SIP trunk with ISDN pbx behind Elastix server

Discussion in 'General' started by Mattz, Jul 8, 2008.

  1. Mattz

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    Hi,

    I'm trying to accomplish the following setup:

    I have:

    - sip account at a provider, national one.
    - Elastix server
    - BRI ISDN card
    - ISDN pbx

    I would like to use the ISDN PBX to call with using my old solution. I have added an ISDN card to a Elastix Box and set it as NT.

    I made a ISDN-crossover cable for between the Elastix server and the ISDN pbx.

    This all works kinda OK, I get a dailtone when I pick up the telephone that is attached to the ISDN PBX, yes and this is a dialtone from the Elastix server what I hear.

    I have setup a SIP trunk with my sip account, and show sip peers shows me an OK.

    now I actually need to accomplish that the number, sip account, that I dail will be forwarded to the ISDNcard/ISDNpbx.

    And when I want to call from my ISDN-PBX that the outside route is using my sip account.

    I'm a little bit confused here because I don' t know for sure of I need more than one trunk for this because of the NT card that' s in it.

    Can someone advise me here or give me some nice examples for my situation ?

    I have tested and searched the whole day about this, but I don't see this setup happening a lot.

    Thanks.
     
  2. MageMinds

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    Just to make sure.

    Your ISDN trunk is configured in the list of trunks right?

    If yes, then. Here what I think.

    You need a Ring Group that will dial ONE extension on your ISDN PBX, if you need ot make more than one phone ring you'll have to setup a ring group on your ISDN PBX and use that extension in the Ring Group. Don't forget to suffix the extension with a # in your Elastix Ring Group. The # tell FreePBX that the number should be dialed using an Outbound Route, we'll see that later.

    You need a Inbound route that will send all incoming SIP calls to that Ring Group.

    You need an Outbound route that will match the extension number on your ISDN PBX.

    Setup a Soft Phone and register it with Elastix and see in the Asterisk CLI (asterisk -vvvvvr) when you dial the extension of your ISDN PBX if it works. if it works then try to call your PSTN-SIP number to see if asterisk dial the ISDN card.

    This will only cover incoming calls. Make the incoming call works first, then we'll see for the outgoing calls.

    I'm not sure yet of the best way to make outbound calls from your ISDN PBX. Asterisk will see those call as incoming calls from the ISDN Trunk and you need you route them automatically to your SIP trunk. DISA won't work... Oh maybe yes, if you dial 9 on your ISDN PBX you can send that to Elastix and the Inbound route with a DID of 9 will automatically send the call to DISA and then you dial as normal. That depend on how your ISDN PBX handle the signalling of the phones, is it digit by digit or as a bunch of digit? For outbound you'll have to test and come back here with results to find more idea.

    Good luck.
     
  3. MageMinds

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    Always keep in mind that Asterisk have to work with two set of trunk and no Extensions.

    Your setup would probably be simpler to configure in pure Asterisk text file with the FreePBX GUI. I'm prety sure there's a way for FreePBX to make it work and if you try to do it directly in Asterisk text file you're on your own, because I'm no expert in that.

    For your outgoing calls you will probably have to use text file and the one you're looking for is /etc/asterisk/extensions_custom.conf in there you'll have to add a [from-isdncardtrunkname-custom] context, this is a context included automatically by FreePBX, look into the extensions_additional.conf to find the right name for your custom context; search for "[from-trunk" and cycle between them to find the one representing your ISDN trunk, in the context you will find on the first line include => "what your looking for"

    In that custom context you'll have to take care of Dialing SIP with the DID received from your ISDN. The exten command will send everithing to a Dial command bypassing everything that the default FreePBX context contain. After the dial you will probably have to add a Hangup() to avoid continuing to execute the default context. This is as far as my knowledge go in Asterisk for now...

    THIS IS A WILD GUEST, DON'T LISTEN TO ME YOU WILL PROBABLY BREAK EVERYTHING WITH THIS. (I think you understand what I'm saying.)
    Code:
    [from-trunk-isdn-custom]
    exten => _.,1,Dial(SIP/siptrunkname/${EXTEN})
    exten => _.,2,HangUp()
    
    <br><br>Post edited by: MageMinds, at: 2008/07/10 02:35
     
  4. MageMinds

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    I just reread your post an see that you get a Elastix dial tone from your ISDN PBX Phones.

    I assume you dial something like 9 to get the Elastix dialtone right?

    Either way I pretty sure that outbound calls will have to be handled manually in a custom context.
     

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