SIP ALG Problems, VOS3000 gateway configuration, VoIP Fraud Prevention, VOS3000 Media Proxy, VOS3000 Call Termination Reasons
Understanding VOS3000 call termination reasons is essential for maintaining a reliable VoIP operation. When calls fail or disconnect unexpectedly, the termination reason in the CDR (Call Detail Record) provides crucial information for diagnosis. This comprehensive reference guide covers all server-side termination reasons, client-side error codes, and provides actionable troubleshooting steps based on the official VOS3000 2.1.9.07 manual documentation.
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Every call processed through VOS3000 generates a CDR record that includes the termination reason. This information is captured at the “Termination Reason” or “Call End Reason” field and indicates why the call ended. Understanding these reasons helps identify patterns, troubleshoot recurring issues, and optimize call success rates.
Navigation in VOS3000 Client: 1. Recent CDR Location: Data Query → Recent CDR Purpose: View recent call records with termination info 2. CDR Query Location: Data Query → CDR Purpose: Search historical CDR with filters 3. Call Analysis Location: Operation Management → Business Analysis → Call Analysis Purpose: Deep dive into specific call signaling Key CDR Fields for Diagnosis: ┌──────────────────────────────────────────────────────────────┐ │ Field │ Information Provided │ ├──────────────────────────────────────────────────────────────┤ │ Termination Reason │ Why the call ended (primary field) │ │ Session Time │ How long call lasted │ │ PDD │ Post dial delay │ │ Caller/Callee IP │ Endpoint addresses │ │ Codec │ Audio encoding used │ │ Setup Time │ When call started │ │ Connect Time │ When call was answered │ │ End Time │ When call terminated │ └──────────────────────────────────────────────────────────────┘
Server-side termination reasons are generated by VOS3000 itself when the softswitch decides to end or reject a call. These reasons indicate specific conditions that prevented call completion. (VOS3000 Call Termination Reasons)
| Termination Reason | Description | Solution |
|---|---|---|
| Account Locked | The account is currently disabled or locked | Check Account Management → unlock account |
| Account Disabled | Account status is disabled | Enable account in Account Management |
| Account Expired | Account validity period has ended | Extend expiry date or add payment |
| Insufficient Balance | Account balance too low for call | Add credit to account |
| No Matching Account | No account found to bill the call | Create account or fix caller ID mapping |
| Termination Reason | Description | Solution |
|---|---|---|
| No Matching Rate | No rate found for the destination prefix | Add rate entry for the destination |
| The Called Not Online | No appropriate device/gateway to accept call | Check gateway status, add routing gateway |
| Connection Limit Exceeded | Maximum concurrent calls reached | Increase line limit on account/gateway |
| Unregistered | Terminal not registered and call not allowed | Check registration, verify device config |
| Termination Reason | Description | Related Parameter |
|---|---|---|
| Response Timeout | Called party did not answer before timeout | SS_TIMEOUT_PHONE_HANGUP, Alerting timer |
| Connection Timeout | No SIP response after max retries | SS_SIP_RESEND_INTERVAL, SS_SIP_SEND_RETRY |
| Proceeding Timeout | No response within proceeding time limit | Setup/Callproceeding timer in gateway |
| Session Timeout | SIP Timer update not received in time | SS_SIP_SESSION_TTL, SS_SIP_NO_TIMER_REINVITE_INTERVAL |
| Connection Establishment Timeout | Connection not established in time | Mapping gateway proceeding timeout |
| Termination Reason | Description | Solution |
|---|---|---|
| Caller’s Number Restricted | Caller ID length exceeds allowed limit | Check SS_CALLERALLOWLENGTH setting |
| Called Number Restricted | Callee number length exceeds limit | Adjust number length settings |
| Caller’s Number Prefix Restricted | Caller ID prefix not accepted by gateway | Update allowed prefixes in gateway |
| Called Number Prefix Restricted | Destination prefix not accepted | Check gateway prefix settings |
| Call Restriction | Call blocked by restrictions (e.g., International) | Update account authorization settings |
| Termination Reason | Description | Configuration |
|---|---|---|
| No-Answer Forwarding by Caller | Caller has no-answer forwarding enabled | Phone Management → Call Forwarding |
| No-Answer Forwarding by Called | Called party has no-answer forwarding | Called phone’s forwarding settings |
