mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
20072 Commits (7aa382b53f74db13174cbd5ac6be03bcc8ae1974)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Tilghman Lesher
|
7aa382b53f |
Merged revisions 272562 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r272562 | tilghman | 2010-06-25 15:17:37 -0500 (Fri, 25 Jun 2010) | 5 lines Make the structure of the table specified before match the queries and results. (closes issue #17557) Reported by: cmaj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272568 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Matthew Nicholson
|
480b7e43ca |
Implemement support for handling multiple documents when sending.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272558 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
David Vossel
|
aa428b8967 |
chan_sip: more accurate retransmissions
RFC3261 states that Timer A should start at 500ms (T1) by default. In chan_sip this value initially started at 1000ms and I changed it to 500ms recently. After doing that I noticed in my packet captures that it still occasionally retransmitted starting at 1000ms instead of 500ms like I told it to. This occurs because the scheduler runs in the do_monitor thread. If a new retransmission is added while the do_monitor thread is sleeping then it may not detect that retransmission for nearly 1000ms. To fix this I just poke the do_monitor thread to wake up when a new packet is sent reliably requiring retransmits. The thread then detects the new scheduler entry and adjusts its sleep time to account for it. Review: https://reviewboard.asterisk.org/r/747 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272557 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Tilghman Lesher
|
041932c6c6 |
Symlink sounds files, to save disk space, when multiple tarballs/checkouts are on the same system.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272533 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Richard Mudgett
|
30888f913d |
Merged revisions 272446 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r272446 | rmudgett | 2010-06-24 16:58:49 -0500 (Thu, 24 Jun 2010) | 10 lines ss_thread calls pri_grab without lock during overlap dial Recent changes to chan_dahdi with relation to overlap dialing call pri_grab without first obtaining a lock. (closes issue #17414) Reported by: pdf Patches: bug17414.patch uploaded by jpeeler (license 325) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272447 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Russell Bryant
|
450f4ff2ee |
Resolve some errors produced during module unload of chan_iax2.
The external test suite stops Asterisk using the "core stop gracefully" command.
The logs from the tests show that there are a number of problems with Asterisk
trying to cleanly shut down. This patch addresses the following type of error
that comes from chan_iax2:
[Jun 22 16:58:11] ERROR[29884]: lock.c:129 __ast_pthread_mutex_destroy:
chan_iax2.c line 11371 (iax2_process_thread_cleanup):
Error destroying mutex &thread->lock: Device or resource busy
For an example in the context of a build, see:
http://bamboo.asterisk.org/browse/AST-TRUNK-739/log
The primary purpose of this patch is to change the thread pool shutdown
procedure to be more explicit to ensure that the thread exits from a point
where it is not holding a lock. While testing that, I encountered various
crashes due to the order of operations in unload_module() being problematic.
I reordered some things there, as well.
Review: https://reviewboard.asterisk.org/r/736/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272370 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
15 years ago |
|
Matthew Nicholson
|
cb22af3ec5 |
Merged revisions 272367 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 This version of the patch only adds AgentComplete for attended transfers. It was already present for blind transfers. ........ r272367 | mnicholson | 2010-06-23 17:33:51 -0500 (Wed, 23 Jun 2010) | 8 lines Send AgentComplete manager events in the event of blind and attended transfers. (closes issue #16819) Reported by: elbriga Patches: app_queue.diff uploaded by elbriga (license 482) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272368 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Tilghman Lesher
|
621a86db18 |
If there is realtime configuration, it does not get re-read on reload unless the config file also changes.
(closes issue #16982) Reported by: dmitri Patches: res_musiconhold.patch uploaded by dmitri (license 1001) Tested by: atis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272332 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Tilghman Lesher
|
9ec4987d3b |
Ensure a NULL file while debugging cannot crash AEL.
