VOS3000 SIP Call Flow – Complete Routing Process with Error Troubleshooting

Understanding VOS3000 SIP call flow is essential for troubleshooting VoIP issues. Every call that passes through VOS3000 follows a specific path from the originating device through the softswitch to the terminating gateway. This guide explains the complete call routing process, identifies common failure points, and provides troubleshooting solutions based on official VOS3000 2.1.9.07 documentation.

📞 Need help troubleshooting VOS3000 routing issues? WhatsApp: +8801911119966

🔄 VOS3000 SIP Call Flow Overview

In VOS3000, call routing is the process of matching an incoming call to a routing rule that defines which outbound gateway should be used. The softswitch acts as the central intelligence, processing SIP signaling, applying business rules, managing billing, and connecting parties. Here’s the complete flow:

📊 Call Flow Diagram

┌─────────────┐    SIP INVITE    ┌─────────────────┐    SIP INVITE    ┌─────────────┐
│   SIP       │ ──────────────▶ │                 │ ──────────────▶ │   Routing   │
│   Client    │                  │    VOS3000      │                  │   Gateway   │
│  (Caller)   │ ◀────────────── │   Softswitch    │ ◀────────────── │  (Vendor)   │
└─────────────┘    SIP 200 OK    └─────────────────┘    SIP 200 OK    └─────────────┘
      │                                │                                │
      │         RTP Media Stream       │       RTP Media Stream        │
      └────────────────────────────────┴────────────────────────────────┘

📋 Step-by-Step SIP Call Flow (VOS3000 SIP Call Flow)

Step 1: SIP Client Registration

Before making calls, SIP clients (phones, softphones, or gateways) must register with VOS3000:

• REGISTER Request: Client sends SIP REGISTER to VOS3000
• Authentication: VOS3000 challenges with 401 Unauthorized
• Credentials: Client provides username/password (mapping gateway credentials)
• Validation: VOS3000 validates against account database
• 200 OK: Registration confirmed, client is now “Online”

If registration fails, check: correct credentials, account status (not locked/disabled), IP address matches gateway configuration, and network connectivity.

Step 2: Call Initiation (SIP INVITE)

When the caller dials a number:

• INVITE Request: SIP client sends INVITE with called number to VOS3000
• SDP Contains: Codec preferences, RTP port for media
• VOS3000 Processing: Identifies calling account from source IP or authentication

Step 3: Prefix Matching & Routing Decision

VOS3000 applies routing logic to determine the destination:

• Number Analysis: Extracts prefix from called number
• Prefix Match: Matches against routing gateway prefix configurations
• Gateway Selection: According to VOS3000 manual, gateways are chosen based on: priority number, ratio of current calls to channels, historical calls, and gateway ID
• LCR Application: If enabled, Least Cost Routing selects lowest-cost matching route
• Rate Application: Billing rate applied based on matched prefix

Step 4: Gateway Selection & Call Forwarding

Based on routing configuration, VOS3000 forwards the call:

• Routing Gateway Prefix: According to VOS3000 manual, “when the number being called is not registered in the system, the call will be routed only to gateways which match the prefix specified”
• Multiple Prefixes: Multiple prefixes can be specified, separated by commas
• Gateway Priority: When multiple gateways match, selection follows priority, load balancing, and capacity rules

Step 5: Call Establishment

The terminating gateway processes the call:

• 100 Trying: Gateway acknowledges INVITE
• 180 Ringing: Destination phone starts ringing
• 200 OK: Call answered, SDP contains destination RTP information
• ACK: VOS3000 confirms call establishment

Step 6: Media Stream (RTP)

After call establishment, audio flows between parties:

• RTP Packets: Media flows between caller and called party
• Media Proxy: VOS3000 can proxy media (configured per gateway)
• Codec Negotiation: Final codec based on SDP negotiation

Step 7: Call Termination & CDR Creation

When the call ends:

• BYE Request: Either party can initiate termination
• 200 OK: Confirmation of termination
• CDR Record: Call Detail Record created with duration, cost, and status
• Billing Update: Account balances updated

⚠️ Common VOS3000 Call Errors & Solutions (VOS3000 SIP Call Flow)

Based on the official VOS3000 2.1.9.07 manual, here are server-side call end reasons and their solutions:

🔴 Response Timeout

Description: The called party did not answer before the timeout limit was reached.

Causes:

• Timeout limit reached (set by “Alerting” signal of Routing Gateway or SS_TIMEOUT_PHONE_HANGUP parameter)
• Destination unreachable or not responding
• Network latency issues

Solutions:

• Adjust timeout parameter in routing gateway configuration
• Check destination gateway connectivity
• Verify network quality and latency
• Review SS_TIMEOUT_PHONE_HANGUP in softswitch parameters

🔴 Connection Timeout

Description: No response to SIP message was received after specified number of trials.

Causes:

• Destination gateway offline or unreachable
• Firewall blocking SIP traffic
• Incorrect gateway IP configuration

Solutions:

• Verify gateway is online (check Online Routing Gateway)
• Confirm firewall allows SIP port (typically 5060)
• Check gateway IP address in configuration
• Adjust SS_SIP_RESEND_INTERVAL and SS_SIP_SEND_RETRY parameters if needed

🔴 Account Locked

Description: The account is disabled or locked.

Causes:

• Account manually disabled by administrator
• Agent account locked (affects sub-accounts)
• Balance insufficient with no overdraft

Solutions:

• Check account status in General Account management
• Verify agent account is active
• Add balance or increase overdraft limit

🔴 Session Timeout

Description: Session expired due to SIP Timer protocol or max duration limit.

