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Quintum A800

Viewing 8 posts - 1 through 8 (of 8 total)
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  • #27497
    Hamy
    Guest

    Here is the scenario:
    A800 Analog
    Calls originated from PBX to IP
    If the call fails over IP it goes to PSTN
    I have 2 questions:
    1- Is it possible to only reroute calls that fail with a specific release cause to PSTN? It seems A800 reroute all the calls (even wrong numbers) to PSTN. Can I prevent it?
    2- When the call reroutes to PSTN, A800 considers the call connected while the call is in progress and it connects after 10 or 20 seconds. The CDR logs the call duration from the moment the call is routed to PSTN and this means an extra charge for the customer. Is there a way to stop this and let the tenor only start the calculation of the call duration when the call is actually connected?
    Any help would be really appreciated.
    Regards

    #27498
    Mark
    Guest

    Hamy,

    1. The Quintum Tenor should only re-route to the PSTN when IP calls fail for the following cause codes 3, 31, 34, 38, 41-45, & 47. It will also re-route on code 17 if partial trunk group is enabled.

    2. This is an issue where you have not setup the answer supervision correctly. In the PSTN trunk group , set supervision to 2 (answer) or 3 (answer & disconnect), set the answerdelay to 120 and make sure progtone is set to 0. For more information on this review the answer supervision document on Quintum’s web site at;
    http://www.quintum.com/support/1G/kb/telco/Answer_Supervision.pdf

    #27499
    Hamy
    Guest

    Dear Mark,
    Thank you for your answer. I have set it like this (PSTN 1):
    1- cassig 1
    2- super 2
    3- answer 120
    4- progtone 0
    and DSP:
    1- asop 3
    It seems I still have the problem. Tenor waits 120 seconds and then starts billing. Anything I am doing wrong?
    System Software Version: P4-2-20-38(LEC) (1733185/0x81C5)
    Boot Software Version: P4-1-3 (180592/0xE814)
    Database Version: 2.08 09-13-2000 (277900)

    #27500
    Hamy
    Guest

    Anybody? Please!

    #27501
    Mark
    Guest

    Try setting the asoptions to 0.

    #27502
    Hamy
    Guest

    Thank you for your answer. Have done that. No success 🙁 .
    Here comes the configuration:
    ———————-
    Unit
    —-
    Unit: 1
    IP Address = 192.168.254.2
    External IP Address = ***************
    Name = ctel1

    Online = 1
    Relay ResetTime = 240
    Relay Reset Number = 2

    TcpKeepAlive = Disabled(0)

    System
    ——
    Country Value = 6
    Country Code = 32
    Area Code = 2
    Minimum DN = 7
    Maximum DN = 7
    International Prefix:
    1: 00
    Long Distance Prefix = 0
    Carrier Selection Prefix:
    Intercom Used = no(0)
    Private DN Used = no(0)
    Interdigit Timeout = 4 sec.

    Contact =
    Location =
    IP Address : of Snmp Trap Server 1 = 0.0.0.0
    IP Address : of Snmp Trap Server 2 = 0.0.0.0
    IP Address : of Snmp Trap Server 3 = 0.0.0.0

    IP Address : Port # of Syslog Server 1 = 0.0.0.0 : 514
    IP Address : Port # of Syslog Server 2 = 0.0.0.0 : 514
    IP Address : Port # of Syslog Server 3 = 0.0.0.0 : 514
    Syslog Facility = 16

    IP Address : Port # of Cdr Server 1 = 0.0.0.0 : 0
    IP Address : Port # of Cdr Server 2 = 0.0.0.0 : 0
    Cdr Password:
    Cdr Format: 0

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

    Primary Time Server: IP Address = 0.0.0.0
    Secondary Time Server: IP Address = 0.0.0.0
    UTC Offset: Unknown

    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 = no(0)

    Dialplan
    ——–
    User Programmable DP = No
    Dialplan table:
    index:Pattern DpType min max nprefix

    System LAN
    ———-
    Subnet Mask = 255.255.255.240
    Default Gateway = 192.168.254.1

    PSTN Trunk Group
    —————-
    PSTN Trunk Group: 1
    Name = pstn1
    Pass Through = yes(1)
    PT Trunk ID = 0
    Provide Call Progress Tone = no(0)
    Busyout = no(0)
    Hunt Algorithm = ascending(0)
    Modem Calls = No(0)
    Direction = outgoing(1)
    DN Used = public
    End Of Dial = yes(1)
    End Of Dial Digit = #
    Add End of Dial Digit = no(0)
    Ivr Type = None
    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 = 1
    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 = 120
    Flash-Hook Signaling = no(0)
    Supervision = answer and disconnect(3)
    Caller Id Detection = no(0)
    dtmf-ontime = 100
    dtmf-offtime = 100
    Channel:
    unit# 1 line# 2: 1

    PBX Trunk Group
    —————
    PBX Trunk Group: 1
    Name = PbxPassThrough1
    Pass Through = yes(1)
    PT Trunk ID = 0
    Provide Call Progress Tone = no(0)
    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 = 16
    Private Number of Digits = 4
    Public Hunt Ldn’s:
    Private Hunt Ldn’s:
    BDN: Index Bdn
    1 02***********
    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: 1,2,3,4,5,6,7,8