| Timed Forwarding | Call matches time-based forwarding | Time period forwarding config |
| On-Busy Forwarding | Called party busy, forwarded | Busy forwarding config |
| Call Forwarding by Called | Unconditional forwarding active | Forwarding unconditional setting |
| Forwarding Loop | Forwarding creates infinite loop | Fix forwarding configuration |
| Do-Not-Disturb from Called | Called has DND enabled | Disable DND or handle in routing |
| Termination Reason | Description | Solution |
|---|---|---|
| Protocol Limit | Server cannot process this protocol type | Check protocol compatibility settings |
| Busy | Called number is busy | Normal termination, may retry later |
| Forcible Hang-Up | Server disconnected session | Check admin actions, system limits |
| Hang-Up by Caller | Caller ended the call normally | Normal termination |
| Hang-Up by Called | Called party ended the call | Normal termination |
| Session Closed by Called | Called closed TCP without hang-up signal | Check network/endpoint stability |
| Session Closed by Caller | Caller closed TCP without hang-up | Check caller network/device |
When calls involve H.323 protocol, termination reasons may include standard H.323 cause codes. These codes follow ITU-T Q.931 recommendations and provide detailed information about call failures.
| Cause Code | Name | Meaning |
|---|---|---|
| 1 | UnallocatedNumber | Number not assigned to any destination |
| 3 | NoRouteToDestination | No route to the called number |
| 6 | ChannelUnacceptable | Channel not acceptable for this call |
| 16 | NormalCallClearing | Call cleared normally |
| 17 | UserBusy | User is busy |
| 18 | NoResponse | No response from user |
| 19 | NoAnswer | User alerted but did not answer |
| 21 | CallRejected | Call was rejected |
| 27 | DestinationOutOfOrder | Destination cannot be reached |
| 28 | InvalidNumberFormat | Number format invalid |
| 34 | NoCircuitChannelAvailable | No channel available |
| 38 | NetworkOutOfOrder | Network not functioning properly |
| 41 | TemporaryFailure | Temporary network problem |
| 42 | Congestion | Network congestion |
| 44 | RequestedCircuitNotAvailable | Requested circuit not available |
| 47 | ResourceUnavailable | Insufficient resources |
| 49 | QoSNotAvailable | Requested QoS not available |
SIP responses follow standard HTTP-style status codes. Understanding these codes helps diagnose problems when they appear in CDR records or Call Analysis.
| Code | Name | Meaning |
|---|---|---|
| 400 | Bad Request | Malformed SIP message |
| 401 | Unauthorized | Authentication required |
| 403 | Forbidden | Request understood but refused |
| 404 | Not Found | User not found |
| 405 | Method Not Allowed | SIP method not allowed |
| 406 | Not Acceptable | Cannot produce acceptable response |
| 407 | Proxy Authentication Required | Proxy authentication needed |
| 408 | Request Timeout | Server could not respond timely |
| 410 | Gone | Resource no longer exists |
| 415 | Unsupported Media Type | Media format not supported |
| 422 | Session Interval Too Small | Session timer too short |
| 423 | Interval Too Brief | Registration interval too short |
| 480 | Temporarily Unavailable | Callee temporarily unavailable |
| 481 | Call Leg/Transaction Does Not Exist | Matching transaction not found |
| 482 | Loop Detected | SIP loop detected |
| 483 | Too Many Hops | Max-Forwards exceeded |
| 484 | Address Incomplete | Number incomplete |
| 485 | Ambiguous | Ambiguous destination |
| 486 | Busy Here | Callee is busy |
| 487 | Request Terminated | Request cancelled |
| 488 | Not Acceptable Here | SDP not acceptable |
| 491 | Request Pending | Request pending for same call |
| 493 | Undecipherable | Cannot decrypt request |
| Code | Name | Meaning |
|---|---|---|
| 500 | Server Internal Error | Unexpected server error |
| 501 | Not Implemented | Function not implemented |
| 502 | Bad Gateway | Invalid response from upstream |
| 503 | Service Unavailable | Service temporarily unavailable |
| 504 | Server Time-out | No response from upstream |
| 505 | Version Not Supported | SIP version not supported |
| 513 | Message Too Large | Message exceeds size limit |
| Code | Name | Meaning |
|---|---|---|
| 600 | Busy Everywhere | All destinations busy |
| 603 | Decline | Call declined everywhere |
| 604 | Does Not Exist Anywhere | User does not exist |
| 606 | Not Acceptable | Session cannot be established |
VOS3000 provides detailed call analysis tools that show the complete SIP/H.323 signaling flow, making it easier to diagnose complex problems.
Enabling Call Tracing: 1. Go to System → Debug Trace 2. Enable tracing (check "On") 3. Set trace length (default: 10 minutes) Using Call Analysis: 1. Navigation → Data Query → CDR 2. Find the problematic call 3. Right-click → Call Analysis 4. View signaling flow: Call Analysis Information: ┌──────────────────────────────────────────────────────────────┐ │ Column │ Information │ ├──────────────────────────────────────────────────────────────┤ │ Serial Number │ Order of SIP messages │ │ Caller Signaling │ SIP messages from/to caller │ │ Callee Signaling │ SIP messages from/to callee │ │ Memo │ VOS3000 internal processing notes │ │ Time │ Timestamp of each message │ └──────────────────────────────────────────────────────────────┘ Export Options: - Export: Save signaling as file - Import: Load saved file for analysis This helps identify: - Where in call flow the failure occurred - What SIP response code was returned - Which side initiated termination - Authentication challenges - SDP negotiation issues
Regular CDR analysis helps identify recurring issues before they become major problems.
CDR Analysis Dashboard (Recommended):
1. ASR (Answer Seizure Ratio)
- Calculate per gateway, destination, account
- Alert threshold: Below 40%
- Indicates: Routing issues, capacity problems
2. ACD (Average Call Duration)
- Monitor for unusual patterns
- Very low ACD: Audio problems, wrong routes
- Very high ACD: Possible fraud
3. PDD (Post Dial Delay)
- High PDD indicates routing issues
- Alert threshold: Above 5 seconds
4. Termination Reason Distribution
- Track % of each termination reason
- Sudden changes indicate new problems
Sample Analysis SQL (for database queries):
SELECT
termination_reason,
COUNT(*) as count,
(COUNT(*) * 100.0 / (SELECT COUNT(*) FROM cdr_table)) as percentage
FROM cdr_table
WHERE start_time >= DATE_SUB(NOW(), INTERVAL 24 HOUR)
GROUP BY termination_reason
ORDER BY count DESC;
The most common reasons are “Hang-Up by Caller” and “Hang-Up by Called” which are normal terminations. For abnormal terminations, “Response Timeout” and “Connection Timeout” are most frequent, usually caused by network issues, firewall problems, or endpoint misconfiguration.
Response Timeout occurs when the called party doesn’t answer (no 180 Ringing or 200 OK). Connection Timeout occurs when SIP messages don’t receive any response after retries. Proceeding Timeout occurs during call setup when 100 Trying is received but no further progress. Session Timeout happens during an established call when session timer updates fail.
This can occur when: the account has credit but the rate for the destination is higher than the balance, there’s a minimum balance requirement configured, or the account’s overdraft limit has been reached. Check rate tables and account settings in Account Management.
This occurs when a call is made to a destination prefix that doesn’t have a corresponding entry in the rate table. Check that rate prefixes cover all destination patterns. Remember that VOS3000 uses longest prefix matching, so ensure appropriate prefix entries exist.
Session Timeout typically indicates NAT binding expiry or SIP Timer issues. Check NAT keep-alive settings (SS_SIP_NAT_KEEP_ALIVE_PERIOD), verify session timer configuration (SS_SIP_SESSION_TTL), and ensure the client supports SIP Session Timers. If the client doesn’t support timers, check SS_SIP_NO_TIMER_REINVITE_INTERVAL for the maximum call duration.
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