Asterisk
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asterisk
mirror of https://github.com/asterisk/asterisk
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7436 Commits
151 Branches
1609 Tags
308 MiB
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7436 Commits (e735f507b72b9c2b71321dc4b2ccc569246f946b)
 

Author SHA1 Message Date
Kevin P. Fleming 52c23802d6 use auto-build for res modules too
20 years ago
Kevin P. Fleming 90cca08247 prepare to use auto-build rules in other Makefiles
20 years ago
Kevin P. Fleming e42c6c4f9e set standard properties on all non-binary files
20 years ago
Tilghman Lesher ada97b3723 Fix compile time warnings
20 years ago
Kevin P. Fleming 99f10b2e26 Merged revisions 9493,9528 via svnmerge from
20 years ago
Mark Spencer 4a80fe5022 Add capability to retrieve list of channel types
20 years ago
Mark Spencer c74856b173 Make IAX2 not try to authenticate when registration info is not changing
20 years ago
Mark Spencer 7dc0bdf35b Fix SMDI to not blow up asterisk when there's no config file.
20 years ago
Kevin P. Fleming bd8d1bf5e3 ignore archives in this directory
20 years ago
Kevin P. Fleming 7a4b31e515 doh!
20 years ago
Kevin P. Fleming ad7f3e78e4 remove last vestiges of pbx_functions
20 years ago
Kevin P. Fleming a38a7eec61 build function modules independently (no more pbx_functions.so)
20 years ago
Kevin P. Fleming c7da92d2ea fix compiler warnings
20 years ago
Kevin P. Fleming 845a5dc75d one more copyright/file header and version string support
20 years ago
Kevin P. Fleming 7b2988821a clean up SMDI support commit:
20 years ago
Matthew Fredrickson 4f803dfda0 Lots of little fixes for doing MSVC compiling codecs in windows (#6022)
20 years ago
Matthew Fredrickson c344781c4b Add boost gain option to microphone, also minor bug fixes (#6388)
20 years ago
Matthew Fredrickson 5dd0b76b2d Minor bugfix for #6386
20 years ago
Matthew Fredrickson af07dc8883 Add smdi support for asterisk (see doc/smdi.txt for config info) (#5945)
20 years ago
Kevin P. Fleming cadfcdfe8e Merged revisions 9404 via svnmerge from
20 years ago
Kevin P. Fleming b40bd71a9a restore 'rfc2833' naming for DTMF mode in chan_sip
20 years ago
Christian Richter fd9fe21d2c * jb can be configed by misdn_set_opt
20 years ago
Christian Richter 0058f40ed5 added Proceeding state, to be more ETSI conform with outgoing setups on the Network side (no release_complete but disconnect or release in proceeding state)
20 years ago
Tilghman Lesher 4f1e5d1aef Bug 6387 - janitor cleanup for linked lists
20 years ago
Tilghman Lesher 154e107d06 Blocked revisions 9326 via svnmerge
20 years ago
Tilghman Lesher 3665cabca3 Blocked revisions 9323 via svnmerge
20 years ago
Russell Bryant b96af335d8 conversions to memory allocation wrappers, remove duplicated error messages,
20 years ago
Christian Richter 091a14754d still some L1 related fixes
20 years ago
Olle Johansson 4d07b89fdd - Change "rfc2833" to "rtp" in sip.conf. Keeping backwards compatibility.
20 years ago
Olle Johansson 24ceb84434 - Adding example on using european time zones in voicemail.conf
20 years ago
Russell Bryant 9f7097b004 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9263 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Russell Bryant 3f37ece516 remove some Makefile targets for things that don't exist in the tree
20 years ago
Russell Bryant 7529e12ab4 remove unnecessary format check in zt_request so that we will actually accept
20 years ago
Russell Bryant e13567b0c9 various code cleanup changes including changing #define'd constants to enums,
20 years ago
Russell Bryant a8d24df108 clarify return value (issue #6434)
20 years ago
Russell Bryant 14906b3cfc Merged revisions 9246 via svnmerge from
20 years ago
Matt O'Gorman 5f0de64f78 Merged revisions 9232 via svnmerge from
20 years ago
Tilghman Lesher 8774e4a7d8 Merged revisions 9233 via svnmerge from
20 years ago
Christian Richter be20a26a35 fix for the L2 Signalling
20 years ago
Christian Richter f354a43bd3 L2 Signalling in PTP-TE Mode fixed
20 years ago
Christian Richter a43c6c0f49 * dozens of white-space removements
20 years ago
Matthew Fredrickson 556ec7c940 Make sure we output the APDU debug on pri debug commands
20 years ago
Christian Richter b42dd639ee default values of jitterbuffer and jitterbuffer_upper_threshold should be > 0, this fixes the tv_fix warnings, because we use ast_read to transmit frames to asterisk in jitterbuffer mode, instead of queueing the audio data with ast_queue_frame.
20 years ago
Kevin P. Fleming 8d8aa07407 minor cleanup
20 years ago
Kevin P. Fleming d87a77b89b rename properties for forward-porting fixes so they don't conflict with developer branches
20 years ago
Tilghman Lesher b347ddf3c6 Merged revisions 9156 via svnmerge from
20 years ago
Russell Bryant 5384e654f7 remove windows-style line endings
20 years ago
BJ Weschke 81125d803e Don't set a global variable if the channel you're trying to set CHANLOCALSTATUS on has gone away already.
20 years ago
Tilghman Lesher 34bcd0e0b6 Bug 6322 - Implementation of SHA1 in Asterisk (plus dialplan function to use it)
20 years ago
Kevin P. Fleming 193b923925 remove obsolete stuff
20 years ago