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${ noResults }
19955 Commits (bfdb973191e4845fb82e90d459a20678bad2a24c)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Paul Belanger
|
7d53dc86d6 |
Notify CLI when modules is loaded / unloaded
(closes issue #17308) Reported by: pabelanger Patches: cli.modules.patch uploaded by pabelanger (license 224) Tested by: pabelanger, russell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262800 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Leif Madsen
|
c17cda109a |
Revert previous WARNING message removal.
Marquis42 suggested a better method of doing what I wanted because I ended up removing the WARNING message for all instances when really I just wanted to remove it for the 'return' keyword, not everything. (issue #17145) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262798 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Leif Madsen
|
881450ec82 |
Remove unnecessary WARNING message in ael/pval.c
(closes issue #17145) Reported by: okrief git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262796 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
David Vossel
|
a0b12a5666 |
Merged revisions 262662 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r262662 | dvossel | 2010-05-12 12:00:04 -0500 (Wed, 12 May 2010) | 11 lines fixes app_meetme dsp error We attempted to detect silence after translating a frame from signed linear. This caused a flooding of errors. To resolve this the code to detect silence was moved before the translation. (closes issue #17133) Reported by: jsdyer ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262744 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Richard Mudgett
|
093dbfdd52 |
Don't crash when destroying chan_dahdi pseudo channels.
Must do a deep copy of the cc_params in duplicate_pseudo(). Otherwise, when the duplicate pseudo channel is destroyed, it frees the original pseudo channel cc_params. The original pseudo channel is then left with a dangling pointer for when the next duplicated pseudo channel is created. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262743 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Richard Mudgett
|
e2336b73ef |
Merged revisions 262657,262660 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r262660 | rmudgett | 2010-05-12 11:46:47 -0500 (Wed, 12 May 2010) | 4 lines Forgot some conditionals around the callrerouting facility help text. JIRA ABE-2223 .......... r262657 | rmudgett | 2010-05-12 11:26:49 -0500 (Wed, 12 May 2010) | 22 lines Add mISDN Call rerouting facility for point-to-point ISDN lines (exchange line) In the case of ISDN point-to-multipoint (multidevice) you can use the mISDN "facility calldeflect" application for call diversions from external (PSTN) to external (PSTN). In that case this is the only way to get rid of the two call legs to the PBX and let the calling number at the C party become the number of the A party. In the case of ISDN point-to-point (exchange line) the call deflection facility may not be used. Instead a call rerouting facility has to be used. This patch for chan_misdn.c is an extension to realize this service (facility rerouting application). It can accept either spelling: "callrerouting" or "callrerouteing". The patch is tested towards Deutsche Telekom and requires a modified version of mISDN from Digium, Inc. Patches: misdn_rerouteing_corrected.patch (Slightly modified.) JIRA ABE-2223 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262661 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Tilghman Lesher
|
1d7a548ae6 |
Ensure the arguments are initialized. Also miscellaneous CG cleanup.
(closes issue #16576) Reported by: uxbod Patches: 20100505__issue16576.diff.txt uploaded by tilghman (license 14) Tested by: uxbod git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262656 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Paul Belanger
|
4b1d9f85a7 |
Convert to AST_CLI_YESNO and AST_CLI_ONOFF
Clean up chan_sip.c to use new AST_CLI functions (closes issue #17287) Reported by: pabelanger Patches: issue17287.patch uploaded by pabelanger (license 224) Tested by: russell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262613 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Richard Mudgett
|
9534f72cb0 |
Dialing an invalid extension causes incomplete hangup sequence.
Revision -r1489 of the libpri 1.4 branch corrected a deviation from Q.931 Section 5.3.2. However, this resulted in an unexpected behaviour change to the upper layer (Asterisk). This change uses pri_hangup_fix_enable() to follow Q.931 Section 5.3.2 call hangup better if the version of libpri supports it. (issue #17104) Reported by: shawkris Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262569 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Tilghman Lesher
|
2c10997e99 |
Move cause 200 to cause 26, as specified in Q.850.
