappeared (issue #8097 reported by eldadran) ........
- ................
-
-2006-10-06 16:43 +0000 [r44566-44579] Luigi Rizzo <rizzo@icir.org>
-
- * configs/sip.conf.sample: document a bit the use of templates.
- They are highly convenient for writing configuration files,
- especially if you have many similar entries, or want to switch
- quickly between different configurations without having to
- comment/uncomment large sections of the files.
-
- * configs/sip.conf.sample: document the "contact" option a bit
- better.
-
- * res/res_limit.c: help old bsd-system which don't have RLIMIT_AS
- and use RLIMIT_VMEM for virtual memory limits.
-
- * main/manager.c, main/http.c: make sure sockets are blocking when
- they should be blocking.
-
- * channels/chan_sip.c, configs/sip.conf.sample: Two things: 1.
- slightly rearrange/simplify the parsing of the argument in
- sip_register. This brings in a patch that has been in Mantis
- (5834) for ages, and is the larger part of the commit; 2.
- implement the "contact" option for peers, similar to the one in
- users.conf: If you put a "contact" option with a non-empty
- argument (e.g. contact=123) in a peer section, asterisk will
- register with the provider as if you had a register=
- username:secret@host/contact line in the general section. The
- latter is a very small is a new feature so i am not putting it in
- the 1.4 branch, although the "contact" option in user.conf is
- already in the 1.4 branch and so it wouldn't be too strange to
- merge it. Note that the implementation of "contact" is much
- simpler than the one in 5834, and limited to a few lines in
- build_peer().
-
-2006-10-06 09:01 +0000 [r44545] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Remove deprecated "incominglimit" config
- option
-
-2006-10-06 06:43 +0000 [r44537] Luigi Rizzo <rizzo@icir.org>
-
- * configs/sip.conf.sample: update example commands to match current
- syntax (does not apply to 1.4)
-
-2006-10-06 02:24 +0000 [r44527] Russell Bryant <russell@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in: regenerate the
- configure script to reflect the latest changes done by Luigi
- Rizzo
-
-2006-10-05 20:13 +0000 [r44503-44516] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c: Fix indenting a bit (issue #8082 reported
- by selsky)
-
- * /, main/file.c: Merged revisions 44502 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r44502 | file | 2006-10-05 15:57:16 -0400 (Thu,
- 05 Oct 2006) | 10 lines Merged revisions 44501 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r44501 | file | 2006-10-05 15:55:41 -0400 (Thu, 05 Oct 2006) | 2
- lines Treat busy control frames as hangup in the file streaming
- core (issue #8097 reported by eldadran) ........ ................
-
-2006-10-05 18:29 +0000 [r44489] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_ael.c: These mods fix a problem pointed out by dgartang,
- where in certain situations, the target of a goto cannot be
- found, even right under your nose. This is because the current
- context is not updated properly, and rather than waste time and
- find why and where the context should have been updated, I just
- use my newly added 'dad' ptrs, and pop until I have either the
- context or extension, and use that instead.
-
-2006-10-05 18:03 +0000 [r44487] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 44486 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44486 | file | 2006-10-05 14:01:51 -0400 (Thu, 05 Oct 2006) | 2
- lines One more T.38 fix! Don't leave a reinvite hanging by a
- thread if the other side is already setup with T.38 ........
-
-2006-10-05 16:11 +0000 [r44477] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, main/app.c: Merged revisions 44476 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44476 | kpfleming | 2006-10-05 11:10:01 -0500 (Thu, 05 Oct 2006)
- | 3 lines don't segfault when an argument without a close
- parenthesis is found stop parsing as soon as that situation
- occurs ........
-
-2006-10-05 15:42 +0000 [r44467] Luigi Rizzo <rizzo@icir.org>
-
- * configure.ac, acinclude.m4: Basically, this commit only
- simplifies configure.ac and makes the mechanism more flexible,
- but otherwise should not affect your build even if you regenerate
- the "configure" script. (Most likely you need to run bootstrap.sh
- as you really need to re-run autoheader for reasons that i do not
- completely understand). If you don't regenerate "configure", of
- course you will see no difference. In detail: - restructure the
- check for mandatory modules to remove some redundant code blocks;
- - extend the AST_EXT_LIB_CHECK so that it can used also for
- checking headers; - define the AST_C_DEFINE_CHECK macro to test
- for #defined symbols; - for the two above macros, add a last
- argument that getscopied into HAVE_$1_VERSION so the source can
- adapt to different versions of the same libraries/header/etc -
- document the above; - document a problem that existed before and
- i did not manage to solve: the 'description' argument to
- AC_DEFINE does not substiture shell variables so you will not see
- the actual values in the comments (in autoconfig.h)..
-
-2006-10-05 02:43 +0000 [r44451] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 44450 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44450 | file | 2006-10-04 22:40:40 -0400 (Wed, 04 Oct 2006) | 2
- lines Don't totally bail out if T.38 was negotiated ........
-
-2006-10-05 01:43 +0000 [r44437] Kevin P. Fleming <kpfleming@digium.com>
-
- * utils/Makefile, /: Merged revisions 44436 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44436 | kpfleming | 2006-10-04 20:42:06 -0500 (Wed, 04 Oct 2006)
- | 2 lines this change was correct, the old version is no longer
- needed ........
-
-2006-10-05 01:40 +0000 [r44435] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, apps/app_read.c, apps/app_waitforring.c, CHANGES,
- apps/app_speech_utils.c: As per ToDo list, I have made it so that
- Wait(), WaitExten(), Congestion(), Busy(), Read(), WaitForRing(),
- will now either actually handle a floating point argument as
- advertised, or has been upgraded to accept a floating point
- [timeout] arg.
-
-2006-10-05 01:30 +0000 [r44434] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 44433 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r44433 | kpfleming | 2006-10-04 20:30:05 -0500
- (Wed, 04 Oct 2006) | 10 lines Merged revisions 44432 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r44432 | kpfleming | 2006-10-04 20:27:57 -0500 (Wed, 04
- Oct 2006) | 2 lines fix Polycom presence notification again
- ........ ................
-
-2006-10-04 23:52 +0000 [r44408-44423] Luigi Rizzo <rizzo@icir.org>
-
- * configure.ac: simplify checks for OSS using AST_EXT_LIB_CHECK;
- remove two repeated blocks using better logic.
-
- * acinclude.m4: small formatting fix
-
- * acinclude.m4: when only checking headers, do not set $1_LIB. Also
- PBX_$1=0 is the default so we don't need to set it explicitly.
-
- * acinclude.m4: document, and extend a bit the macro
- AST_EXT_LIB_CHECK so that it can be used in more places in
- configure.ac
-
- * configure.ac: restore proper CPPFLAGS and LDFLAGS for FreeBSD,
- until a better solution is found. Please do not commit the
- regenerated "configure" file yet, as there are some more
- simplifications to be applied to configure.ac and acinclude.m4 in
- the next few days. For the same reason, i am postponing the
- commit to the 1.4 branch until the above changes are complete.
-
- * utils/Makefile: correct libraries for astman, at least so i
- think...
-
- * Makefile: put linker flags in ASTLDFLAGS where they belong
-
-2006-10-04 21:20 +0000 [r44379-44394] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 44393 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r44393 | kpfleming | 2006-10-04 16:17:30 -0500
- (Wed, 04 Oct 2006) | 11 lines Merged revisions 44392 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r44392 | kpfleming | 2006-10-04 16:15:29 -0500 (Wed, 04
- Oct 2006) | 3 lines remove workaround for old Polycom firmware
- SUBSCRIBE requests add workaround for new Polycom firmware
- SUBSCRIBE requests (bug is known to exist in 2.0.1 firmware)
- ........ ................
-
- * include/asterisk.h, /, main/utils.c: Merged revisions 44390 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44390 | kpfleming | 2006-10-04 16:04:21 -0500 (Wed, 04 Oct 2006)
- | 2 lines make LOW_MEMORY builds actually work ........
-
- * include/asterisk/utils.h, main/autoservice.c, main/dnsmgr.c,
- channels/chan_zap.c, res/res_snmp.c, /, apps/app_meetme.c,
- channels/chan_sip.c, main/utils.c, main/devicestate.c,
- res/res_musiconhold.c, res/res_jabber.c, apps/app_queue.c,
- channels/chan_iax2.c, channels/chan_oss.c, main/cdr.c,
- channels/chan_phone.c, main/manager.c, pbx/pbx_spool.c,
- res/res_smdi.c, channels/chan_skinny.c, channels/chan_h323.c,
- main/http.c, channels/chan_alsa.c, pbx/pbx_dundi.c,
- apps/app_mixmonitor.c, main/asterisk.c, channels/chan_mgcp.c:
- Merged revisions 44378 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44378 | kpfleming | 2006-10-04 14:47:22 -0500 (Wed, 04 Oct 2006)
- | 4 lines update thread creation code a bit reduce standard
- thread stack size slightly to allow the pthreads library to
- allocate the stack+data and not overflow a power-of-2 allocation
- in the kernel and waste memory/address space add a new stack size
- for 'background' threads (those that don't handle PBX calls) when
- LOW_MEMORY is defined ........
