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VOS3000 Llamadas Cortadas Essential: Diagnostico Completo 📞

May 21, 2026May 22, 2026 king

VOS3000 Llamadas Cortadas Essential: Diagnostico Completo 📞

Las VOS3000 llamadas cortadas son uno de los problemas que mas afectan la experiencia del usuario y la percepcion de calidad de un servicio VoIP. 🚫 Cuando una llamada se desconecta prematuramente sin que ninguna de las partes haya colgado, el usuario percibe el servicio como poco confiable, independientemente de la calidad de audio durante la llamada. Comprender las causas y aplicar las soluciones correctas es fundamental para mantener la satisfaccion del cliente. 🔧

En esta guia completa sobre las VOS3000 llamadas cortadas, cubriremos todas las causas posibles de desconexion prematura, desde los temporizadores SIP y RTP hasta los problemas de failover y firewall. Cada seccion incluye tablas de diagnostico, ejemplos y soluciones paso a paso. 🚀

Causas Principales de Llamadas Cortadas 📊

Las VOS3000 llamadas cortadas pueden ser causadas por multiples factores que actuan en diferentes capas del sistema. Identificar la capa donde ocurre el problema es el primer paso para una solucion efectiva. 📋

📊 Causa	Frecuencia	Capa	Sintoma
⏱️ RTP Timeout	⭐⭐⭐⭐⭐ Muy alta	Media	Corte despues de silencio
🔄 Session Timer	⭐⭐⭐⭐ Alta	Senalizacion	Corte a intervalo fijo
🔥 Firewall UDP Timeout	⭐⭐⭐⭐ Alta	Red	Corte despues de X minutos
🔀 Failover/Switch	⭐⭐⭐ Media	Ruteo	Corte con cambio de ruta
📞 Proveedor rechaza	⭐⭐⭐ Media	Terminacion	Corte con codigo SIP
🌐 NAT Timeout	⭐⭐⭐⭐ Alta	Red	Corte en llamadas largas

RTP Timeout: La Causa Mas Comun ⏱️

El RTP timeout es la causa mas frecuente de VOS3000 llamadas cortadas. Cuando VOS3000 detecta que no hay flujo RTP en una direccion durante un periodo determinado, asume que la llamada ha perdido conectividad y la desconecta automaticamente. Esto puede ocurrir cuando un dispositivo entra en silencio prolongado o cuando el flujo RTP se interrumpe por problemas de red. 🔇

Para solucionar problemas de RTP timeout, configure el parametro RTP timeout en VOS3000 con un valor adecuado (tipicamente 30-60 segundos). Un valor muy bajo causara cortes prematuros durante pausas naturales en la conversacion, mientras que un valor muy alto mantendra llamadas fantasma que consumen recursos. Para informacion sobre RTP, consulte nuestra guia de interrupcion RTP del sistema VOS3000. 🔧

SIP Session Timer 🔄

El SIP Session Timer es un mecanismo que mantiene las sesiones SIP activas mediante re-INVITEs periodicos. Si el Session Timer no se renueva correctamente, se produciran VOS3000 llamadas cortadas a intervalos regulares. Este problema es comun cuando los dispositivos no soportan Session Timer o cuando los re-INVITEs son bloqueados por un firewall. ⏱️

Para configurar el Session Timer, acceda a los parametros SIP de VOS3000 y defina el intervalo de sesion (tipicamente 1800 segundos). Para informacion sobre sesiones SIP, consulte nuestra guia de sesion SIP del sistema VOS3000. 📋

Firewall UDP Timeout 🔥

Los firewalls que implementan timeouts UDP cortos son una causa muy comun de VOS3000 llamadas cortadas. Los firewalls mantienen una tabla de conexiones UDP activas, y si no ven trafico en una entrada durante un tiempo determinado (tipicamente 5-30 minutos), eliminan la entrada y bloquean los paquetes posteriores. Esto corta la llamada sin que VOS3000 genere un BYE. 🔥

Para solucionar este problema, configure el SIP NAT keepalive en VOS3000 para enviar paquetes periodicos que mantengan las entradas UDP activas en el firewall. El intervalo de keepalive debe ser menor que el timeout UDP del firewall. Para informacion sobre NAT, consulte nuestra guia de NAT del sistema VOS3000. 🌐

Failover y Cambio de Ruta 🔀

El failover agresivo puede causar VOS3000 llamadas cortadas cuando el sistema detecta degradacion en la ruta actual y conmuta a una ruta alternativa. Si la conmutacion no se realiza correctamente, la llamada existente se corta. Para informacion sobre failover, consulte nuestra guia de failover del sistema VOS3000. 🔄

Para configurar el failover sin afectar llamadas en curso, ajuste los parametros de switch limit y aggressive failover en VOS3000. El switch limit define el umbral de llamadas fallidas antes de cambiar de ruta, mientras que el aggressive failover controla si las llamadas existentes se conmutan o solo las nuevas. Para informacion sobre pasarelas avanzadas, consulte nuestra guia de failover de pasarelas del sistema VOS3000. 🔧

Diagnostico Paso a Paso 🔍

Diagnosticar las VOS3000 llamadas cortadas requiere analizar los CDR y los codigos de finalizacion de las llamadas afectadas. Los codigos de finalizacion proporcionan informacion critica sobre por que se corto la llamada. Para informacion sobre codigos, consulte nuestra guia de codigos de finalizacion del sistema VOS3000. 🔍

📊 Codigo Finalizacion	Significado	Causa Probable	🔧 Solucion
📞 Normal BYE	Una parte colgo	Fin normal de llamada	Verificar con usuario
🔄 RTP Timeout	Sin flujo RTP	Problema de red/media	Ajustar RTP timeout
⏱️ Session Timeout	Sesion expirada	Session Timer no renovado	Configurar keepalive
🔀 Switch/Failover	Cambio de ruta	Failover agresivo	Ajustar switch limit
🚫 Proveedor rechaza	SIP 503/487	Proveedor sin capacidad	Failover a otro proveedor
🔥 Firewall	Sin BYE ni CANCEL	UDP timeout en firewall	Configurar NAT keepalive

Preguntas Frecuentes sobre VOS3000 Llamadas Cortadas ❓

❓ Por que se cortan las llamadas en VOS3000 despues de unos minutos?

