Extensions, Lines & Ring Groups¶

Overview¶

All SIP extensions are managed via the Platform API and delivered dynamically to FreeSwitch viamod_xml_curl. There are no static file extensions.

Extensions¶

Current occupation¶

Extension	Name	Domain:	Users	Type
1000	Staff	sip.xynap.tech	admin (1)	Standard
1001	Staff 2	sip.xynap.tech	andre.willing (3)	Standard
1010	Line Extension 1	sip.xynap.tech	—	Line
1011	Line Extension 2	sip.xynap.tech	—	Line
1012	Line Extension 3	sip.xynap.tech	—	Line
1013	Line Extension 4	sip.xynap.tech	—	Line
21–24	Support/Sales	sip.xynap.tech	—	Short term
webrtc 1	WebRTC Client 1	sip.xynap.tech	—	WebRTC
webrtc 3	WebRTC Client 3	sip.xynap.tech	—	WebRTC

Number plan¶

Area	Use
1000–1099	Regular extensions (employees)
1010-1019	Line Extensions (multiple lines)
21–29	Short choices (Support, Sales etc.)
webrtc *	WebRTC-Only Clients
*99	Test extension → LiveKit SIP Bridge

API management¶

# Alle Extensions auflisten
GET /api/v1/sip/extensions

# Neue Extension anlegen
POST /api/v1/sip/extensions
{
    "extension": "1002",
    "name": "Neuer Mitarbeiter",
    "domain": "sip.xynap.tech",
    "user_id": 5,
    "password": "auto-generated",
    "enabled": true
}

# Extension-Config fuer SIP-Phone abrufen
GET /api/v1/sip/extensions/{id}/config

Password handling

Passwords are generated on the server side and Fernet-encrypted is stored in the database. The/configendpoint returns the password decrypted — this endpoint requires the corresponding RBAC permission.

Lines (multiple lines)¶

Lines allow to associate several phone numbers/trunks of an extension. A line linked:

Extension(e.g. 1010)
DID(incoming call number)
Trunk(external provider)

# Lines verwalten
GET /api/v1/sip/lines
POST /api/v1/sip/lines
{
    "extension_id": 3,
    "did_id": 1,
    "trunk_id": 2,
    "priority": 1
}

Ring Groups¶

Ring Groups distribute incoming calls to multiple extensions simultaneously or successively.

# Ring Groups verwalten
GET /api/v1/sip/ring-groups
POST /api/v1/sip/ring-groups
{
    "name": "Support Team",
    "strategy": "simultaneous",
    "ring_timeout": 30,
    "extensions": [1000, 1001]
}

Strategien:

Strategy	Description
`simultaneous`	All extensions ring simultaneously
`sequence`	Extensions ring in succession
`random`	Random selection

Call Filters¶

Call filters allow black and whitelisting of phone numbers:

GET /api/v1/sip/call-filters
POST /api/v1/sip/call-filters
{
    "name": "Spam-Block",
    "type": "blacklist",
    "pattern": "+49176*",
    "action": "reject"
}

SIP-Phone Provisioning¶

Connection data¶

Parameters	Value
SIP server	`sip.xynap.tech`
Port (UDP)	5060
Port (TLS)	5061
WSS (WebRTC)	`wss://sip.xynap.tech:7443`
Domain:	`sip.xynap.tech`
Transport	UDP, TLS or WSS

Hardware phones (e.g. Yealink, Snom)¶

Settings for a typical SIP phone:

Server:          sip.xynap.tech
Port:            5060
Transport:       UDP
Username:        1000
Password:        (aus API /sip/extensions/{id}/config)
Domain:          sip.xynap.tech
Outbound Proxy:  (leer)
STUN:            nicht noetig (host network)

NAT and STUN

Since FreeSwitch listens directly to the public IP withnetwork_mode: host, no STUN/TURN is required for the server side. Clients behind NAT may need STUN for their own RTP address detection.

Softphones (Linphone, Open-SIP etc.)¶

Same settings like Hardphones. Use Port 5061 for TLS connections and accept thesip.xynap.techcertificate.

WebRTC Softphone¶

The integrated WebRTC softphone in the xynap Platform usesJsSIPvia a WSS connection.

Technical details¶

Parameters	Value
Library	JSIP
Protocol	WSS (WebSocket Secure)
URL	`wss://sip.xynap.tech:7443`
Codecs	Opus (preferably), PCMA, PCMU
SRTP	Mandatory (WSS requires encrypted media)

Acquaintances¶

JsSIP Race Conditions

JsSIP has known race conditions during connection setup. The WebRTC client must wait until the WebSocket connection is fully built up before a REGISTER is sent. The Platform-UI implements a connection-state handler with retry logic.

ICE Timeout

With slow connections, ICE-Gathering can take too long. The timeout is configured for 5 seconds. If the timeout regularly strikes, it can be adjusted in the FreeSwitch-Config.

DIDs (numbers)¶

DIDs (Direct Inward Dialing) assign incoming numbers to the correct routing.

# DIDs verwalten
GET /api/v1/sip/dids
POST /api/v1/sip/dids
{
    "number": "+4934556643378",
    "name": "IVR Hauptnummer",
    "trunk_id": 1,
    "destination_type": "ivr_menu",
    "destination_id": 1
}

Routing objectives¶

Type	Description
`extension`	Directly to the extension
`ring_group`	To a Ring Group
`ivr_menu`	To an IVR menu
`voicemail`	To a Voicemailbox
`external`	Forwarding to external number