-
-2006-10-04 19:33 +0000 [r44336-44377] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael-test/ael-test16 (added), pbx/ael/ael.tab.c,
- pbx/ael/ael.y, pbx/pbx_ael.c, pbx/ael/ael-test/ref.ael-test16
- (added), pbx/ael/ael-test/ael-test16/extensions.ael: These
- changes resolve the problems in bug 8090, where there's a crash
- compiling an empty context
-
- * configs/muted.conf.sample: I've been meaning to add some
- explanation about muted... here it is
-
- * configs/manager.conf.sample: CLI reverbification update to this
- config file
-
- * apps/app_macro.c: Added a warning to the documentation for Macro
- in response to bug 7776
-
-2006-10-04 00:26 +0000 [r44323] Kevin P. Fleming <kpfleming@digium.com>
-
- * Makefile, include/asterisk.h, /, main/asterisk.c, main/loader.c,
- main/term.c: Merged revisions 44322 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44322 | kpfleming | 2006-10-03 19:25:44 -0500 (Tue, 03 Oct 2006)
- | 3 lines ensure that local include files are always used avoid a
- duplicate function name (term_init()) ........
-
-2006-10-03 22:36 +0000 [r44313] Matt O'Gorman <mogorman@digium.com>
-
- * /, channels/chan_gtalk.c, res/res_jabber.c: Merged revisions
- 44312 via svnmerge from
- https://svn.digium.com/svn/asterisk/branches/1.4 ........ r44312
- | mogorman | 2006-10-03 17:35:43 -0500 (Tue, 03 Oct 2006) | 2
- lines fix issue with dialing client without resource. ........
-
-2006-10-03 20:19 +0000 [r44299] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 44298 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r44298 | kpfleming | 2006-10-03 15:18:29 -0500
- (Tue, 03 Oct 2006) | 10 lines Merged revisions 44296 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r44296 | kpfleming | 2006-10-03 15:14:13 -0500 (Tue, 03
- Oct 2006) | 2 lines fix a logic error in my previous fix to the
- queue reload code ........ ................
-
-2006-10-03 20:17 +0000 [r44297] Joshua Colp <jcolp@digium.com>
-
- * CHANGES, apps/app_queue.c: Strat becomes Strategy based on
- feedback from two nameless fellows
-
-2006-10-03 18:47 +0000 [r44287] Paul Cadach <paul@odt.east.telecom.kz>
-
- * /, channels/h323/ast_h323.cxx: Merged revisions 44283,44286 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44283 | pcadach | 2006-10-04 00:30:48 +0600 (Срд, 04 Окт 2006) |
- 1 line Fix preparation of type and presentation of calling number
- ........ r44286 | pcadach | 2006-10-04 00:42:20 +0600 (Срд, 04
- Окт 2006) | 1 line Change default presentation indicator to "user
- provided not screened" if octet 3a missed in CallingPartyNumber
- IE ........
-
-2006-10-03 18:37 +0000 [r44273-44285] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 44284 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44284 | file | 2006-10-03 14:35:55 -0400 (Tue, 03 Oct 2006) | 2
- lines Use VideoSupport instead so it is considered a valid XML
- attribute name. (issue #8075 reported by renemendoza) ........
-
- * CHANGES, apps/app_queue.c: Add 'Strat' manager field to
- QueueParams event. (issue #7704 reported by renemendoza)
-
- * main/channel.c, CHANGES: Add Masquerade manager event which trips
- when a masquerade happens (issue #7840 reported by moy)
-
-2006-10-03 16:42 +0000 [r44263] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael-test/ref.ael-test11, pbx/ael/ael-test/ref.ael-test14,
- pbx/ael/ael-test/ref.ael-test15, pbx/ael/ael-test/ref.ael-ntest9,
- pbx/ael/ael-test/ref.ael-ntest10,
- pbx/ael/ael-test/ref.ael-ntest12, pbx/ael/ael-test/ref.ael-test1,
- pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3,
- pbx/pbx_ael.c, pbx/ael/ael-test/ref.ael-test4,
- pbx/ael/ael-test/ref.ael-test5, pbx/ael/ael-test/ref.ael-test6,
- pbx/ael/ael-test/ref.ael-test7, pbx/ael/ael-test/ref.ael-test8,
- pbx/ael/ael-test/ref.ael-vtest13: These changes correspond to the
- changes to app_stack's Gosub() application
-
-2006-10-03 16:15 +0000 [r44262] Joshua Colp <jcolp@digium.com>
-
- * UPGRADE.txt: First entry! Tell people about the callerid changes
- with manager.
-
-2006-10-03 15:53 +0000 [r44253] Matt O'Gorman <mogorman@digium.com>
-
- * main/udptl.c, funcs/func_rand.c, main/say.c, apps/app_record.c,
- apps/app_test.c, funcs/func_strings.c, apps/app_alarmreceiver.c,
- apps/app_ices.c, channels/chan_iax2.c, main/loader.c,
- res/res_smdi.c, channels/chan_skinny.c, apps/app_zapscan.c,
- apps/app_zapras.c, main/http.c, channels/chan_alsa.c,
- apps/app_externalivr.c, cdr/cdr_odbc.c, main/db.c, main/sched.c,
- apps/app_dial.c, main/pbx.c, channels/chan_agent.c,
- apps/app_disa.c, channels/iax2-provision.c,
- apps/app_talkdetect.c, apps/app_db.c, res/res_monitor.c,
- channels/chan_misdn.c, apps/app_zapbarge.c,
- channels/chan_features.c, apps/app_macro.c, apps/app_voicemail.c,
- apps/app_meetme.c, res/res_musiconhold.c, channels/chan_gtalk.c,
- res/res_jabber.c, main/enum.c, cdr/cdr_csv.c, main/channel.c,
- channels/chan_phone.c, apps/app_osplookup.c, main/manager.c,
- apps/app_mp3.c, res/res_agi.c, main/logger.c, main/app.c,
- main/dns.c, channels/chan_mgcp.c, apps/app_nbscat.c,
- res/res_config_pgsql.c, channels/chan_zap.c, funcs/func_db.c,
- channels/chan_sip.c, apps/app_festival.c,
- apps/app_waitforsilence.c, res/res_adsi.c, res/res_crypto.c,
- apps/app_queue.c, main/rtp.c, cdr/cdr_tds.c,
- channels/chan_jingle.c, apps/app_directed_pickup.c, main/file.c,
- pbx/pbx_dundi.c, channels/chan_nbs.c, main/dsp.c: bug #8076 check
- option_debug before printing to debug channel. patch provided in
- bugnote, with minor changes.
-
-2006-10-03 15:50 +0000 [r44252] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_stack.c: Okay, I can't use ast_app_separate_args for
- that... and add some debugging for murf...
-
-2006-10-03 15:48 +0000 [r44250-44251] Luigi Rizzo <rizzo@icir.org>
-
- * configure.ac: comment the fact that autoconf2.59 is ok to process
- this file, but we want to use 2.60 in case the generated
- "configure" file must me committed back to the repository, so we
- keep differences to a minimum.
-
- * bootstrap.sh: simplify this file
-
-2006-10-03 00:07 +0000 [r44241] Matt O'Gorman <mogorman@digium.com>
-
- * include/asterisk/jabber.h, res/res_jabber.c: 44240 same as but
- without the removing of chan_jingle and such, as I hope to finish
- jingle support for 1.6
-
-2006-10-02 22:02 +0000 [r44231] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_stack.c: Use the standard parsing routines
-
-2006-10-02 20:58 +0000 [r44200-44218] Joshua Colp <jcolp@digium.com>
-
- * configs/queues.conf.sample, doc/channelvariables.txt, CHANGES,
- apps/app_queue.c: Expand setinterfacevar option to also set a
- variable, MEMBERNAME, which contains the member's name. (issue
- #8046 reported by jmls)
-
- * apps/app_dial.c, main/channel.c, apps/app_meetme.c,
- res/res_features.c, apps/app_dumpchan.c, CHANGES,
- apps/app_queue.c: Make callerid fields in Manager events more
- consistent. CallerIDNum for number and CallerIDName for name.
- (issue #7976 reported by suhler)
-
- * /, channels/chan_sip.c: Merged revisions 44215 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r44215 | file | 2006-10-02 16:11:02 -0400 (Mon,
- 02 Oct 2006) | 10 lines Merged revisions 44213 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r44213 | file | 2006-10-02 16:07:59 -0400 (Mon, 02 Oct 2006) | 2
- lines Change the fd on the I/O context in case it changed during
- the reload, which is indeed possible. (issue #7943 reported by
- eclubb) ........ ................
-
- * /, contrib/init.d/rc.redhat.asterisk: Merged revisions 44199 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r44199 | file | 2006-10-02 15:41:39 -0400 (Mon,
- 02 Oct 2006) | 10 lines Merged revisions 44198 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r44198 | file | 2006-10-02 15:39:59 -0400 (Mon, 02 Oct 2006) | 2
- lines We should be using $AST_SBIN instead of hardcoding the path
- for the error message (issue #7942 reported by eclubb) ........
- ................
-
-2006-10-02 19:01 +0000 [r44187-44196] Paul Cadach <paul@odt.east.telecom.kz>
-
- * configs/users.conf.sample, /, pbx/pbx_config.c: Merged revisions
- 44186 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44186 | pcadach | 2006-10-03 00:52:56 +0600 (Втр, 03 Окт 2006) |
- 1 line Missed part of userconf functionality for chan_h323
- ........
