Changes between Version 1 and Version 2 of DeveloperNotesMartin


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Timestamp:
12/07/09 17:41:48 (16 years ago)
Author:
martin
Comment:

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  • DeveloperNotesMartin

    v1 v2  
    1515 * review existing scripts
    1616 * collect all properties into a data dictionary
     17 * central site configuration - need push/pull infrastructure in device
    1718
     19== Dial in Features ==
     20 Features will be configured in dial plans which will be based on customer choices
     21
     22 * calls (either voip or pstn) configured in asterisk to go to a line or a dial group (multple simultaneous rings)
     23 * PABX features (depending on customer purchase selections) may be available. With PBX option gives options to dial one line etc without PABX, all lines will ring.
     24 * voicemail mode may be turned on or off
     25  * set message
     26  * define email storage destinatino
     27  * night mode (all calls to voicemail)
     28 * pickup groups (multiple handsets in one group)
     29  * optionally one phone only rings
     30  * members of the group can pickup calls
     31 * permanent redirection feature - always redirect calls to external number
     32 * call routing if call is to 1300/1800/normal number
     33 * no support for 1900 numbers
     34 * LATER IVR (interactive voice routing)
     35
     36== Dial out features ==
     37 * use first free line
     38 * caller id on/off setting (pstn)
     39 * voip similar
     40 * enter code to override setting
     41 * use multiple registrations for line-based billing
     42 * LATER use account-code prefix on dialed number for billing
     43 * localisation of dialing (automatic add prefix to provide 8-digit dialing)
     44 * monitor internet interface and route to PSTN if necessary (user should be notified)
     45 * if no lines available play notification
     46 * international call barring
     47 * rated service barring (1300/1900 bars)
     48 * manual selection of PSTN
     49 * direct dial to extension
     50
     51== Call handling features ==
     52 * LATER call recording
     53 * conferencing
     54  * dial extension or external line, press code to join conference
     55  * one active conference per device
     56  * LATER multiple conferences per device
     57 * call hold and resume
     58 * LATER call parking
     59 * onhold music preconfigured
     60  * LATER upload/customise onhold music
     61
     62== Other items for consideration ==
     63 * SER forks - choose a platform
     64  * OpenSER
     65  * Camino
     66  * OpenSIPS
     67  * etc
     68 * Routing
     69 * Selection of database
     70 * Cleanup billing and configuration API
     71 * Reseller concept is still desirable for the future even if not required initially
     72 * Capped plans are desirable but not currently supported
     73
     74== Provisioning process ==
     75 * customer selects product
     76 * order is configured and placed with company
     77 * work ticket is raised
     78 * provisioning occurs via commandline initially, eg provision-unit N   (where N is the work id)
     79  * collect order data
     80  * generate configuration kit
     81  * flash new unit in test configuration
     82  * load test scripts
     83  * perfom tests (utilise asterisk api for automate the process, using bench units and customer unit)
     84   * receiving calls
     85   * making calls
     86  * heat unit via long ring time process
     87  * retest
     88  * reprovision for shipping
     89  * generate provisioning report
     90 * LATER GUI/web system to provision units
     91 * configure the voip (billing system) platform by creating customer account etc
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