aboutsummaryrefslogtreecommitdiffstats
path: root/main
AgeCommit message (Collapse)AuthorFilesLines
2010-08-23Merged revisions 283209 via svnmerge from russell1-0/+5
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r283209 | russell | 2010-08-23 08:06:57 -0500 (Mon, 23 Aug 2010) | 2 lines Don't blow up on an invalid AMA flag. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@283210 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-19Merged revisions 282826 via svnmerge from tilghman1-4/+6
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282826 | tilghman | 2010-08-19 09:44:51 -0500 (Thu, 19 Aug 2010) | 2 lines Only output debugging if the debug level is on. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@282827 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-16Merged revisions 282468 via svnmerge from twilson1-0/+10
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r282468 | twilson | 2010-08-16 12:53:44 -0500 (Mon, 16 Aug 2010) | 30 lines Merged revisions 282467 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r282467 | twilson | 2010-08-16 12:32:01 -0500 (Mon, 16 Aug 2010) | 23 lines Merged revisions 282430 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r282430 | twilson | 2010-08-16 12:06:37 -0500 (Mon, 16 Aug 2010) | 16 lines Send a SRCCHANGE indication when we masquerade Masquerading a channel means that the src of the audio is potentially changing, so send a SRCCHANGE so that RTP-based media streams can get a new SSRC generated to reflect the change. Original patch by addix (along with lots of testing--thanks!). (closes issue #17007) Reported by: addix Patches: 1001-reset-SSRC-original-channel.diff uploaded by addix (license 1006) srcchange.diff uploaded by twilson (license 396) Tested by: addix, twilson Review: https://reviewboard.asterisk.org/r/862/ ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@282502 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-15Support for GNU/kFreeBSDtzafrir3-5/+7
kFreeBSD is GNU (with glibc) on to of a FreeBSD kernel. See http://glibc-bsd.alioth.debian.org/porting/PORTING This patch gets Asterisk close to building on Debian kFreeBSD i386, mainly by adding an extra test for __GLIBC__ in one or two (or more) places. OSARCH is set to 'kfreebsd-gnu' DAHDI support (and support for chan_vpb) was not tested. Review: https://reviewboard.asterisk.org/r/858/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@282397 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-12Merged revisions 282098 via svnmerge from rmudgett1-21/+28
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282098 | rmudgett | 2010-08-12 17:06:06 -0500 (Thu, 12 Aug 2010) | 7 lines Separate call completion config parameter allocation and default initialization. If you ever have a need to reset the call completion config parameters to defaults, now you can. And no Virginia, C++ idioms do not always work in C. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@282099 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-12Merged revisions 282066 via svnmerge from russell1-1/+25
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282066 | russell | 2010-08-12 15:41:17 -0500 (Thu, 12 Aug 2010) | 4 lines Add a "core reload" CLI command. Review: https://reviewboard.asterisk.org/r/859/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@282067 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-12Merged revisions 282047 via svnmerge from dvossel2-35/+269
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282047 | dvossel | 2010-08-12 15:15:41 -0500 (Thu, 12 Aug 2010) | 35 lines improved translation paths for wideband codecs The problem I'm addressing is that Asterisk's current method of building the least cost translation paths between codecs does not take into account sample rate. For instance, it was possible for siren14 (a 32khz codec), to contain the a translation path to siren7 (a 16khz audio codec) that goes through slin at 8khz. In this case Asterisk takes a 32khz codec, down samples it to 8khz and then up samples it to 16khz which is terrible regardless if it is computationally less expensive. This patch now builds translation paths that give priority to maintaining the best possible sample rate before taking into consideration computational cost. This patch also adds cli commands to expose what translation paths are actually being used. Changes: 1. Translation paths will never contain a step that changes the sample rate unless absolutely necessary. 2. When choosing the best codec to make two channels compatible. Shared codecs with the highest sample rate are given priority. 3. A new cli command to show all translation paths available for a specific codec 'core show translation paths [codec name]' has been added. 4. 'core show translation' which displays the translation matrix now includes the new higher bit audio codecs in the table. 5. 'core show channel [channel name]' now displays the translation paths if translation is used. (closes issue #16841) Reported by: dvossel Review: https://reviewboard.asterisk.org/r/842/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@282048 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-12Merged revisions 282015 via svnmerge from russell1-8/+8