-
- * /, doc/realtime.txt: Merged revisions 44167 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44167 | pcadach | 2006-10-02 23:16:37 +0600 (Пнд, 02 Окт 2006) |
- 1 line Typo fix ........
-
- * /, channels/chan_h323.c: Merged revisions 44166 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44166 | pcadach | 2006-10-02 23:15:11 +0600 (Пнд, 02 Окт 2006) |
- 1 line Optimization of oh323_indicate(): less locks - less
- problems, plus single exit point ........
-
-2006-10-02 17:54 +0000 [r44153-44172] Joshua Colp <jcolp@digium.com>
-
- * main/logger.c, CHANGES, configs/logger.conf.sample: Add option to
- logger to rename log files with timestamp (issue #8020 reported
- by jmls)
-
- * /, main/io.c: Merged revisions 44169 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r44169 | file | 2006-10-02 13:25:13 -0400 (Mon,
- 02 Oct 2006) | 10 lines Merged revisions 44168 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r44168 | file | 2006-10-02 13:22:27 -0400 (Mon, 02 Oct 2006) | 2
- lines Shrink when current_ioc is unused. It is set to -1 when
- unused, not 0. (issue #7941 reported by eclubb) ........
- ................
-
- * res/res_monitor.c: Get rid of the IS_NULL_STRING macro and use
- ast_strlen_zero instead (issue #8070 reported by wrmem)
-
-2006-10-02 16:00 +0000 [r44152] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/asterisk.c: clean up formatting and conformance to code
- guidelines revert Mark's change that caused a memory leak
- (cap_set_proc() does not free the capability structure so we
- always need to call cap_free())
-
-2006-10-02 15:40 +0000 [r44150] Joshua Colp <jcolp@digium.com>
-
- * configs/queues.conf.sample, CHANGES, apps/app_queue.c: Add option
- 'keepstats' which will keep queue statistics during a reload.
- (issue #7908 reported by jmls)
-
-2006-10-02 04:17 +0000 [r44148] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_stack.c: It makes more sense that in GosubIf that the
- two labels might have different arguments.
-
-2006-10-02 02:38 +0000 [r44145-44147] Mark Spencer <markster@digium.com>
-
- * channels/chan_sip.c, channels/chan_iax2.c: Don't use channel when
- you don't mean a channel
-
- * main/asterisk.c: Uhm yah, not sure who committed this into
- trunk... Anyway, I think this is what was intended...
-
-2006-10-01 19:40 +0000 [r44136] Paul Cadach <paul@odt.east.telecom.kz>
-
- * /, channels/chan_h323.c: Merged revisions 44135 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44135 | pcadach | 2006-10-02 01:32:24 +0600 (Пнд, 02 Окт 2006) |
- 1 line Do not simulate any audio tones if we got PROGRESS message
- ........
-
-2006-10-01 18:30 +0000 [r44112-44126] Russell Bryant <russell@digium.com>
-
- * Makefile, /: Merged revisions 44125 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44125 | russell | 2006-10-01 14:30:06 -0400 (Sun, 01 Oct 2006) |
- 6 lines Fix a problem that cuased AST_DATA_DIR in defaults.h to
- be empty. The cause is that since ASTDATADIR is explicitly
- exported using "export ASTDATADIR" at the top of the Makefile,
- make no longer considers the variable "undefined", so the
- Makefile can't use ?= to set ASTDATADIR if not yet set. (issue
- #8063, reported by akohlsmith, fixed by me) ........
-
- * /, configs/queues.conf.sample: Merged revisions 44111 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r44111 | russell | 2006-10-01 11:20:12 -0400
- (Sun, 01 Oct 2006) | 11 lines Merged revisions 44110 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r44110 | russell | 2006-10-01 11:19:23 -0400 (Sun, 01
- Oct 2006) | 3 lines Fix the name of the "eventmemberstatus"
- option in the sample queues.conf (issue #8065, adamg) ........
- ................
-
-2006-10-01 05:37 +0000 [r44100] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_zapateller.c: Janitor for Zapateller: convert to use
- argument macros
-
-2006-09-30 19:23 +0000 [r44091] Paul Cadach <paul@odt.east.telecom.kz>
-
- * /, main/rtp.c: Merged revisions 44090 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44090 | pcadach | 2006-10-01 01:20:38 +0600 (Вск, 01 Окт 2006) |
- 1 line Allow one-way RTP streams (device->Asterisk) ........
-
-2006-09-30 16:37 +0000 [r44081] Luigi Rizzo <rizzo@icir.org>
-
- * Makefile, main/Makefile, codecs/lpc10/Makefile: merge compile
- fixes from 44080: - with AST_DEVMODE, building codecs/lpc10 fails
- because of lots of warnings, and the configure step in editline
- fails as well. Fix this by removing the -Werror in these steps. -
- on FreeBSD (but probably on other platforms as well), the final
- link of asterisk fails because AST_LIBS was not exported to the
- subdirs Makefiles. Add a proper fix in the top-level Makefile (a
- possible alternative way is to add "export AST_LIBS" near the
- beginning of the file). With this fix, i believe that some of the
- platform-specific conditionals in main/Makefile are redundant
- (because they should be already dealt with in the top level
- Makefile) but i don't have a platform to check.
-
-2006-09-30 16:15 +0000 [r44069-44079] Paul Cadach <paul@odt.east.telecom.kz>
-
- * /, channels/chan_sip.c: Merged revisions 44078 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44078 | pcadach | 2006-09-30 22:12:23 +0600 (Сбт, 30 Сен 2006) |
- 6 lines Fix issue #7928 correctly. Next is a comment of previous
- fix: Issue #7928 - Don't send both 404 and 503. Fix by phsultan
- with a small fix by me, myself or I. Thanks, Philippe! (This was
- caused by my changes to the transaction handling) ........
-
- * /, channels/chan_sip.c: Merged revisions 44068 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44068 | pcadach | 2006-09-30 10:37:39 +0600 (Сбт, 30 Сен 2006) |
- 14 lines Found some buggy SIP clients (phones Planet VIP-153T
- firmware 1.0, Linksys PAP2 firmware 3.1.9(LSc)) which sends ACK
- not on OK message only (when remote party answers) but on RINGING
- message too, so when we send 200 OK message, we get unidentified
- ACK message (because INVITE acknowledged on RINGING message
- already), so 200 OK retransmits within its retransmission
- interval then call gets dropped. If someone else knows how to
- provide workaround for such cases, please, fix it in correct way.
- Thanks to ssh from #asteriskru for provide access to his box to
- study and fix this case. ........
-
-2006-09-29 22:52 +0000 [r44056-44058] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, agi, utils: Merged revisions 44057 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44057 | kpfleming | 2006-09-29 17:51:53 -0500 (Fri, 29 Sep 2006)
- | 2 lines ignore temporary files made by the Makefiles during a
- build ........
-
- * /, codecs/gsm/Makefile, configure.ac, Makefile.moddir_rules,
- Makefile.rules, pbx/Makefile, res/Makefile, channels/Makefile,
- agi/Makefile, codecs/Makefile, utils/Makefile, configure,
- build_tools/embed_modules.xml, codecs/ilbc/Makefile,
- codecs/lpc10/Makefile, main/db1-ast/Makefile: Merged revisions
- 44055 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44055 | kpfleming | 2006-09-29 17:47:40 -0500 (Fri, 29 Sep 2006)
- | 2 lines fix a few build system bugs, and convert Makefiles to
- be compatible with GNU make 3.80 ........
-
-2006-09-29 22:36 +0000 [r44054] Jason Parker <jparker@digium.com>
-
- * /, main/asterisk.c, main/cli.c: Merged revisions 44053 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44053 | qwell | 2006-09-29 15:35:09 -0700 (Fri, 29 Sep 2006) | 3
- lines Fix a bug with the removal of 'atleast' argument to 'core
- verbose' and 'core debug'. Add that argument back in. ........
-
-2006-09-29 21:13 +0000 [r44044] Paul Cadach <paul@odt.east.telecom.kz>
-
- * /, channels/h323/ast_h323.cxx: Merged revisions 44034,44042-44043
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44034 | pcadach | 2006-09-30 02:43:13 +0600 (Сбт, 30 Сен 2006) |
- 1 line Fake display name by called number on incoming calls
- (until passing connected number/connected name is not
- implemented) ........ r44042 | pcadach | 2006-09-30 03:05:43
- +0600 (Сбт, 30 Сен 2006) | 1 line Set TON/PRESENTATION
- information more carefully when no CallingNumber IE available
- ........ r44043 | pcadach | 2006-09-30 03:09:10 +0600 (Сбт, 30
- Сен 2006) | 1 line Compile first, please ........
-
-2006-09-29 20:16 +0000 [r44033] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: Remove locking conflict
-
-2006-09-29 19:16 +0000 [r44024-44025] Paul Cadach <paul@odt.east.telecom.kz>
-
- * /, channels/h323/ast_h323.cxx, channels/h323/ast_h323.h: Merged
- revisions 44022 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44022 | pcadach | 2006-09-30 01:06:55 +0600 (Сбт, 30 Сен 2006) |
- 3 lines Properly pass TON/PRESENTATION information - original
- H323Connection::SendSignalSetup() destroys Q.931 fields. ........