Also cleanup the formatting and add a few more that seem like good candidates. (closes issue #16157) Reported by: wimpy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262513 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Jason Parker
|
d8dea9e76a |
Merged revisions 262421 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r262421 | qwell | 2010-05-11 14:55:42 -0500 (Tue, 11 May 2010) | 11 lines Use a less silly method for modifying a flex-generated file. The sed syntax that was used wasn't actually valid, causing some versions to choke. This is the method that is used in 1.6.x+ for similar changes. (closes issue #16696) Reported by: bklang Patches: 16696-sedfix.diff uploaded by qwell (license 4) Tested by: qwell ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262422 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Paul Belanger
|
663a368a87 |
Improve logging by displaying line number
(closes issue #16303) Reported by: dant Patches: issue16303.patch.v2 uploaded by pabelanger (license 224) Tested by: dant, lmadsen, pabelanger git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262419 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Paul Belanger
|
9c012b460f |
Improve logging information for misconfigured contexts
(closes issue #17238) Reported by: pprindeville Patches: chan_sip-bug17238.patch uploaded by pprindeville (license 347) Tested by: pprindeville git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262414 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Tilghman Lesher
|
c84e7f83c8 |
Merged revisions 262321 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r262321 | tilghman | 2010-05-11 12:22:07 -0500 (Tue, 11 May 2010) | 2 lines Fix issue #17302 a slightly different way (mad props to Qwell) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262330 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Jason Parker
|
344a0f8f7b |
Allow bootstrap script to work on Solaris.
As usual, the way they do things is different, so we need to account for that. automake is versioned ala BSD/Linux, but autoconf is not. We don't actually need to specify a version there, since AC_PREREQ will cover it for us. Things will fail pretty loudly if AC_PREREQ isn't met. (closes issue #16341) Reported by: bklang Patches: opensolaris_bootstrap.sh uploaded by bklang (license 919) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262299 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
David Vossel
|
62067caaab |
fixes PickupChan application
(closes issue #16863) Reported by: schern Patches: app_directed_pickup.c.patch uploaded by schern (license 995) for_trunk.diff uploaded by cjacobsen (license 1029) Tested by: Graber, cjacobsen, lathama, rickead2000, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262240 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
David Vossel
|
351e0e90c5 |
fixes crash in chan_console
There is a race condition between console_hangup() and start_stream(). It is possible for console_hangup() to be called and then the stream thread to begin after the hangup. To avoid this a check in start_stream() to make sure the pvt-owner still exists while the pvt lock is held is made. If the owner is gone that means the channel hung up and start_stream should be aborted. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262236 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Tilghman Lesher
|
618bbdc2ad |
Merged revisions 262151 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r262151 | tilghman | 2010-05-10 11:34:21 -0500 (Mon, 10 May 2010) | 10 lines Allow compilation on Mac OS X 10.4 (Tiger) (closes issue #17297) Reported by: jcovert Patches: 20100506__issue17297.diff.txt uploaded by tilghman (license 14) (closes issue #17302) Reported by: jcovert ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262152 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Tilghman Lesher
|
92a8650677 |
Cleanup a bit more by getting rid of useless version defines. Also make library detection use passed CFLAGS.