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282015 | russell | 2010-08-12 13:03:56 -0500 (Thu, 12 Aug 2010) | 2 lines Put back pointer value output for ast_debug(), such that it is only removed for verbose output. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@282016 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-12Merged revisions 281982 via svnmerge from russell1-12/+12
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r281982 | russell | 2010-08-12 11:33:30 -0500 (Thu, 12 Aug 2010) | 5 lines Remove debugging output from verbose messages. Pointer values to internal objects is not terribly useful to users in the verbose messages about adding extensions and contexts. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@281983 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-12Merged revisions 281913 via svnmerge from jpeeler1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r281913 | jpeeler | 2010-08-11 22:03:37 -0500 (Wed, 11 Aug 2010) | 34 lines Merged revisions 281912 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r281912 | jpeeler | 2010-08-11 22:01:38 -0500 (Wed, 11 Aug 2010) | 27 lines Merged revisions 281911 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r281911 | jpeeler | 2010-08-11 22:00:14 -0500 (Wed, 11 Aug 2010) | 20 lines Ensure SSRC is changed when media source is changed to resolve audio delay. This change causes the SSRC to change right before the channels are bridged, which is what used to happen. It seems that fixes were made to attempt limiting SSRC changes, targeted mainly at sending DTMF. DTMF is not affecting the SSRC with this change. There are two other control frames sent in ast_channel_bridge that probably should also be changed to AST_CONTROL_SRCCHANGE as well, but I'm going to leave this change up to the discretion of resolving issue #17007. For reference - old review implementing new control frame SRCCHANGE: https://reviewboard.asterisk.org/r/540 (closes issue #17404) Reported by: sdolloff Patches: bug17404.patch uploaded by jpeeler (license 325) Tested by: sdolloff ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@281914 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-11Merged revisions 281687 via svnmerge from simon.perreault1-3/+3
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r281687 | simon.perreault | 2010-08-11 09:30:59 -0400 (Wed, 11 Aug 2010) | 9 lines Fix parsing of IPv6 address literals in outboundproxy (closes issue #17757) Reported by: oej Patches: 17757.diff uploaded by sperreault (license 252) sip.conf.diff uploaded by sperreault (license 252) Tested by: oej ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@281688 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-10Merged revisions 281575 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r281575 | russell | 2010-08-10 13:05:07 -0500 (Tue, 10 Aug 2010) | 16 lines Merged revisions 281574 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r281574 | russell | 2010-08-10 13:04:32 -0500 (Tue, 10 Aug 2010) | 9 lines Don't move the time threshold for running scheduled events on every iteration. Instead, only calculate the time threshold each time ast_sched_runq() is called. (closes issue #17742) Reported by: schmidts Patches: sched.c.patch uploaded by schmidts (license 1077) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@281576 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-10Merged revisions 281529 via svnmerge from russell1-1/+7
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r281529 | russell | 2010-08-10 11:21:58 -0500 (Tue, 10 Aug 2010) | 8 lines Resolve a problem with channel name tab completion. Hitting tab without typing any part of a channel name resulted in no results. This now results in getting a full list of active channels, just as it did in previous versions of Asterisk. Review: https://reviewboard.asterisk.org/r/818/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@281530 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-06Merged revisions 281085 via svnmerge from tilghman1-1/+15
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r281085 | tilghman | 2010-08-06 13:57:10 -0500 (Fri, 06 Aug 2010) | 8 lines Fix alignment of stringfields on the SPARC architecture (closes issue #17789) Reported by: Ian Mason Patches: 20100806__issue17789__2.diff.txt uploaded by tilghman (license 14) Tested by: Ian_Mason ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@281086 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-05Merged revisions 281052 via svnmerge from russell1-13/+26
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r281052 | russell | 2010-08-05 08:16:11 -0500 (Thu, 05 Aug 2010) | 16 lines Merged revisions 281051 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r281051 | russell | 2010-08-05 08:11:32 -0500 (Thu, 05 Aug 2010) | 9 lines Cleanup default option value handling for cdr.conf [general]. The default values would differ depending on whether or not cdr.conf exists. That is no longer the case. Apply a default value to the unanswered option. Define all default values as named constants. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@281054 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-05Merged revisions 280984 via svnmerge from tilghman1-23/+8