-
-2006-09-29 18:54 +0000 [r44013] Kevin P. Fleming <kpfleming@digium.com>
-
- * Makefile, codecs/Makefile, utils/Makefile, /, configure,
- include/asterisk/autoconfig.h.in, main/Makefile,
- codecs/gsm/Makefile, configure.ac, Makefile.moddir_rules,
- Makefile.rules, pbx/Makefile, channels/Makefile,
- main/db1-ast/Makefile: Merged revisions
- 43996-43997,44008,44011-44012 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r43996 | kpfleming | 2006-09-29 11:47:05 -0500 (Fri, 29 Sep 2006)
- | 2 lines another cross-compile fix ........ r43997 | kpfleming |
- 2006-09-29 11:52:27 -0500 (Fri, 29 Sep 2006) | 2 lines support
- --without-curl in configure script ........ r44008 | kpfleming |
- 2006-09-29 13:25:49 -0500 (Fri, 29 Sep 2006) | 2 lines don't
- abuse CFLAGS and LDFLAGS for build of Asterisk components,
- because they are also then used for non-Asterisk components (like
- menuselect); use our own variables instead ........ r44011 |
- kpfleming | 2006-09-29 13:40:17 -0500 (Fri, 29 Sep 2006) | 2
- lines missed one conversion to ASTCFLAGS ........ r44012 |
- kpfleming | 2006-09-29 13:49:07 -0500 (Fri, 29 Sep 2006) | 2
- lines yet another place where we were not using the correct
- CFLAGS by default ........
-
-2006-09-29 18:35 +0000 [r44010] Paul Cadach <paul@odt.east.telecom.kz>
-
- * /, channels/h323/ast_h323.cxx, channels/chan_h323.c,
- channels/h323/ast_h323.h, channels/h323/chan_h323.h: Merged
- revisions 44009 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r44009 | pcadach | 2006-09-30 00:30:34 +0600 (Сбт, 30 Сен 2006) |
- 1 line Pass TON/PRESENTATION information too ........
-
-2006-09-29 16:38 +0000 [r43979-43994] Kevin P. Fleming <kpfleming@digium.com>
-
- * Makefile, /: Merged revisions 43993 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r43993 | kpfleming | 2006-09-29 11:38:27 -0500 (Fri, 29 Sep 2006)
- | 2 lines a couple more environment settings that can't leak into
- the menuselect build ........
-
- * /, main/cli.c: Merged revisions 43978 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r43978 | kpfleming | 2006-09-29 08:45:40 -0500 (Fri, 29 Sep 2006)
- | 2 lines proper fix for ast_group_t change ........
-
-2006-09-29 01:36 +0000 [r43954-43961] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: One must remember to unlock their list...
- thanks to Qwell for letting me into his box
-
- * main/pbx.c: Cache the application pointer so we don't have to
- needlessly search for it over and over. This should yield a
- suitable performance increase.
-
-2006-09-28 22:43 +0000 [r43953] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, include/asterisk/lock.h: Merged revisions 43952 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r43952 | kpfleming | 2006-09-28 17:42:18 -0500 (Thu, 28
- Sep 2006) | 2 lines eliminate compiler warning when
- DEBUG_CHANNEL_LOCKS is enabled and users of this header file
- don't also include channel.h ........
-
-2006-09-28 20:13 +0000 [r43945] Jason Parker <jparker@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 43944 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r43944 | qwell | 2006-09-28 13:11:22 -0700 (Thu, 28 Sep 2006) | 4
- lines Fix incorrect argument order for member names, on persisted
- members. Issue 8047, patch by jmls. ........
-
-2006-09-28 18:09 +0000 [r43934] Joshua Colp <jcolp@digium.com>
-
- * main/udptl.c, main/frame.c, /, channels/chan_sip.c,
- funcs/func_timeout.c, apps/app_festival.c,
- apps/app_alarmreceiver.c, channels/iax2-provision.c,
- res/res_musiconhold.c, apps/app_followme.c, channels/chan_iax2.c,
- res/res_monitor.c, apps/app_playback.c,
- include/asterisk/logger.h, res/res_smdi.c, channels/chan_misdn.c,
- channels/chan_skinny.c, apps/app_rpt.c, channels/chan_mgcp.c:
- Merged revisions 43933 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r43933 | file | 2006-09-28 14:05:43 -0400 (Thu, 28 Sep 2006) | 2
- lines Put in missing \ns on the end of ast_logs (issue #7936
- reported by wojtekka) ........
-
-2006-09-28 17:38 +0000 [r43921] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 43919 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r43919 | kpfleming | 2006-09-28 12:35:42 -0500
- (Thu, 28 Sep 2006) | 10 lines Merged revisions 43916 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r43916 | kpfleming | 2006-09-28 12:31:57 -0500 (Thu, 28
- Sep 2006) | 2 lines fix buggy (and overly complex) loop used
- during reload of app_queue for static member list updating
- ........ ................
-
-2006-09-28 17:36 +0000 [r43920] Paul Cadach <paul@odt.east.telecom.kz>
-
- * /, channels/h323/ast_h323.cxx: Merged revisions 43918 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r43918 | pcadach | 2006-09-28 23:34:19 +0600 (Чтв, 28 Сен 2006) |
- 1 line Extend call establishment timeout ........
-
-2006-09-28 17:32 +0000 [r43912-43917] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 43915 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r43915 | file | 2006-09-28 13:31:09 -0400 (Thu, 28 Sep 2006) | 2
- lines Make sure the pvt exists before accessing it again as it
- may have gone away (issue #7562 reported by Seb7 and issue #7939
- reported by sorg) ........
-
- * /, main/cli.c: Merged revisions 43913 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r43913 | file | 2006-09-28 13:14:07 -0400 (Thu, 28 Sep 2006) | 2
- lines Warning be gone! ........
-
- * channels/chan_sip.c: Add jitterbuffer information to sip list
- settings (issue #7945 reported by sergee)
-
-2006-09-28 16:54 +0000 [r43902] BJ Weschke <bweschke@btwtech.com>
-
- * /, apps/app_queue.c: Merged revisions 43899 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r43899 | bweschke | 2006-09-28 12:41:05 -0400
- (Thu, 28 Sep 2006) | 11 lines Merged revisions 43897 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r43897 | bweschke | 2006-09-28 12:37:15 -0400 (Thu, 28
- Sep 2006) | 3 lines app_queue is comparing the device names
- incorrectly while checking their statuses. It's internal list of
- interfaces includes the dial string, while the argument passed to
- this function does not have the dial string (/n for a local
- channel). This causes it to ignore the device state changes
- because it thinks it belongs to none of its members. (#8040
- reported and patch by tim_ringenbach) ........ ................
-
-2006-09-28 16:43 +0000 [r43900] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, main/cli.c: Merged revisions 43898 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r43898 | kpfleming | 2006-09-28 11:38:25 -0500
- (Thu, 28 Sep 2006) | 10 lines Merged revisions 43895 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r43895 | kpfleming | 2006-09-28 11:32:44 -0500 (Thu, 28
- Sep 2006) | 2 lines eliminate compiler warning introduced by
- recent changes ........ ................
-
-2006-09-28 16:19 +0000 [r43894] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 43893 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r43893 | file | 2006-09-28 12:17:36 -0400 (Thu,
- 28 Sep 2006) | 10 lines Merged revisions 43891 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r43891 | file | 2006-09-28 12:13:55 -0400 (Thu, 28 Sep 2006) | 2
- lines Stop the stream after waitstream returns so that our
- formats get restored. (issue #7370 reported by kryptolus)
- ........ ................
-
-2006-09-28 16:01 +0000 [r43888] Paul Cadach <paul@odt.east.telecom.kz>
-
- * /, channels/h323/ast_h323.cxx: Merged revisions 43877 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r43877 | pcadach | 2006-09-28 21:56:21 +0600 (Чтв, 28 Сен 2006) |
- 1 line Fix compiler warning ........
-
-2006-09-28 15:32 +0000 [r43865-43875] BJ Weschke <bweschke@btwtech.com>
-
- * /, apps/app_queue.c: Merged revisions 43873 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r43873 | bweschke | 2006-09-28 11:29:21 -0400
- (Thu, 28 Sep 2006) | 11 lines Merged revisions 43871 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r43871 | bweschke | 2006-09-28 11:18:05 -0400 (Thu, 28
- Sep 2006) | 3 lines Fix race condion crash with get_member_status
- (#7864 - tim_ringenbach reported and patched) ........
- ................
-
- * /, apps/app_queue.c: Merged revisions 43864 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r43864 | bweschke | 2006-09-28 09:24:10 -0400 (Thu, 28 Sep 2006)
- | 3 lines Autopause not working for queue members. (#8042 - jmls
- reported and patch) ........
-
-2006-09-28 13:02 +0000 [r43863] Paul Cadach <paul@odt.east.telecom.kz>
-
- * /, channels/h323/ast_h323.cxx, channels/h323/ast_h323.h,
- include/asterisk/compiler.h: Merged revisions 43861-43862 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r43861 | pcadach | 2006-09-28 18:47:23 +0600 (Чтв, 28 Сен 2006) |
- 1 line Put attribute tag at correct place ........ r43862 |
- pcadach | 2006-09-28 18:58:22 +0600 (Чтв, 28 Сен 2006) | 1 line
- Force remote side to start media on outgoing PROGRESS message
- ........
