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Created with Raphaël 2.2.026Apr117422Mar181311109862128Feb1816151097131Jan25195430Dec21171614873230Nov251918171211987429Oct27262218156528Sep2321201514109873230Aug2926252419161196426Jul2322211916151312982130Jun292523181711108728May27211917141210976430Apr272322212019181615141287643131Mar30292625232120191817151295426Feb25181514121110954127Jan262524232215753131Dec2726232120181713127529Nov2320191512130Oct2524171211954323Sep22212017121187643125Aug24191510876542129Jul282019181530Jun221615111098430May26242322211817161514131265430Apr2927242322201514139874326Mar25242321201998712Jan11120Dec173129Nov26225429Oct2825242216131211975220Sep54131AugAsterisk: Fix a bug leads to tone overlaysdeveldevelFix an issue for checking local Voice mailrelease-6.5release-6.5asterisk: Fix audio files for call waitingasterisk: Fix notification about sip registration status changeApply a patch for endpoint identify change which affects incoming call matchingAsterisk :Caller name presentation should be empty instead of 0 if display name not receivedAsterisk : Refined the caller ID presentation logics for name and numberremove unnecessary patchesAsterisk: config_asterisk.sh - remove the duplicated voice mail call flows in extensions.confAsterisk: Fix a bug for voice mail configurationFix bug for external callbackasterisk: Apply back CallLog changes with fixed regressionsDocumentation: Add asterisk.sipX.directory_number to UCI JSON schema and generated markdown filesAsterisk: Enable and load applications Echo and Recordfreeswitch: fix python locationMerge branch 'openwrt-21.02.2' into develAsterisk: remove the firewall rule settings for SIP and RTP portsrelease-6.4release-6.4asterisk: Revert CallLog changes introduced regressionasterisk: Fix the user in SUBSCRIBE "From" headerAsterisk: change the default tx/rx gain from 4 to 0 as Broadcom suggestasterisk: Remove SIPREFERREDBYHDR variable & fix Referred-by headerasterisk: Changes for remote rtp statistics64r64rasterisk: fixup! Add support for RTCP-XR packets generated by DSP platformasterisk: Add averageFarEndInterarrivalJitter rtp statisticasterisk: Add support for RTCP-XR packets generated by DSP platformAsterisk: Fix a bug in config_asterisk.sh that caused call return failAsterisk: Fix a bug in config_asterisk.sh that caused call return failAsterisk: move uci option dial_out_timeout from section tel_options to numberingplanAsterisk: move uci option dial_out_timeout from section tel_options to numberingplanAsterisk : Documentation update for moving feature codes to prefixinfor64r64Asterisk : Documentation update for moving feature codes to prefixinfoAsterisk: replace the value of facilityaction DND_STATUS with DND_INTERROGATE as per TR-104 in /etc/config/asteriskAsterisk: replace the value of facilityaction DND_STATUS with DND_INTERROGATE as per TR-104 in /etc/config/asteriskMove feature codes in calling_features section to prefixinfo sectionsMove feature codes in calling_features section to prefixinfo sectionsAsterisk: Don't configure firewall rules for SIP and RTP/RTCP ports if the config is incompleteMerge pull request #733 from micmac1/sof-1137-21sofia-sip: update to 1.13.7Asterisk: add input rules in firewall for SIP signaling and RTP/RTCP portsAdd JSON schemas for UBUS, UCI and generated .md files