thanks to everyone for your gracious assistance; it stills wants
plenty of minor adjustments but I now have the core of a nicely working
system
-reed
At 11:56 PM 6/17/2003 -0500, John Laur wrote:
So far, I've only been able to get the XTEN Lite phone working
and I really don't understand how I set it up. I used xten
for every option everywhere (display name, username, password,
and Domain/Realm) and the corresponding section in sip.conf.
I've had no luck getting the SJ Labs soft phone to connect using
a similar blunderbuss method.
[youruser] ;username here and also below...
type=friend;dial both to and from
username=youruser ;same thing as in brackets above
password=password ;password obviously
context=default;or put whatever you want - this is the sip realm too
mailbox=1234 ;for message waiting
host=dynamic ;might be coming from different ip's
callerid=Soft Phone 1234
nat=yes;might be behind a nat
I'm wondering if someone could point me to SIP configuration
examples or education so I can understand what I'm doing. I'm
finding the client configuration more confusing that the *
configs.
Your client will want an auth name or two (use the username for these), a
secret or password (the password), a port number (5060 is the default and
you can change it in the [general] section of sip.conf), maybe a realm
(the context though it is not important for authentication), a sip proxy
address - your asterisk server's ip address, and that should be it. Most
have an option you have to turn on to tell the client to actually register
with the proxy. turn that on and check to see that your client is
connected with 'show sip peers' on the asterisk console. It might also be
helpful to turn on 'sip debug' to see if your client is trying to
register. If you got the x-lite working the others should be easy too..
You'll see..
An example of password protected SIP phone access would also be
very helpful.
see above.
I need to be able to support folks working from home connecting
through the net as well inside the office. I expect NAT to be
a pain.
NAT is not so hard once you get it going. First: make sure your asterisk
server has a public IP address and the ONLY default gateway on the machine
is set to the router for the public ip. Make sure you have set nat=yes in
the corresponding sip.conf entry for the device you're setting up, then
start poking at your client for the settings that say I'm behind a NAT
-- they are designed to make sure the packets source at the same UDP ports
they need to come back to so that the NAT's will open up a pathway back to
the internal device. Some clients do this by default anyway -- On the
X-Lite phone you don't really have to do much of anything -- maybe uncheck
the box that says Send Internal IP though I have found that it doesnt
really matter if nat=yes on the asterisk box. On the cisco 7960 phones,
the following settings work:
nat_enable: 1
nat_address;
voip_control_port: 5060
start_media_port: 16384 ; You can reduce this port range if you
end_media_port: 32766 ; have a picky firewall
nat_received_processing: 1 ; Makes phone re-register if your ip changes
Hope this helps you some...
John
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