GrandStream HT 488

Discussion in '3CX Phone System - General' started by tobiastromm, Aug 1, 2011.

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  1. tobiastromm

    tobiastromm New Member

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    Hi Guys,

    i need a litle help with tones on HT 488.

    I am using it with 3CX for connect my PSTN line. It is working fine for make and receive calls, but sometimes the line still connected after the call is ended.

    I think the problem is the tone.



    I have found this values on World Tone Database but I am insecure to change the values of correct way.







    ...about the value above, how I find it?



    and



    Can someone help me?

    Thank You
     

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  2. tobiastromm

    tobiastromm New Member

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    If Help, GrandStream Support say:

    1 Dear Customer,

    This values you can also get them from your internet telephony service provider ITSP. Remember you are using a VoIP Phone not an analog phone. Have you changed the Call progress tone? Do the default values not work well? Have you tried a factory reset to try the default values. In any case, you can configure them on advanced settings and this is how the syntax of the CPT is:

    Syntax: f1=val[,f2=val[,c=on1/off1[-on2/off2[-on3/off3]]]];
    (Frequencies are in Hz and cadence on and off are in ms)

    Ex: f1=350@-13,f2=440@-13,c=0/0;


    f is equal to frequency and is in hertz so you don't have to put hertz. You can have multiple f.
    @ is to determine the volume by default the majority of the time it is set to -11 so I will recommend to use it that way unless otherwise expecified
    c= cadence and is in ms it should be on time/off time. Here is an example with multiple cadences f1=350@-11,f2=440@-11,c=100/100-100/100-100/100;

    Best Regards,
    Comment by : erodriguez on 2011-07-26 08:45:39
     
  3. leejor

    leejor Well-Known Member

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    It looks like you've come across a few ways of writing the disconnect tone. I would try , f1-425. Choice 1 On= .25, Off= .25

    If current disconnect is used by your PSTN, then that is the most reliable. To test for it, you have to be able to see if the phone line goes open for a second, when the far end hangs up. You can put a current meter in series with the phone (analogue is best). Since you are having issues now, i suspect that it is not implemented by your phone company. You will probably have to rely on the disconnect tone.

    As for the impedance, you may be OK leaving it at 600 ohms. Changing it may fix any echo/level issues that you experience. It won't affect the disconnect.
     
  4. tobiastromm

    tobiastromm New Member

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    Leejor,

    first, thank you for answer.

    I change the values.

    On cadence tone when I put .25 click Update and Reboot the HT 488 change the value to 200, is this correct? F2 frequency i can stay with default value?
     

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  5. leejor

    leejor Well-Known Member

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    I would put in 250 (ms), and listen to the disconnect tone. If it is single tone, then leave the second frequency blank. If the tone sounds like there are , in fact, two tones combined, then you have to determine what the second frequency is. The sites you came across don't mention a second frequency. Running some tests is the only way to know if it is going to work for you.
     
  6. tobiastromm

    tobiastromm New Member

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    Leejor,

    there is a delay before the VoIP phone rings (two rings to start). It is possible to fix it?

    The HT 488 have caller ID for incoming calls?
     
  7. leejor

    leejor Well-Known Member

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    It's been a while since I've used a Grandstream ATA or gateway. I had a couple die on me. I may be wrong but I don't believe that the 488 does support caller I'D. If there are no options for caller I'D type, then it won't be supported.

    As for the delay. There should be an option called something like PSTN answer delay. Because there is no need to wait for caller ID info, it can be shortened to 1 second.
     
  8. tobiastromm

    tobiastromm New Member

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    Leejor,

    i have make some search and it haven't Caller Id for PSTN.

    The PSTN answer delay otion is set to "0".

    It's possible to use HT 503 as a Gateway for 3CX? (It's not listed as a supported gateway, but works?).
     
  9. leejor

    leejor Well-Known Member

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    If that is the case, and you are experiencing a delay, then check the 3CX logs. 3CX may be getting the call from the Grandstream but there may be a reason that it is holding up the call.

    Just because it isn't supported, doesn't mean it doesn't work, it just means it hasn't been tested, and you are "on your own". As the 503 is a replacement for older Grandstream devices it should work, and, it should be easy to set the options using one of the older Grandsteam setups. Most options should use the same terminology. I seem to recall that it does support caller ID.
     
  10. tobiastromm

    tobiastromm New Member

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    Leejor, look the print screen.

    The call is terminated, but two gateways are still connected :(

    PS: I have already make the changes on ATA.

    Any idea?
     

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  11. leejor

    leejor Well-Known Member

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    If you have one gateway connected to another and the call stays up after the conversation has ended, then you need to modify the "disconnect detection" methods used by the device(s).

    There are generally 3 types used to determine that a PSTN call has dropped...silence detection, with a time set to drop he line if it is silent for a given period.