-
-2006-09-28 11:32 +0000 [r43854-43855] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
- channels/misdn/isdn_lib.c: Merged revisions 43852 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r43852 | crichter | 2006-09-28 13:03:05 +0200
- (Do, 28 Sep 2006) | 9 lines Merged revisions 43764 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r43764 | crichter | 2006-09-27 14:51:03 +0200 (Mi, 27
- Sep 2006) | 1 line fixed a bug which led to chan_list zombies,
- when the call could not be properly established in misdn_call.
- also removed the ACK_HDLC stuff which is not really needed.
- ........ ................
-
- * channels/chan_misdn.c, /, channels/Makefile: Merged revisions
- 43775 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r43775 | crichter | 2006-09-27 18:24:51 +0200 (Mi, 27 Sep 2006) |
- 1 line removed the chan_misdn versioning, since asterisk has it's
- own ........
-
-2006-09-28 11:12 +0000 [r43845-43853] Paul Cadach <paul@odt.east.telecom.kz>
-
- * channels/h323/cisco-h225.h, /, channels/h323/ast_h323.cxx,
- main/file.c, channels/h323/cisco-h225.asn,
- channels/h323/cisco-h225.cxx: Merged revisions
- 43635,43843-43844,43846 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r43635 | pcadach | 2006-09-26 03:26:12 +0600 (Втр, 26 Сен 2006) |
- 1 line Fix ASN1 description of non-standard Cisco extensions
- ........ r43843 | pcadach | 2006-09-28 12:01:37 +0600 (Чтв, 28
- Сен 2006) | 1 line Don't treat unknown control frames as voice
- ........ r43844 | pcadach | 2006-09-28 12:02:45 +0600 (Чтв, 28
- Сен 2006) | 1 line Don't warn on HOLD/UNHOLD control frames
- ........ r43846 | pcadach | 2006-09-28 16:51:21 +0600 (Чтв, 28
- Сен 2006) | 1 line Do not open transmit channel until TCS is
- received ........
-
- * channels/h323/ast_h323.cxx, channels/chan_h323.c,
- channels/h323/ast_h323.h, CHANGES, channels/h323/chan_h323.h,
- configs/h323.conf.sample: Handle HOLD/RETRIEVE notifications
-
-2006-09-27 22:01 +0000 [r43827-43836] Joshua Colp <jcolp@digium.com>
-
- * CHANGES: Update CHANGES to reflect libcap capability that was
- added.
-
- * configure, main/Makefile, configure.ac, makeopts.in,
- doc/security.txt, main/asterisk.c: Add ability to set high ToS
- bits as non-root on Linux using libcap (issue #7047 reported by
- maddison)
-
- * apps/app_voicemail.c: Finish up last commit
-
- * apps/app_voicemail.c: Do the directory walk dance instead of
- repeated stat calls as it seems to be faster (issue #7507
- reported by Corydon76)
-
-2006-09-27 20:27 +0000 [r43817] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 43816 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r43816 | tilghman | 2006-09-27 15:21:54 -0500
- (Wed, 27 Sep 2006) | 10 lines Merged revisions 43815 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r43815 | tilghman | 2006-09-27 15:20:35 -0500 (Wed, 27
- Sep 2006) | 2 lines Avoid inability to lock directory log message
- by creating the directory ahead of time. (Issue 7631) ........
- ................
-
-2006-09-27 20:03 +0000 [r43804-43814] Jason Parker <jparker@digium.com>
-
- * main/pbx.c: Add BACKGROUNDSTATUS to Background() Issue #7835,
- original patch by bcnit - redone by me.
-
- * main/pbx.c, /, apps/app_playback.c: Merged revisions 43803 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r43803 | qwell | 2006-09-27 12:44:02 -0700 (Wed, 27 Sep 2006) | 4
- lines Fix an issue with PLAYBACKSTATUS not being set under
- certain circumstances. Fix a minor issue, to make it use the
- filenames that were parsed, instead of the entire argument
- string. Fix Background() to return -1 like Playback(), if no args
- are specified. ........
-
-2006-09-27 19:39 +0000 [r43792-43802] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: I *think* this is the last list in
- chan_iax2
-
- * /, main/rtp.c: Merged revisions 43798 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r43798 | file | 2006-09-27 15:10:59 -0400 (Wed, 27 Sep 2006) | 2
- lines Compensate for out of order packets better if RFC2833
- compensation is turned on. ........
-
- * /, channels/chan_iax2.c: Merged revisions 43783 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r43783 | file | 2006-09-27 13:00:31 -0400 (Wed, 27 Sep 2006) | 2
- lines Get rid of two functions from a time now past (we THINK
- these are from pre-recursive lock time) that may be contributing
- to two open issues on the bug tracker (7562/7939) and that has
- the potential to just make bad things happen if the timing is
- right. ........
-
-2006-09-27 17:00 +0000 [r43785] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Fix some little things
-
-2006-09-27 16:57 +0000 [r43780] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /, res/res_features.c: Merged revisions 43779 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r43779 | russell | 2006-09-27 12:55:49 -0400
- (Wed, 27 Sep 2006) | 50 lines Merged revisions 43778 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r43778 | russell | 2006-09-27 12:54:30 -0400 (Wed, 27
- Sep 2006) | 42 lines Fix a problem that occurred if a user
- entered a digit that matched a bridge feature that was configured
- using multiple digits, and the digit that was pressed timed out
- in the feature digit timeout period. For example, if blind
- transfer is configured as '##', and a user presses just '#'. In
- this situation, the call would lock up and no longer pass any
- frames. (issue #7977 reported by festr, and issue #7982 reported
- by michaels and valuable input provided by mneuhauser and kuj.
- Fixed by me, with testing help and peer review from Joshua Colp).
- There are a couple of issues involved in this fix: 1) When
- ast_generic_bridge determines that there has been a timeout, it
- returned AST_BRIDGE_RETRY. Then, when ast_channel_bridge gets
- this result, it calls ast_generic_bridge over again with the same
- timestamp for the next event. This results in an endless loop of
- nothing until the call is terminated. This is resolved by simply
- changing ast_generic_bridge to return AST_BRIDGE_COMPLETE when it
- sees a timeout. 2) I also changed ast_channel_bridge such that if
- in the process of calculating the time until the next event, it
- knows a timeout has already occured, to immediately return
- AST_BRIDGE_COMPLETE instead of attempting to bridge the channels
- anyway. 3) In the process of testing the previous two changes, I
- ran into a problem in res_features where ast_channel_bridge would
- return because it determined that there was a timeout. However,
- ast_bridge_call in res_features would then determine by its own
- calculation that there was still 1 ms before the timeout really
- occurs. It would then proceed, and since the bridge broke out and
- did *not* return a frame, it interpreted this as the call was
- over and hung up the channels. The reason for this was because
- ast_bridge_call in res_features and ast_channel_bridge in
- channel.c were using different times for their calculations.
- channel.c uses the start_time on the bridge config, which is the
- time that the feature digit was recieved. However, res_features
- had another time, 'start', which was set right before calling
- ast_channel_bridge. 'start' will always be slightly after
- start_time in the bridge config, and sometimes enough to round up
- to one ms. This is fixed by making ast_bridge_call use the same
- time as ast_channel_bridge for the timeout calculation. ........
- ................
-
-2006-09-27 16:49 +0000 [r43777] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Add CLI block and unblock circuit commands
- for SS7.
-
-2006-09-27 16:25 +0000 [r43776] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 43774 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r43774 | file | 2006-09-27 12:23:12 -0400 (Wed, 27 Sep 2006) | 2
- lines Make rfc2833compensate a global option. ........
-
-2006-09-27 12:32 +0000 [r43763] Paul Cadach <paul@odt.east.telecom.kz>
-
- * channels/chan_h323.c: Use ast_strdupa() instead of strdup(),
- thanks to sergee
-
-2006-09-27 04:37 +0000 [r43754-43757] Russell Bryant <russell@digium.com>
-
- * apps/app_voicemail.c: remote an unused buffer in mm_login()
- (issue #8038, selsky) In passing, I have cleaned up some
- formatting to better comply with our guidelines. I have also
- changed one place to use S_OR(), and a couple of places to use
- ast_strlen_zero() as appropriate.
-
-2006-09-27 03:45 +0000 [r43740-43747] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
- pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3,
- pbx/pbx_ael.c, pbx/ael/ael-test/ref.ael-test4,
- pbx/ael/ael-test/ref.ael-test5,
- pbx/ael/ael-test/ael-test11/extensions.ael,
- pbx/ael/ael-test/ref.ael-test6, CHANGES,
- pbx/ael/ael-test/ael-test3/extensions.ael,
- pbx/ael/ael-test/ref.ael-test7,
- pbx/ael/ael-test/ael-test5/extensions.ael,
- pbx/ael/ael-test/ref.ael-vtest13: This commits the changes to AEL
- to use the gosub-with-args from Tilghman to perform macro calls.
- This results in substantially smaller stack footprint, which
- allows macro call depths in excess of 100,000 levels, rather than
- the limit of 7 calls deep, which the Macro app is subject to.
