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2nd dail (quintum)

Viewing 15 posts - 106 through 120 (of 178 total)
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  • #22603
    Alan
    Guest

    I have a Tenor A800, problem is i cant get a dial tone. how can i configure it. Thanks

    #22604
    MikeM
    Guest

    Alan,

    most likely you did not put the tenor online. You need to go to config unit 1# and set online to 1, then submit.

    #22605
    Alan
    Guest

    The Unit is set to Online.

    IP Address = 192.168.1.5
    External IP Address = 0.0.0.0
    Name = tenor

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

    TcpKeepAlive = Disabled(0)

    but still no dial tone

    #22606
    MikeM
    Guest

    When you say no dial tone, do you mean that you do not hear a dial tone when you plug a phone in the PBX port and pick the phone up, or is it that you are not detecting dial tone on the PSTN when you try to send calls out to the PSTN, or are you not getting the 2nd dial tone when you call in from PSTN to Tenor?

    #22607
    Wilson Boyrie
    Guest

    Go to the PBXTG1 .

    It may not be set at all, or you may have a PBXTG1 ,but no channels on it.

    Incluide the channels (ports) that you want dial tone on the PBXTG1.

    Like this:

    Quintum:superior>
    Quintum:superior> config
    config# pbxtg 1
    config pbxtg 1# print
    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 Bypass = 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)
    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
    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: This means that there are no channels asociated to this PBXTG

    config pbxtg 1# enablechan 1 1 This enables channel 1 of the PBX side
    config pbxtg 1# enablechan 1 2 2
    config pbxtg 1# enablechan 1 3 3
    config pbxtg 1# enablechan 1 4 4
    config pbxtg 1# exit
    config pbxtg# exit
    config# submit This will save the changes to permanent memory

    config# pbxtg 1
    config pbxtg 1# print
    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 Bypass = 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)
    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
    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 This means that ports 1,2,3 and 4 are part of this PBXTG.
    config pbxtg 1#

    The only other tie when there is no dial tone is when you have PPOE enabled and the gateway could not log in
    to the ADSL modem or PPOE provider.

    But in that case the gateway keeps rebooting and trying to connect to the internet.

    If you still have troubles, post the entire configuration, minus the I.P. addreeses, and i will take a look.

    Wilson Boyrie.

    #22608
    Alan
    Guest

    Thanks alot Wilson Boyrie, now i have a dial tone on the PBX, i tried the same on PSTN but no dial tone. i how do i go round this?

    Regards

    #22609
    Dave
    Guest

    Hi,
    I have tenors(A400/A800) which is working fine and all of sudden the acd seems to be below 2 mins and also the call duration seems to be below 10 secs.But if i shutdown the box and restart after 20 mins then the ACD starts creeping up for 5mins and after 4 to 5 hrs its all start again creeping down calls, ACD seems very low .Please anybody knows what the problem .If yes then Please help me.

    #22610
    iqbal
    Guest

    hi freinds,
    i m using A800, how can i use this tenor as an intercom if my extention number would be 101 to 108 at tenor every port. pls send me detail configuration.
    at iqbaltith@gmail.com or this site

    thanks
    iqbal

    #22611
    MikeM
    Guest

    iqbal,

    You can find information on this feature on Quintum’s web site under the customer service area. The even have a sample config there. It is directly at;
    http://www.quintum.com/support/1G/kb/telco/Phone-to-phone_dialing.pdf

    If you require more in-depth assistance, you may contact me directly at mike_voip@hotmail.com

    Mike

    #22612
    kan
    Guest

    I have a AS400 Tenor VOIP box. since few days, the Power LED is flickering and i am unable to telnet to VOIP box. Any solutions?

    #22613
    Jun (Quintum)DX2060
    Guest

    I am in the midst of connecting a quintum to a C5CM by Verso Clarent thru H323. Can anyone help me?

    #22614
    Ripper
    Guest

    Hi guys… Can anyone help me? I have a Quintum Tenor DX Series and would like to configure it so that it automatically disconnects a call after a certain no. of minutes… Pls. help///

    #22615
    MikeM
    Guest

    Ripper,

    there is a setting in the IPRG called maxtalktime. You can set this to a value that will equal the number of minutes any call will last.

