>> MIB - Management Information Base

>> Table: voip - (.1.3.6.1.4.1.272.4.33)

voip
OIDNameTypeAccess
.6voipSipsipN
.10CallHistoryMaxEntriesINTEGERRW
.11AdminStatusENUMRW
.22voipStatMIBvoipN
.26TraceAdminStatusENUMRW
.27TraceBufferIDINTEGERR
.28TraceModeENUMRW
.40ProvProfMgmtCmdENUMRW
.41ProvProfMgmtHostDisplayStringRW
.42ProvProfMgmtFileDisplayStringRW
.43ProvProfMgmtPathDisplayStringRW
.44ProvProfMgmtIndexINTEGERRW
.45ProvProfMgmtDescrDisplayStringRW
.47OutboundInterface0INTEGERRW
.48OutboundInterface1INTEGERRW
.49OutboundInterface2INTEGERRW
.1DialLatencyINTEGERRW
.2TerminationENUMRW
.3OperStatusENUMR
.4LastErrorDisplayStringR
.5SBCINTEGERRW
.6DropExtensionDisplayStringRW
.1voipStatRtcpHistoryMaxEntriesINTEGERRW
.20ServerCertIndexINTEGERRW
.21ServerCertSourceENUMRW
.22MaxConcurrentServerTransactionsINTEGERRW
.23MaxServerTransactionsPerSecondINTEGERRW
.24MaxServerTransactionsPerPeerINTEGERRW
.25RoutingPreferenceENUMRW
.26DefaultCountryCodeDisplayStringRW
.27DefaultAreaCodeDisplayStringRW
.28SipPortINTEGERRW
.29SipNatPortINTEGERRW
.30DscpSignallingINTEGERRW
.31IsdnTypeOfNumberENUMRW
.32IpVersionENUMRW

voipSip
description not available
CallHistoryMaxEntries
Maximum number of voipCallHistory entries in memory.

Range: 0 to 65535

AdminStatus
Globally enable or disable VoIP features. Setting this to 'disable' disables all VoIP dependent subsystems including Mediagateway.

Enumerations:

  • disabled (1)
  • enabled (2)
voipStatMIB
description not available
TraceAdminStatus
Globally set VoIP trace buffer feature. 'disabled' no trace active or stop active trace 'enabled' trace buffer is active using voipTraceMode 'delete' delete existing voip trace buffer, voipTraceBufferID will be set to -1 and voipTraceAdminStatus resets to 'disabled'.

Enumerations:

  • disabled (1)
  • enabled (2)
  • delete (3)
TraceBufferID
If a trace buffer is available the used ID is saved here. In trcBufTable an entry with that ID (trcBufID) exists. If value is -1, no trace buffer is available.
TraceMode
Globally set VoIP trace buffer mode. 'state1' dump all registrations and calls (snapshot) 'state2' dump all registrations, calls, transactions and internal DNS results (snapshot) 'events1' trace current state changes and errors 'events2' trace current state changes, errors and all SIP messages 'sip' trace all sent and reveived SIP messages only

Enumerations:

  • state1 (1)
  • state2 (2)
  • events1 (3)
  • events2 (4)
  • sip (5)
ProvProfMgmtCmd
set VoIP provider profile management command. 'put' export profile via tftp 'get' import profile via tftp 'put-local' export profile(s) into dedicated configuration determined via 'voipProvProfMgmtPath' 'get-local' import profile(s) from dedicated configuration determined via 'voipProvProfMgmtPath' 'copy-prof' create custom-defined profile copy from predefined profile determined via 'voipProvProfMgmtIndex'

Enumerations:

  • none (1)
  • put (2)
  • get (3)
  • put-local (4)
  • get-local (5)
  • copy-prof (6)
ProvProfMgmtHost
TFTP host's IP address to send/receive configuration files containing voip provider profile entries.

Length: 0 to 255

ProvProfMgmtFile
TFTP filename to send/receive configuration files containing voip provider profile entries.

Length: 0 to 255

ProvProfMgmtPath
Name of the dedicated configuration in flash to be used for import/export of voip provider profile(s).

Length: 0 to 255

ProvProfMgmtIndex
Index of voipProviderProfileTable entry to be copied into a new voipProviderProfMgmtTable entry - triggered via voipProvProfMgmtCmd copy-prof (5).

Range: 1 to 5000

ProvProfMgmtDescr
Description of the new provider profile entry.

