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Created with Raphaël 2.2.03Jan23Dec222019161513129853129Nov2825242117161198632131Oct28272625211914111074329Sep2827262219161514131211109831Aug3019181711108res_http_media_cache: Do not crash when there is no extensionchan_sip: Remove deprecated module.pbx_app: Update outdated pbx_exec channel snapshots.res_rtp_asterisk: Asterisk Media Experience Score (MES)res_rtp_asterisk: Asterisk Media Experience Score (MES)res_rtp_asterisk: Asterisk Media Experience Score (MES)pbx_app: Update outdated pbx_exec channel snapshots.res_pjsip_transport_websocket: Add remote port to transportres_pjsip_transport_websocket: Add remote port to transportres_pjsip_transport_websocket: Add remote port to transportres_pjsip_transport_websocket: Also set the remote name.pbx_app: Update outdated pbx_exec channel snapshots.res_pjsip_session: Use Caller ID for extension matching.res_pjsip_session: Use Caller ID for extension matching.res_pjsip_session: Use Caller ID for extension matching.pbx_builtins: Remove deprecated and defunct functionality.res_pjsip_sdp_rtp.c: Use correct timeout when put on hold.res_pjsip_sdp_rtp.c: Use correct timeout when put on hold.res_pjsip_sdp_rtp.c: Use correct timeout when put on hold.app_voicemail_odbc: Fix string overflow warning.app_voicemail_odbc: Fix string overflow warning.app_voicemail_odbc: Fix string overflow warning.func_callerid: Warn about invalid redirecting reason.res_pjsip: Fix path usage in case dialing with '@'res_pjsip: Fix path usage in case dialing with '@'streams: Ensure that stream is closed in ast_stream_and_wait on errorstreams: Ensure that stream is closed in ast_stream_and_wait on errorstreams: Ensure that stream is closed in ast_stream_and_wait on errorfunc_callerid: Warn about invalid redirecting reason.func_callerid: Warn about invalid redirecting reason.app_sendtext: Remove references to removed applications.app_sendtext: Remove references to removed applications.res_pjsip: Fix path usage in case dialing with '@'Set CallLog.{i}.X_Vendor_SIPIPAddress for outgoing INVITE after domain name resolvedDocumentation updateUpdate UCI documentation by adding asterisk.sipX.early_mediaPlay stutter dial tone if MWI is enabledUpdate for 20.1.0-rc120.1.0-rc120.1.0-rc1Update for 18.16.0-rc118.16.0-rc118.16.0-rc1Update for 19.8.0-rc119.8.0-rc119.8.0-rc1