- The scope of the following features was expanded to include all the Services within the same Customer
- Directory dialing to include extensions in all the services within a customer
- Voicemail distribution list to include extensions in all the services within a customer
- Broadcast extensions scope to include all the services within a customer
- Extension sim ring targets to include all the services within a customer and also to include 10-digit North American PSTN numbers
- Caller ID (both Number and Name) will not be sent out if a user dials a phone number prefixed by *67.
- Call Forwarding No Answer (CFNA) can be configured (either by Admin or a user from C3Voice Web portal Advanced Call Forwarding) with different ring durations for internal (extension dialed) and external (DID dialed) calls, respectively.
NOTE: Currently CFNA ring targets include Broadcast extensions but not Sim ring targets. This shortcoming will be addressed in the next update.
- C3Voice user can make a callback to the caller who left the message during a voicemail session using an advanced option (follow the instructions as prompted), provided that C3Voice received a caller id number that is either a 10-digit North American phone number or a 4-digit C3Voice extension.
- C3Voice admin can configure a 0-escape target (an extension) for a Service and also an individual extension (which overrides the Service target) so that a caller can escape from a voicemail session to the specified target by pressing the 0 key.
- From a user Web portal application a C3Voice user can maintain a number of voicemail distribution lists. Each distribution list consists of valid extensions and corresponding user names within the service. A user can upload a message from a local directory (e.g. saved voicemail message) and send it out to one or more voicemail distribution lists.
- C3Voice admin can enable sending out a text messaging notification to a specified address when a new voicemail is left for an extension.
- Add a C3Voice Admin tool that enables the following (for RS provisioning).
- Search for DIDs added/deleted for multi-selectable (including all) service contexts since a specific date/time
- Export the results to CSV files (separate files for Added and Deleted DIDs) with (From, To) range columns
- Enable C3Voice Admin and also local users to configure Sim ring targets for internal (4-digit dialing) calls
Voicemail Usage Policy:
System default values have been set as follows:
- Max Messages per user=50
- Max Message Length=300 sec
- Delete after fwd to email=No.
Admin may override the above for different services and extensions within the service from the Service and Extension provisioning windows, respectively.
CNAM:
For a new Service the default value will be the Service’s full name.
For existing Services it remains to be empty. That is the current value, i.e. no CNAM.
Admin may assign any value to the Service CNAM as desired by the customers.
Admin may also override the Service CNAM for a specific extension by its Line Description.
Calling other IVRs define in Service from within an IVR Editor
Any IVR defined in a Service can be called (more specifically Goto) by other IVR from within an IVR Editor.
- VCC CC Admin can Activate or Inactivate queues from Admin UI. Inactivated queues will retain all the historical reporting and call recordings, but can no longer receive calls. Inactivated queues will not be waited during VCC restart.
- VCC queue voicemail calls are diversion inhibited, i.e. they will not be diverted to an agent's personal voicemail box even if s/he did not answer the call. Forcing an agent to Unavailable for Not Responding rule also applies to queue voicemail calls. These updates will prevent an agent, who forgot to sign out during off-hours, from receiving repeated queue voicemail calls that eventually flood out his/her personal voicemail box.
Voicemail storage backend has been migrated from Zimbra mailserver to Asterisk file system.
All customer voicemail accounts and saved contents have been moved.
Reason for migration:
Random Asterisk crash caused by the IMAP Lib that connects Asterisk and Zimbra server (as a voicemail storage server).
IMAP Lib is not an official Digium component.
It is a contribution by University of Washington.
It had a number of shortcomings to begin with and we managed to overcome one after another.
However we could not fix the memory segmentation fault and resulting random crash after spending so much effort.
"C3 Click to Dial" is released internally within Vantage. Download site URL ishttp://192.168.200.150:4445/download
This is an exact duplication of the forthcoming official public download sitehttp://www.vantageip.com/download
After incorporating user feedback and pending approval of EUSLA by General Counsel’s office, the public download site will be good to go.