Length: 0 to 255

OutboundInterface0
If set this interface index value determines the outbound interface associated with the local VoIP service. In conjunction with the (optional) voipOutboundInterface1 and voipOutboundInterface2 it represents a list of interfaces, the aim in this case is to support (VoIP-)interface-autodetection too. Note that within this list no prioritization, best-match selection or fallback policy takes place.
OutboundInterface1
If set this interface index value determines the outbound interface associated with the local VoIP service. In conjunction with the (optional) voipOutboundInterface0 and voipOutboundInterface2 it represents a list of interfaces, the aim in this case is to support (VoIP-)interface-autodetection too. Note that within this list no prioritization, best-match selection or fallback policy takes place.
OutboundInterface2
If set this interface index value determines the outbound interface associated with the local VoIP service. In conjunction with the (optional) voipOutboundInterface0 and voipOutboundInterface1 it represents a list of interfaces, the aim in this case is to support (VoIP-)interface-autodetection too. Note that within this list no prioritization, best-match selection or fallback policy takes place.
DialLatency
The dial delay in seconds after the last entered digit.

Range: 1 to 15

Termination
This value describes the RTP stream routing between SIP phones:

'off' -> RTP streams between phones are routed directly; 'on' -> RTP streams are always terminated in the gateway.

This value can be set to off to reduce systemload. In this case all SIP phones in the network must be able to connect over IP routing. Otherwise the Mediagateway is always endpoint of the RTP stream and the streams are connected in the gateway.

Enumerations:

  • off (1)
  • on (2)
OperStatus
The operational status of the Mediagateway.

Enumerations:

  • down (1)
  • up (2)
  • failed (3)
LastError
The last error message on PABX startup.

Length: 0 to 255

SBC
Set the default Session Border Controller (SBC) behaviour.

A Session Border Controller is a SIP provider account which takes over the PBX functionality of the Mediagateway. The 'DDIMode' of the used SIP provider account has to be anything but 'off'. All internal and external activities are controlled by the SBC. If the SBC is disabled ('-1') or the SBC is not available the Mediagateway uses the internal PBX functionality.

Auto mode ('0') means that a separate SIP account is used for each existing extension entry (so in this case it is not possible to set one global SBC). In auto mode each voipExtensionTable entry needs a matching voipProviderTable/voipSipProviderTable entry. So in auto mode Mediagateway pairs SIP provider accounts and their matching extensions.

Possible values:

-1 -> SBC is disabled; 0 -> Auto mode; 1...999 -> Index of a voipSipProviderTable entry (has to be a SIP trunk provider).

DropExtension
Fallback number which is dialed if no matching call endpoint (extension) for a dialed number is found. Fallback number can be an internal extension or another external call endpoint.

Length: 0 to 255

voipStatRtcpHistoryMaxEntries
Maximum number of voipStatRtcpHistory entries in memory.

Range: 0 to 65535

ServerCertIndex
Index of certificate entry in certTable which is to be used as server certificate for SIPS transport. This parameter is only used if 'ServerCertSource' = 'certtable'.
ServerCertSource
Source of server certificate for SIPS transport:

'internal' -> 'ServerCertIndex' is not used, internal server certificate of SIP stack is used instead; 'certtable' -> use certificate from certTable entry specified by 'ServerCertIndex'.

Default value is 'internal'.

Enumerations:

  • internal (1)
  • certtable (2)
MaxConcurrentServerTransactions
The maximum amount of concurrent server transactions created by incoming INVITE, REGISTER, ... If the limit is reached, new server transactions will be ignored. A value of 0 means no limitation

Range: 0 to 10000

MaxServerTransactionsPerSecond
The maximum amount of server transactions per second created by incoming INVITE, REGISTER, ... If the limit is reached, new server transactions will be ignored. A value of 0 means no limitation

Range: 0 to 10000

MaxServerTransactionsPerPeer
The maximum amount of server transactions per peer created by incoming INVITE, REGISTER, ... If the limit is reached, new server transactions from that peer will be ignored. A value of 0 means no limitation

Range: 0 to 10000

RoutingPreference
The routing preference for the received called address. - local matching local extensions are preferred - routing matching configured routes are preferred

Enumerations:

  • local (1)
  • routing (2)
DefaultCountryCode
The default country code. This value is used by GUI VoIP configuration to set or modify provider specific dialing rules.

Length: 0 to 6

DefaultAreaCode
The default area code. This value is used by GUI VoIP configuration to set or modify provider specific dialing rules.

Length: 0 to 12

SipPort
Standard value of the SIP port.

Range: 0 to 65535

SipNatPort
The external port mapped by NAT. For security reasons this might be different than the internal port (mpsVoIPConfigSipPort).

Range: 0 to 65535

DscpSignalling
The 6 bit DSCP value used in IP header for signalling data.

Range: 0 to 63

IsdnTypeOfNumber
Specifies the type of of number for the calling number in ISDN setup: standard - calling number is not modified and type of number is unknown. specific - calling number is checked for international, national or subscriber. The type of number is set to the result. The calling number is modified if the type is international or national.

Enumerations:

  • standard (1)
  • specific (2)
IpVersion
IP version used for VoIP communication.

Enumerations:

  • ipv4 (1)
  • ipv4-ipv6 (2)


MIB Reference to Software Version 10.2.12 generated on 2023/08/29. Provided by webmaster@bintec-elmeg.com
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