-
- * /, configs/extensions.ael.sample: Merged revisions 43739 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r43739 | murf | 2006-09-26 20:32:47 -0600 (Tue, 26 Sep 2006) | 1
- line This change to extensions.ael was to fix bug 8031; the
- install scripts are causing it to be copied to
- /etc/asterisk/extensions.ael, and because it is a fairly direct
- conversion of the original extensions.conf, the macro and context
- names clash with the existing extensions.conf. So, I put an ael-
- in front of all macros and contexts, and checked every goto and
- macro call. Also, this file compiles under aelparse. ........
-
-2006-09-27 01:39 +0000 [r43733] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: Clean up code and convert last two things
- (firmware/dialplan cache) to linked list macros.
-
-2006-09-26 22:18 +0000 [r43721-43727] Jason Parker <jparker@digium.com>
-
- * apps/app_meetme.c: Fire a manager event when a meetme is
- started/stopped. Issue #7891, patch by suhler.
-
- * apps/app_queue.c: Add QueueSummary manager action. Gives "at a
- glance" information about a single queue, or all queues. Issue
- #8035, patch by rgollent, slightly modified (formatting) by me.
-
-2006-09-26 21:01 +0000 [r43715] Russell Bryant <russell@digium.com>
-
- * /, main/asterisk.c: Merged revisions 43710 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r43710 | russell | 2006-09-26 16:56:42 -0400
- (Tue, 26 Sep 2006) | 17 lines (This was actually BE-65) Merged
- revisions 43708 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r43708 | russell | 2006-09-26 16:49:21 -0400 (Tue, 26 Sep 2006) |
- 7 lines Back in revision 4798, this message was changed from
- using ast_cli() to directly calling write(). During this change,
- checking if this was a remote console was removed. This caused
- this message about using "exit" or "quit" to exit an Asterisk
- console to come up in times where it did not make sense. This
- change restores the check to see if this is a remote console
- before printing the message. (fixes BE-4) ........
- ................
-
-2006-09-26 20:51 +0000 [r43709] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c, include/asterisk/channel.h, .cleancount,
- main/cli.c: Merged revisions 43707 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r43707 | file | 2006-09-26 16:47:26 -0400 (Tue,
- 26 Sep 2006) | 10 lines Merged revisions 43705 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r43705 | file | 2006-09-26 16:38:06 -0400 (Tue, 26 Sep 2006) | 2
- lines Use proper type to represent the group variable (issue
- #8025 reported by makoto) ........ ................
-
-2006-09-26 20:30 +0000 [r43702] Jason Parker <jparker@digium.com>
-
- * CHANGES: update CHANGES file to reflect codec support in
- chan_skinny
-
-2006-09-26 20:26 +0000 [r43701] Russell Bryant <russell@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 43700 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r43700 | russell | 2006-09-26 16:24:39 -0400
- (Tue, 26 Sep 2006) | 14 lines Merged revisions 43699 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r43699 | russell | 2006-09-26 16:23:15 -0400 (Tue, 26
- Sep 2006) | 6 lines When parsing the sections of voicemail.conf
- that contain mailbox definitions, don't introduce a length limit
- on the definition by using a 256 byte temporary storage buffer.
- Instead, make the temporary buffer just as big as it needs to be
- to hold the entire mailbox definition. (fixes BE-68) ........
- ................
-
-2006-09-26 20:20 +0000 [r43696-43698] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_local.c, /: Merged revisions 43697 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r43697 | file | 2006-09-26 16:19:33 -0400 (Tue, 26 Sep
- 2006) | 2 lines Strip options off the argument passed for
- devicestate in chan_local. (issue #8034 reported by pcardozo)
- ........
-
- * main/channel.c, /, main/slinfactory.c, apps/app_chanspy.c: Merged
- revisions 43695 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r43695 | file | 2006-09-26 16:09:41 -0400 (Tue, 26 Sep 2006) | 2
- lines Slight overhaul of the whisper support. 1. We need to
- duplicate the frame from ast_translate 2. We need to ensure we
- always have signed linear coming in for signed linear combining.
- 3. We need to ensure we are always feeding signed linear out. 4.
- Properly store and restore write format when beeping on the
- channel we are whispering on. 5. Properly discontinue the stream
- on the channel for the beep. (issue #8019 reported by
- timkelly1980) ........
-
-2006-09-26 19:37 +0000 [r43677-43687] Kevin P. Fleming <kpfleming@digium.com>
-
- * CHANGES: start a CHANGES file for trunk... no need to force
- people to have to review commit logs after branching
-
- * /, sounds/Makefile: Merged revisions 43676 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r43676 | kpfleming | 2006-09-26 13:34:27 -0500 (Tue, 26 Sep 2006)
- | 2 lines update to use 1.4.3 core sounds, with corrected
- beep/beeperr/tt-monkeys files ........
-
-2006-09-26 18:10 +0000 [r43675] Jason Parker <jparker@digium.com>
-
- * main/frame.c, /, doc/rtp-packetization.txt: Merged revisions
- 43674 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r43674 | qwell | 2006-09-26 11:08:51 -0700 (Tue, 26 Sep 2006) | 4
- lines Issue #8015, patch by Dan Austin. Maximum values were
- incorrect, which is why this is being put in 1.4 ........
-
-2006-09-26 17:25 +0000 [r43667] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_stack.c: Gosub arguments (Issue 7780)
-
-2006-09-26 17:09 +0000 [r43666] Jason Parker <jparker@digium.com>
-
- * main/logger.c, configs/logger.conf.sample: Add optional
- queue_log_name config option for logger.conf, to change the name
- of the queue_log file. Issue #7363, patch by Steve Davies,
- slightly modified by me.
-
-2006-09-26 16:56 +0000 [r43658-43659] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: MailboxExists should be a dialplan
- function, not an application (Issue 7503)
-
- * res/res_limit.c: These three are not defined on all platforms
- that we support
-
-2006-09-26 15:35 +0000 [r43651] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_skinny.c: Merged revisions 43650 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r43650 | qwell | 2006-09-26 08:33:47 -0700 (Tue, 26 Sep
- 2006) | 11 lines Add proper codec support to chan_skinny. Works
- with at least ulaw, alaw, and g729a. This is technically a "new
- feature", but there are justifications for it. I found a bug with
- the recent rtp packetization changes, which caused the media
- setup to fail under certain circumstances, particularly when
- using allow=all, or having no allow= statements (globally or on
- the device). I could have either removed the rtp packetization
- features, or I could add proper codec support (which, without, I
- think most people would consider to be a bug anyways). ........
-
-2006-09-25 22:09 +0000 [r43641-43643] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 43642 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r43642 | tilghman | 2006-09-25 17:07:44 -0500 (Mon, 25 Sep 2006)
- | 2 lines Should have moved these lines up in the merge, instead
- of removing them ........
-
- * /, apps/app_voicemail.c: Merged revisions 43640 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r43640 | tilghman | 2006-09-25 17:04:47 -0500
- (Mon, 25 Sep 2006) | 12 lines Merged revisions 43634 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r43634 | tilghman | 2006-09-25 16:14:41 -0500 (Mon, 25
- Sep 2006) | 4 lines Two bugs when forwarding voicemail (Issue
- 7824): 1) delete=yes was ignored 2) maxmessages was ignored
- ........ ................
-
-2006-09-25 20:30 +0000 [r43627] Paul Cadach <paul@odt.east.telecom.kz>
-
- * /: Block revision 43626 from 1.4 tree - already here
-
-2006-09-25 15:24 +0000 [r43617] Jason Parker <jparker@digium.com>
-
- * /, sounds/Makefile: Merged revisions 43616 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r43616 | qwell | 2006-09-25 08:23:31 -0700 (Mon, 25 Sep 2006) | 4
- lines One more fix for sounds installation - this time for
- portability. Reported to asterisk-dev mailing list. ........
-
-2006-09-25 14:49 +0000 [r43604] Steve Murphy <murf@digium.com>
-
- * formats/format_ogg_vorbis.c: This tiny fix prevents asterisk from
- crashing if trying to play an OGG moh file.
-
-2006-09-25 09:03 +0000 [r43571-43597] Paul Cadach <paul@odt.east.telecom.kz>
-
- * channels/h323/caps_h323.cxx, channels/h323/ast_h323.cxx,
- channels/chan_h323.c, channels/h323/ast_h323.h,
- channels/h323/chan_h323.h, configs/h323.conf.sample: Support for
- negotiation and receiption of Cisco's RTP DTMF
-
- * channels/h323/ast_h323.cxx: Disable fastStart if requested by
- remote side
-
- * /: Block revision 43582
-
- * channels/chan_h323.c, configs/h323.conf.sample: Specify RFC2833
- payload on dtmfmode option rather than dtmfcodec option
- (deprecated)
-
- * channels/h323/ast_h323.cxx, channels/chan_h323.c: DTMF mode is
- bitmask, not valued field
-
- * channels/h323/caps_h323.cxx, channels/h323/caps_h323.h: Define
- Cisco RTP capability
-
- * channels/h323/caps_h323.cxx: Specify non-standard data
- independedly on OpenH323's codec name (it can be easily changed)
-
- * channels/chan_h323.c, channels/h323/chan_h323.h: Define DTMF
- payload types
-
-2006-09-24 15:01 +0000 [r43554-43565] Russell Bryant <russell@digium.com>
-
- * /, channels/iax2-provision.c: Merged revisions 43564 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r43564 | russell | 2006-09-24 10:58:10 -0400 (Sun, 24
- Sep 2006) | 5 lines Fix a CLI command registration issue where an
- erroneous message claiming that "iax2 show provisioning" was
- already registered. This was because this command was registering
- itself as both the command, as well as the command it is
- deprecating. (issue #8022, reported by bjweeks, fixed by myself)
- ........
