General Settings
Configure SIP account usage, the auto-answering feature, DTMF types and network options.
SIP Accounts
NOTE
The multiple SIP accounts feature is not available in the web-based softphone widget.
To add a SIP account:
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Click Add Another Account.
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Select a color that will identify this SIP account and calls made through it in the call history and add the account name.
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Enter your login information as outlined in the Logging In to Softphone article based on the sign-in option you choose:
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Option 1: Enter agent credentials and domain from PBX Stats.
Adding a SIP account: entering agent credentials and domain from PBX Stats
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Option 2: Enter SIP credentials and domain from CommPeak Portal.
Adding a SIP account: entering SIP credentials and domain from CommPeak Portal
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Option 3: Enter credentials and domain from the CommPeak Dialer or Cloud PBX.
Adding SIP account: entering credentials and domain from the Dialer or Cloud PBX
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Save the changes.
When making calls, just expand the drop-down list and select the necessary account.
Making calls: selecting account
Here, you also have options for fixing the disconnection issue for your SIP accounts:
- Click the RECONNECT button to try to connect your SIP account again
- Download error logs to share them with our support
- Delete the SIP account.
Auto-Answer Call
The feature allows agents to set a time limit, after which their softphone device will automatically answer an incoming phone call.
On the Auto-Answer Calls tab, you will see the auto-answer status.
To turn on the auto-answer feature:
- Expand the tab and click the Auto-Answer Calls toggle.
- On the new line that appears, set the number of seconds after which incoming calls will be auto-answered.

Auto-Answer Calls toggle
DTMF Types
In the DTMF type section, you can choose any of the dual-tone multi-frequency signaling (DTMF) types that CommPeak Softphone supports: RFC2833, SIP INFO, and Inband. Just click on the line with the DTMF type you want to choose.

DTMF type section
Network Options
In Network Options, you can enable STUN (Session Traversal Utilities for NAT) that supports call connectivity and RTP (Real-time Transport Protocol). This feature ensures reliable call connections and high-quality audio transmission across different network environments, e.g., home, office, or mobile data.

Network Options
- Slide the toggle to the right to enable STUN.
- In the Resolution timeout, define how long the standalone CommPeak Softphone should wait for a response from the STUN server when trying to determine its public IP and port. A typical value ranges between 1 and 3 seconds.
Updated 9 days ago