IVR IGNORES EXTENSION DIALING AND MORE...

Discussion in 'General' started by striderec, Dec 21, 2008.

  1. striderec

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    Greetings, I've got two problems I'd appreciate you all to help em with:

    1) IVR IGNORES CALLER ATTEMPTS TO DIAL AN EXTENSION (OR AN OPTION)

    Caller hears the IVR announcement perfectly, it asks him/her to either enter the extension # they want to communicate with or dial "1" to hear the options in spanish. Surprise! The IVR _IGNORES_ completely what the user dials.. the message plays till the end and no matter how many times the caller enters an extension number or dials "1" The IVR won't do a thing until Elastix hangs up the call due to timeout. Why is that?


    2) IVR CONFLICT WITH 2 INTERCONNECTED ELASTIX BOXES

    Using an IAX2 trunk I created on both sides, I've successfully connected 2 Elastix Boxes located in different countries. Both see the each other as local boxes and all extensions, routes and trunks work seamlessly like if both offices were just one. Problem is.. When a customer calls a local PSTN DID associated to the Elastix Box of the Office #1 and wants to dial an extension that is located in office #2, the IVR responds with "INVALID EXTENSION" Why is that? The idea of connecting two boxes is to balance the workload, the box in Office #1 is very busy while the box in office #2 is almost unused. Also, the idea is NOT to duplicate extension numbers on both boxes. In box #1 the extensions ranges are the 1XXX, 3XXX and 5XXX. Extensions ranges for box #2 are the 7XXX. I want customers who call to the DID of the Box #1 be allowed by the IVR to call the 7XXX extensions and get transferred instead of getting the "INVALID EXTENSION" error. What can I do?

    Thank you in advance,

    PAUL
     
  2. Patrick_elx

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    for problem #2
    when you receive a call to your DID and answer with the pabx#1 extension, can the remote extension on pabx#2 still contact pabx#1?


    could you log to your pabx (localy or remotely via ssh) and go the cli with:
    asterisk -vvvvvvvvvvr

    then type the proper debug command(s) depending on your trunk like:
    sip set debug
    rtcp set debug
    rtp set debug


    try to call and look at the trace.
    you will see probably more explanation why your transfer is not working.

    for problem #1
    it looks like the dtmf is not received properly. Did you try to call from different kind of phones? Are you sure that the dtmf is sent properly all the way to your pabx and there is no problem somewhere else in the link. I know that does not solve your problem, but I remember loosing hours on a problem I thought was local that was in fact a network issue.
     
  3. striderec

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    Hi Patrick and thank you for a prompt response.


    As for Problem #2, yes, I've tested 3 different phones from 5 different locations, my own hi-tech cell phone included, a friend's cell phone and 3 fixed phones and the same problem persists. I was thinking of a dmtf problem but all extensions are set to the standard rc2833. How could I know that the problem goes elsewhere? The network is fine, We have a dedicated 1.5 Mbps internet access speed in our end that is barely used so what kind of network problems could they be? Any suggestions about where (or how) to check where the problem originates?

    In problem #1, I can definitely access both boxes since I am in charge of them, I'm gonna try and set the verbose to 10 but the problem is the buffer limit of my SSH client. I don't think it can cope to the brutal amount of lines you get with a verbosity set to 10. I'll try it anyway but if you know of another (and faster) workaround for my problem i'd certainly appreciate it.

    Thank you in advance,

    Paul
     
  4. juancarlosg6

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    Hi, i have de same problem #2 when the call arrive to IVR, digit a extension number but some times no call directly to extension, i don now why, mi trunk is by E1.
     

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