2225 Commits (f68807f0639df85c608dc39506703a95e80e6a05)

Author	SHA1	Message	Date
Jeff Peeler	aafc5f4e99	Merged revisions 294639 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r294639 | jpeeler | 2010-11-11 13:31:00 -0600 (Thu, 11 Nov 2010) | 53 lines Merged revisions 294384 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r294384 | jpeeler | 2010-11-09 11:37:59 -0600 (Tue, 09 Nov 2010) | 47 lines Fix a deadlock in device state change processing. Copied from some notes from the original author (Russell): Deadlock scenario: Thread 1: device state change thread Holds - rdlock on contexts Holds - hints lock Waiting on channels container lock Thread 2: SIP monitor thread Holds the "iflock" Holds a sip_pvt lock Holds channel container lock Waiting for a channel lock Thread 3: A channel thread (chan_local in this case) Holds 2 channel locks acquired within app_dial Holds a 3rd channel lock it got inside of chan_local Holds a local_pvt lock Waiting on a rdlock of the contexts lock A bunch of other threads waiting on a wrlock of the contexts lock To address this deadlock, some locking order rules must be put in place and enforced. Existing relevant rules: 1) channel lock before a pvt lock 2) contexts lock before hints lock 3) channels container before a channel What's missing is some enforcement of the order when you involve more than any two. To fix this problem, I put in some code that ensures that (at least in the code paths involved in this bug) the locks in (3) come before the locks in (2). To change the operation of thread 1 to comply, I converted the storage of hints to an astobj2 container. This allows processing of hints without holding the hints container lock. So, in the code path that led to thread 1's state, it no longer holds either the contexts or hints lock while it attempts to lock the channels container. (closes issue #18165) Reported by: antonio ABE-2583 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294640 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	2d6a9d607f	Merged revisions 294429 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294429 | tilghman | 2010-11-09 14:27:23 -0600 (Tue, 09 Nov 2010) | 8 lines Detect GMime properly on systems where gmime flags and libs are configured with pkg-config. (closes issue #16155) Reported by: jcollie Patches: 20100917__issue16155.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294430 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	3f9644b7db	Analog lines do not transfer CONNECTED LINE or execute the interception macros. ... Add connected line update for sig_analog transfers and simplify the corresponding sig_pri and chan_misdn transfer code. Note that if you create a three-way call in sig_analog before transferring the call, the distinction of the caller/callee interception macros make little sense. The interception macro writer needs to be prepared for either caller/callee macro to be executed. The current implementation swaps which caller/callee interception macro is executed after a three-way call is created. Review: https://reviewboard.asterisk.org/r/996/ JIRA ABE-2589 JIRA SWP-2372 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294349 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	375b5b6b09	Merged revisions 294277 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294277 | jpeeler | 2010-11-08 15:58:13 -0600 (Mon, 08 Nov 2010) | 16 lines Fix playback failure when using IAX with the timerfd module. To fix this issue the alert pipe will now be used when the timerfd module is in use. There appeared to be a race that was not solved by adding locking in the timerfd module, but needed to be there anyway. The race was between the timer being put in non-continuous mode in ast_read on the channel thread and the IAX frame scheduler queuing a frame which would enable continuous mode before the non-continuous mode event was read. This race for now is simply avoided. (closes issue #18110) Reported by: tpanton Tested by: tpanton I put tested by tpanton because it was tested on his hardware. Thanks for the remote access to debug this issue! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294278 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	6dfc9dddd8	Avoid valgrind warnings for ast_rtp_instance_get_xxx_address ... The documentation for ast_rtp_instance_get_(local/remote)_address stated that they returned 0 for success and -1 on failure. Instead, they returned 0 if the address structure passed in was already equivalent to the address instance local/remote address or 1 otherwise. 90% of the calls to these functions completely ignored the return address and passed in an uninitialized struct, which would make valgrind complain even though the operation was technically safe. This patch fixes the documentation and converts the get_xxx_address functions to void since all they really do is copy the address and cannot fail. Additionally two new functions (ast_rtp_instance_get_and_cmp_(local/remote)_address) are created for the 3 times where the return value was actually checked. The get_and_cmp_local_address function is currently unused, but exists for the sake of symmetry. The only functional change as a result of this change is that we will not do an ast_sockaddr_cmp() on (mostly uninitialized) addresses before doing the ast_sockaddr_copy() in the get_*_address functions. So, even though it is an API change, it shouldn't have a noticeable change in behavior. Review: https://reviewboard.asterisk.org/r/995/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293803 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	d411ade931	Allow the DAHDI driver to compile, even with a sufficiently older version of libpri. ... Fixes our Bamboo builds. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293046 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	cdc420235e	Allow the DAHDI driver to compile, even with a sufficiently older version of libpri. ... Fixes our Bamboo builds. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292906 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	a37956721c	Add the ability for ast_find_ourip to return IPv4, IPv6 or both. ... While testing chan_gtalk I noticed jabber was using my IPv6 address and not IPv4. When using bindaddr=0.0.0.0 it is possible for ast_find_ourip() to return both IPv6 and IPv4 results. Adding a family parameter gives you the ablility to choose. Since jabber/gtalk/h323 do not support IPv6, we should only return IPv4 results. Review: https://reviewboard.asterisk.org/r/973/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291758 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	f5782280d8	Gtalk enhancements and general code cleanup. ... This patch includes several chan_gtalk enhancements. Two new gtalk.conf options have been added, externip and stunadd. Setting externip allows us to manually specify what the external IP address is outside of a NAT environment. Setting the stunaddr option to a valid stun server allows for that external ip to be retrieved via a STUN server automatically. This external IP is then advertised during call setup as a possible candidate. I have also attempted to clean up chan_gtalk's code so it meets our coding guidelines. During this cleanup I noticed several things that need to be done in the code and made a TODO section at the top of the file. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291192 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	c437bca01b	Merged revisions 290751 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r290751 | qwell | 2010-10-07 15:57:14 -0500 (Thu, 07 Oct 2010) | 16 lines Merged revisions 290750 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r290750 | qwell | 2010-10-07 15:56:04 -0500 (Thu, 07 Oct 2010) | 9 lines Allow PRI to build properly when using --with-pri. Use the directories found for the parent when using lib dependencies. (closes issue #17314) Reported by: tzafrir Patches: 17314-withdeps.diff uploaded by qwell (license 4) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@290752 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	f87133b078	Fixes chan_gtalk to work with gmail client ... This patch was written by Philippe Sultan (phsultan). Thanks for keeping this up to date! git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@290479 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	ddebf12b88	Merged revisions 289798 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r289798 | jpeeler | 2010-10-01 18:01:31 -0500 (Fri, 01 Oct 2010) | 22 lines Merged revisions 289797 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289797 | jpeeler | 2010-10-01 17:58:38 -0500 (Fri, 01 Oct 2010) | 15 lines Change RFC2833 DTMF event duration on end to report actual elapsed time. The scenario here is with a non P2P early media session. The reported time length of DTMF presses are coming up short when sending to the remote side. Currently the event duration is a running total that is incremented when sending continuation packets. These continuation packets are only triggered upon incoming media from the remote side, which means that the running total probably is not going to end up matching the actual length of time Asterisk received DTMF. This patch changes the end event duration to be lengthened if it is detected that the end event is going to come up short. Review: https://reviewboard.asterisk.org/r/957/ ABE-2476 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@289840 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	f48a6f6cfe	More Solaris compatibility fixes ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@289543 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	e00c4dcc6d	Solaris compatibility fixes ... Review: https://reviewboard.asterisk.org/r/942/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@289104 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	2eb6c7395d	Merged revisions 288637 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r288637 | tilghman | 2010-09-23 22:36:01 -0500 (Thu, 23 Sep 2010) | 9 lines Merged revisions 288636 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288636 | tilghman | 2010-09-23 22:20:24 -0500 (Thu, 23 Sep 2010) | 2 lines Solaris compatibility fixes ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@288638 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	ecabd15422	Addition of the FrameHook API (AKA AwesomeHooks) ... So far all our tools for viewing and manipulating media streams within Asterisk have been entirely focused on audio. That made sense then, but is not scalable now. The FrameHook API lets us tap into and manipulate _ANY_ type of media or signaling passed on a channel present today or in the future. This tool is a step in the direction of expanding Asterisk's boundaries and will help generate some rather interesting applications in the future. In addition to the FrameHook API, a simple dialplan function exercising the api has been included as well. This function is called FRAME_TRACE(). FRAME_TRACE() allows for the internal ast_frames read and written to a channel to be output. Filters can be placed on this function to debug only certain types of frames. This function could be thought of as an internal way of doing ast_frame packet captures. Review: https://reviewboard.asterisk.org/r/925/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@287647 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	c9bfde6afd	Add parking extension for non-default parking lots. ... This is a new feature that allows for parking to custom parking lots to be accessed directly, rather than with channel variables or by changing the default parking lot. The extension is set with the parkext option just as the default parking lot is done. Also, the manager action has been updated to optionally allow a specified parking lot. (closes issue #14882) Reported by: vmikhnevych Patches: patch_14882.txt uploaded by mnick (license 874) modified by me Review: https://reviewboard.asterisk.org/r/884/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@286931 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	e72b55f3cf	Merged revisions 286115 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r286115 | twilson | 2010-09-10 15:35:25 -0500 (Fri, 10 Sep 2010) | 23 lines Merged revisions 286059 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286059 | twilson | 2010-09-10 14:25:08 -0500 (Fri, 10 Sep 2010) | 16 lines Inherit CHANNEL() writes to both sides of a Local channel Having Local (/n) channels as queue members and setting the language in the extension with Set(CHANNEL(language)=fr) sets the language on the Local/...,2 channel. Hold time report playbacks happen on the Local/...,1 channel and therefor do not play in the specified language. This patch modifies func_channel_write to call the setoption callback and pass the CHANNEL() write info to the callback. chan_local uses this information to look up the other side of the channel and apply the same changes to it. (closes issue #17673) Reported by: Guggemand Review: https://reviewboard.asterisk.org/r/903/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@286189 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	6b87845350	Merged revisions 285961 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r285961 | tilghman | 2010-09-10 00:31:31 -0500 (Fri, 10 Sep 2010) | 6 lines Another fix for Mac OS X. While trying to fix this the "right" way, I wandered into dependency hell. Two hours later, I backed out, and just removed the offending code. ast_inline_api only goes one level deep and then it breaks. Ouch. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@285962 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	f7c710feed	Merged revisions 285930 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r285930 | tilghman | 2010-09-09 20:16:32 -0500 (Thu, 09 Sep 2010) | 14 lines Merged revisions 285889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285889 | tilghman | 2010-09-09 19:13:45 -0500 (Thu, 09 Sep 2010) | 7 lines Fix Mac OS X build. This also fixes a rather grievous calculation error for the offset of ast_fdset, which was masked on Linux and FreeBSD, because these platforms check the first 256 FDs regardless of the bitmask setting (due to backwards compatibility). ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@285931 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	a766133be6	Merged revisions 285529 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r285529 | qwell | 2010-09-08 15:42:44 -0500 (Wed, 08 Sep 2010) | 1 line Follow coding guidelines in moh rescan fix. Also fix the documentation that got me in trouble. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@285530 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	f7f7eaa3a0	Fix build on FreeBSD 8.0, take 2. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@285336 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	2fc3cf65af	Fix libsrtp -fPIC check for when non-standard prefix is used. ... Thanks to loompek in #asterisk for reporting the issue and testing this patch. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@285161 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	222d4676d1	Add a C++ compatible version of AST_CLI_DEFINE(). ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@285057 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jan Kalab	e7603921cf	Support for calendar events priorities and categories ... Review 880 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284849 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	7e3f95e00a	When optional_api is non-optional, force dependent modules to be loaded. ... (closes issue #17707) Reported by: ira Patches: 20100819__issue17707__asterisk1.8.diff.txt uploaded by tilghman (license 14) Tested by: tilghman Review: https://reviewboard.asterisk.org/r/876/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284610 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	6c61e312c6	Merged revisions 284593,284595 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r284593 | tilghman | 2010-09-01 17:59:50 -0500 (Wed, 01 Sep 2010) | 18 lines Merged revisions 284478 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r284478 | tilghman | 2010-09-01 13:49:11 -0500 (Wed, 01 Sep 2010) | 11 lines Ensure that all areas that previously used select(2) now use poll(2), with implementations that need poll(2) implemented with select(2) safe against 1024-bit overflows. This is a followup to the fix for the pthread timer in 1.6.2 and beyond, fixing a potential crash bug in all supported releases. (closes issue #17678) Reported by: russell Branch: https://origsvn.digium.com/svn/asterisk/team/tilghman/ast_select Review: https://reviewboard.asterisk.org/r/824/ ........ ................ r284595 | tilghman | 2010-09-01 22:57:43 -0500 (Wed, 01 Sep 2010) | 2 lines Failed to rerun bootstrap.sh after last commit ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284597 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	8a112de270	Fix SRTP for changing SSRC and multiple a=crypto SDP lines ... Adding code to Asterisk that changed the SSRC during bridges and masquerades broke SRTP functionality. Also broken was handling the situation where an incoming INVITE had more than one crypto offer. This patch caches the SRTP policies the we use so that we can change the ssrc and inform libsrtp of the new streams. It also uses the first acceptable a=crypto line from the incoming INVITE. (closes issue #17563) Reported by: Alexcr Patches: srtp.diff uploaded by twilson (license 396) Tested by: twilson Review: https://reviewboard.asterisk.org/r/878/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284477 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	b246f8691e	Make the AST_CEL_AMA enum match up with the AST_CDR_ ama flag values. ... Really, having 2 enums for this is silly and error prone, demonstrated by the crash that I hit because there was an assumption in the code that the values in each matched up. However, this is a quick fix to get them to match up so it will work. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@283230 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	9fe871150e	fixes truncated uint64_t value in put_unaligned_uint64_t() function ... (issue #17804) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@282543 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	3c0616589e	Fix our FRACKing issue with chan_iax2 a different way. ... Review: https://reviewboard.asterisk.org/r/861/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@282366 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	48fb2c3276	res_stun_monitor for monitoring network changes behind a NAT device ... Review: https://reviewboard.asterisk.org/r/854 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@282269 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	89872aac58	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: https://origsvn.digium.com/svn/asterisk/branches/1.8@282098 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	44bc8cd334	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: https://origsvn.digium.com/svn/asterisk/branches/1.8@282047 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
	5c1c1b35bd	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: https://origsvn.digium.com/svn/asterisk/branches/1.8@281687 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	20582d0a6d	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: https://origsvn.digium.com/svn/asterisk/branches/1.8@280984 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	0140af581e	Revert r280019 for now - This was poorly executed. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@280020 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	5fb5c90b89	Add ability to use system libedit and update bundled libedit. ... The version of libedit that is bundled with asterisk is old and has some bugs. This patch updates the bundled version of libedit within asterisk, and also updates asterisk to use the system libedit instead if one is available (and pkg-config is available). This review integrates several patches from other users specifically kkm and tzafrir. (closes issue #15929) Reported by: kkm Patches: 015929-astcli-editrc-trunk.240324.diff uploaded by kkm (license 888) (issue #16858) Reported by: jw-asterisk (closes issue #17039) Reported by: tzafrir Patches: 0001-allow-using-system-copy-of-libedit.patch uploaded by tzafrir (license 46) Review: https://reviewboard.asterisk.org/r/807/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@280019 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	df89f8da6a	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: https://origsvn.digium.com/svn/asterisk/branches/1.8@279949 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	a4f4f7443b	Merged revisions 279657 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r279657 | qwell | 2010-07-26 17:59:52 -0500 (Mon, 26 Jul 2010) | 5 lines Really fix sounds Makefile (and make it readableish). There was a rather large syntax error that should have caused ALL versions of GNU make to fail. I don't know how it worked. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@279658 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	585d4c3440	Merged revisions 279561 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r279561 | tilghman | 2010-07-26 14:15:59 -0500 (Mon, 26 Jul 2010) | 2 lines Use a special Makefile for noobs who still have GNU Make 3.80. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@279562 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	bc3b185063	Allow for systems without locale support to be usable. ... A recent change to SIP URI comparison code added a locale-specific string comparison to the mix, and certain systems do not support such functions. This fix allows for those systems to still use Asterisk 1.8 (closes issue #17697) Reported by: pprindeville Patches: asterisk-trunk-bugid17697.patch uploaded by pprindeville (license 347) Tested by: mmichelson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@279504 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	e0557f853e	Check if ast_sockaddr is NULL then return. ... (closes issue #17677) Reported by: outcast Patches: issue0017677.patch uploaded by pabelanger (license 224) Tested by: elguero git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@279280 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	3ab0041118	Merge the realtime failover branch ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278957 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	57a92a6a7c	Allow IPv6 addresses for UDPTL streams. ... Review: https://reviewboard.asterisk.org/r/795 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278908 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	1b2893cab4	Add the full current set of CDR drivers ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278579 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	d6e1c724e5	Remove built-in AES code and use optional_api instead ... Review: https://reviewboard.asterisk.org/r/793/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278538 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	ef95349d1c	Merged revisions 278167 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/trunk@278272 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	b4e18d5660	Add load priority order, such that preload becomes unnecessary in most cases ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278132 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	6fa79e8f77	Make ACLs IPv6-capable. ... 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: https://origsvn.digium.com/svn/asterisk/trunk@277814 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago