[Asterisk-Users] Calls between Asterisk servers using SIP?What about IAX (got it working w/ IAX but I have questions)

[Gabriel Afana]

Does anybody have any experience with capabilities here?  I need to know if IAX is able to handle more than that.  I think I might just benchmark this with a bunch of .call files between servers to see how they are handled.

Any input?

- Gabriel Afana

  ----- Original Message ----- 
  From: Umair Bari 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Sent: Tuesday, March 07, 2006 3:30 AM
  Subject: Re: [Asterisk-Users] Calls between Asterisk servers using SIP?What about IAX (got it working w/ IAX but I have questions)


  Hello Gabriel,

  IMHO, using IAX between * servers is a good choice, I dont see any problem in it. Actually I used it for sometime and never encounter any issue, but i had max 5 concurrent connections.
   
  regards,

  Umair bari
   
  On 3/7/06, Gabriel Afana <asterisk at gafana.com> wrote: 
    Hi everyone,
       I just spend the last two hours trying to get two asterisk boxes to
    transfer calls between eachother using SIP.  I dont know why but I *could 
    not* get the calls to authenticate!  I think I got everything setup.

       There was Server A and Server B.  I was trying to place a call from a
    users registered on Server A to a user regsitered on Server B.  I setup the 
    registration info for Server A and even had Server A registering
    successfully to Server B.  However, whenever I would hand off the calls from
    server A to Server B, it would *always* say it failed to authenticate 
    (passwords did not match).  Here was my setup:

    SERVER A:
    register => serga:test at 216.152.244.81

    [to_80]
    username=serga
    type=friend
    secret=test 
    host=216.152.244.81
    disallow=all
    allow=ulaw
    user=phone
    usereqphone=yes
    canreinvite=yes
    regseconds=0
    cancallforward=yes
    dtmfmode=rfc2833
    disallow=all
    allow=ulaw 
    insecure=very
    trunk=yes


    SERVER B:
    [serga]
    type=friend
    username=serga
    trunk=yes
    notransfer=yes
    secret=test
    context=302
    host=dynamic
    qualify=yes



    DIALPLAN ON SERVER A: 
    exten => 302,1,Dial(SIP/to_80/302 at to_80,30,r)

    It always says authentication failed.  However I always noticed it showed
    the user as 301 at 216.152.244.70.  This is the extension of the phone I am 
    calling from.  It seems it is trying to authenticate the actual phone I am
    calling from on Server A, and not Server A itself.  Was I doing something
    wrong?

    I tried doing this with IAX and within 5 minutes I had it all working!!  I 
    feel it was too easy :-)   However, this brings up a big question.........Is
    IAX very reliable for this?  I've heard from people that I should not use
    IAX under any condition because it really is not very
    reliable/thourough/consistant...etc.  I am trying to start a VOBB company 
    and will obviosly need a reliable setup.  I am thinking to have all phones
    register to the servers via SIP and maybe just have all the servers transfer
    calls between eachother via IAX.  Does this sound like a correct setup? 

    - Gabe


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