    IP Trunk Group
    —————
    Incoming IP call delete digits = 0
    Incoming IP call prefix =
    Outgoing IP call delete digits = 0
    Outgoing IP call prefix =
    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 = -4dB
    Tx Gain = -2dB
    Line: 2
    Law = uLaw(0)
    Rx Gain = 0dB
    Tx Gain = 0dB
    Guard Time = 0mS
    DAA Start Up = Enabled

    Bandwidth Management
    ——————–
    Time of Day Maximum Bandwidth:
    Day = 0(Sunday)
    Hour = 00 * * * * * *
    Hour = 06 * * * * * *
    Hour = 12 * * * * * *
    Hour = 18 * * * * * *
    Day = 1(Monday)
    Hour = 00 * * * * * *
    Hour = 06 * * * * * *
    Hour = 12 * * * * * *
    Hour = 18 * * * * * *
    Day = 2(Tuesday)
    Hour = 00 * * * * * *
    Hour = 06 * * * * * *
    Hour = 12 * * * * * *
    Hour = 18 * * * * * *
    Day = 3(Wednesday)
    Hour = 00 * * * * * *
    Hour = 06 * * * * * *
    Hour = 12 * * * * * *
    Hour = 18 * * * * * *
    Day = 4(Thursday)
    Hour = 00 * * * * * *
    Hour = 06 * * * * * *
    Hour = 12 * * * * * *
    Hour = 18 * * * * * *
    Day = 5(Friday)
    Hour = 00 * * * * * *
    Hour = 06 * * * * * *
    Hour = 12 * * * * * *
    Hour = 18 * * * * * *
    Day = 6(Saturday)
    Hour = 00 * * * * * *
    Hour = 06 * * * * * *
    Hour = 12 * * * * * *
    Hour = 18 * * * * * *

    Gatekeeper Administration
    ————————-

    Endpoint Authorization Type = 0 (None)

    Allowed Endpoints
    IP Mask
    1 192.168.254.2 255.255.255.255

    Barred Endpoints
    IP Mask
    No Barred Endpoints Configured

    Gatekeeper System
    —————–
    Zone Name =
    Border Element IP Address(prim) = 192.168.254.2
    Border Element IP Address(sec) = 0.0.0.0
    Discovery IP Address = 192.168.254.2
    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 = **********************:1720
    1:1* Public LDN priority(0)
    2:2* Public LDN priority(0)
    3:3* Public LDN priority(0)
    4:4* Public LDN priority(0)
    5:5* Public LDN priority(0)
    6:6* Public LDN priority(0)
    7:7* Public LDN priority(0)
    8:8* Public LDN priority(0)
    9:9* Public LDN priority(0)

    DSP

    Voice Coding algorithm = 81
    Voice Information Field size = 192 bits
    Silence Suppression = Enable(1)
    Minimum Jitter buffer = 60 msec
    Maximum Jitter buffer = 300 msec
    Receive Gain (PCM -> IP) = -2 dB
    Transmit Gain (IP -> PCM) = -4 dB
    Digit Relay = 0
    Fax Relay Type = 1
    T.38 Fax Low Speed Data Redundancy = 0
    T.38 Fax High Speed Data Redundancy = 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 = 0
    Answer Supervision Options = 0
    Disconnect Supervision Option = 0

    AutoSwitch
    ———-
    Auto Threshold = 50

    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 =
    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= No(0)
    RTP Verification= No(0)

    Do Lightweight RRQ = no(0)

    Radius User
    ———–

    host p 0.0.0.0
    authenticationport p 1812
    accountingport p 1813

    host s 0.0.0.0
    authenticationport s 1812
    accountingport s 1813

    retry = 3
    timeout = 5
    accountingtype = 0
    billingvendor 0
    sharedsecret

    IVR
    ——

    Primary File Server: IP Address = 0.0.0.0
    Secondary File Server: IP Address = 0.0.0.0
    timeout: 5

    Enabled Languages: None

    CID Translation Table
    ——
    Caller ID Translation Table
    Index Pattern Replacement

    Radius Endpoint
    —————

    host p 0.0.0.0
    authenticationport p 1812
    accountingport p 1813

    host s 0.0.0.0
    authenticationport s 1812
    accountingport s 1813

    retry = 3
    timeout = 5
    idtype = 0
    passwordtype = 0
    sharedsecret
    ————————–
    When I dial 02*********** tenor waits 120 seconds and then starts billing. Anything I can do about it?
    Best regards

    #27503
    Hamy
    Guest

    Anyone? I really nedd to get this to work. Thank you for answering.

    #27504
    Mark
    Guest

    Try setting the cassig to 1. I do not think this will change the answer problem, but it is a conflict as you cannot have forward disconnect when supervision is set to both answer and disconnect.

    After this, without going through logs and testing, it would be hard to determine what is happening.

Viewing 8 posts - 1 through 8 (of 8 total)
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