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  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.

 

 

 

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