Las VOS3000 llamadas cortadas despues de unos minutos son tipicamente causadas por firewall UDP timeout. Los firewalls eliminan las conexiones UDP inactivas despues de un periodo determinado (5-30 minutos). Si no hay trafico SIP de mantenimiento durante la llamada, la entrada se elimina y los paquetes posteriores se bloquean, cortando la llamada. La solucion es configurar SIP NAT keepalive en VOS3000 para enviar paquetes periodicos que mantengan la conexion activa en el firewall. 🔥

❓ Como evito que las llamadas se corten por RTP timeout?

Para evitar las VOS3000 llamadas cortadas por RTP timeout, ajuste el parametro RTP timeout en VOS3000 a un valor adecuado (30-60 segundos). Tambien verifique que el media proxy este habilitado para las pasarelas donde los dispositivos estan detras de NAT. Si el RTP timeout esta muy bajo, las pausas naturales en la conversacion pueden activar el timeout. Si esta muy alto, las llamadas fantasma consumiran recursos innecesariamente. ⏱️

❓ El failover puede cortar llamadas existentes?

Si, el failover agresivo en VOS3000 puede cortar llamadas existentes cuando el sistema detecta degradacion en la ruta y conmuta a una ruta alternativa. Para evitar que el failover afecte llamadas en curso, configure el parametro aggressive failover para que solo afecte nuevas llamadas, no las existentes. Para informacion detallada sobre failover, consulte nuestra guia de failover del sistema VOS3000. 🔀

❓ Como verifico por que se corto una llamada en VOS3000?

Para determinar la causa de las VOS3000 llamadas cortadas, consulte los registros CDR en VOS3000. Los CDR contienen el codigo de finalizacion (release cause) que indica por que se corto la llamada. Busque el campo release cause en el CDR y comparelo con la tabla de codigos de finalizacion. Los codigos mas comunes para llamadas cortadas son RTP timeout, session timeout y SIP 503. Para informacion sobre CDR, consulte nuestra guia de registros CDR avanzados del sistema VOS3000. 📋

❓ Que es el SIP NAT keepalive y como ayuda?

El SIP NAT keepalive es un mecanismo que envia paquetes SIP periodicos (tipicamente cada 20-30 segundos) para mantener las entradas NAT activas en los firewalls y routers. Sin keepalive, las VOS3000 llamadas cortadas ocurren porque el firewall elimina la entrada UDP asociada con la llamada. Para configurar NAT keepalive en VOS3000, acceda a los parametros SIP y establezca el intervalo de keepalive. Un intervalo de 20-30 segundos es adecuado para la mayoria de los firewalls. 🌐

❓ Las llamadas se cortan siempre a los 32 segundos, que significa?

Si las VOS3000 llamadas cortadas ocurren consistentemente a los 32 segundos, es casi seguro que se debe a un problema de negociacion de codec o a un firewall que bloquea el flujo RTP. El temporizador de 32 segundos es tipico del SIP INVITE timeout o del primer re-INVITE. Verifique que los codecs esten configurados correctamente en ambas pasarelas y que los puertos RTP esten abiertos en el firewall. 🎵

Conclusion 🏆

Las VOS3000 llamadas cortadas son un problema multifactorial que requiere un diagnostico sistematico basado en los codigos de finalizacion y el analisis de los flujos SIP y RTP. Con las configuraciones correctas de RTP timeout, Session Timer, NAT keepalive y failover, puede reducir drasticamente la tasa de llamadas cortadas y mejorar la experiencia del cliente. 🚀

Para soporte profesional en la resolucion de problemas de llamadas cortadas, contactenos por WhatsApp al +8801911119966. Tambien puede descargar la ultima version desde vos3000.com/downloads. Para continuar aprendiendo, explore nuestros articulos sobre calidad QoS del sistema VOS3000 y registros CDR avanzados. 🤝

Para consultas, contactenos por WhatsApp al +8801911119966. 📱

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For professional VOS3000 installations and deployment, VOS3000 Server Rental Solution:

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

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VOS3000 Call End Reasons: Complete Error Code Reference Guide

April 7, 2026April 7, 2026 king

VOS3000 Call End Reasons: Complete Error Code Reference Guide

VOS3000 call end reasons provide the definitive diagnostic information for understanding why VoIP calls fail, disconnect, or terminate abnormally. Every call processed through VOS3000 generates detailed termination data in the CDR (Call Detail Record) that identifies the exact reason for call completion. This comprehensive reference guide documents all server-side termination reasons, client error codes, SIP response mappings, and H.323 cause codes based on the official VOS3000 2.1.9.07 manual.

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🔍 Understanding VOS3000 Call End Reasons

Reference: VOS3000 2.1.9.07 Manual, Section 4.5 (Pages 244-249)

When a call terminates in VOS3000, the system records the termination reason in multiple CDR fields. Understanding these reasons is essential for maintaining call quality, optimizing routes, and troubleshooting recurring issues. The termination reason indicates whether the call ended normally, was rejected, timed out, or failed due to system configuration.