Causes:

• SIP Timer protocol not receiving update signals
• Session exceeded maximum duration (SS_SIP_NO_TIMER_REINVITE_INTERVAL)

Solutions:

• Check SIP Timer compatibility between endpoints
• Review session timeout parameters
• Verify NAT keepalive is configured

🔴 Caller/Called Number Restricted

Description: Number length or prefix violates restrictions.

Causes:

• Number length exceeds SS_CALLERALLOWLENGTH parameter
• Prefix not allowed by gateway prefix control

Solutions:

• Adjust number length limit in system parameters
• Configure caller/callee prefix control in gateway settings
• Check rewrite rules are applied correctly

🔴 Unregistered

Description: The terminal is not registered and not allowed to make calls.

Causes:

• Device not registered with VOS3000
• Registration expired
• Incorrect registration credentials

Solutions:

• Verify device registration in Online Phone section
• Check registration settings on device
• Confirm credentials match account configuration

🔴 Connection Limit Exceeded

Description: Maximum number of concurrent calls reached.

Causes:

• Line limit reached for gateway or account
• Capacity limit of server reached

Solutions:

• Increase line limit in gateway configuration
• Upgrade to higher capacity server
• Review concurrent call patterns and optimize routing

🔴 The Called Not Online

Description: No appropriate device to accept this call (no matching routing gateway).

Causes:

• No routing gateway configured for the destination prefix
• All matching gateways offline
• Prefix not configured in any gateway

Solutions:

• Configure routing gateway with appropriate prefix
• Check gateway online status
• Verify prefix configuration matches destination numbers

🔴 Proceeding Timeout

Description: No response received from server within time limit.

Causes:

• “Setup” and “Callproceeding” parameters in routing gateway exceeded
• Gateway processing delay

Solutions:

• Adjust proceeding timeout in routing gateway settings
• Check gateway performance and processing capacity

🔴 Forwarding Loop

Description: Wrong configuration caused forwarding route to have loops.

Causes:

• Circular forwarding configuration
• Incorrect call forwarding rules

Solutions:

• Review call forwarding settings in phone management
• Eliminate circular forwarding paths
• Check no-answer, on-busy, and timed forwarding rules

📊 Troubleshooting VOS3000 Call Issues (VOS3000 SIP Call Flow)

Step 1: Check CDR Records

Navigate to Data Query > Recent CDR or CDR to view call records. Important fields:

• Call End Reason: Shows why the call terminated
• Caller/Callee: Verify correct numbers
• Gateway: Confirm routing gateway used
• Duration: Check if call was established

Step 2: Check Gateway Status

Navigate to Operation Management > Gateway Operation > Gateway Status to verify:

• Gateway is online and registered
• Current concurrent calls vs line limit
• Network quality indicators

Step 3: Analyze Routing Configuration

Check these settings:

• Routing gateway prefix matches destination
• Gateway priority and capacity settings
• Caller/Callee rewrite rules applied correctly
• Prefix control allows the number pattern

Step 4: Check Account Status

Verify in Account Management > General Account:

• Account is active (not locked/disabled)
• Balance is sufficient
• Overdraft limit covers call cost

Step 5: Review System Parameters

Check relevant softswitch parameters:

• SS_TIMEOUT_PHONE_HANGUP – Ring timeout
• SS_SIP_RESEND_INTERVAL – SIP retry interval
• SS_SIP_SEND_RETRY – Number of SIP retries
• SS_CALLERALLOWLENGTH – Max number length

🔗 Related Resources (VOS3000 SIP Call Flow)

• 📖 VOS3000 Error Codes Complete Reference
• 📖 VOS3000 Call Routing Explained
• 📖 VOS3000 LCR Configuration Guide
• 📖 VOS3000 RTP Media Troubleshooting
• 📖 VOS3000 Routing Guide
• 📖 VOS3000 Troubleshooting Guide

❓ Frequently Asked Questions (VOS3000 SIP Call Flow)

How do I check why a call failed?

Check the CDR (Call Detail Record) in Data Query section. The “Call End Reason” field shows why the call terminated. Use this to identify routing, authentication, or timeout issues.

Why are calls going to the wrong gateway?

Check routing gateway prefix configuration. VOS3000 routes based on prefix matching. Verify the gateway prefix matches your destination numbers and check gateway priority settings.

How do I fix one-way audio?

One-way audio is typically caused by NAT/firewall issues. Enable media proxy in gateway settings, ensure RTP ports are open, and configure NAT keepalive. See our RTP Media Troubleshooting guide.

What causes high PDD (Post Dial Delay)?

High PDD can be caused by network latency, slow gateway response, or DNS resolution delays. Check network quality, gateway performance, and consider using IP addresses instead of hostnames.

How can I improve ASR?

Analyze failed calls in CDR, identify common failure reasons, optimize routing paths, remove failing gateways, and ensure proper timeout configurations. Monitor gateway performance regularly.

📞 Get Help with VOS3000 Routing Issues (VOS3000 SIP Call Flow)

Experiencing call routing problems or errors in your VOS3000 system? Our experts can help diagnose issues, optimize routing configuration, and improve your ASR/ACD metrics. We provide professional VOS3000 support and optimization services.

📱 WhatsApp: +8801911119966

Contact us for VOS3000 troubleshooting, routing optimization, and professional support! (VOS3000 SIP Call Flow)

📞 Need Professional VOS3000 Setup Support?

For professional VOS3000 installations and deployment:

📱 WhatsApp: +8801911119966
🌐 Website: www.vos3000.com
🌐 Blog: multahost.com/blog
📥 Downloads: VOS3000 Downloads