    CPC, sometimes called something else, it's Called Party Control, it means that when the called party hangs up, the central office (if this is used in your area) will send a short disconnect on the phone line. Some ATA's can detect this and then drop the line. Some ATA's can also detect a current reversal sent buy the phone company when the called party answers, this changes back when they hang up. Once again your phone company has to support this as does your ATA

    The third method, and probably the most common, is the detection of a tone Or combination/pattern send by the phone company if you keep the receiver off-hook after the called (or calling) party has hung up. This, of course, varies, from country to country, phone company to phone company, as you've discovered. This method will usually work if set up correctly.

    I'm lucky as my PSTN provides CPC, so I haven't had to do much experimenting with other disconnect detection methods other than disabling silence detection completely, as it drops low level calls.

    If your Grandstream(s) are actually , still, seizing the line after the disconnect tone has been sent, then you may not have it configured properly to recognise the tone sent. Confirm that, indeed, a tone is being sent.
     
  12. tobiastromm

    tobiastromm New Member

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    Leejor,

    sorry about the delay.

    After some call I can see this on 3CX Logs:

    Is something Wrong? (about ring delay)

    Code:
    14:53:10.578  [CM503008]: Call(3): Call is terminated
    14:52:55.609  Currently active calls - 1: [3]
    14:52:28.703  [CM503003]: Call(3): Call to sip:101@10.0.0.1 has failed; Cause: 487 Request Terminated; from IP:10.0.0.16:43224
    14:52:28.687  [CM503003]: Call(3): Call to sip:108@10.0.0.1 has failed; Cause: 487 Request Cancelled; from IP:10.0.0.86:5060
    14:52:28.578  [CM503007]: Call(3): Device joined: sip:103@10.0.0.85:5060;user=phone
    14:52:28.578  [CM503007]: Call(3): Device joined: sip:10002@10.0.0.88:5062;user=phone
    14:52:28.578  [CM505001]: Ext.103: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream HT487 1.1.0.45 DevId 000b821512e6] PBX contact: [sip:103@10.0.0.1:5060]
    14:52:28.578  [CM503002]: Call(3): Alerting sip:103@10.0.0.85:5060;user=phone
    14:52:24.187  [CM505001]: Ext.101: Device info: Device Identified: [Man: Counterpath;Mod: Bria;Rev: General] Capabilities:[reinvite, no-replaces, unable-no-sdp, recvonly] UserAgent: [Bria Professional release 2.4 stamp 49381] PBX contact: [sip:101@10.0.0.1:5060]
    14:52:24.187  [CM503002]: Call(3): Alerting sip:101@10.0.0.16:43224;rinstance=bd204141d4d3654f
    14:52:24.000  [CM503025]: Call(3): Calling RingAll800:101Ext.101102Ext.102103Ext.103104Ext.104108Ext.108109Ext.109@[Dev:sip:108@10.0.0.86:5060;user=phone]
    14:52:24.000  [CM503025]: Call(3): Calling RingAll800:101Ext.101102Ext.102103Ext.103104Ext.104108Ext.108109Ext.109@[Dev:sip:103@10.0.0.85:5060;user=phone]
    14:52:24.000  [CM503025]: Call(3): Calling RingAll800:101Ext.101102Ext.102103Ext.103104Ext.104108Ext.108109Ext.109@[Dev:sip:101@10.0.0.16:43224;rinst
    ance=bd204141d4d3654f]
    14:52:23.953  [CM503004]: Call(3): Route 1: RingAll800:101Ext.101102Ext.102103Ext.103104Ext.104108Ext.108109Ext.109@[Dev:sip:101@10.0.0.16:43224;rinstance=bd204141d4d3654f,Dev:sip:103@10.0.0.85:5060;user=phone,Dev:sip:108@10.0.0.86:5060;user=phone]
    14:52:23.953  [CM503010]: Making route(s) to <sip:800@10.0.0.1:5060>
    14:52:23.953  [CM505002]: Gateway:[HT 488 J] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream HT488 1.0.3.96 FXO] PBX contact: [sip:10002@10.0.0.1:5060]
    14:52:23.953  [CM503001]: Call(3): Incoming call from 10002@(Ln.10002@HT 488 J) to <sip:800@10.0.0.1:5060>
    14:52:23.953  [CM503012]: Inbound out-of-office hours rule (unnamed) for 10002 forwards to DN:800
     
  13. leejor

    leejor Well-Known Member

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    What actually happened to the call on this post? It shows trunk 10002 and extension 103 being joined.

    I'd have to look at my own logs to confirm but I seem to recall that unanswered extensions in a ring group do create a fail message when the call has been answered by one of the others.
     
  14. tobiastromm

    tobiastromm New Member

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    Leejor

    The call is received by HT 488 arrived to a Ring Group (800 GTodos) and extension 103 answers the call.

    Thank You.
     
  15. leejor

    leejor Well-Known Member

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    OK...I don't understand what he issue is. Is the call failing, or are you just concerned about the logs?
     
  16. tobiastromm

    tobiastromm New Member

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    Leejor,

    it's about what I say before.

    "there is a delay before the VoIP phone rings (two rings to start). It is possible to fix it?" and you ask me for the system log.
     
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