📊 Where to Find VOS3000 Call End Reasons

📍 Location	🧭 Navigation Path	📋 Information Available
Recent CDR	Data Query → Recent CDR	Latest calls with termination reason column
CDR Query	Data Query → CDR	Historical CDR with advanced filtering
Call Analysis	Operation Management → Business Analysis	Detailed SIP/H.323 signaling trace
Debug Trace	System → Debug trace	Real-time SIP message capture
Database Direct	MySQL vos3000_cdr table	Raw CDR data for custom analysis

📋 Complete VOS3000 Server-Side Termination Reasons

Reference: VOS3000 2.1.9.07 Manual, Section 4.5 (Pages 244-249)

Server-side termination reasons are generated by VOS3000 when the softswitch itself terminates or rejects the call. These reasons provide the most detailed diagnostic information as they indicate the exact condition that prevented call completion. (VOS3000 Call End Reasons)

🚨 Termination Reason	📋 Description	🔍 Typical Cause	✅ Solution	📖 Page
Account Locked	The account is currently locked/disabled	Admin locked account, security trigger	Account Management → Change Status to Active	244
Account Disabled	Account status is disabled	Account deactivated by admin	Enable account in account management	244
Account Expired	Account validity period has ended	Expiry date passed	Extend expiry date (set to 2099-12-31)	244
Insufficient Balance	Account balance too low for call	Balance < rate for destination	Add credit to account	244
No Matching Account	No account found to bill the call	Caller ID not mapped to account	Create account or fix caller ID mapping	245
Account Not Online	Called account not registered	Phone endpoint offline	Check phone registration status	245

🚨 Termination Reason	📋 Description	🔍 Typical Cause	✅ Solution	📖 Page
NoAvailableRouter	No routing gateway available	No gateway configured for destination	Add routing gateway with correct prefix	245
Not Found	Destination number not found	Prefix mismatch in gateway	Check gateway prefix configuration	245
No Matching Rate	No rate found for destination	Rate table missing prefix	Add rate entry for destination prefix	245
The Called Not Online	Called device not reachable	Gateway offline or no capacity	Check gateway status and capacity	245
Connection Limit Exceeded	Maximum concurrent calls reached	Line limit reached on account/gateway	Increase line limit or add capacity	246
Unregistered	Endpoint not registered	Registration expired or failed	Verify device registration settings	246
Rate Table Not Found	No rate table assigned	Account missing rate table	Assign rate table to account	246

🚨 Termination Reason	📋 Description	⚙️ Related Parameter	✅ Solution	📖 Page
Response Timeout	No answer from called party	SS_TIMEOUT_PHONE_HANGUP	Increase alerting timeout, check endpoint	246
Connection Timeout	No SIP response after retries	SS_SIP_SEND_RETRY, SS_SIP_RESEND_INTERVAL	Check network, firewall, gateway status	246
Proceeding Timeout	No 100 Trying progress	Gateway proceeding timeout	Check gateway connectivity	247
Session Timeout	Session timer refresh failed	SS_SIP_SESSION_TTL	Check session timer config, NAT keep-alive	247
Connection Establishment Timeout	Call setup incomplete	Mapping gateway timeout	Check gateway response time	247
Alerting Timeout	Ring timeout exceeded	Gateway alerting timer	Increase ring timeout in gateway config	247

🔴 Number Restriction Termination Reasons

🚨 Termination Reason	📋 Description	🔍 Trigger	✅ Solution	📖 Page
Caller’s Number Restricted	Caller ID length exceeds limit	SS_CALLERALLOWLENGTH exceeded	Adjust caller length parameter	247
Called Number Restricted	Callee number too long	Number length limit	Adjust called number length settings	247
Caller’s Number Prefix Restricted	Caller prefix not allowed	Gateway prefix filter	Update allowed prefixes in gateway	248
Called Number Prefix Restricted	Destination prefix blocked	Gateway prefix restriction	Check gateway prefix settings	248
Call Restriction	Call blocked by policy	Account authorization settings	Update account call authorization	248

🚨 Termination Reason	📋 Description	⚙️ Configuration	📖 Page
No-Answer Forwarding by Caller	Caller has no-answer forwarding	Phone Management → Call Forwarding	248
No-Answer Forwarding by Called	Called has no-answer forwarding	Called phone forwarding settings	248
Timed Forwarding	Time-based forwarding active	Time period forwarding config	248
On-Busy Forwarding	Called busy, forwarded	Busy forwarding config	248
Call Forwarding by Called	Unconditional forwarding	Forwarding unconditional setting	249
Forwarding Loop	Infinite forwarding loop	Forwarding chain creates loop	249
Do-Not-Disturb from Called	Called has DND enabled	DND setting on phone	249

🔴 Protocol and System Termination Reasons

🚨 Termination Reason	📋 Description	💡 Notes	📖 Page
Protocol Limit	Server cannot process protocol	Check protocol compatibility	249
Busy	Called number is busy	Normal termination	249
Forcible Hang-Up	Server terminated session	Admin action or system limit	249
Hang-Up by Caller	Caller ended normally	Normal termination	249
Hang-Up by Called	Called party ended call	Normal termination	249
Session Closed by Called	TCP closed without BYE	Network/endpoint issue	249
Session Closed by Caller	Caller TCP closed	Check caller network	249
Codec Mismatch	No common codec	Enable transcoding	249

📞 SIP Response Code to VOS3000 Mapping

Reference: VOS3000 2.1.9.07 Manual, Section 4.5.2 (Page 247-248)

When SIP responses are received from endpoints or gateways, VOS3000 maps these standard SIP codes to internal termination reasons. Understanding this mapping helps correlate SIP traces with CDR data. (VOS3000 Call End Reasons)

📊 SIP 4xx Client Error Mapping

📞 SIP Code	📋 SIP Name	🔄 VOS3000 Mapping	⚡ Typical Cause
400	Bad Request	Protocol Error	Malformed SIP message
401	Unauthorized	Auth Failed	Invalid credentials
403	Forbidden	Account Locked / IP Blocked	Access denied by policy
404	Not Found	Not Found	User/destination unknown
405	Method Not Allowed	Protocol Limit	SIP method rejected
406	Not Acceptable	Codec Mismatch	SDP negotiation failed
407	Proxy Auth Required	Auth Failed	Proxy authentication needed
408	Request Timeout	Response Timeout	No SIP response received
415	Unsupported Media	Codec Mismatch	Media format rejected
422	Session Interval Too Small	Session Timer Error	Session timer too short
480	Temporarily Unavailable	Unregistered / Not Online	Endpoint unavailable
484	Address Incomplete	Number Restricted	Incomplete number
486	Busy Here	Busy	User busy
487	Request Terminated	Cancelled	Call cancelled
488	Not Acceptable Here	Codec Mismatch	SDP negotiation failed

