DISA Problem

Discussion in 'General' started by pti2000, Nov 13, 2010.

  1. pti2000

    Joined:
    Mar 27, 2008
    Messages:
    77
    Likes Received:
    0
    Disa does not work on Elastix 2.0 When you call in the return dial tone is constant and when you dial a number the system give you an error. Does any one have a fixed for this?
     
  2. andyshawn

    Joined:
    Apr 3, 2009
    Messages:
    113
    Likes Received:
    0
    Have you checked the DTMF settings? Are you dialing into the system via the PSTN or via a VoIP handset?
     
  3. pti2000

    Joined:
    Mar 27, 2008
    Messages:
    77
    Likes Received:
    0
    The system has a sip trunk with 8 channel, Now this works well with the older version, but not 2.0
    any way I used my cell phone to call into the system via DISA and call out via the PBX
    but I can't get it to work on this version.
     
  4. andyshawn

    Joined:
    Apr 3, 2009
    Messages:
    113
    Likes Received:
    0
    Ok then. That sounds like there might be a bug in the new elastix code then. Report it in the bug section, so the developers can test it out and fix the problem.
     
  5. Bob

    Bob

    Joined:
    Nov 4, 2007
    Messages:
    2,400
    Likes Received:
    1
    pti2000,

    Andy's suggestion of DTMF is not off the mark.

    Have you confirmed that DTMF is working on your system (just because your SIP phones work, offers no assurance that your DTMF is working.

    For testing, setup an IVR on your system, call it using a Mobile and see if you can press the numbers for the IVR and check to see if this works. If not, look at you DTMF settings.

    Another failure that can occur with DTMF is poor line quality (e.g. SIP connection that is not ideal, this can fail the DTMF straight away).

    So before you throw in a bug report (which I fully recommend you do)...just try to lessen the variables...this is not to say DISA is not possibly faulty, just check all options that you can....

    Also, you would have rebuilt your system to use Elastix 2.0. No matter how much you think you have replicated the setup, there is always something missed (codecs etc)....I still make the mistakes....

    Regards

    Bob
     
  6. dicko

    Joined:
    Oct 24, 2008
    Messages:
    4,099
    Likes Received:
    0
    To further elucidate Bob and Andy's input, Cell phones invariable use GSM, that is just the nature of the technology, so any attempt to use any inband signalling will allways fail miserably, 64k > 32k (or vise versa) = bad dtmf it's just fysix . if you don't use rfc2833 you are doomed to failure, If your VSP does the unthinkable and tries to transcode to inband (strangely some do) then you are SOL :) check with "sip debug" at the asterisk CLI to make sure that the SIP signalling is apparent to both parties
     
  7. pti2000

    Joined:
    Mar 27, 2008
    Messages:
    77
    Likes Received:
    0
    Ok it looks like its the DTMF, Because I setup an IVR and I called in with a cell but could not make contact when I press a number. How do I fix this, where do I change the dtmf setting for this?
     
  8. dicko

    Joined:
    Oct 24, 2008
    Messages:
    4,099
    Likes Received:
    0
    Probably with:-

    dtmfmode=rfc2833

    in all legs of the call, particularly the one to the cell-phone, as stated before, some VSP's don't honor that and the user will have to dig deep into his phone to extend the dtmf length, truly bogus I agree but all VSP's are not equal, try another one before you go further. (JM2CWAE)
     
  9. Bob

    Bob

    Joined:
    Nov 4, 2007
    Messages:
    2,400
    Likes Received:
    1
    Good diagnosis!! ;)

    Now the painful part, and as Dicko said...not all of them are created equal..

    You may need to work with your VSP and determine the right settings. Ring and talk to them about your issue, they might have already come across it with another user and will be able to give you some answers....

    As Dicko mentioned, the codec could play a part as well. The less transcoding between codecs the better. Please note CODEC's and DTMF are different things, but the codec and transcoding can have an effect (as I mentioned, so can line noise and poor transcoding).

    For almost all setups, I stick the GSM codec for all external voice, so this means setting your Trunks to disallow = all and allow=GSM, so that it is the only codec negotiated. If your VSP has a way of locking the codec (as we said they are not all equal), then do that also. The most likely DTMF mode is rfc2833, but try to confirm this with your VSP....

    At least you are now heading in the right direction.

    Regards

    Bob
     
  10. pti2000

    Joined:
    Mar 27, 2008
    Messages:
    77
    Likes Received:
    0
    Ok in my trunk setting for the sip provider I am using in peer detail this what they require

    disallow=all
    allow=ulaw
    canreinvite=no
    context=from-trunk
    dtmfmode=auto

    Now in all the versions before Elastix 2.0 works fine with these settings
    any way can you be more specific on where and what I need to change and or maybe play with to fixed this.
     

Share This Page