Softphone — Browser-Based Calling

What Is the Softphone?

The softphone is a browser-based SIP (Session Initiation Protocol) phone built into the portal. It allows agents to make and receive calls directly from their browser without any physical phone hardware. Features include:

• Inbound and outbound SIP calls.
• Click-to-dial from ticket and contact pages.
• Call recordings stored in MinIO.
• Full call log with direction, duration, linked ticket, and linked contact.

Prerequisites

An admin must configure SIP credentials:

1. Go to Admin → Settings → Telephony.
2. Enter:
   • SIP Server — your SIP proxy hostname (e.g. sip.yourpbx.com).
   • SIP Username — your SIP account username.
   • SIP Password — your SIP account password.
   • SIP Realm — the authentication realm.
3. Click Save.

The softphone widget appears in the bottom-right corner of the portal once SIP credentials are configured.

Making a Call

From the Softphone Widget

1. Click the phone icon in the bottom-right to open the softphone.
2. Type a phone number in the dial pad.
3. Click the green Call button.

Click-to-Dial from a Contact

On a CRM contact page, click the phone icon next to any phone number. The call is logged against the contact.

Click-to-Dial from a Ticket

On the ticket detail page, if the linked contact has a phone number, a click-to-dial button appears. The call is logged against the ticket and the contact.

Receiving Calls

When an inbound call arrives, the softphone widget shows a ringing notification with the caller's number. If the number matches a CRM contact, their name is shown. Click Answer to take the call or Decline to reject it.

Call Log

Every call is logged in Calls → Call Log. Each entry shows:

• Direction (Inbound / Outbound).
• Remote number and duration.
• Status (Answered, Missed, Failed).
• Linked ticket and contact (if applicable).
• Recording playback (if recorded).

Troubleshooting

• No audio — check browser microphone permissions (click the padlock icon in the address bar → allow microphone).
• Calls connecting but one-way audio — typically a NAT traversal issue. Ensure your SIP server has STUN/TURN configured.
• Registration failed — verify SIP credentials in Admin → Settings → Telephony. Check firewall rules allow outbound SIP (UDP/TCP 5060) and RTP (UDP 10000-20000) from the portal server.