(closes issue #17215) Reported by: vazir Patches: 20100518__issue17215.diff.txt uploaded by tilghman (license 14) Tested by: tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272260 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Paul Belanger
|
90c850b5b1 |
Fix previous merge. ast_test_flag != ast_test_flag64
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272259 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Paul Belanger
|
affec518d6 |
Merged revisions 272255 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r272255 | pabelanger | 2010-06-23 16:57:01 -0400 (Wed, 23 Jun 2010) | 12 lines First caller into a dynamic conference now enter pin once. If MeetMe is configured to use dynamic conference numbers, then the first caller (which creates the conference) had to enter the PIN number twice. (closes issue #15878) Reported by: shawkris Patches: issue15878.patch uploaded by pabelanger (license 224) Tested by: pabelanger ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272257 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Terry Wilson
|
9328d05d90 |
Update configure when changing autconf m4 files...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272256 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Terry Wilson
|
e7d3a5edfa |
Honor the --with-${library}=path for AST_EXT_TOOL_CHECK
(closes issue #16991) Reported by: pprindeville Patches: with_netsnmp.patch.txt uploaded by twilson (license 396) Tested by: twilson Review: https://reviewboard.asterisk.org/r/739/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272254 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Paul Belanger
|
c9a0c500ae |
Correct manager variable 'EventList' case.
(closes issue #17520) Reported by: kobaz Patches: manager.patch uploaded by kobaz (license 834) Tested by: lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272252 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Paul Belanger
|
f044ccbb16 |
Add localization support for Spanish
(closes issue #17548) Reported by: cjacobsen Patches: say.conf.sample.diff uploaded by cjacobsen (license 1029) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272243 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Tim Ringenbach
|
c6b7eae5e6 |
Add new AMI command LocalOptimizeAway.
This command lets you request a "/n" local channel optimize itself out of the way anyway. Review: https://reviewboard.asterisk.org/r/732/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272218 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Tilghman Lesher
|
48ae8ded89 |
D'oh! Defaultenabled FTL.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272150 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Tilghman Lesher
|
7cd0856d00 |
Recorded merge of revisions 272147 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r272147 | tilghman | 2010-06-23 13:40:28 -0500 (Wed, 23 Jun 2010) | 5 lines Backport part of revision 136715 to fix callerid in voicemail text files (IMAP only). (closes issue #16945) Reported by: mneuhauser ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272148 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Terry Wilson
|
2bcef29e11 |
Don't start the sla thread unless we realy need it
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272146 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Tilghman Lesher
|
cae6fa66ed |
Load all lines from realtime, not just the first one.
(closes issue #17144) Reported by: nahuelgreco Patches: 20100513__issue17144__trunk.diff.txt uploaded by tilghman (license 14) Tested by: tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272145 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Terry Wilson
|
7938510af9 |
Make sure reload updates SLA config
Even if there are no stations or trunks defined, we need to start the sla thread to make sure we get the reload event. Also, when doing a reload we need to remove the existing trunks and stations or they end up hanging around. (closes issue #16818) Reported by: mbonin Patches: sla_reload.patch uploaded by twilson (license 396) Tested by: twilson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272109 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Mark Michelson
|
2c798f321a |
Add extra protection for reinvite glare scenario.
Testing proved that if Asterisk sent a connected line reinvite, and the endpoint to which the reinvite were being sent sent a reinvite, Asterisk would not properly respond with a 491 response. The reason is that on connected line reinvites, we set the dialog's invitestate to INV_CALLING to prevent Asterisk from sending a rapid flurry of connected line reinvites. For other reinvites we do not do this. Because of the current invitestate, when Asterisk received the reinvite, we interpreted this as a spiraled INVITE, and thus did not behave properly. The fix for this is to not enter the loop detection or spiral logic in handle_request_invite if the channel state is currently up. This way, no mid-call reinvites will be misinterpreted, no matter what the nature of the reinvite may have been. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272090 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Russell Bryant
|
746d8e6013 |
Don't try to lock/unlock an uninitialized lock on a dahdi_pri.
