Wednesday, August 7, 2013

ShoreTel v13.1 SIP and BIAMP

At the beginning of the year we expanded and built a new set of training/conference rooms.  We employed GrayBow ( to handle the Audio/Visuals of the room.

One such feature is the use of Crestron, Revolabs for conference phones.  We have had several issues with programming the system to dial, maintain connections longer then 30 some minutes.  GrayBow has been great sending techs out and working with the vendors to resolve the issue (hoping the new firmware release fixes all the issues).  Below documents how we had to configure our ShoreTel System to acknowledge the BIAMP card in the new system.

ShoreWare Director Configuration

  1. Access ShoreTel Director via Web Browser (http://[DirectorIP]/shorewaredirector/login.asp)
    • Default Login: admin / changeme
  2. Create a SIP Profile
    1. Navigate to Administration > IP Phones > SIP Profiles
    2. Click New
      1. Name: [We used BIAMP]
      2. User Agent: biamp.*
      3. Priority: 100
      4. System Parameters:
        • OptionsPing=0
        • SendEarlyMedia=0
        • MWI=none
        • 1CodecAnswer=1
        • StripVideoCodec=0
      5. Custom Parameters:
      6. Click Save
  3. Create a Codec List (We skipped as already configured for other SIP System)
    1. Navigate to Administration > Call Control > Codec Lists
    2. Click New
      1. Name: [We used BIAMP_codecs]
      2. Under Choose Codecs Select: AAC_LC/32000
      3. Click Add >>
      4. Under Codec List Members Select: AAC_LC/32000
      5. Click Move Up >> until at the top
      6. Click Save
  4. Configure Switch Port (Again we skipped as already configured for other SIP System)
    1. Navigate to Administration > Platform Hardware.. > Voice Switches.. > Primary
    2. Select desired switch name to configure SIP on
    3. Determine desired port > Change Port Type to 100 SIP Proxy
      • Check your switch manual for more details
    4. Update Description for easy Identification later
    5. Click Save
  5. Configuring Site (Again we skipped as already configured for other SIP System)
    1. Navigate to Administration > Sites > [Desired Site Name]
    2. Update Following Sections:
      1. Bandwidth
        1. Admission Control Bandwidth: 1544 kbps 
        2. Intra-Site Calls: LS_Codecs
        3. Inter-Site Calls: LS_Codecs
        4. FAX and Modem Calls: Fax Codecs - High Bandwidth
      2. SIP Proxy
        1. Virtual IP Address: [Designate static IP for SIP Proxy]
        2. Proxy Switch 1: [Select defined switch]
        3. Proxy Switch 2: [Select defined switch]
    3. Click Save
    4. Create SIP User Extension
      1. Navigate to Administration > Users > Individual Users
      2. Click Go to the right of Add new user at site [desired site name]
      3. Complete Following Fields
        1. First Name: [RoomNames]
        2. Number: [System may auto-populate otherwise enter desired extension]
        3. License Type [Select Extension and Mailbox]
          • You can come back after completion to set to Extension-Only
        4. Primary Phone Port > IP Phones: Any IP Phone
        5. Click Save
      4. Scroll to Bottom and update following:
        1. Client Username: [Keep things simple same as First Name]
        2. SIP Password: [Define password / By default nonassigned]
        3. Click Save
      5. Click Save

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