| 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: |
| OperStatus |
| The operational status of the Mediagateway. Enumerations: |
| 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 |
| 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)
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| 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: |
| 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: |
| IpVersion |
| IP version used for VoIP communication. Enumerations: |