(closes issue #17309) Reported by: stuarth git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262102 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Tilghman Lesher
|
49b292babf |
Use CPPFLAGS to pass PTHREAD_CFLAGS for vpb only
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262048 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Alec L Davis
|
dd3343c33d |
VoicemailMain and VMauthenticate, allow escape to the 'a' extension when a single '*' is entered
Where a site uses VoicemailMain(mailbox) the users have to be at their own extension to clear their voicemail, they have no way of escaping VoicemailMain to allow entry of new boxnumber. This patch, allows a site to include to 'a' priority in the VoicemailMain context, to allow an escape. If the 'a' priority doesn't exist in the context that VoicemailMain was called from then it acts as the old behaviour. Reported by: alecdavis Tested by: alecdavis Patch vm_a_extension.diff2.txt uploaded by alecdavis (license 585) Review: https://reviewboard.asterisk.org/r/489/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262005 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Tilghman Lesher
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6a683a1ee8 |
Fix build on Linux
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261964 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Tilghman Lesher
|
03e1608c29 |
Double free crash
(closes issue #17245) Reported by: thedavidfactor Patches: 20100426__issue17245.diff.txt uploaded by tilghman (license 14) Tested by: murraytm git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261917 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Tilghman Lesher
|
13f15cae67 |
Use the detected pthread building flags in every place, instead of hardcoding -lpthread.
We nicely detect the right flags on each system for building Asterisk with pthreads, then ignore it for every other build option that requires us to build with pthreads. This caused some items to return a false negative. Also cleanup some minor naming issues that caused "library library" redundancy in the output. (closes issue #17303) Reported by: stuarth Patches: 20100507__issue17303.diff.txt uploaded by tilghman (license 14) Tested by: stuarth git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261913 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Leif Madsen
|
d73dc3be2d |
Update UPGRADE-1.6.txt stating insecure=very has been removed.
(closes issue #17282) Reported by: stuarth Tested by: stuarth git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261867 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Jeff Peeler
|
356b090875 |
Fix deadlock in sig_pri when hanging up.
The pri_dchannel thread currently violates locking order by locking the private and then attempting to queue a frame, which needs to lock the channel. Queueing a frame is unneccesary though and is actually a regression since sig_pri. All the places that currently use ast_softhangup_nolock now will just set the softhangup value directly as before. (closes issue #17216) Reported by: lmsteffan Patches: bug17216.patch uploaded by jpeeler (license 325) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261866 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Richard Mudgett
|
6f03bf4a42 |
Some code optimizations.
* Made more places use pri_queue_control() instead of pri_queue_frame() and a local frame variable. * Made pri_queue_frame() use sig_pri_lock_owner(). pri_queue_frame() no longer releases the libpri access lock unless it is required. * Made the pri_queue_frame() and pri_queue_control() parameter list similar to sig_pri_lock_owner(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261822 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Jeff Peeler
|
8312f25b13 |
Merged revisions 261735 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r261735 | jpeeler | 2010-05-06 15:10:59 -0500 (Thu, 06 May 2010) | 8 lines Only allow the operator key to be accepted after leaving a voicemail. Or rather disallow the operator key from being accepted when not offered, such as after finishing a recording from within the mailbox options menu. ABE-2121 SWP-1267 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261736 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Jason Parker
|
8d42a80bed |
Merged revisions 261608 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r261608 | qwell | 2010-05-06 11:56:02 -0500 (Thu, 06 May 2010) | 4 lines Use the versioned MOH tarballs, now that we have them. This makes for more reproducibility. Prompted by a discussion in #asterisk-dev ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261609 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Tilghman Lesher
|
ba9b0d95e6 |
Permit more lines within a SIP body to be parsed.
The example given within the related issue showed 120 lines, which was mostly a result of the body being XML. (closes issue #17179) Reported by: khw git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261560 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Russell Bryant
|
be77c44d45 |
Add test case for removing random elements from a heap.
I modified the original patch for trunk to use the unit test API. (issue #17277) Reported by: cappucinoking Patches: test_heap.diff uploaded by cappucinoking (license 1036) Tested by: cappucinoking, russell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261500 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Russell Bryant
|
12631bc3a0 |
Fix handling of removing nodes from the middle of a heap.
