Blog

1. Potential dialplan corruption problem (missing DIDs) has been fixed.  
2. Personal Speed Dial List Web App has been added. A user can configure 2-digit speed dial list (00-99); can click-to-dial from the app; can dial *1XX (00-99) from his/her MAXvoice phone.
3. Phone restart flag behavior has been changed such that it is presented only if the configuration file was changed but the primary extension was not. The latter requires a manual restart.
4. Delete Voicemail Message behavior has been changed such that it takes effective only if an Email Forward destination is configured. 

1. Asterisk voicemail configuration has been migrated to Asterisk Realtime Architecture (ARA), meaning no more voicemail reloads.
2. All existing customers' voicemail configuration files have been migrated to matching MAXvoice DB table records under ARA.
3. A new Web App Click-to-Dial extension has been added. Using this Web App a user can designate a specific instance of his/her extension to be used for click-to-dial.
4. Click-to-Dial API internals but not externals have been modified to support the Click-to-Dial extension Web App. 

• C3Voice Asterisk Servers: updated to 1.8.30.0  but without loading a kernel timer module, i.e. noload=>res_timing_timerfd.so
• C3VCC: updated with 1.8.30.0 compliant AMI Redirect Action modification and support for agents using a broadcast extension - agents are prompted to select a specific instance
• C3App: updated with MAXadmin bug fixes related to Caller ID Blacklist and Block Sending Caller ID Edit in the Service page

1. MAXadmin will include all customer provisioning Web apps - IVR Manager, Hoteling Phone Manager, Hunt Group Step Config, SAC Config
2. Dial by Name IVR will support both search by First and Last Names
   
3. Admin UI phone tab entries will be sortable and searchable by the column values
4. OBP address field will accommodate FQDN
5. Panasonic phone configuration bug that resulted in missing SIP lines setting was fixed.
6. C3VCC call transfer bug that resulted in calls being sent to voicemail when no agent joined, irrespective of configured destination, was fixed.

1. Asterisk Real-time Architecture: Asterisk SIP Peers (phones and extensions) configuration was changed from file-based to DB-based. There is no need for SIP reload after adding/deleting/changing phones and phone extensions
2. Caller name (CNAM) lookup support using TNSI CNAM Dip service was added
3. Minor Enhancements were made to IVR Editor
4. Logo Changes: New logos MAXcoreplatform, MAXvoice and MAXadmin replaced old logos

1. Install and configure SNMP monitoring support for Asterisk and Digium MIBs
2. Add support for individual IVRs defining their own Schedules instead of sharing one for the Service.

   If IVR’s own schedules are defined, they will replace the Service’s schedules in its entirety, i.e. no partial substitution or addition.

3. Add support for customer phone display logo upload for the Polycom phone models: IP450, IP550, VVX300, VVX400, VVX500, VVX600

Image Format Details

Polycom VVX Series

Image Format: JPEG (Note: Progressive/Multiscan JPEG not supported)

Polycom SoundPoint IP Series

Image Format: JPEG (Note: Progressive/Multiscan JPEG not supported)

Resolution Recommendations

The recommended image format is JPEG. The file size must be less than 600KB.

The image resolution should not exceed the highest supported by the lowest end phone.

For example, if a Service has VVX 300, VVX 600 and IP 550 phones, the image resolution should be no greater than 208x104, the highest supported by the lowest end phone VVX 300.

1. MAXadmin - a provisioning Web application for Local Admin has been added to the list of user Web applications.
2. Custom phone display logo upload for Cisco 7940 and 7960 phones has been added (90x56 pixels, 8-bit per pixel, bmp file).
3. IVR generation template makes the menu choice repeats up to 5 times. NOTE: Existing IVRs need to be modified manually.
4. Schedule and Holiday items are listed in a chronological order.
5. Default MWI tone for Polycom phones was changed to silence.

1. Asterisk SIP compliance enforcement is normalized (pedantic=no changed to yes). It is possible because all the Cisco phones have been upgraded to 8.7, which complies to the latest SIP spec.
2. In RS, a suppressed logic that detects calls to a destination internal to C3Voice and sends it back to C3Voice is removed. It is possible because of item 1 above.
3. Phone States used in a number of Admin pages are updated by Asterisk AMI events passed by C3VoiceProxy. SIP SHOW PEER is used only when a specific phone is selected.
4. Soft Attendant Console application is stabilized and performance vastly improved.
5. Sending Caller ID is customized as below:
   1. Admin can block sending caller id for all external calls from any line in a Service
   2. Admin can block sending caller id for all external calls from a specific line in a Service
   3. Admin can configure different caller ids for external and internal calls
6. Configuration for Polycom SPIP 650 and 670 phones with 1, 2 and 3 expansion modules (sidecars) is supported.

1. C3Voice Hoteling
   1. Admin provisioning of two new phone types: Hoteling, Virtual
   2. Admin assigning a virtual phone to a user
   3. User association/disassociation of a virtual phone to/from a hoteling phone of the same model from Web portal
   4. C3Voice automatic disassociation of all virtual phones from hoteling phones at the end of a day   
2. C3Voice Hoteling Phone Manager Application: User of the application, typically Customer Local Admin, can view and disassociate any virtual phone from a hoteling phone defined in the Service