How to use RDNIS variable

Discussion in 'General' started by tecnoclic, Oct 28, 2009.

  1. tecnoclic

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    Hi,
    Anybody knows how to use RDNIS variable. In others words, how to read this variable (my Voip Provider sends it, I see when I make a sip debug in CLI> ) to route some incoming calls to its IVRs.
    Explanation:
    Let's suppose a person dialed 5551699. It number my Voip Provider sends and asterisk shows like:
    RDNIS is 5551699. So this call I route to IVR-1. But if another person dialed 5551688, this one I route to IVR-2.
    Anybody knows how to resolve it?
    Note: the phones (5551699 and 5551688 are forwarding to my asterisk DID: 5551677)
    Thank you
     
  2. tecnoclic

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    What kind of Forum is it?
    I've written 10 post and all without any response. The only answered questions in this forum is:
    What's the default password of elastix web administrator?
    Answer: palosanto
    How can I create a new extension?
    Answer.....
    How to change the password of web interface?
    Answer....
    When some "Elastix GURU" see this kind of questions, inmediatly writes the answer.

    But when some people have a different question, that is not in FAQ over wide internet, those "GURUS" keep silence....
    WHERE THE ELASTIX GURUS????

    LAMERS YOU ALL!!!!
     
  3. smitch

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    tecnoclic,

    not tryin to jump on ya. BUT. do you know how to search google? Using a search term of "rdnis asterisk" returns lots of hits. the first hit from voip-info will give you the required info. to use it, you will have to do some custom dialplan scripting. additionally, there is wonderful book called Elastix without Tears in the download section of this site. READ IT. Another must read is Asterisk: The Future of Telephony. It can be downloaded in pdf form here:
    http://www.asteriskdocs.org/ You will gain alot of valuable knowledge.

    Last bit of advice.

    INVEST IN YOURSELF BEFORE EXPECTING THIS FORUM TOO!!!!!

    (p.s. and then maybe someone will answer your posts)
     
  4. ramoncio

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    Calm down, tecnoclic, and wellcome to the forum.

    There aren't many Elastix 'gurus' as you say. And they might be busy, so please be nice and patient.

    Meanwhile you can search Google, as smitch wisely says, look for your own answers and post them here when you have the solution, so you can help others.

    I even didn't know what RDNIS was.

    You are asking about a service that is not implemented into Elastix, so I think it is quite normal that people haven't even heard about it, so it's difficult to get an answer.

    A couple of years ago it was nearly impossible to get someone to answer ANY question about Elastix. At least now, most trivial questions, and many advanced ones get their answers here.

    And last, if you are a new member of the community and you start yealing like that, you won't get many answers. This is a community, and we don't get paid, so people might or might not answer you at their will.

    You can always call Palosanto paid support, there you can find the real 'gurus' who will answer for sure.
     
  5. tecnoclic

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    Hi.
    Ramoncio, this service is implemented into Elastix. I say it, because when I make a debug sip command in CLI, among all data appears:
    RDNIS is ######## it number is the DNIS (in asterisk is called RDNIS).
    So my question is: how to get this variable to route some incoming calls?
    Thanks
     
  6. dicko

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    Your logic is flawed,, what you see is in the sip headers, so could many other things be that are not supported by Asterisk. SIP - (Session Initiation Protcol) can be used for anything from SMS messages to HD-TV. In fact social security numbers might well be embedded in the rtp stream (audio) of the calls, that does not mean that that information is "implimented"

    I might point out that Elastix is a wrapper around FreePBX, which is a wrapper around Asterisk, RDNIS is rudimentarily implimented in Asterisk:

    http://www.voip-info.org/wiki/view/RDNIS

    and

    http://www.voip-info.org/wiki/view/Aste ... c+callerid

    To which site you were previously pointed too. So your statement "is implemented into Elastix" although incorrect, could perhaps be changed to "is implemented a little in Asterisk",

    You will have to go upstream from Elastix (which does not directly impliment it), to FreePBX (which does not directly impliment it), to Asterisk, where you will have to learn how to write a custom context to change the way YOU handle caller ID, to do what you want with it, having done that feel free to add it, following the normal constraints (perhaps extensions_override_freepbx.conf) to FreePBX, then it will be available in Elastix.

    dicko
    (a LAMER when it comes to appeasing others' sense of entitlement)
     
  7. michellesullivan

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    Very simple.

    Create a custom context in extensions_custom.conf and add the lines:

    [custom-get-cid-sip]
    exten => _.,1,Noop(Fixing DID with RDNIS number (${CALLERID(RDNIS)}))
    exten => _.,n,Goto(from-trunk,${CALLERID(RDNIS)},1)

    (note in this case I called my custom context 'custom-get-cid-sip')

    Set the context in your PEER details for the trunk to the new/custom context.

    Put the number sent in the RDNIS line in the DID field for an incoming route, and voila you have an inbound route based on the RDNIS (which in my case tells me the phone number the calling party dialed to get to me) Multiple numbers on the same SIP trunk will require multiple inbound routes (one for each number.) Multiple trunks can use the same custom context, but each inbound number will require a new inbound route. If you edit the config files yourself you could use a pattern match to pick them up if you so desire.

    Best,

    Michelle
     

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