This bug surfaced in 1.6.2 and does not affect code in any other released version of Asterisk. It manifested itself as SIP qualify not happening when it should, causing peers to go unreachable. This was debugged down to scheduler entries sometimes not getting executed when they were supposed to, which was in turn caused by an error in the heap code. The problem only sometimes occurs, and it is due to the logic for removing an entry in the heap from an arbitrary location (not just popping off the top). The scheduler performs this operation frequently when entries are removed before they run (when ast_sched_del() is used). In a normal pop off of the top of the heap, a node is taken off the bottom, placed at the top, and then bubbled down until the max heap property is restored (see max_heapify()). This same logic was used for removing an arbitrary node from the middle of the heap. Unfortunately, that logic is full of fail. This patch fixes that by fully restoring the max heap property when a node is thrown into the middle of the heap. Instead of just pushing it down as appropriate, it first pushes it up as high as it will go, and _then_ pushes it down. Lastly, fix a minor problem in ast_heap_verify(), which is only used for debugging. If a parent and child node have the same value, that is not an error. The only error is if a parent's value is less than its children. A huge thanks goes out to cappucinoking for debugging this down to the scheduler, and then producing an ast_heap test case that demonstrated the breakage. That made it very easy for me to focus on the heap logic and produce a fix. Open source projects are awesome. (closes issue #16936) Reported by: ib2 Tested by: cappucinoking, crjw (closes issue #17277) Reported by: cappucinoking Patches: heap-fix.rev2.diff uploaded by russell (license 2) Tested by: cappucinoking, russell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261496 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Tzafrir Cohen
|
6b2e51b1ed |
When failing to configure, don't destroy 'cfg' twice
Fixes a crash when some config section had an incorrect channel config. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261451 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Richard Mudgett
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f0a7adb309 |
Avoid a crash on SS7 channels.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261405 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Russell Bryant
|
865bdbc954 |
Restore previous asterisk.conf syntax, where the directories aren't commented out.
This fixes some breakage in the test suite, that uses the contents of asterisk.conf to discover the install layout on the system. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261364 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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David Vossel
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f16625b7a0 |
fixes sip native transfer
The Refer-To header field containing the Replaces header in the URI was not being decoded properly. This caused invalid parsing between the caller id field and the domain resulting in a failed transfer. (closes issue #17284) Reported by: dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261316 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Paul Belanger
|
35eeb71ead |
Merged revisions 261274 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r261274 | pabelanger | 2010-05-05 12:42:22 -0400 (Wed, 05 May 2010) | 12 lines Registration fix for SIP realtime. Make sure realtime fields are not empty. (closes issue #17266) Reported by: Nick_Lewis Patches: chan_sip.c-realtime.patch uploaded by Nick Lewis (license 657) Tested by: Nick_Lewis, sberney Review: https://reviewboard.asterisk.org/r/643/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261314 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Mark Michelson
|
a6ea125e7c |
Prevent unnecessary warnings when getting rtpsource or rtpdest.
If a recognized media type was present, but the media type was not enabled for the channel, then a warning would be emitted. For instance, attempting to get CHANNEL(rtpsource,video) on a call with no video would cause a warning message to appear. With this change, the warning will only appear if the stream argument is not recognized as being a media type that can be specified. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261313 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Paul Belanger
|
d7ff67179d |
'queue reset stats' erroneously clears wrapuptime configuration.
Resets each member's lastcall to 0 now. (closes issue #17262) Reported by: rain Patches: wrapuptime_reset_fix.diff uploaded by rain (license 327) Tested by: rain git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261232 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Paul Belanger
|
b2f59bea24 |
New 'manager show settings' CLI command.
See the CHANGES file for more details. (closes issue #16343) Reported by: pabelanger Patches: issue16343.patch.v5 uploaded by pabelanger (license 224) Tested by: pabelanger, tilghman, lmadsen Review: https://reviewboard.asterisk.org/r/630/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261180 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Paul Belanger
|
3cea79e5fd |
New static asterisk.conf.sample file.
