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Provide individualized 911 caller id that overrides the default Service caller id
Provide a new “Emergency Caller ID” field (mandatory) for individual phone line configuration as follows.
The field will have 3 selectable choices: Service Caller ID (default selection), External Caller ID (show only if configured), Other (for user input field, if selected this field cannot be empty) |
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Force a new user to change the password after an initial log in with a default password |
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Fix a critical bug in IMAP Client Lib that it does not release TCP sockets when done. In the future in order to address scalability also make it open sockets for only a number of accounts at a time, close them when done, then open sockets for other accounts, close them when done, etc. until all accounts updates are done. |
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Make Steps (used in DID and Hunt Group) editable.
Currently Steps can be deleted but not editable.
A Ring step may include quite many numbers (in case of sim ring).
Any change in the sim ring list now requires deleting the whole step and then retype or copy/paste all the phone numbers again. |
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New conferencing server architecture to prevent depletion of conference rooms by reservationless conferences.
If conference is enabled for a service, present a new parameter “Reservationless room” (in addition to “room”). Make the max value 50 for now. It may increase or decrease in the future.
Set aside rooms exclusively for reservationless conferences. They cannot use rooms for scheduled conferences. By the same token scheduled conferences cannot use rooms for reservationless conferences.
There will be no change in room allocation logic, either scheduled or reservationless. The only difference is they use their own rooms.
Add safeguarding:
If a new reservationless conference creation exceeds the assigned reservationless rooms, bring up a warning pop-up and do not accept it.
If a new scheduled conference creation exceeds the assigned scheduled rooms (anytime during the conf period plus minus 15 min grace and buffer periods), bring up a warning pop-up and do not accept it.
If a reserved conference does not end after a grace period (because there is still a participant) but the same conference room is scheduled to be used by another conference, forcibly end the overdue conference. |
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Conferencing room provisioning safeguards:
Do not allow the reduction in the number of reservationless conference rooms if the number becomes less than the currently configured number of reservationless conferences at the time.
Do not allow the reduction in the total number of conference rooms if the number for scheduled conferences becomes less than the maximum number of the concurrently scheduled conferences at the time. |
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Major bug or deficiency fixes:
Remove + from the outbound PSTN call pattern match. This prevents DID calls (with + prefix) from matching the outbound PSTN call pattern and erroneously becoming an outbound call. Also remove +1 from calling number when deliver it to the called phone, so that the user can do call back to the calling number.
Make ignoresdpversion=yes in Asterisk sip.conf. This makes Asterisk not to enforce SDP session version check so that non SDP session version complying Cisco 79xx phone's re-INVITE (hold and un-hold) would not result in one-way audio. |
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Add Status and Audit functions to each Queue (for VCC) just like phone extensions |
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C3Voice will generate a phone specific configuration file (identified by a phone's MAC address in the file name) based upon 1) a template specific to the phone's model, 2) customer/service specific parameters the phone belongs to, e.g. Outbound Proxy and 3) individual phone's parameters, e.g. extension, display name, etc. The generated config file will be copied to the appropriate directory in the configuration server. For phones using HTTP download, the directory will be service specific, e.g. /var/prov/http/vantag-vantag. For phones using TFTP download, the root directory will be shared by all services, e.g. /var/prov/tftp. See the attched documents for further details. |
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Previously called firmware files (sip.ld, sip.ver, 000000000000.cfg, sip.cfg) will be manually loaded so firmware update need to be removed from C3Voice. When new configuration or change to an existing configuration is initiated by super-user C3Voice generates and copy the above mentioned 4 configuration files to the network mounted prov directory in the configuration server. |
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Expand the scope of CC from Service to Customer
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Manually migrate Vantage Learning CCs and conduct exhaustive testing |
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