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no dialtone detected

  • This topic has 17 replies, 1 voice, and was last updated 17 years, 1 month ago by MikeM to Hasan Mahmud Riyad.
Viewing 15 posts - 1 through 15 (of 18 total)
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  • #30817

    Hi everyone,

    I need some help with my quintum
    I’m unable to detect dialtone (test o 1 v) RJ11 is connected on pstn port 1.

    i’m in France, so must i set a specific dialtone frequency ? If yes, How ?

    When i try to make a call from a SIP phone, i can see in logs :

    CH : 433608:[2: 1] sent message to cas: Setup
    CAS : 433609:Received Message
    CAS : 433609:mtype = 128
    CAS : 433609:CAS: ISDN Message
    CAS : 433609:[2,1] Process ISDN Message: imsg=0.
    CAS : 433609:[2,1] Set Disc States
    CAS : 433609:LocalDiscState = 0
    CAS : 433609:RemoteDiscState = 0
    CAS : 433609:In decode: message code = 0
    CAS : 433609:[2,1] Decode SETUP
    CAS : 433609:[2,1] casProc: idle-SETUP
    CAS : 433609:[2,1] Send Seizure to Line
    AB : 433609:[2,01]: t 1111
    CAS : 433609:[2,1] Send ABCD = 15
    CAS : 433609:[2,1] Start Timeout, ID = 1
    CAS : 433609:[2,1] New State = 12
    CAS : 434109:Timeout Event = 1
    CAS : 434109:Stopped Timeout T = 1
    CAS : 434109:[2,1] casProc: o1-T1
    CAS : 434109:[2,1] No Responce to Seizure
    CAS : 434109:RemoteDiscState = 0
    CAS : 434109:[2,1] casProc: Disc Resp Proc
    CAS : 434109:[2,1] Send Disc-Idle to Line
    AB : 434109:[2,01]: t 0000
    CAS : 434109:[2,1] Send ABCD = 0
    CAS : 434109:[2,1] Start Delay, ID = 9
    CAS : 434109:[2,1] New State = 20
    CAS : 434189:Delay Event = 9
    CAS : 434189:Stopped Delay D = 9
    CAS : 434189:[2,1] casProc: x5x-D9
    CAS : 434189:Sent Msg to Usid = 44
    CAS : 434189:[2,1] Sent RELCOMP to CH
    CAS : 434189:[2,1] New State = 0
    CH : 434189:[2: 1] received message from cas: RelComp
    CH : 434189:OBCSM[6]: Release from peer=0xd82c04 cause=0x29 redir=.
    CH : 434189:TBCSM[6]: Release complete from peer=0xd8a80c.
    CH : 434189:OBCSM[6]: Trying another route.
    CH : 434189:udp disconnect: 8 11

    thank you for any help ๐Ÿ™‚

    MikeM to Bert


    Not sure if there is a big difference in the ring tone in France than any other place. One thing you can try is to connect the pstn port to the pbx port with a straight cable and run the test. This will at least say that the port is good or not.

    Another thing to make sure is that there is no voice mail on the analog line that gives a stutter dial tone.

    You did not mention the model type, but I am guessing from the log output and the command structure you gave for the test that it is a gen 1 unit (A400/A800) and not a Gen 2 product (AS/AX). There is no settings for different countries in the gen 1 product for the dial tone or type of analog line. There is a feature in gen 2 for line template and that may help.

    Finally, you mentioned SIP. I am guessing that you downloaded the software from Quintum to support SIP on the gen 1 unit. I have not heard too much on this software yet, but it is beta and there may be some issues with the SIP side.


    Sorry for mistakes
    Yes it is a gen 1 Analog Quintum A800

    I call with a SIP phone, throught a nextone sswitch witch “translate” SIP to H.323.

    I tried to link pstn and pbx : test is OK

    i’m not sure if quintum call the good number (I suppose in dn:068196xxxx, dn means Dialed Number)

    Another thing : i set ‘waitdialtone’ to 0 (no), so when i try to call with SIP phone, the nextone route well to my quintum, i can see the green pstn port led flashing , but the called phone doesn’t ring …

    does A800 check for something before really call the “end phone” ?

    Thank you for any help ๐Ÿ™‚


    MikeM to Bert


    If you have dialtonedetect set to Yes, then dial delay will do nothing. When the tenor receives a call from IP, it will route it to the first available pstn line (based on your hunt setting). It will go off-hook on the line and try to detect a standard dialtone. If it cannot detect this, then the call will be disconnected. You can try to disable dialtonedetect and set dialdelay for 500 (500ms) and see if that helps.



    dialtonedetect is set to 0 (no)

    the issue is :

    – if i run a test ( test o v), the dialtone is not detected.

