با سلام
من یک تنور axt800 دارم که برای ارتباط با pstn استفاده میشه
همه چی درست کار میکنه اما وقتی از بیرون زنگ میزنن کالر ایدی anonymouse نمایش داده میشه
اشکال از خط نیست چون قبلا dinstarداشتم و به خاطر کیفیت کم با tenor عوض کردم و تو dinstar مشکل کالر آیدی نداشتم.
dial plan و cas signallin و trunk group رو براتون میذارم.
-DialPlan-1
/*----------------
MAXDNlength: 30 (max DN length)
MINDNlength: 1 (min DN length)
LongDistancePrefix: (Not Set) (Numeric string)
CarrierPrefixPattern: (Not Set) (101xxxx;x is 0 to 9)
INTernationaLPrefix[1..8]: (Not Set)
IntercomEnable: 0 (No; default)
InterDigitTimeout: 4 (secs; default)
DialPlanCountry: 255 (No dial plan,default)
ProgressToneCountry: 0 (US)
AllowUnmatchedNumbers: 1 (enable)
CountryCode: (Not Set) (digits 1 to 7)
AreaCode: (Not Set) (Numeric String)
------------------*/
-VoiceCodec-1
/*----------------
name: (Not Set) (name)
CodecVoiceCoding: 9 (G.711 Mu-Law)
CodecPayloadSize: 20 (ms)
------------------*/
-VoiceCodec-2
/*----------------
name: (Not Set) (name)
CodecVoiceCoding: 8 (G.711 A-Law)
CodecPayloadSize: 20 (ms)
------------------*/
-CASSignalingGroup-line
/*----------------
id: line (read only)
name: (Not Set) (name)
ORientation: 0 (Slave, User; default)
SignalingType: 1 (Loop Start)
GuardTime: 2 (numeric,default 10 ms)
Ring12Delay: 4 (numeric;seconds)
DialToneDetect: 1 (Yes, dial tone detect before di)
DialDelay: 500 (ms delay; 0-default)
AnswerDelay: 0 (seconds,)
FlashhookSignal: 0 (No)
ForwardDisconnectDelay: 2 (ms delay,)
CallerIDDetection: 1 (FSK or DTMF)
SilenceBasedDisconnect: 0 (seconds)
ToneBasedSupervision: 1 (Disconnect)
MAXRingFrequency: 40 (numeric,default 50Hz)
MINRingFrequency: 15 (numeric,default 15Hz)
DisconnectSupervisionOptions: 3 (on,off interval)
RingValidation: 1 (enabled)
DTMFONTime: 50 (50-1000ms,def 100)
DTMFOFFTime: 50 (50-1000ms,def 100)
RXGain: -2 (db)
TXGain: 0 (db)
Impedance: 0 (600 ohms,default)
LineTemplate: 0 (See CLI guide)
LocalLoopType: 0 (EIA Type 0, default)
DialMode: 0 (DTMF Tone Dial)
------------------*/
-SIPSignalingGroup-1
/*----------------
SIPInfoFormat: 0 (Nortel,default)
ProxyFailoverBehavior: 1 (fail over)
PrimarySIPServerPort: 5060 (portnumber,def 5060)
PrimarySIPServer: 10.20.30.1 (IP address)
SecondarySIPServerPort: 5060 (port number,def 5060)
SecondarySIPServer: (Not Set) (IP address)
OutboundPrimaryServerPort: 5060 (port number,def 5060)
OutboundPrimaryServer: (Not Set) (IP address)
OutboundSecondaryServerPort: 5060 (port number,def 5060)
OutboundSecondaryServer: (Not Set) (IP address)
RegisterExpiryTime: 30 (seconds,def 300)
AllowOnlyProxyCalls: 1 (from proxy,default)
SDPin180Ringing: 1 (send SDP)
SIPServerInFromHdr: 1 (use SIP Server IP)
SDPin183Progress: 1 (yes,default)
IncludeQuintumHeader: 0 (disable)
UserNameInContact: 0 (default,contact info not sent)
RequestReTransmitCount: 11 (default 11)
UserNameAsURI: 0 (disable,default)
MaxForwards: 70 (default 70)
PRACKMethod: 1 (supported)
Send180Ringing: 1 (enabled)
Send183Progress: 1 (enabled)
SipPstnInterworking: 0 (disabled)
