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authorAlfred E. Heggestad <aeh@db.org>2016-03-12 18:24:21 +0100
committerAlfred E. Heggestad <aeh@db.org>2016-03-12 18:24:21 +0100
commit1b10113475504cec501b2f6d2e18204e635d871f (patch)
tree5a7c4c88ddf89b5781cd0cd84df356e43d098349 /docs
parent8120bb5a4e7d1d2da630cf901a1555bc13dfba1b (diff)
baresip release v0.4.18
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+2016-03-12 Alfred E. Heggestad <aeh@db.org>
+
+ * Version 0.4.18
+
+ * GIT URL: https://github.com/alfredh/baresip.git
+ * GIT tag: v0.4.18
+ * NOTE: Requires libre v0.4.14 or later
+ Requires librem v0.4.7 or later
+
+ * baresip-core:
+ - call: fix SIP INFO with dtmf-relay (thanks Gary Metalle)
+ - ua: add event UA_EVENT_CALL_CLOSED for ua_hangup()
+
+ * selftest:
+ - add tests for answer a call and hangup
+
+ * Modules:
+
+ * alsa: fix potential crash (thanks Gary Metalle)
+
+ * audiounit: fix compilation for iOS (issue #91)
+
+ * avcodec: fix compilation for FFmpeg 3.0
+
+ * avformat: fix compilation for FFmpeg 3.0
+
+ * gtk: always handle incoming calls (thanks Charles Lehner)
+
+ * h265: fix compilation for FFmpeg 3.0
+
+ * menu: add config 'menu_bell off/on' to enable Bell alert
+ add command 'A' for switch audio device (thanks AlexMarlo)
+
+ * v4l2_codec: add list of encoders (fixes #99)
+
+
2016-01-17 Alfred E. Heggestad <aeh@db.org>
* Version 0.4.17