📊 SIP 5xx Server Error Mapping

📞 SIP Code	📋 SIP Name	🔄 VOS3000 Mapping	⚡ Typical Cause
500	Server Internal Error	Protocol Error	Remote server error
502	Bad Gateway	Gateway Error	Invalid response from upstream
503	Service Unavailable	NoAvailableRouter	Server overloaded/maintenance
504	Server Time-out	Connection Timeout	No response from upstream

📞 H.323 Cause Codes Complete Reference

Reference: VOS3000 2.1.9.07 Manual, Section 4.5.3 (Page 248-249)

For H.323 protocol calls, VOS3000 uses ITU-T Q.931 cause codes. These standard cause codes provide detailed information about call failures in traditional telephony and H.323-based VoIP systems.

📊 H.323 Q.931 Cause Codes Table (VOS3000 Call End Reasons)

🔢 Code	📋 Name	📝 Meaning	🔄 VOS3000 Action
1	Unallocated Number	Number not assigned to any destination	Record as Not Found
2	No Route to Transit Net	No route through transit network	Record as NoAvailableRouter
3	No Route to Destination	No route to called number	Record as NoAvailableRouter
6	Channel Unacceptable	Channel not acceptable for call	Record as Connection Limit
7	Call Awarded	Call awarded, delivered to endpoint	Normal call handling
8	Preemption	Call preempted by higher priority	Record as Forcible Hang-Up
16	Normal Call Clearing	Call cleared normally	Record as Hang-Up by Caller/Called
17	User Busy	User is busy	Record as Busy
18	No User Responding	No response from user	Record as Response Timeout
19	No Answer from User	User alerted but no answer	Record as Response Timeout
20	Subscriber Absent	Subscriber not available	Record as Unregistered
21	Call Rejected	Call was rejected	Record as call rejection
22	Number Changed	Number has been changed	Record as Not Found
26	Non-Selected User Clearing	User not selected for call	Normal termination
27	Destination Out of Order	Destination cannot be reached	Record as Gateway Error
28	Invalid Number Format	Number format invalid	Record as Number Restricted
29	Facility Rejected	Facility rejected	Record as Protocol Error
30	Response to STATUS ENQUIRY	Response to status inquiry	Status response
31	Normal, Unspecified	Normal event, no specific cause	Normal termination
34	No Circuit Available	No channel available	Record as Connection Limit
38	Network Out of Order	Network not functioning	Record as Gateway Error
41	Temporary Failure	Temporary network problem	Record as temporary error
42	Switch Congestion	Network congestion	Record as Connection Limit
44	Requested Circuit Not Available	Requested circuit unavailable	Record as Connection Limit
47	Resource Unavailable	Insufficient resources	Record as Connection Limit
49	QoS Not Available	Requested QoS not available	Record as Protocol Error
50	Requested Facility Not Subscribed	Facility not subscribed	Record as Call Restriction
52	Outgoing Call Barred	Outgoing calls barred	Record as Call Restriction
54	Incoming Call Barred	Incoming calls barred	Record as Call Restriction
57	Bearer Capability Not Authorized	Capability not authorized	Record as Account Locked
58	Bearer Capability Not Available	Capability not available	Record as Codec Mismatch
63	Service Not Available	Service not available	Record as NoAvailableRouter
65	Bearer Capability Not Implemented	Capability not implemented	Record as Codec Mismatch
69	Requested Facility Not Implemented	Facility not implemented	Record as Protocol Limit
70	Only Restricted Digital Bearer	Only restricted digital available	Record as Codec Mismatch
79	Service Not Implemented	Service not implemented	Record as Protocol Limit
81	Invalid Call Reference	Call reference invalid	Record as Protocol Error
88	Incompatible Destination	Destination incompatible	Record as Codec Mismatch
95	Invalid Message	Message invalid	Record as Protocol Error
96	Mandatory IE Missing	Mandatory element missing	Record as Protocol Error
97	Message Not Implemented	Message type not implemented	Record as Protocol Limit
98	IE Not Implemented	Information element not implemented	Record as Protocol Limit
99	Invalid IE Contents	Information element invalid	Record as Protocol Error
100	Invalid Message for State	Message invalid for call state	Record as Protocol Error
101	Recovery on Timer Expiry	Timer expired	Record as Response Timeout
102	Protocol Error	Protocol error unspecified	Record as Protocol Error
111	Protocol Error, Unspecified	Unspecified protocol error	Record as Protocol Error

🔧 CDR Analysis and Diagnosis Flow

🔍 VOS3000 Call End Reasons Diagnosis Workflow

STEP 1: IDENTIFY THE PROBLEM
├── Open CDR Query (Data Query → CDR)
├── Filter by time period, account, or gateway
├── Look at "Termination Reason" column
└── Note patterns: same error repeatedly?