SIPNoConnectTimeout: 180 (seconds,def 180)
SessionTimer: 0 (disabled)
UserAgentHeader: Quintum/1.0.0 (0-31 char,def Quintum1.0.0)
MWIServer: (Not Set) (IP address)
MWIServerPort: 5060 (Numeric,default 5060)
MWISubscribeDuration: 60 (minutes,def 60)
MWIFailureRetryDelay: 30 (minutes,def 30)
CLIHeaderType: 0 (disabled)
UseFromContact: 0 (DN)
SIPTelephoneEvents: 1 (Append "/1" to SDP attribute)
SIPUseDNinRegister: 1 (Use DN from channel map in Regi)
DirectGWRouting: 0 (disable)
ConferenceFactoryName: (Not Set) (0-31 charac,name)
ConferenceFactoryServer: (Not Set) (server name)
ConferenceFactoryPort: 5060 (port num, def 5060)
SIPRegisterOnBootup: 0 (disable)
SDPinInvite: 1 (send SDP)
------------------*/
-UserAgent-101
/*----------------
id: 101
Enabled: 1 (enabled)
ListenPort: 5060 (port,default 5060)
PrimaryUser: tenor (username)
PrimaryPassWord: tenor (up to 31 char)
SecondaryUser: (Not Set) (username)
SecondaryPassWord: (Not Set) (up to 31 char)
Contacts[1..2]: (Not Set)
MWIUserName: (Not Set) (name,15 char)
MWIPassword: (Not Set) (password,15 char)
------------------*/
-TrunkCircuitRoutingGroup-line
/*----------------
id: line
name: (Not Set)
OverlapDial: 0 (No, default)
DIRection: 2 (Bi-direction;default)
ProgressTone: 0 (Off;default)
EndOfDial: 1 (Enable)
EndOfDialDigit: # (default;#,*,0-9)
AddEndOfDialDigit: 0 (No;default)
ForcedRoutingNum: 1000 (A Dialed Number)
ForcedRoutingNumType: 0 (Public; default)
TrunkID: (Not Set) (Numeric string)
TrunkIDDelivery: 0 (No Delivery)
TrunkIDDeliveryFormat: cid (0-9,*,#,literal character)
HUNTAlgorithm: 0 (Ascending)
ModemBypass: 0 (Disabled)
PassThroughEnable: 0 (No; default)
PassThroughID: 0 (Idx of a LCRG;default)
MaxHopoffCallsAllowed: -1 (default)
BusyOut: 0 (Ring Back)
TwoStageDialing: 0 (No)
ProvideAutoSwitchProgressTone: 0 (No,Default)
IVRType: 0 (No IVR; default)
Hairpinning: 0 (No,Default)
EnableExternalRouting: 0 (No; Default)
MaxTalkTime: 0 (Mins,default 0)
InBoundAccessLevel: 0 (0 to 127)
OutBoundAccessLevel: 0 (0 to 127)
StopAcctID: 0 (IP Addr,default)
CallerIDType: 0 (Caller ID)
Play1700Prompt: 0 (No;Default)
PrefiXTrunkID: 0 (No;Default)
AnswerDisconnectTone: 0 (disabled,default)
OutboundANINumDigits: 0 (caller id sent out)
CallTerminationINDication: 0 (disable)
EnableAutoSwitch: 0
AutoSwitchNumberType: 0 (DID received)
AutoSwitchNumber: (Not Set) (DN)
CallWaiting: 0 (disable)
HolD: 0 (disable)
UnattendedTRansfer: 0 (disable)
AttendedTRansfer: 0 (disable)
CallWaitToConfKeystroke: 22 (keystroke,def 22)
HopoffNumberDirAttached[1]: HopoffNumberDirectory-1
HopoffNumberDirAttached[2..16]: (unspecified)
UserAgentAttached: UserAgent-101
CallerIDXlateDirAttached: (unspecified)
OutboundCallerIDXlateDirAttach: (unspecified)
InboundDNISXlateDirAttached: (unspecified)
------------------*/
-ToneProfile-1
/*----------------
id: 1
name: (Not Set)
DiscToneFreq1: 480 (100-2000 Hz,default 480 Hz)
DiscToneFreq2: 620 (100-2000 Hz,default 600 Hz)
DiscToneONTime: 250 (0-2000 ms, default 0 ms)
DiscToneOFFTime: 250 (0-2000 ms, default 0 ms)
------------------*/