Generic selectors
Exact matches only
Search in title
Search in content
Post Type Selectors

Tenor disconnect command

Viewing 5 posts - 1 through 5 (of 5 total)
  • Author
    Posts
  • #27509
    Saleh
    Guest

    hello

    can we configure the tenor so that a user can press * for example to disconnect the call?

    we have a proplem that many times: when a user call and end the call, the tenor dose not end the call

    #27510
    Mark
    Guest

    Saleh,

    Unfortunately no such command exists. If you are having a problem with the line disconnecting, most likely this is due to an incorrect configuration for disconnect supervision. You may want to try to configure the unit for disconnect tone (config pstntg 1# supervision 1) and make sure that the cassignaling is set to loop start only.

    You may want to review the document on Quintum’s Web site that discusses disconnect supervision. It can be found at;
    http://www.quintum.com/support/1G/kb/telco/Disconnect_Supervision.pdf

    #27511
    Telcom Products
    Guest

    Enabling disconnect supervision may rememdy this problem
    mattg@telcomproducts.com

    #27512
    Saleh
    Guest

    thanks all for your replay

    here is my current setup, i enabled the “disconnect supervision” is my config looks fine to you?

    ==================================

    Unit
    —-
    Unit: 1
    IP Address = 192.168.1.100
    External IP Address = 66.66.66.66
    Name = dammam-office

    Online = 1
    Relay ResetTime = 60
    Relay Reset Number = 10

    TcpKeepAlive = Disabled(0)

    System
    ——
    Country Value = 2
    Country Code =
    Area Code =
    Minimum DN = 7
    Maximum DN = 20
    International Prefix:
    1: 00
    Long Distance Prefix =
    Carrier Selection Prefix:
    Intercom Used = no(0)
    Private DN Used = no(0)
    Interdigit Timeout = 4 sec.

    Ring Frequency = 20 Hz(0)
    PSTN Ring Sensitivity = Normal(0)

    Disc Tone Frequency: 480 Hz (min) : 620 Hz (max)
    Disconnect On/Off Time: 250 mSec(on) : 250 mSec(off)

    Call Indication Tone = None(0)
    Disable GUI = yes(1)

    PSTN Trunk Group
    —————-
    PSTN Trunk Group: 1
    Name = PstnPassThrough1
    Pass Through = no(0)
    PT Trunk ID = 0
    Provide Call Progress Tone = yes(1)
    Busyout = no(0)
    Hunt Algorithm = ascending(0)
    Modem Calls = No(0)
    Direction = both(2)
    DN Used = public
    End Of Dial = yes(1)
    End Of Dial Digit = #
    Add End of Dial Digit = no(0)
    Ivr Type = Prepaid calling card
    Ivr Card Digit Length = Variables
    Disabled Voice Prompt 0
    Ivr Preauthentication = 0
    Ivr Language = None
    MultiSession code: = None
    IVR Access Number = None
    Ivr Allow DID = no(0)
    Ivr Answer Delay = 0
    Retry Counter = 1
    CID = From Interface(0)
    External Routing Request = no(0)
    Auto Switch Enable = no(0)
    Forced IP Routing # = none
    Trunk ID(Account Code) = none
    Trunk ID Delivery = none
    2 Stage Dial = No
    Translate Inbound Caller ID = no(0)
    Relay Caller ID = yes(1)
    IP Extension = no(0)
    Maximum LAM Calls Allowed = 4
    LAM: Index Pattern Replacement NumberType
    Cas Signaling Type = loop start fwd disconnect(6)
    Cas Orientation = user(0)
    Dial Tone Detect = yes(1)
    Dial Delay Timeout = 1000
    Answer Delay Timeout = 0
    Flash-Hook Signaling = no(0)
    Supervision = disconnect(1)
    Caller Id Detection = no(0)
    dtmf-ontime = 100
    dtmf-offtime = 100
    Channel:
    unit# 1 line# 2: 1,2,3,4

