Cisco SPA942 (SDSC)

Description

This resource allows to connect a Cisco SPA942 IP phone.

Release notes

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Resource configuration interface

Resource configuration parameters

• Dialplan: String compatible with the digit map feature of MGCP (cfr RFC3435) which indicate when auto-dial should be triggered
• SIP server feature:
  • SIP reinvite: Indicate if SIP reinvite should be enabled, disabled or should not overwrite the global SIP reinvite setting
  • Qualify SIP devices registration: In order to monitor the SIP device availability, the Telephony server can send SIP OPTIONS messages (every 60s). If the device does not respond within the defined delay, it will be considered as off-line. Possible values are:
    • Default: Keep the default setting defined in the Telephony server module (for example Asterisk 1.2.x)
    • No: Disable Qualify SIP registration for this device
    • Yes: Enable Qualify SIP registration for this device, with a default timeout value of 2000ms
    • Suggested number: Enable Qualify SIP registration for this device, with the chosen timeout value (expressed in milliseconds).
  • Force specific codec: Indicates if the codec below should be used or if the global codec list should be used
  • First codec: Preferred codec to use during call setup
  • Second codec: First fallback codec to be used during a call setup
  • Third codec: Second fallback codec to be used during a call setup
  • Registration refresh period: period after which the phone must refresh its SIP registration
• Advanced options
  • Network Jitter Level - Determines how jitter buffer size is adjusted by the SPA942. Jitter buffer size is adjusted dynamically.The minimum jitter buffer size is 30 milliseconds or (10milliseconds + current RTP frame size), whichever is larger, for all jitter level settings. However, the starting jitter buffer size value is larger for higher jitter levels. This setting controls the rate at which the jitter buffer size is adjusted to reach the minimum. Select the appropriate setting: low, medium, high, very high, or extremely high. Defaults to high.
  • Silence Supp Enable - To enable silence suppression so that silent audio frames are not transmitted, select yes. Otherwise, select no.
• Switch configuration
  • RTP Port Min - Bottom of the range that RTP will use. Default is port 16384.
  • RTP Port Max - Top of the range that RTP will use. Default is port 16482.
  • SIP ToS/DiffServ Value - ToS/DiffServ field value in UDP IP packets carrying a SIP message.
  • RTP ToS/DiffServ Value - ToS/DiffServ field value in UDP IP packets carrying RTP data.
  • SIP CoS Value - CoS value for SIP messages.
  • RTP CoS Value - CoS value for RTP data.
  • Enable VLAN - Enables the VLAN feature.
  • VLAN ID - VLAN id to be user by the phone.