[Asterisk-Users] reject connect from iaxtel.com

Rich Adamson radamson at routers.com
Mon Jan 5 18:45:14 MST 2004


> I have problem trying to receive incoming calls from iaxtel.com. The 
> error message is " rejected connect from ip address - <iaxtel.com>.
> 
> I have set up the iax.conf file as follow:
> 
> port=5036
> allow=gsm
> register=>dkwok:password at iaxtel.com
> 
> [dkwok]
> type=friend
> context=from_iaxtel
> 
> My extensions.conf is as follows:
> [from_iaxtel]
> exten => 17001813482,1,Dial(sip/1001,20,tr)
> 
> iax2 show registry
> 69.73.19.178:4569  dkwok  203.219.xxx.xxx:1200   60   Registered
> 
> But when connection is attempted, the console says :
> File chan_iax2.c, Line 4301 (socket_read): Rejected connect attempt from 
> 69.73.19.178.
> 
> Any pointer will be appreciated.
> 
> Also when iaxtel put call through local iax server does it send username 
> password and is it matched up with the client section in iax.conf?

Kind of looks like you have a couple of problems based on the limited
config info provided above.

In iax.conf, you need to define a context for the incoming iaxtel call
that includes the "type=user", "username=" and "secret=". 
The "context=from_iaxtel" is where the incoming calls are sent (within
the extensions.conf file), and in your example shown above, the incoming
call can only send the digits "17001813482". If anything else is
sent (including nothing), nothing will happen. In the simplest case
you probably want that to be something like:
 exten => s,1,Dial(SIP/1001,20,tr)
and that essentially means regardless of what was sent for digits
from iaxtel, send the call to x1001.

For outgoing calls to iaxtel, you need something in your dialplan
(with extensions.conf) that looks something like:
 exten => _1800NXXXXXX,1,Dial(IAX2/dkwok:password at iaxtel.com/${EXTEN}@iaxtel)
which means, when you dial 1-800-Nxx-xxxx from your sip phone,
send this call via iax2 protocol to iaxtel.com using the 
username=dkwok and password=password.

It would be in your best interest to surf around the wiki at
 http://www.voip-info.org/tiki-index.php
as there is a lot of good asterisk config data available.

Rich








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