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- [PS-1152] - Report in CDR the trunk name, if any, used to route the call
- [PS-1137] - Manage two different VoIP account with same phone number but different security model (????!!!!)
- [PS-1099] - Show contact name for incoming call on desktop phones
- [PS-1109] - Document "account locked" option in PrivateServer
- [PS-1179] - Long list of registered users
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- [PS-1117] - configurable list of groups
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- [PS-1125] - Add shell script to monitor connection quality
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- [PS-1134] - Improve user experience on RTP timeout
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- [PS-1142] - Set a datetime format in asterisk logs
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- [PS-1146] - Audio tones and messages play is configurable depending on caller
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- [PS-1163] - Clean up yum repository list
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- [PS-1169] - Set meaningful names for
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- PrivateServer services
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- [PS-1176] - Invalidate licence per user basis upon ACTIVATE
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- request
Fixed Bugs
- [PS-486] - Re-invite a user in a 3-way calling hangs up the call
- [PS-1097] - Call duration is zero for calls routed on SIP trunk
- [PS-1115] - Hide the user lock check box unless the lock is set
- [PS-1140] - Missing caller name is not filled with owner for calls directed to trunk
- [PS-1141] - Call to not-existent user reports to clients "temporary unavailable" SIP-480
- [PS-1144] - When trunk cannot secure call, a temporary unavailable message is reported to clients
- [PS-1145] - Wrong or missing DTMF relayed to trunk
- [PS-1155] - Translate received SIP 480 with SIP 480, instead of SIP 503
- [PS-1158] - Blind transfer setting is replicated in ast_config table
- [PS-1175] - Group delete fails
- [PS-1182] - Changing the IP address on the server triggers an authentication error afterward
- [PS-1183] - NTP configuration doesn't change the messages after a new action has been performed