Grandstream SIP phones and PTSN

Discussion in 'General' started by rs232c, Sep 6, 2009.

  1. rs232c

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    I have the Grandstream GXP2000 SIP Phone properly connected by ethernet cable to the Elastix server with static IP's. I do not use a SIP trunk, only PTSN lines. I have already created the extensions, trunks, inbound and outbound routes according to Elastix Without Tears. Because there is no SIP accounts I have not setup any account information on the phones themselves.

    Is there anyone that can instruct me on anything I need to do to program these phones themselves to link up with PTSN lines?

    Thank You
     
  2. rejil.rajan

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    Hi

    So you mean to say that, these phones can communicate with each other using Elastix server. If this is true, you will just need to create a trunk to the PSTN as mentioned in the Elastix without Tears and create a route which will be used by these phones to make calls out through the PSTN
     
  3. rs232c

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    This means there is nothing to program into the phones themselves beyond a static IP that will link up to the incoming and outgoing routes.

    The phones are set up correctly.


    Thank You
     
  4. rejil.rajan

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    I hope you have configured the Phones to communicate with the Elastix Server, that is why I asked if your able to call between the phones dialing there extensions
     
  5. rs232c

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    Sorry, I missed that -

    Yes, I can call between each extension as the Elastix server is not required to do that. What I am not sure about is do I program each phone separately by it's own IP setup page to link up to any PTSN line (i.e., program the phone to communicate with the Elastix server). This assumes setting up Account1 through Account4 and/or Ext1 through Ext2 as needed.


    Thank You
     
  6. rejil.rajan

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    Yes, you will have to setup the account giving the username and password as per the extension you created in the Elastix Server, you will need to create SIP extensions in the server to do the same.

    The SIP proxy on the grandstream device should be given as the Elastix Server IP. This link should help you do that

    http://www.asteriskguru.com/tutorials/g ... phone.html
     
  7. rs232c

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    I did not use the extension number as the SIP User ID number, I used the description label instead. Now I am without issue for dialing in and out.


    Thank You


    Resolved.
     

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