Capabilities-at-a-Glance

Scout's SIP implementation supports the following capabilities when they are available on the SIP server.

Capability Description
Controls
Patch

A method for the console operator to connect calls to multiple other radio or telephone endpoints.

Activity History with integrated Instant Recall Recorder Tools that allow the console operator to review past conversations for analysis or clarification.
Transfer Transfer a call to a third party as either a blind transfer or an attended transfer.
Automatic Ring Down A line that immediately dials a DTMF string to a specific endpoint when touched if the line is in idle or disconnect state.
Hold Place a line on hold to stop the audio from being transmitted from or received at the Scout console.
Dial Tone Dial Tone button provides a dial tone for the endpoint; commonly referred to as Recall.
Hookflash Flash button that simulates quickly hanging up then picking up again to send a hook-flash signal to the SIP server.
Voicemail Visual indication on console alerts dispatcher that a voicemail is waiting.
Forward Forward individual or groups of SIP lines to a SIP extension, voicemail box, or external number.
Do Not Disturb Activate Do Not Disturb on a console to block audible call indications for all endpoints on the console. All visual and audible call indications cease to endpoints configured as private when the console is in Do Not Disturb. Private endpoints reject calls or redirect calls to another endpoint.
Subscriber Signaling
Caller ID Subscriber Unit ID displayed on Scout screen on the endpoint pad and in Activity History; Class caller ID via gateway or SIP signaling.
Emergency Call Inbound emergency calls for an endpoint configured as an emergency line; all calls from the endpoint are emergency calls.
DTMF Dual-Tone Multi-Frequency (DTMF) signaling, using 0 – 9, A – D, * and #.
High Availability Adding the High Availability driver to SIP endpoints allows for fast failover of an active SIP call from one VPGate to another without dropping the call or losing a significant amount of audio.
Redundant Registrars Using primary and secondary registrars; if primary fails, secondary registrar attempts to send the outbound call. Uses UDP and TCP transport protocols.
Registrar Authentication Providing the option for primary and secondary registrars to have unique authentication user names and passwords.
Registrar Refresh Time User-defined registrar refresh time for the primary and secondary registrars.
Direct Connect Configuration option to have an endpoint connect directly without going through a registrar.
Outbound Proxy Supports using an outbound proxy with a choice to force SIP Signaling through the outbound proxy.
Secondary Dialing Secondary dialing using RTP-embedded digits; Use to dial a second string of digits after connecting to the primary number; for example, an extension number.
Codec Encoding Encoding for audio codecs including: G.711, G.729a, G.726-16, G.726-24, G.726-32 and G.726-40.