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

Gabriel Afana asterisk at gafana.com
Tue Mar 7 03:12:59 MST 2006


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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