    #22616
    Alan
    Guest

    Hi Mike, hope u can get back to me on this. you had helped me a while back on my quintum A800 which did not have a dial tone. the same problem has come up again. could you please take a look at my config

    Welcome to Tenor Multipath Switch RS-232 Server (A002-008C33)———
    authenticationport s 1812

    Quintum:tenor> Password: Thank you. Type ? for help the configngvendor 0
    p
    config# printretf8990)

    UnitR
    ——
    Unit: 1 Address = 0.0.0.0 (A002-008C33
    IP Address = xxx.xxx.xxx.xxx
    Secondary File Server: I
    External IP Address = 0.0.0.0 Thank you. Type ? for help
    Name = tenor 5

    Online = 1

    Enable
    Relay ResetTime = 240onfig
    Relay Reset Number = 2on TableQuintum:tenor>

    TcpKeepAlive = Disabled(0)ID Translation Table

    Systemome to
    ——Index
    Country Value = 0lacementA002-008C
    Country Code = 1

    Radius E
    Area Code =

    Minimum DN = 7m:tenor> Passw
    Maximum DN = 7.0gain
    International Prefix:p 1812

    Welcome
    1: 011ntingport p 181
    Long Distance Prefix = 1ost s 0.0.0.0
    Carrier Selection

    Mode
    IP Address : Port # of Syslog Server 2 = 0.0.0.0 : 514
    DN Used = public
    En
    IP Address : Port # of Syslog Server 3 = 0.0.0.0 : 514t = #
    Add End of Dial Digit = no(
    Syslog Facility = 16
    Ivr Type

    IP Address : Port # of Cdr Server 1 = 0.0.0.0 : 0
    CID = Trunk ID(1)
    Extern
    Relay Caller ID = yes(1)
    Secondary Time Server: IP Address = 0.0.0.0
    Public Number of Digits = 8
    UTC Offset: Unknownof Digits = 4

    Disc Tone Frequency: 480 Hz (min) : 620 Hz (max)ivate Hunt Ldn’s:
    BDN: Index
    Disconnect On/Off Time: 250 mSec(on) : 250 mSec(off)tart(1)

    Call Indication Tone = None(0)
    Disable GUI = no(0)Signaling = yes(1)

    Dialplan
    ——–
    User Programmable DP = Nook Min = 200
    Dialplan table:

    index:Pattern DpType min

    dtmf-of
    —————-
    PSTN Trunk Group: 11: 1,2,3,4

    Name = PstnPassThrough11 1
    Pass Through = yes(1)
    config p
    PT Trunk ID = 0
    Comman
    Provide Call Progress Tone = no(0)
    Busyout = no(0)me: name {set n
    Hunt Algorithm = ascending(0)
    Modem Calls = No(0)um {passthru trunk:
    Direction = both(2)
    DN Used = public

    End Of Dial = yes(1)thru trunk id}
    End Of Dial Digit = #

    Add End of Dial Digit = no(0)progress tone: 1=yes, 0=no}
    Ivr Type = None
    CID = From Interface(0)
    multipat
    External Routing Request = no(0)

    Auto Switch En
    Maximum LAM Calls Allowed = 4ing, 1=outgoing, 2=both}
    LAM: Index Pattern Replacement NumberType
    dnused: num {incoming ca
    Cas Signaling Type = loop start(1)
    Cas Orientation = user(0) dial digit used: 1=yes, 0=no}
    Dial Tone Detect = yes(1)
    end
    Dial Delay Timeout = 1000 0-9, * or #}
    Answer Delay Timeout = 0
    addenddialdigit: n
    Flash-Hook Signaling = no(0)ing calls: 1=yes,
    Supervision = none(0)
    Caller Id De
    PBX Trunk Group
    —————snumber: str {a
    PBX Trunk Group: 1ormat}
    Name = PbxPassThrough1

    Pass Through = yes(1): 0=no, 1=2nd dialton
    PT Trunk ID = 0lingcard
    Provide Call Progress Tone = yes(1)

    Multipath = yes(1)ount0, 4=ANI type1
    Hunt Algorithm = ascending(0)unt1
    Modem Calls = No(0)
    Direction = both(2)=2nd dial prompt ty
    DN Used = publicrompt type 2, 9=
    End Of Dial = yes(1)
    End Of Dial Digit = #
    Add End of Dial Digit = no(0)
    pincode: str {pin co
    Ivr Type = None9 or x}
    Partial TG = no(0)
    CID = Trunk ID(1)tialtg: num {part
    External Routing Request
    dn+
    Trunk ID Delivery = none
    externalr
    2 Stage Dial = Nonal routing reque
    Translate Inbound Caller ID = no(0)
    Relay Caller ID = yes(1)accessnumber: str {acces
    IP Extension = no(0)}
    Public Number of Digits = 8
    ivranswe
    Private Number of Digits = 4 timeout in msec}
    Public Hunt Ldn’s:
    Private Hunt Ldn’s: ivraccountlength
    BDN: Index Bdnlength: 1-20=l
    Cas Signaling Type = loop start(1)
    ivrpinlengt
    Cas Orientation = net(2)ngth}
    Flash-Hook Signaling = yes(1) ivrcardlength: num {card length: 1-2
    Flash-Hook Min = 200
    iv
    Flash-Hook Max = 700

    unit# 1 line# 1: 1,2,3,4 4=Spanish

    IP Trunk Group

    —————German,
    Incoming IP call delete digits = 0 6=Arabic,
    Incoming IP call prefix = 7=R
    Outgoing IP call delete digits = 0
    ivrpreauth: num {
    Outgoing IP call prefix = method: 0=no , 1=yes}
    Prefix Trunk ID = no(0)
    Default Trunk = No
    ivral
    External Routing Request = no(0) does not match ivraccessnumber:
    Display Information ID = Tenor-Gateway