This small changes prevents destroy_all_channels() from accessing a lock on an unused dahdi_pri struct, resolving a ton of ERRORs that get spewed out when shutting Asterisk down gracefully. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272052 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
David Vossel
|
3f00e3ff03 |
fixes issue with 'dialplan remove extension blah' segfaulting with tab completion
(closes issue #17440) Reported by: kobaz git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272014 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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David Vossel
|
1509737580 |
ignore CANCEL request after having already received final response to INVITE
RFC 3261 section 9 states that a CANCEL has no effect on a request to a UAS that has already given a final response. This patch checks to make sure there is a pending invite before allowing a CANCEL request to be processed, otherwise it responds to the CANCEL with a "481 Call/Transaction Does Not Exist". Review: https://reviewboard.asterisk.org/r/697/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271977 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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David Vossel
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3a875d8524 |
minor fixes for white/black event filters
This fixes a ref count leak in event filters and checks for a filter container allocation failure during session creation. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271905 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Matthew Nicholson
|
5f45ca4d50 |
Merged revisions 271902 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r271902 | mnicholson | 2010-06-22 12:31:57 -0500 (Tue, 22 Jun 2010) | 8 lines Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is set. This is necessary to keep the ref count correct. (closes issue #16815) Reported by: rain Patches: chan_sip-unref-fix.diff uploaded by rain (license 327) (modified) Tested by: rain ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271903 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Jeff Peeler
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42c24b585a |
Add regular expression filtering for manager events.
This patch as documented in the sample config allows one to optionally apply white, black, or both types of filtering to manager events. The new 'eventfilter' option is set per user. (closes issue #14861) Reported by: fnordian Patches: eventfilter3.patch uploaded by fnordian (license 110), modified by me Review: https://reviewboard.asterisk.org/r/673/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271868 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Russell Bryant
|
294c78a27e |
Resolve some errors that occur on a graceful shutdown.
Don't Finalize() if Initialize() did not succeed. This resulted in an error about trying to Finalize() an invalid handle. Also trim some trailing whitespace while in the area. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271867 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Russell Bryant
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9cc7c55578 |
Change the method of retrieving the Asterisk version string.
Using this method makes it so res_fax doesn't have to be rebuilt on every svn update. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271833 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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David Vossel
|
6d82dbb905 |
fixes attended transfer behavior when both transferee and transferer hung up
If both the transferer and transferee of a attended transfer hangup before the new channel picks up, the new channel should be hung up as well as it has no endpoint to talk to. This mirrors the expected behavior used in 1.4. (closes issue #17444) Reported by: corruptor git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271831 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Matthew Nicholson
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519b5a09e4 |
Updated the CHANGES file documenting the addition of a configurable port in the dundi config file.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271764 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Matthew Nicholson
|
01507f4ab7 |
Merged revisions 271761 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r271761 | mnicholson | 2010-06-22 09:49:36 -0500 (Tue, 22 Jun 2010) | 9 lines Allow users to specify a port for dundi peers. (closes issue #17056) Reported by: klaus3000 Patches: dundi-peerport-patch-trunk.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271762 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Matthew Nicholson
|
9bbeb945e8 |
Merged revisions 271689 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r271689 | mnicholson | 2010-06-22 07:52:27 -0500 (Tue, 22 Jun 2010) | 8 lines Modify chan_sip's packet generation api to automatically calculate the Content-Length. This is done by storing packet content in a buffer until it is actually time to send the packet, at which time the size of the packet is calculated. This change was made to ensure that the Content-Length is always correct. (closes issue #17326) Reported by: kenner Tested by: mnicholson, kenner Review: https://reviewboard.asterisk.org/r/693/ ........ This change also adds an ast_str_copy_string() function (similar to ast_copy_string), that copies one ast_str into another, properly handling embedded nulls. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271690 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Tilghman Lesher
|
e3873889b8 |
Conflict kqueue on OS X, since it doesn't work there yet, anyway.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271657 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
David Vossel
|
d4bbf88e96 |
add speex 16khz sample frame so codec cost can be calculated
(closes issue #17534) Reported by: fabled Patches: speex-wb-sample.diff uploaded by fabled (license 448) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271625 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Jeff Peeler
|
b3281ac725 |
Merged revisions 271552 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r271552 | jpeeler | 2010-06-21 15:37:47 -0500 (Mon, 21 Jun 2010) | 7 lines Do not use sizeof to calculate size of a heap allocated character array. Change left out from 271399. (closes issue #16053) Reported by: diLLec ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271554 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
David Vossel
|
462da0585e |
fixes crash when From header URI is missing "sip:"
(closes issue #17437) Reported by: klaus3000 Patches: sip_crash uploaded by dvossel (license 671) Tested by: klaus3000 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271553 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
David Vossel
|
1a7e1aee5e |
fixes logic error introduced by slin16 sip support
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271551 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Tilghman Lesher
|
63fd368411 |
Add new application for declining counting words in multiple languages.