-
- * /, channels/chan_iax2.c: Merged revisions 43553 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r43553 | russell | 2006-09-24 09:53:35 -0400
- (Sun, 24 Sep 2006) | 12 lines Merged revisions 43552 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r43552 | russell | 2006-09-24 09:50:30 -0400 (Sun, 24
- Sep 2006) | 4 lines Check to see if the channel that is
- activating the IAXPEER function is actually an IAX2 channel
- before proceeding to process it to avoid crashing. (issue #8017,
- reported by admott, fixed by myself) ........ ................
-
-2006-09-24 12:15 +0000 [r43539-43546] Paul Cadach <paul@odt.east.telecom.kz>
-
- * main/rtp.c: Small Cisco's RTP DTMF update
-
- * channels/chan_h323.c: Avoid possible deadlock on channel
- destruction
-
- * main/rtp.c: Correct behavior on Cisco's DTMF
-
-2006-09-22 23:46 +0000 [r43525-43526] Kevin P. Fleming <kpfleming@digium.com>
-
- * /: file forgot one :-)
-
- * Makefile, /: Merged revisions 43524 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r43524 | kpfleming | 2006-09-22 18:44:47 -0500 (Fri, 22 Sep 2006)
- | 2 lines don't output the 'build complete' message when the
- target being run is already going to do an installation ........
-
-2006-09-22 23:34 +0000 [r43522] Joshua Colp <jcolp@digium.com>
-
- * /: You see nothing...
-
-2006-09-22 22:13 +0000 [r43519] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_skinny.c: Merged revisions 43518 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r43518 | qwell | 2006-09-22 15:12:12 -0700 (Fri, 22 Sep
- 2006) | 4 lines Allow chan_skinny.so to be unloaded properly.
- Remove reload support, since it doesn't actually...work. ........
-
-2006-09-22 21:34 +0000 [r43506-43507] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_ael.c: This commits a change to return
- MODULE_LOAD_FAILURE on error, and SUCCESS (instead of 0) when all
- goes well for bug 8004
-
- * pbx/pbx_ael.c: As per bug 8004, we now return
- AST_MODULE_LOAD_DECLINE when we can't read extensions.ael
-
-2006-09-22 20:33 +0000 [r43495-43500] Paul Cadach <paul@odt.east.telecom.kz>
-
- * channels/chan_h323.c: Move from h.323 to h323 command prefix
-
- * channels/chan_h323.c: Fix compilation warnings
-
- * channels/h323/compat_h323.h: Use own factory for our
- OpalMediaFormats too
-
- * channels/h323/caps_h323.cxx, channels/h323/compat_h323.h: Fix our
- capability's factory
-
-2006-09-22 17:26 +0000 [r43493] Jason Parker <jparker@digium.com>
-
- * /, main/cli.c: Merged revisions 43492 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r43492 | qwell | 2006-09-22 10:25:05 -0700 (Fri, 22 Sep 2006) | 2
- lines Make sure we explicitly set the CLI command to not be
- deprecated, if it isn't. ........
-
-2006-09-22 16:43 +0000 [r43488-43490] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, sounds/Makefile: Merged revisions 43489 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r43489 | kpfleming | 2006-09-22 11:42:46 -0500 (Fri, 22 Sep 2006)
- | 2 lines use rebuilt extra sounds ........
-
- * main/channel.c, /: Merged revisions 43486 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r43486 | kpfleming | 2006-09-22 10:51:13 -0500 (Fri, 22 Sep 2006)
- | 2 lines all the Linux systems I have don't use '__m_count' for
- this field, so I don't know where this came from... ........
-
-2006-09-22 15:50 +0000 [r43483-43485] Russell Bryant <russell@digium.com>
-
- * channels/chan_misdn.c, /: Merged revisions 43482 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r43482 | russell | 2006-09-22 11:42:44 -0400 (Fri, 22
- Sep 2006) | 3 lines return AST_MODULE_LOAD_DECLIDE if mISDN could
- not be configured (issue #8006, Mithraen) ........
-
-2006-09-22 14:58 +0000 [r43479-43480] Luigi Rizzo <rizzo@icir.org>
-
- * include/asterisk/threadstorage.h: compatibility fix: use
- "attribute_XXX" instead of *__attribute__ ((XXX)) so we can
- handle compiler/os dependencies in our compiler.h
-
- * channels/chan_sip.c: style fix: move variable declaration at the
- beginning of the block.
-
-2006-09-22 14:04 +0000 [r43478] Russell Bryant <russell@digium.com>
-
- * main/frame.c, /: Merged revisions 43477 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r43477 | russell | 2006-09-22 10:02:58 -0400 (Fri, 22 Sep 2006) |
- 3 lines Suppress a compiler warning about the use of a
- potentially uninitialized variable. It couldn't actually happen,
- though. ........
-
-2006-09-22 04:54 +0000 [r43472] Paul Cadach <paul@odt.east.telecom.kz>
-
- * channels/h323/caps_h323.cxx: Add missing include
-
-2006-09-22 03:09 +0000 [r43470] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_skinny.c: Merged revisions 43469 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r43469 | qwell | 2006-09-21 20:01:16 -0700 (Thu, 21 Sep
- 2006) | 4 lines First shot at unload_module in chan_skinny.. More
- to come. ........
-
-2006-09-21 23:55 +0000 [r43467] Matt O'Gorman <mogorman@digium.com>
-
- * /, include/asterisk/jabber.h, channels/chan_gtalk.c,
- res/res_jabber.c: Merged revisions 43466 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r43466 | mogorman | 2006-09-21 18:50:56 -0500 (Thu, 21 Sep 2006)
- | 2 lines updates for better compontent support ........
-
-2006-09-21 23:29 +0000 [r43463-43465] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_odbc.c, configs/res_odbc.conf.sample: Merged revisions
- 43464 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r43464 | tilghman | 2006-09-21 18:24:41 -0500 (Thu, 21 Sep 2006)
- | 2 lines Twould help if we actually documented how the new
- features in res_odbc actually work. (Oops) ........
-
- * res/res_limit.c (added): Set process limits without restarting
- Asterisk
-
-2006-09-21 22:53 +0000 [r43461] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c, channels/chan_iax2.c: Oh look more changes,
- but these are my own! (Clean up module load functions)
-
-2006-09-21 22:44 +0000 [r43460] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c: Suppress compiler warnings
-
-2006-09-21 22:32 +0000 [r43459] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_alsa.c: Clean up chan_alsa load module function
- (issue #8000 reported by Mithraen)
-
-2006-09-21 22:23 +0000 [r43458] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/acl.h, doc/ip-tos.txt, channels/chan_sip.c,
- doc/mp3.txt, doc/ael.txt, doc/channelvariables.txt, main/acl.c:
- And some deprecated APIs and modifications to documentation
-
-2006-09-21 22:23 +0000 [r43455-43457] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_oss.c: Merged revisions 43456 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r43456 | file | 2006-09-21 18:21:40 -0400 (Thu, 21 Sep 2006) | 2
- lines Some more clean up in the load function for chan_oss (issue
- #8002 reported by Mithraen with minor mods by moi) ........
-
- * /, channels/chan_mgcp.c: Merged revisions 43454 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r43454 | file | 2006-09-21 18:12:09 -0400 (Thu, 21 Sep 2006) | 2
- lines Clean up chan_mgcp's module load function (issue #8001
- reported by Mithraen with mods by moi) ........
-
-2006-09-21 21:59 +0000 [r43452] Tilghman Lesher <tlesher@digium.com>
-
- * doc/ip-tos.txt, channels/chan_local.c, channels/chan_sip.c,
- res/res_features.c, channels/chan_agent.c, res/res_convert.c,
- res/res_crypto.c, res/res_musiconhold.c, channels/chan_iax2.c,
- channels/chan_oss.c, channels/chan_skinny.c,
- channels/chan_features.c, res/res_agi.c, channels/chan_h323.c,
- channels/chan_alsa.c, apps/app_settransfercapability.c (removed),
- res/res_indications.c, pbx/pbx_config.c, res/res_odbc.c,
- channels/chan_mgcp.c: Lots more removal of deprecated things
-
-2006-09-21 21:22 +0000 [r43451] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, main/Makefile, build_tools/strip_nonapi (added): Merged
- revisions 43450 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r43450 | kpfleming | 2006-09-21 16:21:29 -0500 (Thu, 21 Sep 2006)
- | 2 lines add another attempt to strip non-API symbols from the
- final binary... script will need to be extended to work on
- non-Linux systems ........
