asterisk

Commit Graph

Author	SHA1	Message	Date
Russell Bryant	4723a780ca	Add a todo item for CEL. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282798 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	5729763b89	Merged revisions 282543 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282543 \| dvossel \| 2010-08-17 14:34:06 -0500 (Tue, 17 Aug 2010) \| 4 lines fixes truncated uint64_t value in put_unaligned_uint64_t() function (issue #17804) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282544 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	557ad25f07	Merged revisions 282366 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282366 \| tilghman \| 2010-08-13 23:53:58 -0500 (Fri, 13 Aug 2010) \| 4 lines 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/trunk@282367 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	0f8eaa6299	Merged revisions 282269 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282269 \| dvossel \| 2010-08-13 15:03:56 -0500 (Fri, 13 Aug 2010) \| 4 lines 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/trunk@282270 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	8bc5bf82df	Merged revisions 282098 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@282099 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	bbb32fe33e	Merged revisions 282047 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@282048 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
	a491cac965	Merged revisions 281687 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@281688 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	af43e57821	Merged revisions 280984 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@280985 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	395a35900a	Merged revisions 279949 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@279951 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	189839c307	Merged revisions 279658 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r279658 \| qwell \| 2010-07-26 18:03:38 -0500 (Mon, 26 Jul 2010) \| 12 lines 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/trunk@279659 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	c3a1e55de2	Merged revisions 279562 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r279562 \| tilghman \| 2010-07-26 14:18:26 -0500 (Mon, 26 Jul 2010) \| 9 lines 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/trunk@279564 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	dd9428666d	Merged revisions 279504 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279504 \| mmichelson \| 2010-07-26 11:04:09 -0500 (Mon, 26 Jul 2010) \| 14 lines 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/trunk@279533 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	222b96c2e2	Merged revisions 279280 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279280 \| pabelanger \| 2010-07-24 14:18:43 -0400 (Sat, 24 Jul 2010) \| 8 lines 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/trunk@279285 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 years ago
Tilghman Lesher	76d4bbacd1	Merged revisions 277738 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277738 \| tilghman \| 2010-07-17 11:59:11 -0500 (Sat, 17 Jul 2010) \| 5 lines Remove uclibc cross-compile triplet, as uclibc has a working fork()... it's only uclinux that does not. (closes issue #17616) Reported by: pprindeville ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277775 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	a7c92fad28	Merged revisions 277568 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@277773 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	e5c4c90064	Allow xmllint to be used for XML docs validation. xmllint seems to be more commonly available since it comes with libxml2. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277703 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	fe9e0e672e	Finally, a method that really fixes the assertions in chan_iax2.c related to cancelling lagid. No, replacing usleep(1) with sched_yield() did not have an effect. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277484 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	753fa3b278	Define LLONG_MAX on systems that do not have it. (closes issue #17644) Reported by: pprindeville git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276769 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	832d1296c6	Remove the old stub files, preferring the optional_api method. (closes issue #17475) Reported by: tilghman Review: https://reviewboard.asterisk.org/r/695/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276490 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	cf7bbcc4c6	Expand the caller ANI field to an ast_party_id Expand the ani field in ast_party_caller and ast_party_connected_line to an ast_party_id. This is an extension to the ast_callerid restructuring patch in review: https://reviewboard.asterisk.org/r/702/ Review: https://reviewboard.asterisk.org/r/744/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276393 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	ec37ffbdaf	ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char tag; }; struct ast_party_dialed { struct { char str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	da8450323f	Kill some startup warnings and errors and make some messages more helpful in tracking down the source. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275105 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	9aa4771a8d	Merged revisions 275021 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275021 \| russell \| 2010-07-09 10:33:08 -0500 (Fri, 09 Jul 2010) \| 4 lines Document that a leading and trailing slash is expected for test categories. Also, emit a warning if a test is registered without one of these. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275022 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	e10dcd165b	Extend length limit on country name in indications.conf. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274907 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	cd4ebd336f	Add IPv6 to Asterisk. This adds a generic API for accommodating IPv6 and IPv4 addresses within Asterisk. While many files have been updated to make use of the API, chan_sip and the RTP code are the files which actually support IPv6 addresses at the time of this commit. The way has been paved for easier upgrading for other files in the near future, though. Big thanks go to Simon Perrault, Marc Blanchet, and Jean-Philippe Dionne for their hard work on this. (closes issue #17565) Reported by: russell Patches: asteriskv6-test-report.pdf uploaded by russell (license 2) Review: https://reviewboard.asterisk.org/r/743 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274783 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Eliel C. Sardanons	a1b89a6a50	Implement AstData API data providers as part of the GSOC 2010 project, midterm evaluation. Review: https://reviewboard.asterisk.org/r/757/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274727 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	f4d96da591	Merged revisions 273793 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273793 \| tilghman \| 2010-07-02 16:36:39 -0500 (Fri, 02 Jul 2010) \| 9 lines Have the DEADLOCK_AVOIDANCE macro warn when an unlock fails, to help catch potentially large software bugs. (closes issue #17407) Reported by: pdf Patches: 20100527__issue17407.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/751/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273830 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	25156a326d	Remove unnecessary if test in CV_DSTR() git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273198 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	522ea689c4	Misc doxygen cleanup in config.h git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273197 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	0f9e53b6ef	Exclude libical for insufficient versions. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273055 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
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
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
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	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
David Vossel	fcb055fb4e	addition of G.719 pass-through support (closes issue #16293) Reported by: malcolmd Patches: g719.passthrough.patch.7 uploaded by malcolmd (license 924) format_g719.c uploaded by malcolmd (license 924) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270940 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	81c15adfa2	Add distributed devicestate via the XMPP protocol. (closes issue #15757) Reported by: Marquis Patches: distributed_devstate-XMPP.txt uploaded by lmadsen (license 10) Tested by: Marquis, lmadsen, marcelloceschia Review: https://reviewboard.asterisk.org/r/351/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270519 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	41821558c7	Reverting patch and reopening issue #16155 , as patch breaks FreeBSD / OSX builds. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270151 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	f19405516d	Use pkg-config to find gmime libraries This way the libraries can be found even if they are in non-standard locations. (closes issue #16155) Reported by: jcollie Patches: 0008-change-configure.ac-to-look-for-pkg-config-gmime-2.0.patch uploaded by jcollie (license 412) Tested by: jsmith, tilghman, pabelanger git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270042 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	05c46771ca	Resolve an invalid memory read on an event. Valgrind pointed out that attempting to get an IE value from an event that has no IEs produces an invalid memory read past the end of the event. Thanks to mmichelson for pointing the problem out to me and then testing the fix. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269417 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	3c26b511e0	Fix build on Mac OS X (and maybe FreeBSD, too) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269119 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	c672763af8	Fix some doxygen warnings. (closes issue #17336) Reported by: snuffy Patches: doxygen-fixes1.diff uploaded by snuffy (license 35) Tested by: russell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268969 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	857814f435	Add SRTP support for Asterisk After 5 years in mantis and over a year on reviewboard, SRTP support is finally being comitted. This includes generic CHANNEL dialplan functions that work for getting the status of whether a call has secure media or signaling as defined by the underlying channel technology and for setting whether or not a new channel being bridged to a calling channel should have secure signaling or media. See doc/tex/secure-calls.tex for examples. Original patch by mikma, updated for trunk and revised by me. (closes issue #5413) Reported by: mikma Tested by: twilson, notthematrix, hemanshurpatel Review: https://reviewboard.asterisk.org/r/191/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268894 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago

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2195 Commits (da683f0cc0ce4fe3b683a703dfe6cfe8eaefd38e)