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r280984 | tilghman | 2010-08-05 02:46:36 -0500 (Thu, 05 Aug 2010) | 22 lines Merged revisions 280983 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r280983 | tilghman | 2010-08-05 02:40:47 -0500 (Thu, 05 Aug 2010) | 15 lines Merged revisions 280982 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r280982 | tilghman | 2010-08-05 02:28:33 -0500 (Thu, 05 Aug 2010) | 8 lines Change context lock back to a mutex, because functionality depends upon the lock being recursive. (closes issue #17643) Reported by: zerohalo Patches: 20100726__issue17643.diff.txt uploaded by tilghman (license 14) Tested by: zerohalo ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@280985 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-02Merged revisions 280628 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r280628 | tilghman | 2010-08-02 09:41:46 -0500 (Mon, 02 Aug 2010) | 2 lines Make this a little more deterministic... we want the latest value, not just a 1 somewhere. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@280629 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-02Merged revisions 280624 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r280624 | tilghman | 2010-08-02 09:27:20 -0500 (Mon, 02 Aug 2010) | 2 lines Apparently, the values in makeopts are sometimes 1:1 and sometimes 1. Compensate for this. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@280626 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-29Merged revisions 280450 via svnmerge from dvossel1-0/+3
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r280450 | dvossel | 2010-07-29 14:13:27 -0500 (Thu, 29 Jul 2010) | 25 lines Merged revisions 280449 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r280449 | dvossel | 2010-07-29 14:05:25 -0500 (Thu, 29 Jul 2010) | 18 lines Merged revisions 280448 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r280448 | dvossel | 2010-07-29 14:04:23 -0500 (Thu, 29 Jul 2010) | 12 lines fixes issue with translator frame not getting freed A translator frame even if it local storage so the translation path can be freed. This issue prevented g729 licenses from being freed up. (closes issue #17630) Reported by: manvirr Patches: encoder_fix.diff uploaded by dvossel (license 671) Tested by: manvirr, dvossel ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@280459 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-29Merged revisions 280391 via svnmerge from russell1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r280391 | russell | 2010-07-29 11:25:43 -0500 (Thu, 29 Jul 2010) | 2 lines Don't blow up if get_codec() was not provided in the RTP glue. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@280395 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-29Merged revisions 280307 via svnmerge from mnicholson1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r280307 | mnicholson | 2010-07-29 08:56:35 -0500 (Thu, 29 Jul 2010) | 11 lines Merged revisions 280306 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r280306 | mnicholson | 2010-07-29 08:45:11 -0500 (Thu, 29 Jul 2010) | 2 lines Implement support for ast_channel_queryoption on local channels. Currently only AST_OPTION_T38_STATE is supported. ABE-2229 Review: https://reviewboard.asterisk.org/r/813/ ........ Additionally, pass AST_CONTROL_T38_PARAMETERS control frames through generic bridges. This change appears to have been unintentionally left out of rev 203699. ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@280308 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-27Merged revisions 279949 via svnmerge from dvossel2-1/+21
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r279949 | dvossel | 2010-07-27 15:57:00 -0500 (Tue, 27 Jul 2010) | 31 lines Merged revisions 279946 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r279946 | dvossel | 2010-07-27 15:54:32 -0500 (Tue, 27 Jul 2010) | 24 lines Merged revisions 279945 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r279945 | dvossel | 2010-07-27 15:33:40 -0500 (Tue, 27 Jul 2010) | 19 lines remove empty audiohook write list on channel If a channel has an audiohook write list created on it, that list stays on the channel until the channel is destroyed. There is no reason to keep that list on the channel if it becomes empty. If it is empty that just means we are doing needless translating for every ast_read and ast_write. This patch removes the audiohook list from the channel once it is detected to be empty on either a read or write. If a audiohook is added back to the channel after this list is destroyed, the list just gets recreated as if it never existed to begin with. (closes issue #17630) Reported by: manvirr Review: https://reviewboard.asterisk.org/r/799/ ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@279951 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-27Merged revisions 279817 via svnmerge from dvossel1-2/+4
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279817 | dvossel | 2010-07-27 11:09:15 -0500 (Tue, 27 Jul 2010) | 2 lines fix sip transaction match with authentication, fix confusing log message when using getaddrinfo ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@279818 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-27Merged revisions 279636,279815 via svnmerge from russell1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279636 | russell | 2010-07-26 16:53:30 -0500 (Mon, 26 Jul 2010) | 2 lines Ignore a control subclass of -1 in ast_waitfordigit_full(). ........ r279815 | russell | 2010-07-27 11:06:58 -0500 (Tue, 27 Jul 2010) | 4 lines Support "channels" in addition to "channel" in chan_dahdi.conf. Review: https://reviewboard.asterisk.org/r/804 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@279816 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-27Merged revisions 279726 via svnmerge from pabelanger1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279726 | pabelanger | 2010-07-26 21:53:38 -0400 (Mon, 26 Jul 2010) | 9 lines Use ast_sockaddr_setnull() when http is not enabled. Otherwise, ast_tcptls_server_start() will still start http. (closes issue #17708) Reported by: pabelanger Patches: http.patch uploaded by pabelanger (license 224) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@279727 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-25Merged revisions 279390 via svnmerge from tilghman1-4/+6
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279390 | tilghman | 2010-07-25 12:32:21 -0500 (Sun, 25 Jul 2010) | 8 lines Don't assume qlog is open. (closes issue #17704) Reported by: vrban Patches: issue17704.patch uploaded by pabelanger (license 224) Tested by: vrban ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@279391 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-24Merged revisions 279273 via svnmerge from pabelanger1-6/+4
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279273 | pabelanger | 2010-07-24 13:36:42 -0400 (Sat, 24 Jul 2010) | 6 lines Default sin_family to AF_INET for TCP / TLS Bindaddress. Otherwise, 'manager show settings' will generate errors if manager is not enabled. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@279274 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-23Merged revisions 278981 via svnmerge from tilghman1-6/+13
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r278981 | tilghman | 2010-07-23 11:42:25 -0500 (Fri, 23 Jul 2010) | 8 lines Avoid race with consolethread on shutdown (on parallel processors). (closes issue #17080) Reported by: sybasesql Patches: 20100721__issue17080.diff.txt uploaded by tilghman (license 14) Tested by: sybasesql ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278982 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-23Merge the realtime failover branchtilghman1-42/+101
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278957 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-23Allow IPv6 addresses for UDPTL streams.mmichelson1-102/+70
Review: https://reviewboard.asterisk.org/r/795 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278908 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-22Add method for finding XML doc files for systems that don't support GLOB_BRACE.jpeeler1-5/+62
In particular, Solaris and perhaps others do not support the above mentioned GNU extension. In this case the paths are simply expanded without the braces and the calls to glob are made separately. Note: I could not explain memory allocation failures that were being reported from within libxml itself when making calls to glob without using GLOB_NOCHECK. This is the only reason why that flag is being used. (closes issue #15402) Reported by: snuffy Patches: bug_xmlpatt-v3.diff uploaded by snuffy (license 35), modified by me git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278708 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-22Merged revisions 278618 via svnmerge from mmichelson1-1/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r278618 | mmichelson | 2010-07-22 09:55:04 -0500 (Thu, 22 Jul 2010) | 13 lines Allow PLC to function properly when channels use SLIN for audio. If a channel involved in a bridge was using SLIN audio, then translation paths were not guaranteed to be set up properly since in all likelihood the number of translation steps was only 1. This patch enforces the transcode_via_slin behavior if transcode_via_slin or generic_plc is enabled and one of the formats to make compatible is SLIN. AST-352 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278620 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-21Remove built-in AES code and use optional_api insteadtwilson4-2059/+0
Review: https://reviewboard.asterisk.org/r/793/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278538 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-21Update documentation for 'comebacktoorigin' in featuers.conf.russell1-10/+12
The documentation for this option did not match the code. Fix that along with some minor cleanups to the code along the way. Document a slight change in behavior (to something that was previously undocumented) in UPGRADE.txt. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278425 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-20Separate queue_log arguments into separate fields, and allow the text file ↵tilghman1-66/+178
to be used, even when realtime is used. (closes issue #17082) Reported by: coolmig Patches: 20100720__issue17082.diff.txt uploaded by tilghman (license 14) Tested by: coolmig git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278307 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-20Merged revisions 278167 via svnmerge from tilghman2-1/+28
