Personnel

  1. Contact Person: Rob Tuck
  2. Software Development: Rob Tuck
  3. Hardware Development: Rick Crispin

Schedule for Voicemail -> email interface

This section of the schedule pertains only to providing voicemail to email capabilities to faculty and students. This is an initial stage of the internal VOIP-based extension system scheduled for deployment by September 2007.

Costs

We need to keep track of all costs requiired and savings obtained to establish this new phone service.

Billing

We need to couple the Asterisk records to the SAS billing info. This requires serious software development and constrains when a full internal VOIP will be ready.

Notes

Establishing voicemail -> email is being done at a much faster pace than anticipated. Other commitments may need to be reduced.

The initial test setup should use a dedicated set of 4 lines in a hunt group to handle student/faculty voicemail (6450 -> 6451 -> 0592 -> 2685). The last line (2685) will always be busy. Peak student and faculty use must be measured as well as the number of callers who get a busy signal. The last number in the group of four will allow us to keep track of how many calls are refused by a three line service - the fourth line is only for data collection. Acceptable rejection rates (e.g., 1%, 5%, 10%) need to be established rejection rates need to be monitored and reported.

VOIP phones must be able to run on the same connection network as a work station.

Initial test for "typical" VOIP will use four lines in a dedicated hunt group (6430 -> 3167 -> 2558 -> 4787). Unlike the voicemail setup all lines will be availalbe. We need to keep call records but will not be able to record rejection rates since there is no "always-busy" line at the end of the hunt group.

Student extensions

Faculty and staff keep their extensions

Logistics of phone line location

Logistics of voice mail

Attendant protocols

Open Issues

Virtual phones?

Incoming caller ID

Outgoing caller ID

Private list of all cell phone numbers

POE: power over ethernet

Known Problems

  1. OTRS queue is needed
  2. Voice mail playback is garbled using VOIP phone
  3. call is delayed 2-3 rings compared to partner system
  4. Echo on VOIP phones
    • 9/26/06: does not affect VOIP/VOIP calls, only affects VOIP/POTS calls.
    • 9/26/06: tuning Asterisk software makes significant improvement
  5. auto attendant not present, spelling name fails, 0 doesn't go to operator
    • 9/26/06: # to invoked attendent disabled
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