STEP 2: CLASSIFY ERROR TYPE
├── Account Issue?
│   ├── Account Locked → Account Management
│   ├── Insufficient Balance → Balance Management
│   └── Account Expired → Extend Expiry Date
│
├── Routing Issue?
│   ├── NoAvailableRouter → Add Gateway
│   ├── Not Found → Check Prefix Config
│   └── No Matching Rate → Add Rate Entry
│
├── Network Issue?
│   ├── Connection Timeout → Firewall/Connectivity
│   ├── Response Timeout → Gateway Response
│   └── Session Timeout → NAT/Session Timer
│
└── Protocol Issue?
    ├── Codec Mismatch → Enable Transcoding
    ├── Protocol Limit → Check Compatibility
    └── Protocol Error → Debug Trace Analysis

STEP 3: ANALYZE CALL TRACE
├── Find problematic call in CDR
├── Right-click → Call Analysis
├── Review SIP/H.323 message sequence
└── Identify exact point of failure

STEP 4: IMPLEMENT SOLUTION
├── Apply fix based on diagnosis
├── Document changes made
├── Test with sample calls
└── Verify in CDR that issue resolved

STEP 5: MONITOR
├── Check ASR improvement
├── Monitor for recurrence
└── Adjust configuration as needed

💰 VOS3000 Support Services

Need professional assistance with VOS3000 call failure diagnosis? Our team provides comprehensive VOS3000 support including CDR analysis, error code troubleshooting, and system optimization.

📦 Service	📝 Description	💼 Scope
CDR Analysis	Deep dive into call failure patterns	Error patterns, ASR/ACD analysis
Configuration Audit	Review VOS3000 settings	Parameters, gateway, routing
Technical Support	24/7 remote assistance	Troubleshooting, optimization

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📖 VOS3000 SIP Call Flow Complete Guide
📖 VOS3000 ASR ACD Analysis
📖 VOS3000 Gateway Analysis Reports
📖 VOS3000 Account Management
📥 VOS3000 Downloads – Manual and Software

❓ Frequently Asked Questions

What is the difference between “Response Timeout” and “Connection Timeout”?

Response Timeout occurs when the called party does not answer within the configured alerting time (phone rings but nobody picks up). Connection Timeout occurs when VOS3000 sends SIP INVITE but receives no SIP response at all after multiple retries, typically indicating network connectivity issues, firewall blocks, or the gateway being offline.

Why do I see different error codes for similar failures?

VOS3000 records different termination reasons based on the exact point of failure and the protocol used. A SIP 404 Not Found from a gateway produces “Not Found” while a routing table miss produces “NoAvailableRouter”. The distinction helps pinpoint whether the issue is with routing configuration or the remote endpoint’s response.

How can I filter CDR to see only failed calls?

In CDR Query, you can filter by Session Time = 0 to see calls that never connected, or create custom queries against the database to filter by specific termination reasons. The vos3000_cdr MySQL table contains all termination information for advanced analysis.

What causes “Codec Mismatch” errors?

Codec Mismatch occurs when the caller and callee have no common audio codec. This typically happens when one side only supports G.711 (alaw/ulaw) and the other only supports G.729 without transcoding enabled. Enable transcoding in VOS3000 or configure compatible codecs on both endpoints.

How do I interpret H.323 cause codes in CDR?

H.323 cause codes follow ITU-T Q.931 standard. Common codes include 16 (normal call clearing), 17 (user busy), 18 (no response), 21 (call rejected), and 34 (no circuit available). Use the complete cause code table in this guide to interpret specific values in your CDR.

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VOS3000 Session Timer: Complete Easy Guide to SIP Keep-Alive Configuration

April 7, 2026April 7, 2026 king

VOS3000 Session Timer: Complete Guide to SIP Keep-Alive Configuration

VOS3000 session timer is a critical mechanism for maintaining call stability and preventing “zombie calls” that consume system resources. Based on RFC 4028 specifications, the session timer functionality in VOS3000 2.1.9.07 ensures that active VoIP sessions are properly monitored while failed or hung calls are detected and cleaned up automatically. This comprehensive guide covers all session timer parameters, NAT keep-alive configuration, and troubleshooting procedures based on the official VOS3000 manual.

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🔍 What is VOS3000 Session Timer?

Reference: VOS3000 2.1.9.07 Manual, Section 4.1.3 (Page 213)

The VOS3000 session timer implements the SIP Session Timer mechanism defined in RFC 4028. This protocol extension addresses a fundamental problem in SIP-based VoIP systems: the inability to detect when a call has failed at one endpoint while the other endpoint believes the call is still active. These “zombie calls” can persist indefinitely, consuming system resources, occupying call capacity, and causing billing discrepancies.

📊 The Zombie Call Problem

🚨 Scenario	❌ Without Session Timer	✅ With Session Timer
Endpoint Power Failure	Call remains “active” indefinitely in system	Session expires, call terminated cleanly
Network Disconnection	No notification, resources wasted	Refresh fails, session cleaned up
Device Crash	Zombie call persists for hours/days	Maximum session duration enforced
NAT Timeout	One-way audio, confused state	Session refresh detects failure
Billing Impact	Incorrect CDR duration, revenue loss	Accurate call termination timing

⚙️ VOS3000 Session Timer Parameters Complete Reference

Reference: VOS3000 2.1.9.07 Manual, Section 4.3.5.2 (Pages 229-239)

VOS3000 provides a comprehensive set of session timer parameters that control how the softswitch monitors and maintains active SIP sessions. These parameters are configured in the System Parameters section and affect all SIP-based communications.

📊 Core Session Timer Parameters Table

⚙️ Parameter	📊 Default	📏 Range	📝 Description	📖 Manual Page
SS_SIP_SESSION_TTL	600	60-86400 sec	Detecting SIP connected status interval (Session-Expires value)	230
SS_SIP_SESSION_UPDATE_SEGMENT	2	2-10	Divisor for refresh interval calculation (TTL/segment)	230
SS_SIP_SESSION_TIMEOUT_EARLY_HANGUP	0	0-3600 sec	Terminate session before actual timeout (margin)	230
SS_SIP_NO_TIMER_REINVITE_INTERVAL	7200	0-86400 sec	Maximum call duration for non-timer endpoints	230
SS_SIP_SESSION_MIN_SE	90	90-3600 sec	Minimum session expires value per RFC 4028	231

📊 Session Timer Refresh Calculation

📐 Session Timer Refresh Interval Formula

Refresh Interval = SS_SIP_SESSION_TTL ÷ SS_SIP_SESSION_UPDATE_SEGMENT

Example with Defaults:	600 ÷ 2 = 300 seconds (5 minutes)
First Refresh Attempt:	At 5 minutes into the call
Session Expires If:	No response to refresh within TTL period

📡 NAT Keep-Alive Configuration Deep Dive

Reference: VOS3000 2.1.9.07 Manual, Section 4.1.2 (Pages 212-213)

NAT (Network Address Translation) devices maintain binding tables that map internal private IP addresses to external public addresses. These bindings have a timeout period, typically ranging from 30 to 300 seconds depending on the device. When a binding expires without traffic, incoming calls cannot reach the endpoint behind NAT.

