iax voipcheap.com

Discussion in 'UK VOIP' started by ¬Stephen Hammond, Dec 12, 2006.

  1. Apparently iax is better is it possible to use iax with voipcheap.com ?
     
    ¬Stephen Hammond, Dec 12, 2006
    #1
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  2. ¬Stephen Hammond

    Ivor Jones Guest

    IAX is better for what..? IAX stands for Inter-Asterisk-eXchange and is
    used for talking between/to/from Asterisk boxes.

    However, while it is theoretically easier to set up, in that it only uses
    a single port (usually 4569) rather than the myriad of port numbers used
    by SIP, it isn't necessarily "better" as such. I and a friend in North
    Wales are currently scratching our fingers down to bone level trying to
    figure out why an IAX ATA at my end is suffering no end of problems
    connecting to his Asterisk PBX whereas a similar SIP unit is working
    fine..!

    Ivor
     
    Ivor Jones, Dec 12, 2006
    #2
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  3. Can your friend connect to it locally using IAX?
     
    David Quinton, Dec 13, 2006
    #3
  4. ¬Stephen Hammond

    Ivor Jones Guest

    "David Quinton"
    Yes. We're beginning to think it's a firmware bug in the IAX ATA -
    http://www.atcom.cn/En_products_AG188.html as someone else is having
    exactly the same problem connected to a totally different Asterisk box.

    The device is very new on the market, we're waiting on a reply from the
    manufacturers.


    Ivor
     
    Ivor Jones, Dec 13, 2006
    #4
  5. ¬Stephen Hammond

    Tim Guest

    IAX is very proprietary.

    Even though the source code is open, it is still a specific to asterisk
    thing.

    Tim
     
    Tim, Dec 13, 2006
    #5
  6. ¬Stephen Hammond

    Ivor Jones Guest

    Maybe that's why it's called Inter-ASTERISK-eXchange ;-)

    If anyone's interested, the system we're using is CNET (see www.ckts.info)

    Ivor
     
    Ivor Jones, Dec 13, 2006
    #6
  7. Have you tried the manufacturers forums?

    http://bbs.pa1688.com/ (I know it's not a PA1688-based ATA, but that's
    just the url of the site).
     
    Thomas Kenyon, Dec 13, 2006
    #7
  8. ¬Stephen Hammond

    Ivor Jones Guest

    Yes, already looked there. Nothing specific to our problem. It doesn't
    help that CNET isn't exactly a mainstream use of VoIP..!

    Ivor
     
    Ivor Jones, Dec 13, 2006
    #8
  9. ¬Stephen Hammond

    David Cook Guest

    Try IDEFISK from http://www.asteriskguru.com. If this works then
    unfortunately it is your ATA.

    Best regards.
     
    David Cook, Dec 14, 2006
    #9
  10. ¬Stephen Hammond

    Ivor Jones Guest

    Thanks, I'll give that a look. The problem is, some of the trouble we're
    having is intermittent, i.e. a call attempt will fail two or three times,
    then succeed on the next try.

    Ivor
     
    Ivor Jones, Dec 15, 2006
    #10
  11. Is there a qualify= line in the iax.conf entry on the remote host?
    Are you registering to the remote host or are you using static hosts?
     
    Thomas Kenyon, Dec 15, 2006
    #11
  12. ¬Stephen Hammond

    Ivor Jones Guest

    I don't know about qualify= as I'm not at the remote end and my knowledge
    of Asterisk/Linux is virtually non-existent..! However I will ask and let
    you know. What should be in that line..?

    Using registration as you would to a normal VoIP provider.

    Incidentally, I've tried the DIAX softphone on my laptop and that works
    fine.


    Ivor
     
    Ivor Jones, Dec 15, 2006
    #12
  13. ¬Stephen Hammond

    ale.cx Guest

    Asterisk in most configurations logs most or all of what it does in
    /var/log/asterisk. You can also increase debugging with 'iax2 debug'
    from the console. Annoyingly enough it's not possible to debug an
    individual IAX peer or trunk in the way you can with SIP :-(
     
    ale.cx, Dec 15, 2006
    #13
  14. ¬Stephen Hammond

    ale.cx Guest

    Asterisk in most configurations logs most or all of what it does in
    /var/log/asterisk. You can also increase debugging with 'iax2 debug'
    from the console. Annoyingly enough it's not possible to debug an
    individual IAX peer or trunk in the way you can with SIP :-(
     
    ale.cx, Dec 15, 2006
    #14
  15. ¬Stephen Hammond

    ale.cx Guest

    Asterisk in most configurations logs most or all of what it does in
    /var/log/asterisk. You can also increase debugging with 'iax2 debug'
    from the console. Annoyingly enough it's not possible to debug an
    individual IAX peer or trunk in the way you can with SIP :-(
     
    ale.cx, Dec 15, 2006
    #15
  16. In this situation, normal is only defined by your setup, if you both
    have static IPs then you wouldn't want to register. (registrations are
    quite uncommon with iax, so it wouldn't surprise me if the registration
    code in the ATA hasn't been exhaustively tested).

    The only VoIP providers I use that will use iax to terminate to you,
    (gradwell and voiptalk) do not support registrations. (afaik).

    It would also be prudent (if the ATA supports it) to use RSA
    authentication. (if you don't know, ask).
     
    Thomas Kenyon, Dec 15, 2006
    #16
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