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AgeCommit message (Expand)Author
2012-01-09Replace direct access to channel name with accessor functionsTerry Wilson
2012-01-09Fix joinable thread terminating without joiner memory leak in chan_iax.c.Richard Mudgett
2012-01-09Fix shutdown handling of sqlite3 astdb.Walter Doekes
2012-01-09Multiple revisions 350127-350128Richard Mudgett
2012-01-09Make Asterisk -x command line parameter imply -r parameter presence.Richard Mudgett
2012-01-09Prevent SLA settings from getting wiped out on reloadKinsey Moore
2012-01-06Don't leak CID in From header when presentation=unavailableTerry Wilson
2012-01-06Fix lua goto detection to prevent unexpected behavior with confbridgeKinsey Moore
2012-01-06Fix memory leaks in app_followme find_realtime().Richard Mudgett
2012-01-05Fix premature free'ing of the frame committed in r349608Matthew Jordan
2012-01-05Make not assume that the cel_sqlite3_custom SQL table primary key is AcctId.Richard Mudgett
2012-01-05Make pbx_config.c use Gosub instead of Macro call for stdexten.Richard Mudgett
2012-01-05Allow playback of formats that don't support seekingKinsey Moore
2012-01-05Fix an issue where dsp.c would interpret multiple dtmf events from a single k...Jonathan Rose
2012-01-05Ensures Asterisk closes when receiving terminal signals in 'no fork' mode.Jonathan Rose
2012-01-04Fix for ConfBridge config parser unlocking channel mutex too many timesMatthew Jordan
2012-01-04Free successfully translated frame in fax_gateway_framehookMatthew Jordan
2012-01-04Fix segfault in chan_dahdi for CHANNEL(dahdi_span) evaluation on hangup.Richard Mudgett
2012-01-04Make debian init script conform to the LSB standardKinsey Moore
2012-01-04Update autosupport script and man pageKinsey Moore
2012-01-04Adds Subscription-State header to notify with call completion. per RFC3265Jonathan Rose
2012-01-04Fix documentation for SayNumber to reflect the fact that language is changed ...Jonathan Rose
2011-12-31Fix some minor formatting issues based on coding guidelines.Russell Bryant
2011-12-31Constify tag argument in REF_DEBUG related code.Russell Bryant
2011-12-29Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loopMatthew Jordan
2011-12-28Use ast_audiohook_write_list_empty to determine if our lists are empty insteadSean Bright
2011-12-28Tell Subversion to gnore the 'astdb2bdb' binary file if it exists.Kevin P. Fleming
2011-12-28Improve T.38 gateway V.21 preamble detection.Kevin P. Fleming
2011-12-27Fix timing source dependency issues with MOHMatthew Jordan
2011-12-27Once an audiohook is attached to a channel, we continue to transcode all of theSean Bright
2011-12-23Allow overriding of IMAP server settings on a user by user basisMatthew Jordan
2011-12-23INFO/Record request configurable to use dynamic featuresJonathan Rose
2011-12-23chan_sip autocreatepeer=persist option for auto-created peers to survive reloadJonathan Rose
2011-12-23Merged revisions 349045 via svnmerge from Sean Bright
2011-12-23Fix missing doc tags found while fixing ASTERISK-18689Kinsey Moore
2011-12-23Fix extension state callback references in chan_sip.Richard Mudgett
2011-12-22Fix for memory leaks / cleanup in cel_pgsqlMatthew Jordan
2011-12-22Fix segfault on answer.Damien Wedhorn
2011-12-22Add Asterisk TestSuite event hooks to support ConfBridge testingMatthew Jordan
2011-12-22Allow packetization vaules > 127Terry Wilson
2011-12-21Make codecs/speex ignore *.i files also.Richard Mudgett
2011-12-21Make apps/confbridge ignore *.i files also.Richard Mudgett
2011-12-20Fix chan_iax2 to not report an RDNIS number if it is blank.Richard Mudgett
2011-12-20This adds support for setting several safe_asterisk parameters usingMatthew Nicholson
2011-12-19Fix crashes on other platforms caused by interference from Darwin weak symbol...Richard Mudgett
2011-12-19Update documentation for MESSAGE_SEND_STATUS variable.Leif Madsen
2011-12-19Add a separate buffer for SRTCP packetsTerry Wilson
2011-12-18Correct two flaws in sip.conf.sample related to AST-2011-013.Kevin P. Fleming
2011-12-16Clean-up on isle five for __ast_request_and_dial() and ast_call_forward().Richard Mudgett
2011-12-16Voicemail with the saycid option will now play a caller's name based on cid i...Jonathan Rose
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