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tenor A800

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Viewing 15 posts - 166 through 180 (of 207 total)
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  • #22903 Reply
    Saroj
    Guest

    Hello all,

    Can somebody help me with my tenor A800 ??? As soon as the calls hit the gateway, the Fast Busy signal comes and gets disconnect. The disconnect cause is showing the normal call clearing (16).

    Please help me out.

    Regards,
    Saroj

    #22904 Reply
    MikeM to Saroj
    Guest

    Saroj,

    there could be many reasons for this and without any further troubleshooting it would be difficult to say what is the cause. If, as soon as call hits the gateway you get fast busy, but then the cause code is 16, I find that somewhat strange as typically in this case you may get a cause 41, or 34 or 47.

    This would require additional trouble shooting, take logs, look at config and such.

    If you would like assistance on this, please contact me at mike_voip@hotmail.com and we can discuss my services.

    Mike M
    mike_voip@hotmail.com

    #22905 Reply
    Mizanur Rahman
    Guest

    Dear sir,
    Can you send me tenor A800 tarmination configaraton to netmeeting from i can using from my PC.

    Thanks

    Mizaur

    #22906 Reply
    MikeM to Mizanur Rahman
    Guest

    Mizanur,

    There are many things to consider for doing term.

    If you would like assistance in configuration, you may contact me directly at mike_voip@hotmail.com and we can discuss my services.

    MikeM
    mike_voip@hotmail.com

    #22907 Reply
    Saroj
    Guest

    Thanks MikeM,

    I got the problem fixed by tech guys from HK.

    #22908 Reply
    shayan
    Guest

    Can anyone tell me how can i eliminate de Border Element.

    #22909 Reply
    shayan
    Guest

    i have AXT800 and i m facing low ASR
    i want to make Gatekeeper and Gateway both
    i want to make my Computer Gatekeeper using GNUGK
    And my quintum as a GW
    Please anyone is here who can help me to configure this setup
    even i can pay u for that

    #22910 Reply
    Riaz to MikeM
    Guest

    Hello Mike,

    Thanks The Tenor A800 is now working and the reason for not being able to configure it was the Serial port mismatch due to the upgraded versions of available computers. I used a USB to serial convertor and it worked.

    Thanks for the assistance and hope we shall chat again on MSN soon.

    With Kindest Regards
    Riaz Malik

    #22911 Reply
    Saroj
    Guest

    Hello everyone,

    I am once again back with a simple problem for you guys. Can somebody tell me how to make phone calls from the tenor A800 gateway which is available in the LAN. I am trying to make the call from the PC from where I can connect to the tenor.

    Thanks in advance.

    Regards,
    Saroj

    #22912 Reply
    MikeM to Saroj
    Guest

    Saroj,

    If you are simply looking to connect a phone to the A800 and make a call out over IP, you need only configure a static route with the destination IP address and add a * as the DN so this way any calls you dial will go to that IP address. You may want to make sure that you have the country set to 1, countrycode removed, areacode removed, ldprefix removed and intlpref 1 removed. These parameters are all found under the config sys prompt.

    If you need further assistance you can contact me at mike_voip@hotmail.com to discuss my services.

    thanks
    Mike M
    mike_voip@hotmail.com

    #22913 Reply
    Suman
    Guest

    Hello all,

    This is strange happening in my gateway. Previously the calls came as here in Port 4 but now strangely it is coming like the other ports. The sad part is this active(16) are shown as false calls and dropped. Please help.

    Thanks in advance.

    Quintum:roc> ch calls

    cas: t [c8f004 c9e80c] 4 (2, 1)
    cas: t [c10404 c5480c] 4 (2, 2)
    cas: t [c86004 cd740c] 4 (2, 3)
    cas: t [d80804 be400c] 5 (2, 4)
    cas: t [c59004 c5b00c] 4 (2, 6)
    cas: t [c13c04 cdb00c] 4 (2, 7)
    cas: t [d03c04 ca200c] 4 (2, 8)
    h323: orig calls = 7 term calls = 0
    h323: number of calls in stack = 7
    CAS Call Blocks:
    t (2, 1) active(16)
    t (2, 2) active(16)
    t (2, 3) active(16)
    t (2, 4) active-disc rcvd(17)
    t (2, 6) active(16)
    t (2, 7) active(16)
    t (2, 8) active(16)

    #22914 Reply
    Suman
    Guest

    Here is my configuration.

    Unit
    —-
    Unit: 1
    IP Address = xxx.xxx.xxx.xxx
    External IP Address = 0.0.0.0
    Name = nepalroc

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

    TcpKeepAlive = Disabled(0)

    System
    ——
    Country Value = 3
    Country Code =
    Area Code =
    Minimum DN = 7
    Maximum DN = 20
    International Prefix:
    Long Distance Prefix =
    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: key
    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 = 255.255.255.0
    Default Gateway = xxx.xxx.xxx.xxx

    PSTN Trunk Group
    —————-
    PSTN Trunk Group: 1
    Name = PstnPassThrough1
    Pass Through = yes(1)
    PT Trunk ID = 0
    Provide Call Progress Tone = no(0)
    Busyout = no(0)
    Hunt Algorithm = ascending-round-robin(2)
    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 = 8
    LAM: Index Pattern Replacement NumberType
    Cas Signaling Type = loop start rev battery(5)
    Cas Orientation = user(0)
    Dial Tone Detect = yes(1)
    Dial Delay Timeout = 1000
    Answer Delay Timeout = 0
    Flash-Hook Signaling = no(0)
    Supervision = none(0)
    Caller Id Detection = no(0)
    dtmf-ontime = 100
    dtmf-offtime = 100
    Channel:
    unit# 1 line# 2: 1,2,3,4,5,6,7,8

    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: 1,2,3,4,5,6,7,8

    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
    No Allowed Endpoints Configured

    Barred Endpoints
    IP Mask
    No Barred Endpoints Configured

    Gatekeeper System
    —————–
    Zone Name = npRoc
    Border Element IP Address(prim) = xxx.xxx.xxx.xxx
    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
    No Static Routes configured

    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) = 0 dB
    Transmit Gain (IP -> PCM) = 0 dB
    Digit Relay = 1
    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 = 0
    Answer Supervision Options = 0
    Disconnect Supervision Option = 0
    Answer Supervision Delay = 0 (disabled)

    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 = Yes(1)
    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

    #22915 Reply
    Mikem to Suman
    Guest

    You may not have the disconnect supervision and/or answer supervision set correctly on your unit. Verify these settings using the documentation on Quintum’s web site. Go to their web site and search for answer supervision and then search for disconnect supervision and check your settings against the recommendation there.

    Mike M
    mike_voip@hotmail.com

    #22916 Reply
    MikeM to suman
    Guest

    Also,

    Please do not copy your entire config to these forums as they are long enough without everyone having to read through your config.

    Again, as stated, check out the docs on quintum’s web site because in the pstntg area you definitely do not have the correct settings for answer and disconnect supervision. I would try setting signaling to just loop start, supervision to 3 (both answer and disconnect) and answer delay to 120.

    Mike M
    mike_voip@hotmail.com

    #22917 Reply
    Suman
    Guest

    Hello MikeM,

    Changed back to the settings as per your instruction. But the calls are all dropping and the major problem, even if the calls drop the billing is charged. 🙁

    Thanks in advance.

Viewing 15 posts - 166 through 180 (of 207 total)
Reply To: Reply #22746 in tenor A800
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