-
-2006-09-21 21:17 +0000 [r43442-43449] Tilghman Lesher <tlesher@digium.com>
-
- * main/udptl.c, main/pbx.c, main/frame.c, main/translate.c,
- apps/app_queue.c, main/config.c, main/rtp.c,
- apps/app_setcdruserfield.c (removed), main/cli.c, main/channel.c,
- main/manager.c, main/file.c, main/http.c, main/logger.c,
- main/astmm.c, main/image.c, main/asterisk.c: Remove deprecated
- CLI apps from the core
-
- * /, apps/app_url.c: Merged revisions 43445 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r43445 | tilghman | 2006-09-21 15:22:43 -0500 (Thu, 21 Sep 2006)
- | 2 lines Fix documentation to reflect how Url() really works
- ........
-
- * apps/app_setcallerid.c, apps/app_voicemail.c: More removal of
- deprecated stuff
-
- * main/pbx.c, main/manager.c, UPGRADE.txt: Remove 1.4 changes from
- UPGRADE.txt, remove deprecated callerid field, remove deprecated
- SetGlobalVar app
-
- * /, apps/app_rpt.c: Merged revisions 43441 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r43441 | tilghman | 2006-09-21 14:43:32 -0500 (Thu, 21 Sep 2006)
- | 2 lines Oops, missed the merge breakage ........
-
-2006-09-21 19:42 +0000 [r43440] Kevin P. Fleming <kpfleming@digium.com>
-
- * makeopts.in: fix this so chan_zap links properly again
-
-2006-09-21 19:35 +0000 [r43439] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_language.c (removed), funcs/func_moh.c (removed),
- apps/app_lookupcidname.c (removed), funcs/func_md5.c,
- apps/app_hasnewvoicemail.c (removed), funcs/func_blacklist.c
- (added), apps/app_random.c (removed), funcs/func_vmcount.c
- (added), res/res_realtime.c (added), apps/app_lookupblacklist.c
- (removed), apps/app_realtime.c (removed), apps/app_queue.c:
- Remove deprecated apps and funcs
-
-2006-09-21 19:27 +0000 [r43437] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c, main/channel.c, /, channels/chan_sip.c,
- include/asterisk/rtp.h, include/asterisk/channel.h, main/rtp.c:
- SS7 marked the start of an open season for trunk again but here's
- something minor - abstract early bridging into the technology so
- that we don't always assume they use RTP and try it.
-
-2006-09-21 19:22 +0000 [r43436] Kevin P. Fleming <kpfleming@digium.com>
-
- * configure: regenerated at PCadach's request
-
-2006-09-21 19:18 +0000 [r43429-43434] Paul Cadach <paul@odt.east.telecom.kz>
-
- * acinclude.m4: Check for 64-bit OpenH323/PWLib versions too,
- thanks to Mithraen (please, re-build configure script)
-
- * channels/h323/caps_h323.cxx: Declare our own media formats to not
- rely on OpenH323 configuration
-
- * channels/h323/caps_h323.cxx, channels/h323/ast_h323.cxx,
- channels/chan_h323.c, channels/h323/caps_h323.h: Introduce Cisco
- G.726-32 capability (g726aal2 form)
-
-2006-09-21 18:42 +0000 [r43427-43428] Matthew Fredrickson <creslin@digium.com>
-
- * configure: Update configure
-
- * channels/chan_zap.c, build_tools/menuselect-deps.in,
- include/asterisk/autoconfig.h.in, configure.ac, makeopts.in,
- configs/zapata.conf.sample: Merge in SS7 changes.... need to
- still cleanup zapata.conf
-
-2006-09-21 17:06 +0000 [r43411-43423] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_rpt.c: Merged revisions 43422 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r43422 | tilghman | 2006-09-21 12:04:40 -0500
- (Thu, 21 Sep 2006) | 10 lines Merged revisions 43420 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r43420 | tilghman | 2006-09-21 12:01:48 -0500 (Thu, 21
- Sep 2006) | 2 lines Whitespace change... really just an excuse to
- test repotools ........ ................
-
- * /: Last merge should not have brought in the 1.2 props
-
- * /, configure, configure.ac, cdr/cdr_tds.c: Merged revisions 43410
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r43410 | tilghman | 2006-09-21 11:31:59 -0500
- (Thu, 21 Sep 2006) | 10 lines Merged revisions 43409 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r43409 | tilghman | 2006-09-21 11:18:19 -0500 (Thu, 21
- Sep 2006) | 2 lines TDS 0.64 updates ........ ................
-
-2006-09-21 16:09 +0000 [r43403-43406] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, main/Makefile: Merged revisions 43405 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r43405 | kpfleming | 2006-09-21 11:08:03 -0500 (Thu, 21 Sep 2006)
- | 2 lines remove this change... it requires binutils 2.17
- ........
-
- * /: remove extraneous property
-
-2006-09-20 23:20 +0000 [r43397] Jason Parker <jparker@digium.com>
-
- * /, build_tools/make_version: Merged revisions 43396 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r43396 | qwell | 2006-09-20 16:19:25 -0700 (Wed, 20 Sep
- 2006) | 2 lines fix minor typo in the way version is handled
- ........
-
-2006-09-20 23:02 +0000 [r43393] Kevin P. Fleming <kpfleming@digium.com>
-
- * /: this has been manually merged
-
-2006-09-20 Kevin P. Fleming <kpfleming@digium.com>
-
- * Asterisk 1.4.0-beta1 released.
-
diff --git a/main/cdr.c b/main/cdr.c
index cabf3fc7a..d66ec767a 100644
--- a/main/cdr.c
+++ b/main/cdr.c
@@ -431,13 +431,6 @@ void ast_cdr_free(struct ast_cdr *cdr)
while (cdr) {
struct ast_cdr *next = cdr->next;
- char *chan = S_OR(cdr->channel, "<unknown>");
- if (!ast_test_flag(cdr, AST_CDR_FLAG_POSTED) && !ast_test_flag(cdr, AST_CDR_FLAG_POST_DISABLED))
- ast_log(LOG_NOTICE, "CDR on channel '%s' not posted\n", chan);
- if (ast_tvzero(cdr->end))
- ast_log(LOG_NOTICE, "CDR on channel '%s' lacks end\n", chan);
- if (ast_tvzero(cdr->start))
- ast_log(LOG_NOTICE, "CDR on channel '%s' lacks start\n", chan);
ast_cdr_free_vars(cdr, 0);
ast_free(cdr);
@@ -1017,10 +1010,6 @@ static void post_cdr(struct ast_cdr *cdr)
chan = S_OR(cdr->channel, "<unknown>");
check_post(cdr);
- if (ast_tvzero(cdr->end))
- ast_log(LOG_WARNING, "CDR on channel '%s' lacks end\n", chan);
- if (ast_tvzero(cdr->start))
- ast_log(LOG_WARNING, "CDR on channel '%s' lacks start\n", chan);
ast_set_flag(cdr, AST_CDR_FLAG_POSTED);
if (ast_test_flag(cdr, AST_CDR_FLAG_POST_DISABLED))
continue;
diff --git a/main/channel.c b/main/channel.c
index fdcdc83d4..6133674cb 100644
--- a/main/channel.c
+++ b/main/channel.c
@@ -1345,6 +1345,11 @@ void ast_channel_free(struct ast_channel *chan)
/* Destroy the jitterbuffer */
ast_jb_destroy(chan);
+
+ if (chan->cdr) {
+ ast_cdr_detach(chan->cdr);
+ chan->cdr = NULL;
+ }
ast_mutex_destroy(&chan->lock_dont_use);
@@ -1657,6 +1662,7 @@ int ast_hangup(struct ast_channel *chan)
ast_cdr_end(chan->cdr);
ast_cdr_detach(chan->cdr);
+ chan->cdr = NULL;
}
ast_channel_free(chan);
diff --git a/main/pbx.c b/main/pbx.c
index 8846a9e72..5eb255014 100644
--- a/main/pbx.c
+++ b/main/pbx.c
@@ -6972,6 +6972,7 @@ static int ast_pbx_outgoing_cdr_failed(void)
ast_cdr_end(chan->cdr);
ast_cdr_failed(chan->cdr); /* set the status to failed */
ast_cdr_detach(chan->cdr); /* post and free the record */
+ chan->cdr = NULL;
ast_channel_free(chan); /* free the channel */
return 0; /* success */
diff --git a/main/poll.c b/main/poll.c
index 731dbcefb..823d0cbd4 100644
--- a/main/poll.c
+++ b/main/poll.c
@@ -268,19 +268,19 @@ int poll
fd_set except_descs; /* exception descs */
struct timeval stime; /* select() timeout value */
int ready_descriptors; /* function result */
- int max_fd; /* maximum fd value */
+ int max_fd = 0; /* maximum fd value */
struct timeval *pTimeout; /* actually passed */
FD_ZERO (&read_descs);
FD_ZERO (&write_descs);
FD_ZERO (&except_descs);
- assert(pArray != (struct pollfd *) NULL);
-
/* Map the poll() file descriptor list in the select() data structures. */
- max_fd = map_poll_spec(pArray, n_fds,
- &read_descs, &write_descs, &except_descs);
+ if (pArray) {
+ max_fd = map_poll_spec (pArray, n_fds,
+ &read_descs, &write_descs, &except_descs);
+ }
/* Map the poll() timeout value in the select() timeout structure. */
pTimeout = map_timeout(timeout, &stime);