- Apr 26, 2022
-
-
Wenpeng Song authored
Fix a bug that two call progress tones are being played simultaneously
-
- Apr 11, 2022
-
-
Add Hangup() at the end of the Voice mail extension in order to avoid calling the local phone number after closing the voice mail application by pressing "#".
-
- Apr 07, 2022
-
-
Grzegorz Sluja authored
-
- Apr 04, 2022
-
-
Grzegorz Sluja authored
-
Do some manipulations with users with a UK telephone number, e.g. +441473000281.
-
- Mar 22, 2022
-
-
Hemlata authored
-
- Mar 18, 2022
-
-
Hemlata authored
-
- Mar 13, 2022
-
-
Sukru Senli authored
-
- Mar 11, 2022
-
-
- Mar 10, 2022
-
-
Remove the unused asterisk.extensionX.boxnumber reference
-
Valid external callback enable config: asterisk.set1.ccbs_enable='1' asterisk.set1.internal_service='0'
-
Grzegorz Sluja authored
-
- Mar 09, 2022
-
-
Hemlata authored
-
- Mar 08, 2022
-
-
- Mar 06, 2022
-
-
Sukru Senli authored
-
Sukru Senli authored
-
- Mar 02, 2022
-
-
Yalu Zhang authored
/etc/firewall.sip is also removed. Rules for those SIP and RTP ports shall be configured in firewall directly.
-
Grzegorz Sluja authored
-
- Mar 01, 2022
-
-
Grzegorz Sluja authored
-
- Feb 28, 2022
-
-
Yalu Zhang authored
-
- Feb 18, 2022
-
-
Grzegorz Sluja authored
-
- Feb 16, 2022
-
-
Grzegorz Sluja authored
-
Grzegorz Sluja authored
-
Grzegorz Sluja authored
-
- Feb 15, 2022
-
-
- Feb 10, 2022
-
-
- Feb 09, 2022
-
-
Hemlata authored
-
Asterisk: replace the value of facilityaction DND_STATUS with DND_INTERROGATE as per TR-104 in /etc/config/asterisk
-
- Feb 07, 2022
-
-
The purpose is to remove vendor extensions in TR-104 and use PrefixRange and FacilityAction instead.
-
- Feb 01, 2022
-
-
Yalu Zhang authored
- Also change the listening TCP port from 5061 to 5060 which is the same as UDP as per RFC3261 by default. - Remove firewall rules when stopping asterisk
-
- Jan 31, 2022
-
-
micmac1 authored
[21.02] sofia-sip: update to 1.13.7
-
Sebastian Kemper authored
Contains DOS fix, see [1]. Converted to AUTORELEASE. [1] https://github.com/signalwire/freeswitch/issues/1518 Signed-off-by:
Sebastian Kemper <sebastian_ml@gmx.net> (cherry picked from commit f04a5a23)
-
- Jan 25, 2022
-
-
Yalu Zhang authored
Also fix an error in /lib/voice/config_asterisk.sh for default RTP port ranges.
-
- Jan 19, 2022
-
-
Hemlata authored
-
- Jan 05, 2022
-
-
Hemlata authored
-
Grzegorz Sluja authored
-
Grzegorz Sluja authored
-
- Jan 04, 2022
-
-
micmac1 authored
[21.02] asterisk-chan-dongle: update to the latest git HEAD
-
Shaleen Jain authored
The upstream commit 6073c91fcf0a46a1525d500c274fa5ab96af7dda broke non-quectel devices due to the channel driver unable to complete initialization. This is fixed in the latest upstream commit. Signed-off-by:
Shaleen Jain <shaleen@jain.sh> (cherry picked from commit 409e7977)
-
The bind directive in pjsip.conf has never been working correctly. This commit fixes the regexp to it matches the layout in the template config.
-