📊 NAT Keep-Alive Parameters Table

⚙️ Parameter	📊 Default	📏 Range	📝 Function	📖 Page
SS_SIP_NAT_KEEP_ALIVE_MESSAGE	HELLO	Text string	Content of NAT keep-alive UDP packet	212
SS_SIP_NAT_KEEP_ALIVE_PERIOD	30	10-86400 sec	Interval between keep-alive transmissions	212
SS_SIP_NAT_KEEP_ALIVE_SEND_INTERVAL	500	1-10000 ms	Delay between individual keep-alive packets in batch	212
SS_SIP_NAT_KEEP_ALIVE_SEND_ONE_TIME	3000	1-10000	Number of keep-alive packets sent per batch cycle	212

🔄 How NAT Keep-Alive Works in VOS3000

VOS3000 NAT Keep-Alive Operation Flow:
=======================================

SCENARIO: Endpoint behind NAT firewall
┌─────────────────────────────────────────────────────────────────────────────┐
│                                                                             │
│  ENDPOINT                    NAT DEVICE                   VOS3000 SERVER    │
│  (192.168.1.100)            (Public IP)                  (Softswitch)       │
│                                                                             │
│  1. REGISTER ───────────────────────────────────────────────────────────►  │
│     (Via: 192.168.1.100)                                                    │
│                                                                             │
│  2. VOS3000 Records:                                                         │
│     - Received IP: Public NAT IP                                            │
│     - Received Port: NAT mapped port                                        │
│     - Contact: Internal IP (via Contact header)                             │
│                                                                             │
│  3. NAT BINDING TABLE:                                                       │
│     Internal: 192.168.1.100:5060 → External: PublicIP:45678                │
│                                                                             │
│  4. KEEP-ALIVE MESSAGE (every 30 seconds):                                  │
│     ◄─────────────────────────────────────────────────────────────────────  │
│     UDP packet "HELLO" to PublicIP:45678                                    │
│                                                                             │
│  5. NAT BINDING REFRESHED:                                                   │
│     - Timer resets to 30+ seconds                                           │
│     - Binding remains active                                                │
│                                                                             │
│  6. INCOMING CALL:                                                           │
│     ◄─────────────────────────────────────────────────────────────────────  │
│     INVITE reaches endpoint successfully!                                   │
│                                                                             │
└─────────────────────────────────────────────────────────────────────────────┘

IMPORTANT: If SS_SIP_NAT_KEEP_ALIVE_MESSAGE is empty, keep-alive is DISABLED!

🔧 VOS3000 Session Timer Configuration Guide

📍 Navigation to System Parameters

Step	Navigation Path	Action
1	Operation management	Click main menu
2	Softswitch management	Select softswitch node
3	Additional settings	Right-click → Additional settings
4	System parameter tab	Find session timer parameters
5	Modify values	Edit desired parameters
6	Apply changes	Click OK to save

⚙️ Recommended Configuration by Deployment Type

🏢 Scenario	⏱️ SESSION_TTL	🔄 SEGMENT	🚫 NO_TIMER_INTERVAL	📡 NAT_PERIOD
Standard VoIP Wholesale	600 (10 min)	2	0 (disabled)	30 sec
Call Center Operations	900 (15 min)	3	14400 (4 hrs)	20 sec
Mobile/Unstable Networks	300 (5 min)	2	3600 (1 hr)	15 sec
Enterprise PBX	1200 (20 min)	2	28800 (8 hrs)	30 sec
High-Security Environment	180 (3 min)	2	1800 (30 min)	10 sec

📊 Session Timer Message Flow Diagram

VOS3000 Session Timer - Complete Call Flow with Refresh:
=========================================================

CALLER                          VOS3000                         CALLEE
  │                               │                               │
  │  1. INVITE                    │                               │
  │  Session-Expires: 600         │                               │
  │  Min-SE: 90                   │                               │
  │──────────────────────────────►│                               │
  │                               │  2. INVITE (forwarded)        │
  │                               │  Session-Expires: 600         │
  │                               │──────────────────────────────►│
  │                               │                               │
  │                               │  3. 200 OK                    │
  │                               │  Session-Expires: 600         │
  │                               │◄──────────────────────────────│
  │  4. 200 OK                    │                               │
  │  Session-Expires: 600         │                               │
  │◄──────────────────────────────│                               │
  │                               │                               │
  │  5. ACK                       │                               │
  │──────────────────────────────►│  6. ACK                       │
  │                               │──────────────────────────────►│
  │                               │                               │
  │           ═════════════════════════════════════════           │
  │           ║    CALL ACTIVE - AUDIO FLOWING           ║        │
  │           ═════════════════════════════════════════           │
  │                               │                               │
  │  [5 minutes into call]        │                               │
  │                               │                               │
  │  7. UPDATE (session refresh)  │                               │
  │  Session-Expires: 600         │                               │
  │◄──────────────────────────────│                               │
  │  8. 200 OK                    │                               │
  │  Session-Expires: 600         │                               │
  │──────────────────────────────►│                               │
  │                               │  9. UPDATE (session refresh)  │
  │                               │──────────────────────────────►│
  │                               │  10. 200 OK                   │
  │                               │◄──────────────────────────────│
  │                               │                               │
  │           ═════════════════════════════════════════           │
  │           ║    SESSION REFRESHED SUCCESSFULLY       ║        │
  │           ═════════════════════════════════════════           │
  │                               │                               │
  │  [If refresh fails]           │                               │
  │                               │                               │
  │  11. BYE (session timeout)    │                               │
  │◄──────────────────────────────│  12. BYE (session timeout)    │
  │                               │──────────────────────────────►│
  │                               │                               │
  │  CDR: Termination Reason = "Session Timeout"                 │
  │                               │                               │