    Line=no,
    —-s}
    Line: 1[str] < "**" or "##" or no stri Law = uLaw(0) Rx Gain = -4dB Tx Gain = -2dB DAA Start Up = Enabledtime, 3= Gatekeeper Administration ------------------------- Barred Endpointsn [remote-line [ IP Maskels), r No Barred Endpoints Configured emote-lin Gatekeeper System -----------------numdigit: num {nu Zone Name = NRb pbx for pub nu Border Element IP Address(prim) = 217.199.146.179 prvnumdigit: num {num Border Element IP Address(sec) = 0.0.0.0 Discovery IP Address = 0.0.0.0dn: num {delete bdn at the spe Gatekeeper Password = LRQ returns all candidates(0) setbdn: pattern {add bdn: Maximum LRQ Hops = 0?} WAN Call Limit = 0 (disabled) huntpubldn: LCF/LRJ V3plus = 1e Border Element -----------------untprvldn: index Static Routing No Static Routes configurednk group 1 DSP --- Voice Coding algorithm = 81 current settings Voice Information Field size = 192 bits: Move one level back in config mode Silence Suppression = Enable(1) exit!: Ge Minimum Jitter buffer = 60 msec Maximum Jitter buffer = 300 msec config pbxtg 1# disable Receive Gain (PCM -> IP) = -2 dB
    config pbxtg 1#disablecha
    Transmit Gain (IP -> PCM) = -4 dB
    config pbxtg 1#disablechan 1 3
    Digit Relay = 0
    c
    Fax Relay Type = 0lechan 1 4
    Fax Maximum Rate = 144config pbxtg 1#disable
    Fax Playout FIFO nominal delay = 600
    Command Error
    config
    Fax Modem Coding =

    AutoSwitch
    config
    ———-ablechan 1
    Auto Threshold = 50
    config

    H323 Gatewaychan 1 3
    —————–
    config pb
    Primary GK Address = 0.0.0.0
    config
    Primary Auto Discovery = 1
    C
    Secondary GK Address = 0.0.0.0g pbxtg 1#enablechan 1 6
    Secondary Auto Discovery = 0nd Error
    config
    H.323 ID =ablechan 1
    Register DN = Register as GW Prefixes(1)rror
    config pbxtg 1#enablec
    Ignore Bandwidth in ACF = no(0)
    Command Error
    Default H245 Tunneling = yes(1)
    config pbxtg# exit
    Round Trip Delay = 0 submit
    One Stage Dialing = 0
    Command Error
    RTP Verification= No(0)tipath = yes(1)

    Do Lightweight RRQ = no(0)g(0)

    Radius User Calls = No
    ———–

    host p 0.0.0.0(2)
    authenticationport p 1812
    End Of Di
    accountingport p 1813
    End Of Dial

    host s 0.0.0.0
    authenticationport s 1812no(0)
    accountingport s 1813one
    Pa

    retry = 3no(0)
    timeout = 5
    CID = Tr
    accounting
    timeout: 5
    2 Stage D

    Enabled Languages: Noneslate Inbound Caller ID = no

    CID Translation Table
    Relay
    ——ID = y
    Caller ID Translation Table
    IP Extension = no(0)
    Index Pattern ReplacementDigits = 8

    Radius Endpointber of Digits =
    —————

    authenticationport s 1812
    accountingport s 1813

    retry = 3
    timeout = 5
    idtype = 0
    passwordtype = 0
    sharedsecret

    Product Name: Tenor Analog A400 Multipath Switch – 4 ports (Rev. B)
    Gatekeeper Status: Mini
    GK Calls Allowed: 8
    Feature Bit Status: -PS/+RB/-ER
    Languages allowed: 1
    Serial Number: A002-008C33
    Ethernet Address: 00-30-E1-00-8C-33
    IP Address: xxx.xxx.xxx.xxx
    Subnet Mask: 255.255.255.0
    Default Gateway: xxx.xxx.xxx.xxx
    System Software Version: P4-2-20-40(LEC) (1733826/0xD5B6)
    Boot Software Version: P4-1-3 (180592/0xE814)
    Database Version: 2.08 09-13-2000 (277900)

    config#

    #22617
    MikeM
    Guest

    Alan,

    Which side are you not getting dialtone from, PSTN or PBX? Is it when you dial in or out? Please explain a little. Also, you may contact me at mike_voip@hotmail.com as it is better to send the config as an attachment there then take up all the space on the forum for it.

    Mike

Viewing 15 posts - 106 through 120 (of 178 total)
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