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Setup Analog Stenophone Environment using CUCM 11.5

Paki
Beginner
Beginner

Dear Members,

Hope this message find you well.

SCENARIO 1:

Please recall an old PSTN communication system where an employee / staff used to have an analog stenophone set (as one in attached pictures) with 3 buttons on the phone connected with each other through jumpers. These were exchange, hold & steno buttons. The employee / staff phone & officer phone were connected through an analog PSTN Rosette Box using two separate drop cables.

  1. When an employee / staff used to receive a call from an outside PSTN on his PSTN number (& not on extension number), he used to press the hold button & then inform the officer that he had a call & then he used to release the hold button & pressed the transfer button. The call was established between calling party & the officer. After the conversation & when the officer hung-up the phone, the line was automatically shifted to the employee / staff.
  2. In addition to this, the employee / staff had an option to transfer the line directly to the officer so that if he / she wasn't available, the officer could directly received & made a call.

Since we have VoIP phones now-a-days, we have call transfer button hard coded on the phone.

SCENARIO 2:

Now imagine a scenario where there is a CUCM 11.5, a Cisco Router as VoIP SIP Gateway & 10 x Cisco 7821 VoIP phones installed & connected to form a VoIP communication system.

QUESTION:

The question is how we can establish above-mentioned old analog PSTN functionality (Scenario 1(a & b)) on VoIP system (Scenario 2)? So that the call is directly received on employee / staff phone & the officer isn't disturbed at all.

OPTIONS AVAILABLE:

  1. Busy Lamp Field (BLF):     Share one number between two phones but the hitch is that when employee / staff phone will ring, the officer phone will also ring. This will disturb the officer.
  2. Hunt Group / Line Hunt:    Incoming call from outside PSTN number will first arrive at employee / staff phone, if phone isn't answered after a certain criterion, the call will be forwarded to the officer number or next number. Again, the officer will be disturbed. He can be either in meeting or he is a high designation officer who should not receive a call directly from outside PSTN number (can be an official protocol requirement). 
  3. After Hour Call Setup:     What if after office timings, the employee / staff isn't available on seat? If the employee / staff is not available on seat, then officer cannot attend call in his room. He will have to go the employee / staff desk because BLF & Hunt Group are neither configured nor can be configured.

LIMITATIONS:     

  1. Hunt Group / Line Hunt cannot be configured on two VoIP phones using same extension / landline number. (If you think it can be done, let me know please).
  2. Cisco 7821 is a dual-line phone. At line 1, extension number is configured for internal communication. At line 2, VoIP SIP landline number is configured for direct communication by the officer. There is no extra button at which line could be transferred.

Your response is highly appreciated.

Thank you.

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