Voice Settings covers tenant-wide configuration for the voice channel — how calls are handled and timed out, agent workspace options, and the underlying SIP and audio behavior used to connect calls.
- Navigate to
Admin (System) > System Settings > Voice
- Each field is listed under its section heading, described below
| Field | Description |
|---|
| Default No Answer Time (Skill) | How long a skill call rings an agent before being re-routed |
| Default No Answer Time (Direct) | How long a direct call rings an agent before being re-routed |
| Queue is Plugged | Blocks new calls from entering queue when enabled |
| Callback Prompt Timeout | How long a caller has to respond to the callback offer prompt |
| Hold Music | The audio played to callers on hold and in queue |
| Agent Max Lines | The maximum number of concurrent calls per agent |
| Enable API Trigger for Internal Calls | Fires API triggers for internal (extension to extension) calls |
| Field | Description |
|---|
| Contact Center PBX Group | The PBX group whose members are treated as contact center agents |
| Hide Skill Enable | Hides the per-skill enable toggles in the agent dashboard |
| UC Agent New Call Button | Shows the New Call button for UC agents in the agent dashboard |
| WebRTC Enabled | Allows agents to take calls in the browser |
| WebRTC Connection Timeout | How long to wait for a WebRTC agent connection before giving up |
| Field | Description |
|---|
| SIP Registration Interval | How often SIP handsets re-register with the phone system |
| SIP Keepalive OPTIONS Interval | How often to send a lightweight SIP OPTIONS ping to keep the connection to the phone system alive through firewalls and NAT. Set to 0 to disable; recommended around 15 seconds for UDP |
| SIP Keepalive TCP/TLS Timeout | How often to send a keepalive on TCP/TLS connections to the phone system to hold the connection open through firewalls and NAT. Set to 0 to disable; recommended around 120 seconds. Takes effect the next time the connection is established |
| RTP Start Port | The start of the UDP port range used for call audio (RTP) |
| RTP End Port | The end of the UDP port range used for call audio (RTP) |
| RTP Audio Codec Preference | The preferred audio codec order for calls |
| DTMF Detection Mode | How keypad (DTMF) tones are detected on calls |
| Delay after DTMF tone | The audio delay applied after a DTMF tone is detected |
| SIP Contact Header | The format of the SIP Contact header sent to the phone system |
| SIP P-Asserted-Identity | The format of the SIP P-Asserted-Identity header sent to the phone system |
| Bridge two SIP parties instead of REFER | Keeps two SIP parties on Xima's media bridge instead of sending a SIP REFER to the phone system; enable for switches that reject the REFER used to complete a transfer between two external parties |
| Web Phone ICE Servers | Which ICE servers agent web phones are offered — see below |
| Web Phone Force Relay Agents | Agents whose web phone always uses TURN relay servers with media forced through the relay, whatever the tenant-wide Web Phone ICE Servers setting says — for individual agents behind strict egress firewalls that block the media servers' rotating IPs, when the rest of the tenant connects fine |
| SIP Capacity Overflow Destination | Where calls are sent when all SIP capacity is in use |
| Dial Digit Send Delay | The delay between sent dial digits |
| Option | Behavior |
|---|
| STUN (Recommended) | Web phones connect directly to the media server |
| STUN + TURN | A direct connection is preferred, with relay servers available as a fallback |
| TURN | Relay servers are offered, but a direct connection may still be used |
| TURN + Force Relay | Media always goes through relay servers, for agents behind strict egress firewalls that block the media servers' rotating IPs |
| Field | Description |
|---|
| Logging Level | The verbosity of voice service logs |
