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* call: add call_id accessorAlfred E. Heggestad2018-01-17
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* test: test all audio tx modesAlfred E. Heggestad2017-11-26
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* add support for specifying sample format (#317)Alfred E. Heggestad2017-11-09
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | API: ausrc and auplay - add config items for ausrc/auplay format: ausrc_format s16|float auplay_format s16|float - audio.c: convert audio samples to/from signed 16-bit Modules: alsa add test for sample format FLOAT rst: add support for FLOAT sample format audiounit: add support for FLOAT sample format coreaudio: check for signed 16-bit audio format oss: check for signed 16-bit sample format winwave: check for S16LE pulse: add support for FLOAT sample format sndio: check for S16 format gst1: check sample format aufile: check sample format aubridge: check sample format gst: check sample format opensles: check for S16 sample format jack: check sample format alsa: remove usage of local config test: change samples to void pointer test: change sample type to void pointer
* test: remove useragent password for un-authenticated testsAlfred E. Heggestad2017-07-09
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* ua: add ua_progress() and testcaseAlfred E. Heggestad2017-06-03
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* add RTP Header extension for Client-to-Mixer Audio Level Indication (#264)Alfred E. Heggestad2017-06-02
| | | | | | | | | | | | | | | | * add RTP Header extension for Client-to-Mixer Audio Level Indication https://tools.ietf.org/html/rfc6464 requires libre from this commit or later: https://github.com/creytiv/re/commit/1544a1e375c76a80084b411d21b0431f95e9cdfb * fix warnings * fix warnings * minor cleanup
* test: added a mock audio-codecAlfred E. Heggestad2017-04-25
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* aucodec: move list of aucodec to struct baresip (#221)Alfred E. Heggestad2017-03-04
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* test: fix build when USE_VIDEO is offAlfred E. Heggestad2016-12-07
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* test: added testcase for call with videoAlfred E. Heggestad2016-12-07
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* use KEYCODE_REL instead of 0x00Alfred E. Heggestad2016-07-31
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* test: fix warnings about unused parametersAlfred E. Heggestad2016-07-31
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* test: add test for sending DTMF-digitsAlfred E. Heggestad2016-07-31
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* add '@' command to switch calls using line numbersAlfred E. Heggestad2016-07-26
| | | | | | | | | | | 1. for a given UA, each call has a unique line number starting from 1 2. the list of calls is sorted in an arbitrary order, but the last list element indicates "current call" 3. the '@' command takes a numeric argument which is the line-number of the wanted call
* added config 'call_max_calls' to limit maximum number of calls per uaAlfred E. Heggestad2016-07-23
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* call: add sequential line-numbers for multiple calls (ref #141)Alfred E. Heggestad2016-07-23
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* add support for rtp_timeout and redialAlfred E. Heggestad2016-07-19
| | | | | | | | | | | | | | | | 1. Added support for RTP timeout. The feature is disabled by default and can be enabled with config "rtp_timeout N" where N is the number of seconds of RTP inactivity. If this is detected, the call is closed with a "special" SIP reason code of 701. 2. Added support for automatic re-connect in the menu module. This can be enabled by setting the 2 config items: redial_attempts 3 redial_delay 5 This work was contributed by Sveriges Radio. Thanks goes to Ola Palm and Jim Eld.
* aucodec: split srate into srate and crate (Clock Rate)Alfred E. Heggestad2016-05-01
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* add test for SIP with DNSAlfred E. Heggestad2016-04-02
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* ua: add UA_EVENT_CALL_CLOSED on ua_hangupAlfred E. Heggestad2016-02-28
| | | | | | thanks to Gary Metalle for testing ref. issue #111
* test: add testcases for call hangup from both sidesAlfred E. Heggestad2016-02-14
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* test: use milliseconds for re_main_timeout valueAlfred E. Heggestad2016-02-14
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* selftest: silent output, add optional verbose -vAlfred E. Heggestad2016-01-09
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* test: remove asserts for ua/callAlfred E. Heggestad2015-12-01
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* test: fix warningAlfred E. Heggestad2015-11-01
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* call: check address-family of incoming SDP offerAlfred E. Heggestad2015-10-29
| | | | this fixes #79
* test: added 2 call testsAlfred E. Heggestad2015-10-29