🚨 Session Timer Troubleshooting Guide

📊 Common Problems and Solutions

🚨 Symptom	🔍 Root Cause	✅ Solution	📖 Reference
Calls drop at exactly 30 seconds	NAT binding timeout, not session timer	Enable NAT keep-alive, reduce period to 15-20s	Page 212
Calls drop at 5-minute intervals	Session refresh failing	Check if endpoint supports re-INVITE/UPDATE	Page 213
“422 Session Interval Too Small” error	Session-Expires below minimum	Increase SS_SIP_SESSION_MIN_SE or TTL	Page 231
No incoming calls after idle period	NAT binding expired	Verify NAT keep-alive is enabled and working	Page 212
Re-INVITE rejected with 491	Glare condition (simultaneous re-INVITEs)	Normal – VOS3000 will retry automatically	Page 213
Zombie calls still occurring	Session timer not negotiated	Check NO_TIMER_REINVITE_INTERVAL setting	Page 230

🔧 Debug Trace Analysis for Session Timer

VOS3000 Debug Trace - Session Timer Analysis:
==============================================

Step 1: Enable Debug Trace
Navigation: System → Debug trace
Enable: Check "On"
Set duration: 10-30 minutes

Step 2: Look for Session Timer Headers in SIP Messages:
───────────────────────────────────────────────────────

INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK12345
From: ;tag=abc123
To: 
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: 
Session-Expires: 600;refresher=uac    ← SESSION TIMER HEADER
Min-SE: 90                            ← MINIMUM SESSION EXPIRES
Content-Type: application/sdp
Content-Length: ...

Step 3: Check 200 OK Response:
──────────────────────────────
SIP/2.0 200 OK
...
Session-Expires: 600;refresher=uac    ← CONFIRMED SESSION TIMER
...

Step 4: Look for Session Refresh Messages (UPDATE or re-INVITE):
────────────────────────────────────────────────────────────────

UPDATE sip:[email protected]:5060 SIP/2.0
...
Session-Expires: 600                    ← REFRESHING SESSION
...

Step 5: If No Session Timer Headers Found:
──────────────────────────────────────────
- Endpoint does not support RFC 4028
- VOS3000 will use SS_SIP_NO_TIMER_REINVITE_INTERVAL
- Maximum call duration will be enforced

📊 Session Timer vs NAT Keep-Alive Comparison

📊 Aspect	⏱️ Session Timer	📡 NAT Keep-Alive
Primary Purpose	Detect failed calls, prevent zombie sessions	Maintain NAT bindings for incoming calls
RFC Standard	RFC 4028 (SIP Session Timer)	NAT traversal best practices
Protocol Used	SIP re-INVITE or UPDATE messages	UDP packets or SIP messages
When Active	During active call (after 200 OK)	While endpoint is registered
Direction	Bidirectional (negotiated refresh)	Server to endpoint (unidirectional)
Default Interval	600 seconds (10 minutes)	30 seconds
Failure Result	Call terminated, CDR updated	Incoming calls may fail
Endpoint Support Required	Yes (RFC 4028 compliance)	No (transparent to endpoint)

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📦 Service	📝 Description	💼 Includes
VOS3000 Installation	Complete server setup	OS, VOS3000, Database, Security
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❓ Frequently Asked Questions about VOS3000 Session Timer

What happens if an endpoint doesn’t support session timer?

VOS3000 will use the SS_SIP_NO_TIMER_REINVITE_INTERVAL parameter to limit the maximum call duration. This ensures that zombie calls cannot persist indefinitely even when the endpoint doesn’t support RFC 4028. Set this value based on your business requirements (default is 7200 seconds or 2 hours).

Why are my calls dropping exactly at 30 seconds?

30-second call drops are almost always caused by NAT binding timeout, not session timer issues. The solution is to enable NAT keep-alive by setting SS_SIP_NAT_KEEP_ALIVE_MESSAGE to a value like “HELLO” and reducing SS_SIP_NAT_KEEP_ALIVE_PERIOD to 15-20 seconds. Also check if SIP ALG is enabled on your router (it should be disabled).

What is the difference between re-INVITE and UPDATE for session refresh?

Both methods can be used for session refresh. UPDATE is generally preferred because it doesn’t modify the SDP session parameters, while re-INVITE also renegotiates media. VOS3000 automatically selects the appropriate method based on endpoint capabilities and configuration.

How do I calculate the optimal session timer refresh interval?

The refresh interval equals SS_SIP_SESSION_TTL divided by SS_SIP_SESSION_UPDATE_SEGMENT. With defaults (600 ÷ 2 = 300 seconds), VOS3000 sends a refresh every 5 minutes. For mobile networks, consider 300 ÷ 2 = 150 seconds for faster failure detection.

Can session timer prevent billing fraud?

Session timer helps prevent zombie calls that could result in incorrect CDR durations, but it’s not a fraud prevention mechanism. For fraud protection, implement proper account limits, IP restrictions, and monitor for unusual calling patterns using VOS3000’s built-in reports.

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SIP ALG Problems, VOS3000 gateway configuration, VoIP Fraud Prevention, VOS3000 Media Proxy, VOS3000 Call Termination Reasons

VOS3000 Call Termination Reasons: Complete Troubleshooting Reference Guide

March 31, 2026March 31, 2026 king

VOS3000 Call Termination Reasons: Complete Troubleshooting Reference Guide

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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🔍 Understanding Call Termination in VOS3000

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.