(closes issue #16869) Reported by: chappell Patches: app_say_counted-20100317.c uploaded by chappell (license 8) Tested by: chappell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271520 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Jeff Peeler
|
54f2dfc91c |
Merged revisions 271399 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r271399 | jpeeler | 2010-06-18 14:28:24 -0500 (Fri, 18 Jun 2010) | 11 lines Fix crash when parsing some heavily nested statements in AEL on reload. Due to the recursion used when compiling AEL in gen_prios, all the stack space was being consumed when parsing some AEL that contained nesting 13 levels deep. Changing a few large buffers to be heap allocated fixed the crash, although I did not test how many more levels can now be safely used. (closes issue #16053) Reported by: diLLec Tested by: jpeeler ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271483 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
David Vossel
|
3f9c6bb3bc |
file.c was truncating audio file formats to the lower 32bits.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271341 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Jeff Peeler
|
0effac1e66 |
Recorded merge of revisions 271335 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r271335 | jpeeler | 2010-06-18 13:33:17 -0500 (Fri, 18 Jun 2010) | 13 lines Eliminate deadlock potential in dahdi_fixup(). (This is a backport of 269307, committed to trunk by rmudgett.) Calling dahdi_indicate() when the channel private lock is already held can cause a deadlock if the PRI lock is needed because dahdi_indicate() will also get the channel private lock. The pri_grab() function assumes that the channel private lock is held once to avoid deadlock. (closes issue #17261) Reported by: aragon ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271336 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
David Vossel
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846050f698 |
fixes some coding guideline issue
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271300 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
David Vossel
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a1fe641a38 |
retransmit response to BYE requests until timer J expires
According to RFC 3261 section 17.2.2, which describes non-INVITE server transaction, when a dialog enters the Completed state it must destroy the dialog after Timer J (T1*64) fires. For a BYE transaction Asterisk terminates the dialog immediately during sip_hangup() when it should be waiting T1*64 ms. This results in some odd behavior. For instance if Asterisk receives a BYE and transmits a 200ok in response, if the endpoint never receives the 200ok it will retransmit the BYE to which Asterisk responds with a "481 Call leg/transaction does not exist" because the dialog is already gone. To resolve this I made a function called sip_scheddestroy_final(). This differs slightly from sip_schedestroy() in that it enables a flag that will prevent the destruction from ever being rescheduled or canceled afterwards. It also prevents the pvt's needdestroy flag from being set which triggers the destruction of the dialog within the do_monitor thread(). By using this function we are guaranteed destruction will not occur until the scheduled time. This allows Asterisk to respond to any possible retransmits for a dialog after we process the initial BYE request for T1*64 ms. Other changes: I removed two instances where sip_cancel_destroy is used right before calling sip_scheddestroy. sip_scheddestroy always calls sip_cancel_destroy before scheduling the new destruction so it is completely unnecessary. Review: https://reviewboard.asterisk.org/r/694/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271262 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
David Vossel
|
ba3d1ad680 |
adds support for slin16 in sip
(closes issue #16153) Reported by: kfister Patches: 16153-1.6.2.0-rc5.patch uploaded by kfister (license 912) slin16.sip.patch.1 uploaded by malcolmd (license 924) Tested by: kfister, malcolmd git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271261 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
David Vossel
|
b00f58da25 |
adds speex 16khz audio support
(closes issue #17501) Reported by: fabled Patches: asterisk-trunk-speex-wideband-v2.patch uploaded by fabled (license 448) Tested by: malcolmd, fabled, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271231 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Jeff Peeler
|
0ef5550742 |
Change expected operation from error to debug message
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271192 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Matthew Nicholson
|
a087394ae4 |
Blocked revisions 271123 via svnmerge
........ r271123 | mnicholson | 2010-06-17 10:11:27 -0500 (Thu, 17 Jun 2010) | 7 lines Set sin_family in ast_get_ip_or_srv() and removed the 'last' member of the ast_dnsmgr_entry struct. (closes issue #15827) Reported by: DennisD Patches: (modified) dnsmgr_15827.patch uploaded by chappell (license 8) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271124 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Paul Belanger
|
531290385c |
option w[(secs)] incorrectly capitalized in xmldoc
(closes issue #17516) Reported by: karlfife git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271089 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |