asterisk users - Jun 2003 - newbie needs SIP config examples -- especially soft phones

Hi,

I'm experimenting with the dev kit lite and now past the USB
unpleasantness it's working great with standard phones and
lines.

The priority right now is getting soft phones (under Windows
XP) working well.

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.

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.

An example of password protected SIP phone access would also be
very helpful.

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.

thanks,
-reed

> 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

You can also find some examples here for sip.conf and other Asterisk
config files provided by jtodd@loligo.com

http://volume.fox-den.com/asterisk/current/

There are some funny monkey sounds at
http://volume.fox-den.com/asterisk/ in the sounds directory.

Great for scaring people that wouldn't expect to hear a monkey when they
hit a prompt.

John Haigh

-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Reed Wade
Sent: Wednesday, June 18, 2003 12:18 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] newbie needs SIP config examples -- especially
soft phones




Hi,

I'm experimenting with the dev kit lite and now past the USB
unpleasantness it's working great with standard phones and lines.

The priority right now is getting soft phones (under Windows
XP) working well.

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.

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.

An example of password protected SIP phone access would also be very
helpful.

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.

thanks,
-reed




_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users

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
>
>
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users@lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users