Asterisk Phone Registration

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

  1. rs232c

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    I am trying to register an extention with Asterisk. I created a user on the Elastix web pages and entered the user information into the extention under Account1. I have no SIP service with my Grandstream GXP2000. I did not modify any configuration files directly besides the two places listed. I did add the server information, etc, as required.

    What else must I do to have Account1 register with Asterisk or at least show me where it attempted and can get an error message?
     
  2. jgutierrez

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    From the server, are you able to make a ping to your phone's ip address? Have you entered correctly the extension, password? What does you get when you execute from shell:
    asterisk -rx "sip show peers"
     
  3. rs232c

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    This is what I get for asterisk -rx "sip show peers"

    Name/username Host Dyn Nat ACL Port Status
    104 (Unspecified) D N 0 UNKNOWN
    103 (Unspecified) D N 0 UNKNOWN
    102 (Unspecified) D N 0 UNKNOWN
    101 (Unspecified) D N A 0 UNKNOWN
    4 sip peers [Monitored: 0 online, 4 offline Unmonitored: 0 online, 0 offline]
     
  4. dicko

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    Greetings rs232c:

    Having watched your posts over the last couple of days, it occurs to me that nobody has suggested that you read "Elastix Without Tears" , I hereby suggest you do so , your problems seem very typical of many new users, and this publications will I believe largely answer your questions in a wholesale rather than a retail fashion.

    I notice you take your handle from a very well known and understood communication protocol, VOIP is largely an analog of that protocol, in other words RTS won't be raised unless DSR is also raised, "Elastix Without Tears" largely enables the DSR signal in the warm-ware port involved (dev/tty/you in this case) , your trunks must agree as to baud word length and stop bits and hand-shaking, then RI will be responded to by DSR which will accept DTR which will open the session.

    I hope you appreciate my allegorical whimsy, but I can assure you that when you get the signaling protocols right, the data will flow from RX to TX and vice versa (asynchronously))

    regards

    dicko

    p.s.

    don't try and short pins, it don't work in VOIP all signals require attention.
     
  5. rs232c

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    Thank you. I have read it completely and followed it's instructions no less than four times.


    The issue has been resolved, I had to put the extention number into the phone extension setup pages for the SIP User id user not the display name label that I created for it. Once that was done, everything came to life and I can now dial in and out.

    Now it's time to setup the features.

    Resolved.
     
  6. dicko

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    I'm glad you got there,

    Reading is preparatory to comprehension. Comprehension will make programming the features trivial. Welcome to Elastix/Asteriskthe.
     

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