Index |
A unique value for each trunk group. Range: 1 to 65535 |
Type |
This value determines the type of the VoIP provider. Enumerations: |
Mode |
This value determine the mode of the VoIP. Enumerations: |
Descr |
This variable contains a textual description of the VoIP provider. Length: 0 to 20 |
Bundle |
This variable contains the number of the bundle for this provider. |
Domain |
This variable contains name, port and transport of the registrar. example: 192.0.0.1:5060;tcp or phone30.bintec.de:5060;upd Length: 0 to 64 |
IpAddress |
This variable contains IP adress of the registrar. |
UserAccount |
This variable contains the user account. Length: 0 to 64 |
AuthName |
This variable contains authorisation name. Length: 0 to 64 |
AuthPassword |
This variable contains authorisation password. Length: 0 to 64 |
Facilities |
Special facilities of the IP provider. internationalNumber: Generate international phone number. CLIR: 1 -> Deactivate number suppression. notRegistered: Not registered with VoIP provider. logInProxy: Allow login of a proxy. useAreaCode: add area code to dialed number noHoldRetrieve: no hold/retrieve message to the provider replaceIntPrefix: replace intern. prefix (e.g. '00) with '+' clrProvBind: clear provider binding natPing: If set to 0 no NAT ping packets will be send. Set to 1 periodic NAT ping packets will be send. earlyConnect: allow early SIP connect srtp: allow SRTP negotiation noT38: 1 -> FAX via T.38 disabled callDeflection: redirect an unanswered call to another endpoint during the call setup phase (code 302) sendUpdate: send SIP UPDATE sendUserAccinPpreferred:Always send user account in P-Preferred-identity header field sendUserAccinPasserted: Always send user account in P-Asserted-identity header field verifyTlsCa: Check peer certificate with local CAs verifyTlsHostname: Check peer certificate with contacted hostname. video: If set to 1 video will be negotiated between IP endpoints. bnc: If set to 1, bnc (bulk number contact) will be used for outgoing registration checkServerSrc: If set to 1, incoming INVITE must be sent from registrar address. getCalledNumberFromRequestUri: For incoming calls the called party number is retrieved from request URI instead of To header. fetchProvBind: Initial registration uses 'fetch bindings'. This is a query for existing bindings. mediasec: Send 'mediasec' information in REGISTER to tell that SRTP is used in calls. srtp must be set too. singleCodecAnswer: Send answer (SDP) with a single audio codec only. notRegisteredOptions: Additional option for 'notRegistered'. If set, send periodical SIP message OPTIONS. Enumerations: - internationalNumber (0)
- clir (1)
- notRegistered (3)
- logInProxy (4)
- useAreaCode (6)
- noHoldRetrieve (7)
- replaceIntPrefix (8)
- clrProvBind (9)
- natPing (10)
- earlyConnect (12)
- srtp (13)
- noT38 (14)
- callDeflection (15)
- sendUpdate (16)
- sendUserAccinPpreferred (17)
- sendUserAccinPasserted (18)
- verifyTlsCa (19)
- verifyTlsHostname (20)
- video (21)
- bnc (22)
- checkServerSrc (23)
- getCalledNumberFromRequestUri (24)
- fetchProvBind (25)
- mediasec (26)
- singleCodecAnswer (27)
- notRegisteredPeriodicOptions (28)
|
Network |
This variable contains the special location index. Range: 1 to 18 |
InviteTimeout |
This variable contains the length of the end of dial timer. |
AdminStatus |
VoIP Provider is active or not active. Set to delete, to discard the whole entry. Enumerations: - enable (1)
- disable (2)
- delete (3)
|
OperationStatus |
This variable contains the actual operation status. notregistered : client is not registered registering : notregistered, registration pending registered : client is registered reregistering : registered, reregistration pending Enumerations: - notregistered (1)
- registering (2)
- registered (3)
- reregistering (4)
|
TestCommand |
This variable contains the current status. Set to test, to test the VoIP provider. Automatically set to 'idle' when complete. Enumerations: |
TestResult |
This variable contains the current status. Enumerations: - idle (1)
- progress (2)
- success (3)
- error (4)
|
CurrentCalls |
This variable contains the current calls. |
ReregisterTimer |
This variable contains reregistering timer in seconds. Range: 1 to 86400 |
Codecs |
The supported Codecs of the Provider set one or more of the following bit values: ulaw (1) alaw (2) g729 (4) g726 (8) g726_16 (16) g726_32 (32) g726_40 (64) g728 (128) g723_63 (256) g723_53 (512) g729_e (1024) gsm (2048) dtmf (4096) Default ulaw, alaw, g.729 and dtmf. Range: 0 to -1 |
CodecOrder |
Sorting the codecs default -> system default order quality -> highest quality first lowest -> lowest bandwidth first highest -> highest bandwidth first Enumerations: - default (1)
- quality (2)
- lowest (3)
- highest (4)
|
Proxy |
This variable contains name, port and transport of the outbound proxy. example: 192.0.0.1:5060;tcp or phone30.bintec.de:5060;upd Length: 0 to 64 |
STUNServer |
This variable contains name and portof the STUN server. example: 192.0.0.1:5060 or phone30.bintec.de:5060 Length: 0 to 64 |
MaxSimCalls |
This variable contains the maximum of simultaneous connection for this provider. 0 = no limitation. |
ReplaceSrc |
This variable contains the source string for substitution of the incoming number prefix. If the incoming number prefix is equal to this string, the incoming number prefix will be replaced with the string in mpsTrunkGroupVoIPReplaceDest. Length: 0 to 16 |
ReplaceDest |
String substitution, if the incoming number prefix is equal to the mpsTrunkGroupVoIPReplaceSrc string. Length: 0 to 16 |
IfIndex |
Reference to the interface index mpsDeviceIfIndex of the mspDeviceTable . Range: 1 to 2147483647 |
CodecProperties |
A reference to mpsCodecPropertyTable for codec properties. If '0' use system default. Range: 0 to 65535 |
Realm |
The Realm/Domain of the provider. This domain is used e.g. at registration or authentication instead of registration domain. Length: 0 to 64 |
FromRealm |
The From Realm/Domain of the provider. This domain is used after the '@' in the From information element in the SIP header. Length: 0 to 64 |
CLIP |
Use the phone number instead of the user account in the SIP header to indidcate the caller identity. This is provider dependent: display : transfer in DISPLAY of the FROM header field. user : transfer in USER (account) field in the FROM header field. p-preferred-identity : transfer in the P-Preferred-Identity header field. p-asserted-identity : transfer in the P-Asserted-Identity header field. Enumerations: - display (0)
- user (1)
- p-preferred-identity (2)
- p-asserted-identity (3)
|
CLIR |
Type of number suppression in outgoing SIP calls. This is provider dependent: display : anonymous in display of the From header field. user : anonymous in user (account) field in the From header field. domain : anonymous.invalid in domain of the From header field. privacy-header : Privacy header field with value 'header'. privacy-user : Privacy header field with value 'user'. privacy-id : Privacy header field with value 'id'. Enumerations: - display (0)
- user (1)
- domain (2)
- privacy-header (3)
- privacy-user (4)
- privacy-id (5)
|
ProfileNo |
This variable is only used internally and managed by TR069 daemon. An entry gets assigned to a VoIP Profile by it. Profile number 1 means that entry isn't modifiable by GUI. Profile 3 is even not not shown in GUI and gets only used if no other provider entry is available. A value of 0 means 'any other profile' |
SipProvProfile |
Index to the SIP provider profile (voipProviderProfileIndex) which was used to create this VoIP provider entry. This variable is used internally by GUI. (0 means undefined) |