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r278167 | tilghman | 2010-07-20 15:59:06 -0500 (Tue, 20 Jul 2010) | 4 lines Do not queue up DTMF frames while a call is on hold. (Fixes ABE-2110) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278272 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-20Add load priority order, such that preload becomes unnecessary in most casestilghman3-6/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278132 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-20Merged revisions 278023 via svnmerge from tilghman1-5/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r278023 | tilghman | 2010-07-20 11:37:18 -0500 (Tue, 20 Jul 2010) | 7 lines Off-by-one error (closes issue #16506) Reported by: nik600 Patches: 20100629__issue16506.diff.txt uploaded by tilghman (license 14) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278024 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-19Merged revisions 277906 via svnmerge from jeang1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277906 | jeang | 2010-07-19 15:16:36 -0500 (Mon, 19 Jul 2010) | 7 lines Avoid trying to pickup a parked extension before the park operation is completed. A crash could occur if the extension is picked up while the parking extension is being announced. Testing pu->notquiteyet while searching for a parked extension resolves this crash. (ABE-2418) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277945 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-19Make ACLs IPv6-capable.mmichelson3-51/+229
ACLs can now be configured to match IPv6 networks. This is only relevant for ACLs in chan_sip for now since other channel drivers do not support IPv6 addressing. However, once those channel drivers are outfitted to support IPv6 addressing, the ACLs will already be ready for IPv6 support. https://reviewboard.asterisk.org/r/791 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277814 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-17Merged revisions 277568 via svnmerge from tilghman1-1/+30
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277568 | tilghman | 2010-07-16 16:54:29 -0500 (Fri, 16 Jul 2010) | 8 lines Since we split values at the semicolon, we should store values with a semicolon as an encoded value. (closes issue #17369) Reported by: gkservice Patches: 20100625__issue17369.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277773 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Merged revisions 277625 via svnmerge from tringenbach1-0/+13
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277625 | tringenbach | 2010-07-16 17:43:39 -0500 (Fri, 16 Jul 2010) | 9 lines Save and restore AST_FLAG_BRIDGE_HANGUP_DONT on attended transfer. ast_bridge_call() clears AST_FLAG_BRIDGE_HANGUP_DONT. But during an attended transfer, ast_bridge_call() is called for a second bridge on the same channel, and it clears that flag, which still needs to get set for when the original ast_bridge_call() gets control back and checks it. Review: https://reviewboard.asterisk.org/r/741 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277657 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Finally, a method that really fixes the assertions in chan_iax2.c related to ↵tilghman1-6/+21
cancelling lagid. No, replacing usleep(1) with sched_yield() did not have an effect. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277484 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Merged revisions 277327 via svnmerge from mnicholson1-1/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277327 | mnicholson | 2010-07-16 13:30:22 -0500 (Fri, 16 Jul 2010) | 8 lines Interpret device state AST_DEVICE_UNKNOWN as extension state AST_EXTENSION_NOT_INUSE. (closes issue #16035) Reported by: francesco_r Patches: pbx.c.patch uploaded by viniciusfontes (license 978) Tested by: francesco_r, agx, lawbar ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277331 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Merged revisions 277261 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277261 | tilghman | 2010-07-16 13:04:11 -0500 (Fri, 16 Jul 2010) | 5 lines If variable gotten is not set, will segfault on Solaris. (closes issue #17636) Reported by: bklang ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277263 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Print f->subclass.integer instead of f->subclass.mnicholson1-1/+1
(fix build breakage introduced in r277250) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277262 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Merged revisions 277247 via svnmerge from mnicholson1-0/+13
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277247 | mnicholson | 2010-07-16 12:29:57 -0500 (Fri, 16 Jul 2010) | 4 lines For pass through DTMF tones, measure the actual duration between the begin and end packets on the wire. If it is detected to be less than AST_MIN_DTMF_DURATION, trigger dtmf emulation. AST-362 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277250 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Avoid crashing when installing a duplicate translation path with a lower cost.seanbright1-0/+1
(closes issue #17092) Reported by: moy Patches: translate.rev254273.patch uploaded by moy (license 222) Tested by: moy git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277143 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Formatting changesoej2-8/+17
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277102 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Fix build on FreeBSDtilghman1-1/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276871 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-07-02 01:56:30 +0000