thanks!, was Re: [Asterisk-Users] newbie needs SIP config examples -- especially soft phones

[Reed Wade]

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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