    – if i launch a voip call from my SIP phone, the quintum receive the request, with a valid num to dial. I see the little green light of the pstn port. i can wait 5 minuts, the phone called will never ring (and i hear nothing in the SIP phone).
    The proper line is connected to the first pstn port, with a standard phone wire (when i plug a ‘real’ phone on this wire, i can hear dialtone and i can call who i want).

    So i really don’t understand this issue, since apparently there is nothing else to configure.

    But still no dialtone detected ๐Ÿ™

    i know i’m not a ‘Quintum Master’, but sincerely I really don’t know what i’m missing … ๐Ÿ™

    Thank anyone for any help ๐Ÿ˜‰

    MikeM to Bert


    It does not sound like you are missing anything here. For some reason, the tenor is just not detecting your dialtone. You said that you have dialtone detect set to 0, make sure you have dial delay set to 500 or 1000. Another test is to put a standard phone into the PBX port 1 of Tenor, set the pbxtg and pstntg passthru to 1 for yes and try a call from the phone to the pstn. Also, try putting the tenor in bypass. Finally, make sure that your phone line is set for dtmf and not rotary dial.

    Other than that, the only way I can help is if I were to make some test calls from my unit to your unit and see what I hear as I may be able to tell about the dial tone, and other tones from my experience.

    If you would like me to do this, please contact me at




    I found the cause of the issue : i used a standard phone wire to link pstn #1 and the phone catch. I tried with another wire from quintum one with a “double end”, and it works … ๐Ÿ™‚

    MikeM to Bert

    Glad to hear that you got that straightened out. I did see your IM yesterday as well, but I was quite busy.


    As I’m unable to send new message (it says that msg is ok and I just have to refresh, but in fact my msg never appeared), I post my question here :

    About test cmd (in A800):
    I know I have to make a test o. I want to use the line 5.

    So I do that :
    test o 5 v. -> dialtone detected.
    test d 061210xxx
    All digits sent.
    And the called phone doesn’t ring.

    But as I remember, I must have Received Digit ‘0’
    Received Digit ‘6’
    Received Digit ‘1’
    All digit sent

    Someone knows why I’m don’t have the Received digit ‘x’ plz ?

    Another thing, what about if the pstn line is RNIS ??? does the quintum still detect tone ?

    Mikem to Bert

    yes, there seems to be a problem with MSN last night. I think it is cleared up now.

    As for your question, typically you would see digit received for each one. I cannot say why you are not seeing this for all the digits. Additionally, when you use the test o command, you do not need to put a v at the end.

    For RNIS, I am sorry to say that I do not know what that means. If you could give me the description, I could tell you whether it is supported.



    Sorry RNIS is digital lines in France (ISDN).

    About msg I can’t send, it is about this forum, not Msn ๐Ÿ™‚

    Well as I can see, in France, ISDN boxes provide analog lines via their ISDN network.

    And I’m pretty sure my pb comes from that. Does Quintum able to use this kind of “analogic lines” ??

    MikeM to Bert


    so we are talking about BRI lines in France, no the analog tenor does not support BRI lines, only analog, so you must provide a isdn terminal adapter to change the isdn to analog or you can purchase a Tenor BX unit which is a BRI supported Tenor.


    Well, I plugged the Quintum on a *real* analog line (as I don’t know about analog lines provided by an ISDN equipment).

    I test and get dialtone. I have the correct num log (Received digit …)

    But called phone still not ring.

    So my question is can we use Quintum A800 on the Z* interfaces of a ISDN PSTN access ?

    Is some specific settings (e.g. tone frequency) ?

    Thanx for any help

    MikeM to Bert


    Again, you need to get an adapter to change the ISDN line to standard analog. If you have one of these adapters, they should provide you with 2 analog ports on the back side since each BRI line has 2 B channels for voice. If you have this, you should be able to connect the tenor to the analog port of the adapter and treat it like a standard analog line.

    Now I do not know if there is any configuration that you will need to do on the adapter.



    Hmmm… Still unable to post new thread in this forum ๐Ÿ™

    So I’ll use this one.

    I would like to know if there is a way to block caller ID on a quintum DX2060 ? If yes, how ?

    thanx for any help!


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