    PBX Trunk Group
    —————
    PBX Trunk Group: 1
    Name = PbxPassThrough1
    Pass Through = yes(1)
    PT Trunk ID = 0
    Provide Call Progress Tone = yes(1)
    Multipath = yes(1)
    Hunt Algorithm = ascending(0)
    Modem Calls = No(0)
    Direction = both(2)
    DN Used = public
    End Of Dial = yes(1)
    End Of Dial Digit = #
    Add End of Dial Digit = no(0)
    Ivr Type = None
    Partial TG = no(0)
    CID = Trunk ID(1)
    External Routing Request = no(0)
    Auto Switch Enable = no(0)
    Forced IP Routing # = none
    Trunk ID(Account Code) = none
    Trunk ID Delivery = none
    2 Stage Dial = No
    Translate Inbound Caller ID = no(0)
    Relay Caller ID = yes(1)
    IP Extension = no(0)
    Public Number of Digits = 7
    Private Number of Digits = 4
    Public Hunt Ldn’s:
    Private Hunt Ldn’s:
    BDN: Index Bdn
    Cas Signaling Type = loop start(1)
    Cas Orientation = net(2)
    Flash-Hook Signaling = yes(1)
    Flash-Hook Min = 200
    Flash-Hook Max = 700
    Supervision = none(0)
    Caller Id Generation = no(0)
    dtmf-ontime = 100
    dtmf-offtime = 100
    Channel:
    unit# 1 line# 1: 1,2,3,4

    IP Trunk Group
    —————
    Incoming IP call delete digits = 0
    Incoming IP call prefix =
    Outgoing IP call delete digits = 0
    Outgoing IP call prefix = 00010
    Prefix Trunk ID = no(0)
    Default Trunk = No
    External Routing Request = no(0)
    Display Information ID = Tenor-Gateway

    Line
    —-
    Line: 1
    Law = uLaw(0)
    Rx Gain = -8dB
    Tx Gain = -6dB
    Line: 2
    Law = uLaw(0)
    Rx Gain = 0dB
    Tx Gain = 0dB
    Guard Time = 0mS
    DAA Start Up = Enabled

    Gatekeeper System
    —————–
    Zone Name =
    Border Element IP Address(prim) = 192.168.1.100
    Border Element IP Address(sec) = 0.0.0.0
    Discovery IP Address = 0.0.0.0
    Gatekeeper Password =
    LRQ returns all candidates(0)
    Maximum LRQ Hops = 0
    WAN Call Limit = 0 (disabled)
    LCF/LRJ V3plus = 1
    Gatekeeper Option Flags:
    Use IP Header Address = no(0)
    Ridgeway ARQ = no(0)

    Border Element
    —————–
    Static Routing
    Static Route #1
    RouteName =
    Gkmode = Destination is a Gateway (0)
    CallSignalAddress = 80.80.80.80:1720
    1:* Public LDN priority(1)
    2:* Private LDN priority(1)

    DSP

    Voice Coding algorithm = 68
    Voice Information Field size = 160 bits
    Silence Suppression = Enable(1)
    Minimum Jitter buffer = 60 msec
    Maximum Jitter buffer = 300 msec
    Receive Gain (PCM -> IP) = 6 dB
    Transmit Gain (IP -> PCM) = 8 dB
    Digit Relay = 0
    Fax Relay Type = 0
    Fax Maximum Rate = 144
    Fax Playout FIFO nominal delay = 600
    Fax Modem Coding = 0
    Fax Modem Voice Information Field size = 0 bits
    Idle Time = 30
    Answer Supervision Options = 0
    Disconnect Supervision Option = 1

    H323 Gateway
    —————–
    Primary GK Address = 0.0.0.0
    Primary Auto Discovery = 1
    Secondary GK Address = 0.0.0.0
    Secondary Auto Discovery = 0
    H.323 ID = XXX
    Register DN = Register as GW Prefixes(1)
    Ignore Bandwidth in ACF = no(0)
    Default H245 Tunneling = yes(1)
    Round Trip Delay = 0
    One Stage Dialing = 0
    RRQ Interval = 0
    H323 Interop Flags:
    H225 GW Protocol = h323(0)
    Do BRQ = 0
    SlaveSessionId0 = 0
    AllowFastStartOnly = No(0)
    RRQv3plus= No(0)
    ProgressInd Alert= No(0)
    StartH245Flag= No(0)
    Automatic Ext IP Update= Yes(1)
    RTP Verification= No(0)

    Do Lightweight RRQ = no(0)

    ==================================

    thanks

    #27513
    Mark
    Guest

    Saleh,

    did you get a chance to read the document I gave you in my previous email? If this analog unit is not installed in the US or Canada, then it is not configured correctly. Loop Start Fwd Disconnect is typically only supported in the US and Canada. Please try setting the cassignaling to loop start only (1) and then set the supervision to 1 (disconnect tone). This should help.

Viewing 5 posts - 1 through 5 (of 5 total)
  • The forum ‘Voice over IP’ is closed to new topics and replies.