I have recently gotten asterisk running on my linux server and tested two xlite client connections on my lan, so calls internal to the lan work, but not with my VOIP phone. The lan is nat'd behind a cisco pix, where I have opened ports 5060 (sip), 5004 and 8000 (rtp) and 4569 (iax). I got a dial-tone when the phone was set to its factory default however once I assigned an ip to the phone and filled in the settings (referencing the manual) I lost the dial-tone so I'm assuming that I can't test connections to the phone on my lan, which I would have liked to do. The status showed:
System Up Time: 0 day(s) 0 hour(s) xx minute(s) - but the phone is up only when NAT is disabled in sip.conf
Registered: No
PPPoE Link Up: disabled
NAT: detected NAT type is blank, or when NAT is enabled in sip.conf I get 'detected NAT type is UDP blocked'
NAT Mapped IP: 0.0.0.0
After giving up on an internal LAN test I enabled NAT traversal in the Grandstream config pages and defined the STUN server as ip.of.my.pix:5060, and, alternately, ip.of.my.pix:4569 - neither one results in a dial tone or connection.
The error that shows up in asterisk is:
Apr 11 15:23:07 NOTICE[26727]: chan_iax2.c:7411 socket_read: Registration of 'my-FWD-account-number' rejected: 'Registration Refused' from: '192.246.69.186'
So it looks like the phone is trying to register with FWD, which is good, I think...but I have no idea where the non-routable ip is coming from, since it does not show up in the phone's config pages, which I'm viewing in a web browser, and it isn't in my asterisk config files either.
Here's my iax.conf:
[iaxfwd]
type=user
context=incoming
auth=rsa
inkeys=freeworlddialup
register => 123456:password@iax2.fwdne
t.net
and my sip.conf:
sip.conf
[phone]
type=friend
secret=welcome
qualify=yes
nat=no or yes - no dial tone
host=dynamic or a static ip, asterisk error is the same
canreinvite=no
context=incoming
and my extensions.conf:
[incoming]
exten => 2005,1,Dial(SIP/xlite_clie
nt)
exten => 2006,1,Dial(SIP/phone)
exten => 2008,1,Dial(SIP/iaxfwd)
[phone]
exten => 2006,1,Dial(SIP/phone)
[iaxfwd]
exten => 2008,1,Dial(SIP/iaxfwd)
Let me know what I need to do to establish a working connection on this phone, thanks. I'm using the Grandstream budgetone pdf manual and the O'Reilly Asterisk book as resources, so I've tried the RTFM approach but it's not working :(