inbound DTMF tones program

Discussion in 'General' started by jim.christou, Mar 22, 2009.

  1. jim.christou

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    Hello everyone

    I am not a linux user but i run asterisk before and now elastix

    I know the very basic staff about libux programing and the same for asterisk etc.
    I want to ask if there is anybody that know how can I do the following

    I want ta call through my mobile phone to a gsm gateway I have connected to elastix asterisk auto accept the call (answer it) and wait for a dtmf tone. Then I press numbers on my mobile phone (asterisk recognize them) and auto call through another trunk to the destination number that I pressed.

    the general idea is that i can use my mobile phone as a virtual sip phone (from almost anywhere) that is on my network and make calls through anyone of the trunks I have.

    If someone know something or want to help me build this please let me know.
    Feel free to ask any further questions you want

    Thank you very much
     
  2. danardf

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    Hi.

    I think that's the DISA function.

    You must have an inbound route with your GSM number.
    When your GSM number is detected, you can routed this call to a DISA function.
    Into the DISA, your must compose a PIN code (1234 for exemple), and to compose your phone extension number (100 for exemple).
    Like that, now, your gsm is the number 100.
    You can compose all the extensions, services, ...Etc. Like an extension into your office.
    If you hangup the line, your gsm don't keep the number 100.
     
  3. jim.christou

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    Hello there

    Thank you for your fast answer
    I want to ask further I've elastix 1.5 if I will do what you told me the gsm gateway will be used only for that So if I call to it from my mobile will become an extension (virtually), but if someone else will call to gsm gateway will I accept the call like a simple inbound call?

    and at the "Context(Experts Only)", "Allow Hangup" what i have to do? at Context what other options do I have??

    Also Sorry for my english

    Thank you very much
     
  4. ramoncio

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    But you must make sure your gsm gateway passes the dtmf tones ok to asterisk.
    I bought a chinese brand at eBay and it does not work, asterisk doesn't detect the dtmf tones properly, and the overall quality is quite poor.
     
  5. danardf

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    if you want, each call incoming by your GSM gateway can use the DISA, and not the anonymous. The PIN code is a secret code. So, if the caller don't know the PIN code, the call will be hangup.

    For information, the 1.3.2 is stable, and for instant, the 1.5 is RC4 only and not a stable version (must be coming soon)
     
  6. ramoncio

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    You can have an IVR with a hidden option sending you to DISA, callback or whatever you like.
    You will have to pay for the mobile call to the gsm gateway in case of using DISA. And you will have to pay for the mobile call using any of your trunks (usually VoIP) in case of callback.
    And of course you can use the GSM gateway to receive inbound calls when not busy.
     
  7. jim.christou

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    look basicly I don't have gsm gateway exactly

    I have an FCT (gsm to analog phone) device wich I have connected to a pstn gateway ( {gsm2analog}->PSTN to voip) device (grandstream HandyTone 488) and connect it to the elastix

    Shall it work like this

    furthermore off-topic I bought 2 the one of them is not responding lan,wan port and i cannot fix that at all do you know what to do? grandstream doesn't support it any more

    about the cost i have a program from my provider that these two numbers are free of charge for each other

    so i call to gsm gateway free then i make my landscape calls from there (voip) also for free and i have to pay nothing am I right???

    Thank you again for your help
     
  8. jim.christou

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    and another thing how can i set the ivr have its options but after timeout if nothing pressed forward it to a ring group???
     
  9. ramoncio

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    Usually GSM2analog devices are not very good quality, and its support is very poor.
    People usually recommend gsm2sip devices, though they are quite more expensive.
    You will have to test your gateway in order to see if it works in your environment.
    You can enable dtmf debug and see if they get to asterisk ok.
     
  10. ramoncio

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    Option t is for timeout.
    Option i is for invalid number pressed.
     
  11. jim.christou

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    So In the boxes I don't write only the numbers for the ivr and the i, t options right?

    what other options there are available??
     
  12. ramoncio

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    Yep, you can put a hidden option for disa, for example dialing 378.
    and the t and i options sending the call to a ring group.

    To enable dtmf debug, you must edit logger.conf and add dtmf at the end, like this:

    console => notice,warning,error,verbose,dtmf

    (an restart asterisk, of course)

    Your gsm gateway will use inbound dtmf mode, so depending on its voice quality you will have better or worse results. If you can, try to configure it to use g711a or g711u, instead of a high compression codec as g729 or gsm. The same for your grandstream gateway.
     

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