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Markus Gothe
openwrt-telephony
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3ada8545c1f082a39c2a68d3c9221965154104df
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20
4238
4244dnd
64r
alternative-call-feature
alternative_call_feature
ast1811
ast_cleanup
calllog
devel
default
protected
devel-old
feature_4244_dnd
feature_4599_howler
for-14.07
for-15.05
lede-17.01
master
openwrt-18.06
openwrt-19.07
owrt_ast_18_1_1
pj2101
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Asterisk: Fix a bug leads to tone overlays
devel
devel
Fix an issue for checking local Voice mail
release-6.5
release-6.5
asterisk: Fix audio files for call waiting
asterisk: Fix notification about sip registration status change
Apply a patch for endpoint identify change which affects incoming call matching
Asterisk :Caller name presentation should be empty instead of 0 if display name not received
Asterisk : Refined the caller ID presentation logics for name and number
remove unnecessary patches
Asterisk: config_asterisk.sh - remove the duplicated voice mail call flows in extensions.conf
Asterisk: Fix a bug for voice mail configuration
Fix bug for external callback
asterisk: Apply back CallLog changes with fixed regressions
Documentation: Add asterisk.sipX.directory_number to UCI JSON schema and generated markdown files
Asterisk: Enable and load applications Echo and Record
freeswitch: fix python location
Merge branch 'openwrt-21.02.2' into devel
Asterisk: remove the firewall rule settings for SIP and RTP ports
release-6.4
release-6.4
asterisk: Revert CallLog changes introduced regression
asterisk: Fix the user in SUBSCRIBE "From" header
Asterisk: change the default tx/rx gain from 4 to 0 as Broadcom suggest
asterisk: Remove SIPREFERREDBYHDR variable & fix Referred-by header
asterisk: Changes for remote rtp statistics
64r
64r
asterisk: fixup! Add support for RTCP-XR packets generated by DSP platform
asterisk: Add averageFarEndInterarrivalJitter rtp statistic
asterisk: Add support for RTCP-XR packets generated by DSP platform
Asterisk: Fix a bug in config_asterisk.sh that caused call return fail
Asterisk: Fix a bug in config_asterisk.sh that caused call return fail
Asterisk: move uci option dial_out_timeout from section tel_options to numberingplan
Asterisk: move uci option dial_out_timeout from section tel_options to numberingplan
Asterisk : Documentation update for moving feature codes to prefixinfo
r64
r64
Asterisk : Documentation update for moving feature codes to prefixinfo
Asterisk: replace the value of facilityaction DND_STATUS with DND_INTERROGATE as per TR-104 in /etc/config/asterisk
Asterisk: replace the value of facilityaction DND_STATUS with DND_INTERROGATE as per TR-104 in /etc/config/asterisk
Move feature codes in calling_features section to prefixinfo sections
Move feature codes in calling_features section to prefixinfo sections
Asterisk: Don't configure firewall rules for SIP and RTP/RTCP ports if the config is incomplete
Merge pull request #733 from micmac1/sof-1137-21
sofia-sip: update to 1.13.7
Asterisk: add input rules in firewall for SIP signaling and RTP/RTCP ports
Add JSON schemas for UBUS, UCI and generated .md files
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