This simply moves the functionality from the Makefile (cleaning it up) into an external asterisk.conf.samples file. Also updates formatting (easier to read) and grammar changes to asterisk.conf.samples. (closes issue #17027) Reported by: pabelanger Patches: 0017027.asterisk.conf.v6.patch uploaded by pabelanger (license 224) Tested by: qwell, lmadsen, pabelanger, chappell Review: https://reviewboard.asterisk.org/r/616/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261124 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Tilghman Lesher
|
6a0ea1d79e |
Merged revisions 261093-261094 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r261093 | tilghman | 2010-05-04 18:36:53 -0500 (Tue, 04 May 2010) | 7 lines Protect against overflow, when calculating how long to wait for a frame. (closes issue #17128) Reported by: under Patches: d.diff uploaded by under (license 914) ........ r261094 | tilghman | 2010-05-04 18:47:08 -0500 (Tue, 04 May 2010) | 2 lines Add a tiny corner case to the previous commit ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261095 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Mark Michelson
|
fc652b869a |
Add new possible value to autopause option to allow members to be autopaused in all queues.
See the CHANGES file and queues.conf.sample for more details. (closes issue #17008) Reported by: jlpedrosa Patches: queues.autopause_en_review.diff uploaded by jlpedrosa (license 1002) Review: https://reviewboard.asterisk.org/r/581/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261051 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Richard Mudgett
|
159f0d4b24 |
The inalarm flag is not passed up from the sig_analog and sig_pri submodules.
The CLI "dahdi show channel" command was not correctly reporting the InAlarm status. The inalarm flag is now consistently passed between chan_dahdi and submodules. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261007 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Jeff Peeler
|
9db934a869 |
Merged revisions 260923 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260923 | jpeeler | 2010-05-04 13:46:46 -0500 (Tue, 04 May 2010) | 12 lines Voicemail transfer to operator should occur immediately, not after main menu. There were two scenarios in the advanced options that while using the operator=yes and review=yes options, the transfer occurred only after exiting the main menu (after sending a reply or leaving a message for an extension). Now after the audio is processed for the reply or message the transfer occurs immediately as expected. ABE-2107 ABE-2108 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260924 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Jason Parker
|
7eac4cfb03 |
Merged revisions 260801 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260801 | qwell | 2010-05-04 10:49:27 -0500 (Tue, 04 May 2010) | 1 line Fix fallout from removing from configure script. Pointed out by philipp64 on #asterisk-dev ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260802 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Jeff Peeler
|
8ddd92f823 |
Add new admin features to meetme: Roll call, eject all, mute all, record in-conf
This patch adds the following in-conference admin DTMF features: *81 - Roll call (or simply user count if INTROUSER isn't enabled) *82 - Eject all non-admins *83 - Mute/unmute all non-admins *84 - Start recording the conference on the fly FWIW, this code uses newly recorded prompts. (closes issue #16379) Reported by: rfinnie Patches: meetme-enhancements-232771-v1.patch uploaded by rfinnie (license 940) modified slightly by me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260757 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Paul Belanger
|
624c2449ff |
Merged revisions 260661-260662 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260661 | pabelanger | 2010-05-03 12:41:30 -0400 (Mon, 03 May 2010) | 10 lines non-root make install PREFIX=/tmp fails. Prepend libdir when executing mkpkgconfig allowing non-root installs to work. (closes issue #17268) Reported by: pabelanger Patches: issue17268.patch uploaded by pabelanger (license 224) Tested by: pabelanger ........ r260662 | pabelanger | 2010-05-03 12:54:41 -0400 (Mon, 03 May 2010) | 3 lines Should have removed /usr/lib/ part. Thanks Qwell. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260663 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Leif Madsen
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3a1cabb13f |
Merged revisions 260569 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260569 | lmadsen | 2010-05-03 09:57:39 -0500 (Mon, 03 May 2010) | 1 line Minor typo pointed out by pabelanger on IRC. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260570 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Eliel C. Sardanons
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caa2eff30c |
Avoid making AstData depend on libxml2 to compile.
We have some functions inside the AstData API to get the tree in XML form, but it is not required at the moment to compile asterisk and we can disable that part of the API if we don't have libxml2 support. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260521 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |