SIP Trunking

Connect your existing PBX systems to the cloud with PantheonCloud's enterprise-grade SIP Trunking services for reliable voice communications.

Overview

SIP Trunking is a modern telecommunication technology that replaces traditional PRI or analog phone lines with Internet Protocol (IP) based connections. It allows businesses to extend their PBX systems to connect with the Public Switched Telephone Network (PSTN) through the internet. PantheonCloud's SIP Trunking solution provides cost-effective, scalable, and high-quality voice connectivity with enterprise-grade reliability and advanced features.

Scalable

Easily scale your communication capacity as your business grows.

Cost-Effective

Significant cost savings compared to traditional telephony services.

Global Reach

Connect with customers and partners worldwide with local presence.

Key Features

High-Quality Voice

Crystal clear voice quality with enterprise-grade QoS and redundant routing.

HD Voice (G.722) support
Automatic jitter buffer management
Network monitoring and optimization
Redundant global POPs

Unlimited Channels

Scale your concurrent call capacity without purchasing additional hardware.

Pay-as-you-go pricing model
Elastic capacity scaling
No channel limitations
Real-time call analytics

Enterprise Security

Advanced security features to protect your voice communications.

Transport Layer Security (TLS)
Secure Real-time Transport Protocol (SRTP)
IP authentication and ACLs
Fraud detection and prevention

Advanced Call Features

Rich calling features to enhance your business communications.

Caller ID management
Auto attendant and call routing
Call recording and monitoring
Emergency calling (E911)

PBX Integration

Seamlessly connect with your existing PBX systems and telephony infrastructure.

Compatible with all major PBX systems
Simple configuration via API or portal
SIP and IAX protocol support
Technical integration support

Real-time Analytics

Comprehensive call analytics and reporting for better business insights.

Detailed call history and logs
Performance monitoring
Usage statistics and trends
Customizable reports

API Endpoints

Create SIP Trunk

POST

https://api.bitpanyun.com/v1/sip-trunks

Request Body

JSON Example

{
  "name": "Main Office Trunk",
  "description": "Primary SIP trunk for main office",
  "authentication_type": "ip",
  "source_ips": ["203.0.113.1", "203.0.113.2"],
  "domains": [],
  "codecs": ["g711u", "g711a", "g722"],
  "max_concurrent_calls": 100,
  "emergency_enabled": true,
  "voice_settings": {
    "dtmf_mode": "rfc2833",
    "rtcp_mux": true,
    "rtcp_fb": true
  },
  "properties": {
    "department": "IT",
    "location": "Headquarters"
  }
}

Response

JSON Example

{
  "trunk_id": "trunk-567890",
  "name": "Main Office Trunk",
  "description": "Primary SIP trunk for main office",
  "sip_uri": "sip:trunk-567890@sip.bitpanyun.com",
  "status": "ACTIVE",
  "authentication_type": "ip",
  "source_ips": ["203.0.113.1", "203.0.113.2"],
  "domains": [],
  "codecs": ["g711u", "g711a", "g722"],
  "max_concurrent_calls": 100,
  "created_at": "2023-07-15T10:30:00Z",
  "updated_at": "2023-07-15T10:30:00Z"
}

Get SIP Trunk Information

GET

https://api.bitpanyun.com/v1/sip-trunks/{trunk_id}

Response

JSON Example

{
  "trunk_id": "trunk-567890",
  "name": "Main Office Trunk",
  "description": "Primary SIP trunk for main office",
  "sip_uri": "sip:trunk-567890@sip.bitpanyun.com",
  "status": "ACTIVE",
  "authentication_type": "ip",
  "source_ips": ["203.0.113.1", "203.0.113.2"],
  "domains": [],
  "codecs": ["g711u", "g711a", "g722"],
  "max_concurrent_calls": 100,
  "emergency_enabled": true,
  "voice_settings": {
    "dtmf_mode": "rfc2833",
    "rtcp_mux": true,
    "rtcp_fb": true
  },
  "current_stats": {
    "active_calls": 24,
    "total_calls_today": 156,
    "last_incoming_call": "2023-07-15T14:30:22Z",
    "last_outgoing_call": "2023-07-15T14:28:10Z"
  },
  "created_at": "2023-07-15T10:30:00Z",
  "updated_at": "2023-07-15T10:30:00Z"
}

Create Inbound Route

POST

https://api.bitpanyun.com/v1/sip-trunks/{trunk_id}/routes/inbound

Request Body

JSON Example

{
  "name": "Sales Department Inbound Route",
  "description": "Route all sales department numbers",
  "phone_numbers": ["+14155551234", "+14155555678"],
  "destination": {
    "type": "sip",
    "sip_uri": "sales@company-pbx.local"
  },
  "features": {
    "call_recording": true,
    "caller_id_override": false,
    "voicemail_enabled": false
  },
  "time_restrictions": {
    "enabled": true,
    "business_hours": {
      "monday": {"start": "09:00", "end": "17:00"},
      "tuesday": {"start": "09:00", "end": "17:00"},
      "wednesday": {"start": "09:00", "end": "17:00"},
      "thursday": {"start": "09:00", "end": "17:00"},
      "friday": {"start": "09:00", "end": "17:00"}
    },
    "after_hours_destination": {
      "type": "voicemail",
      "voicemail_id": "vm-sales"
    }
  }
}

Response

JSON Example

{
  "route_id": "route-in-789012",
  "trunk_id": "trunk-567890",
  "name": "Sales Department Inbound Route",
  "description": "Route all sales department numbers",
  "phone_numbers": ["+14155551234", "+14155555678"],
  "destination": {
    "type": "sip",
    "sip_uri": "sales@company-pbx.local"
  },
  "status": "ACTIVE",
  "created_at": "2023-07-15T10:30:00Z",
  "updated_at": "2023-07-15T10:30:00Z"
}

Error Codes

Code	Description	Resolution
SIP_001	Invalid authentication credentials	Verify your IP addresses or authentication details are correctly configured
SIP_002	Trunk limit exceeded	Reduce concurrent calls or contact support to increase your limits
SIP_003	Call failed - network issues	Check your network connectivity and firewall settings
SIP_004	Number not found	Verify the phone number is properly formatted and allocated to your account
SIP_005	Codec negotiation failed	Ensure common codecs are enabled between your PBX and our service
SIP_006	Insufficient funds	Add funds to your account or adjust your service plan
SIP_007	Unauthorized access attempt	Check your security settings and whitelist authorized IP addresses