asterisk users - Mar 2005 - How to register two SIP phones ( e.g. Windows Messenger) from different subnet to *

Hi All,

I have two SIP softphones(Windows Messenger) running on different subnet
(Phone-1 on IP XXX.XXX.25.ABC & Phone-2 on IP XXX.XXX.15.XYZ) and my 

Asterisk Server is running on IP XXX.XXX.25.PQR.Because of some security
issues both the subnets are completely isolated ( U cant even PING from 

one to other) and  I want to connect Phone-1 & Phone-2 to the *. 

How can I proceed? Please do help in this regard

Thanks and Regards,

Firdosh.

You you can reach the * server from both location, it's straight
forward, standard registration; from your post I understand that this
is NOT possible as * is in the XXX.XXX.25 subnet and phone 2 is on
XXX.XXX.15.* which is completely isolated).

So given your current config, no, you can't do it. Not unless you
somehow can route traffic from XXX.XXX.15.* to XXX.XXX.25.* (and
back). You can, however, more the * server into a third subnet that is
reachable by both phones (even if the phone cannot reach each other
directly).

--Luki

On Sat, 2005-03-12 at 07:42, Luki wrote:
> Firdosh,
> 
> there were couple typos on my last email, but that's essentially what
> I said. There are two ways of doing it -- but neither will work given
> you current setup.
> 
> 1) Phone A talks directly to B.
> 2) Both Phone A and B talk to a common point C. Point C proxies
> traffic between A and B, because A and B cannot see each other
> directly.
> 
> You you can't have both clients on the same subnet, then you need a
> third subnet C that is reachable from both A and B. Asterisk runs in
> subnet C and proxies the traffic between A and B.
> 
> --Luki

Hi All,

I have a dedicated * server at 172.16.200.150 and my two windows
messenger clients are at 172.16.25.X & 172.16.15.X. Now the server is
visible to both the subnets.Both the users/clients[say msn1 & msn2] are
configured. Then call is made from one user to another. After the callee
receives/accepts the call, neither of users able to hear anything. Sip
debug shows 200 OK for the call.Do I have to "register=>" the
users, if
yes kindly mail the register string.

Here are the sip.conf and extensions.conf

sip.conf
---------
[msn1]
type=friend
host=dynamic
context=default
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
canreinvite=yes
nat=yes


[msn2]
host=dynamic
type=friend
context=default
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
canreinvite=yes

extensions.conf
----------------
[default]
exten => msn1, 1, Dial(SIP/msn1, 20)
exten => msn2, 1, Dial(SIP/msn2, 20)



Thanks and regards,

Firdosh