📊 Where to Find Termination Information (VOS3000 Call Termination Reasons)

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 (VOS3000 Call Termination Reasons)

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

🔴 Number Restriction Termination Reasons

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

🔴 Protocol and System Termination Reasons (VOS3000 Call Termination Reasons)

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

📋 H.323 Device Error Codes (VOS3000 Call Termination Reasons)

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.

📊 Common H.323 Cause Codes (VOS3000 Call Termination Reasons)

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 Error Codes Reference

SIP responses follow standard HTTP-style status codes. Understanding these codes helps diagnose problems when they appear in CDR records or Call Analysis.

📊 SIP 4xx Client Error Codes (VOS3000 Call Termination Reasons)

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

📊 SIP 5xx Server Error Codes

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

📊 SIP 6xx Global Failure Codes

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

🔧 Using Call Analysis for Troubleshooting

VOS3000 provides detailed call analysis tools that show the complete SIP/H.323 signaling flow, making it easier to diagnose complex problems.

⚙️ Call Analysis Tool Usage

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

📊 CDR Analysis for Pattern Detection

Regular CDR analysis helps identify recurring issues before they become major problems.

⚙️ Key Metrics to Monitor

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;

📖 VOS3000 Easy Troubleshooting Guide
📖 VOS3000 FAQ Based on Official Manual
📖 How to Stop Illegal Calls in VOS3000
📖 VOS3000 Security and Hacking Prevention
📖 VOS3000 Downloads

❓ Frequently Asked Questions

What is the most common call termination reason?

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.

How do I differentiate between timeout types?

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.

Why do I see “Insufficient Balance” for accounts with credit?

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.

What causes “No Matching Rate” errors?

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.

How do I troubleshoot intermittent “Session Timeout” errors?

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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VOS3000 Llamadas Cortadas Essential: Diagnostico Completo 📞

Table of Contents

Causas Principales de Llamadas Cortadas 📊

RTP Timeout: La Causa Mas Comun ⏱️

SIP Session Timer 🔄

Firewall UDP Timeout 🔥

Failover y Cambio de Ruta 🔀

Diagnostico Paso a Paso 🔍

Preguntas Frecuentes sobre VOS3000 Llamadas Cortadas ❓

❓ Por que se cortan las llamadas en VOS3000 despues de unos minutos?

❓ Como evito que las llamadas se corten por RTP timeout?

❓ El failover puede cortar llamadas existentes?

❓ Como verifico por que se corto una llamada en VOS3000?

❓ Que es el SIP NAT keepalive y como ayuda?

❓ Las llamadas se cortan siempre a los 32 segundos, que significa?

Conclusion 🏆

📞 Need Professional VOS3000 Setup Support?

VOS3000 Call End Reasons: Complete Error Code Reference Guide

Table of Contents

🔍 Understanding VOS3000 Call End Reasons

📊 Where to Find VOS3000 Call End Reasons

📋 Complete VOS3000 Server-Side Termination Reasons

🔴 Account-Related Termination Reasons

🔴 Routing-Related Termination Reasons (VOS3000 Call End Reasons)

🔴 Timeout-Related Termination Reasons

🔴 Number Restriction Termination Reasons

🔴 Forwarding-Related Termination Reasons

🔴 Protocol and System Termination Reasons

📞 SIP Response Code to VOS3000 Mapping

📊 SIP 4xx Client Error Mapping

📊 SIP 5xx Server Error Mapping

📞 H.323 Cause Codes Complete Reference

📊 H.323 Q.931 Cause Codes Table (VOS3000 Call End Reasons)

🔧 CDR Analysis and Diagnosis Flow

🔍 VOS3000 Call End Reasons Diagnosis Workflow

💰 VOS3000 Support Services

🔗 Related Resources (VOS3000 Call End Reasons)

❓ Frequently Asked Questions

What is the difference between “Response Timeout” and “Connection Timeout”?

Why do I see different error codes for similar failures?

How can I filter CDR to see only failed calls?

What causes “Codec Mismatch” errors?

How do I interpret H.323 cause codes in CDR?

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📞 Need Professional VOS3000 Setup Support?

VOS3000 Session Timer: Complete Guide to SIP Keep-Alive Configuration

Table of Contents

🔍 What is VOS3000 Session Timer?

📊 The Zombie Call Problem

⚙️ VOS3000 Session Timer Parameters Complete Reference

📊 Core Session Timer Parameters Table

📊 Session Timer Refresh Calculation

📐 Session Timer Refresh Interval Formula

📡 NAT Keep-Alive Configuration Deep Dive

📊 NAT Keep-Alive Parameters Table

🔄 How NAT Keep-Alive Works in VOS3000

🔧 VOS3000 Session Timer Configuration Guide

📍 Navigation to System Parameters

⚙️ Recommended Configuration by Deployment Type

📊 Session Timer Message Flow Diagram

🚨 Session Timer Troubleshooting Guide

📊 Common Problems and Solutions

🔧 Debug Trace Analysis for Session Timer

📊 Session Timer vs NAT Keep-Alive Comparison

💰 VOS3000 Installation and Support Services

🔗 Related Resources

❓ Frequently Asked Questions about VOS3000 Session Timer

What happens if an endpoint doesn’t support session timer?

Why are my calls dropping exactly at 30 seconds?

What is the difference between re-INVITE and UPDATE for session refresh?

How do I calculate the optimal session timer refresh interval?

Can session timer prevent billing fraud?

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VOS3000 Call Termination Reasons: Complete Troubleshooting Reference Guide

Table of Contents

🔍 Understanding Call Termination in VOS3000

📊 Where to Find Termination Information (VOS3000 Call Termination Reasons)

📋 Server-Side Termination Reasons (VOS3000 Call Termination Reasons)

🔴 Account-Related Termination Reasons (VOS3000 Call Termination Reasons)