- Admin has an option to select one of the three call waiting indicator sound: beep (default), ring or silent, for the Polycom phones belonging to a Service
- If a hunt group does not have a terminal step (Goto/IVR/Voicemail) and all steps (ring with Dial application) were unsuccessful (timeout or trunk busy), the caller will hear a busy tone
- C3 Admin can block sending a caller ID for a specific extension by configuring its external caller ID empty.
- An SLA can be mapped to an instance of a BANG broadcast/shared extension.
- If a caller dials a non-existing extension, s/he will hear an announcement saying “that extension does not exist.”
- C3Voice Conference Enhancement:
- Change reservationless conference to start when the first participant joins
- Add read-only (for copy and paste to any desktop application) text box for conference info
- Add RESTful API for conference add/edit/delete that can be used by any 3rd party application
- C3Voice Enhancement:
- Add Silence as an option for Music on Hold (MOH)
- In Phone Configuration Server (CF) for a Service specific directory, create symbolic links to default firmware file locations only if symbolic links do not already exist
- In CF change directory and file permissions as requested by Vantage Engineering
- Enable SLA mapping to an old BANG shared extension
- Comment out minsecs = 3 in voicemail.conf so that any length of voicemail is recorded
- Comment out operator = yes in voicemail.conf so that even if a caller presses digit '0' accidentally the call will not be released
- C3Voice Shared Line Appearance: Enable it to be callable from DID and HG steps and IVR
- C3Voice Phone Configuration Server: Make message button mapping and phone display logo a Service parameter
- C3Voice enhancement and adjustment: Make voicemail module scalable and stable; Support Ad hoc call recording for external calls; Make diversion inhibition having higher precedence over call forwarding; Restrict voicemail forwarding outside the Customer’s context.
- C3VCC enhancement and adjustment: Add a routine that clears “dead” queue calls caused by a missing event; Properly handle Hold/Un-hold events from phones other than Cisco’s with the current SIP load
- SLA support
- Service specific MoH support
- Service specific timezone incorporated into phone configuration file creation
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Fix the Asterisk voicemail module memory leak that caused frequent crash and restart: |
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Add VCC queues to IVR editor |
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Conference bridge enhancement:
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Make C3Voice including Asterisk DB backup script following the similar we did for VCC |
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Place two professional quality C3 Logos into Web portal pages as specified by the stakeholders.Extension Dialing Ring Duration: |
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Add templates for the following phone models: SPIP 550 and 650 (see the attached for the full list). |
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Secure the configuration file ownership per sprint review suggestion |
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Add templates for the following phone models: SPIP 450, 650 with Expansion Module, 670 and 6000 |
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Apply the same DB backup script we did for VCC (for BW) |
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Add JMX Manager to C3VCC |
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Provide individualized 911 caller id that overrides the default Service caller id |
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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. |
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New conferencing server architecture to prevent depletion of conference rooms by reservationless conferences. |
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Conferencing room provisioning safeguards: |
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Major bug or deficiency fixes: |
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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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- Open source IMAP client TCP sockets (to Zimbra VM server) depletion problem has been fixed
- Multiple time zone support added
- Admin selectable Emergency caller id added to individual phone line
- First time C3Voice web-portal users are forced to change the default password
- DID and Hunt Group Steps became editable
- New conferencing server architecture implemented to prevent depletion of conference rooms by reservationless conferences
- Removed + from the outbound PSTN call dial pattern match. This prevents DID calls (with + prefix) from matching the outbound PSTN call pattern and erroneously becoming an outbound call.
- Made ignoresdpversion=yes in Asterisk sip.conf. This makes Asterisk not to enforce SDP session version check so that non-complying Cisco 79xx phone's re-INVITE (hold and un-hold) would not result in one-way audio.
- Added a number of minor fixes
- Provided daily support to Vantage Engineering
- (Extra) Changed VCC reporting data feed to SC per changes in SC data handling
- Uploaded production C3VCC and fully regression tested, except for call recording