Hello,
I'm trying to comprehend loose routing concept and I have
a question that concerns me.
As far as I understand loose routing says that if there're Route
headers in a message it should be forwarded according to the URIs
set in Route headers.
I thought that this is true only within a dialog, but RFC3261 (part 16.6) says:
"Requests establishing a dialog may contain a preloaded Route header field."
Also SER manual says: " the failure not to include loose routing in your scripts
may lead to infinite loops. Make sure that you include the following script
fragment immediately after request sanity checks" and provide the following
piece of code:
if (loose_route()) {
t_relay();
break;
};
which as far as I understand unconditionally forwards message if Route header
is present.
So I'm wondering what about security? If I follow this guidelines how I would
shield my PSTN gateway if anyone can construct message and
pre-load it with URI of my gateway and all my proxies must honor it.
For example I have a PSTN gateway on ip address 10.1.1.5 and proxy
on 10.1.1.10 that supposed to interface outside world.
So I guess if someone construct a message like this:
INVITE sip:12345@somewhere.com SIP/2.0
...
Route: <sip:12345@10.1.1.5;lr>
my proxy will forward it to PSTN gateway and it will make outbound call.
Is this true? Please enlighten me on this.
Thank you,
Michael
Hi there
I'm having trouble registrering with my SER build (0.9.2), and using the
auth_mysql ser.cfg from ONsip.org. It is as if SER isn't recieving any
of the messages from my UA, and I've been wondering where I might have
gone wrong. One thing I've found different from a former build (0.8.14),
I have up and running is the socks which SER is listening on. When i do
a serctl ps on the 0.9.2 i get this:
ser@sipsite:~/running/ser0.9.0/sbin> serctl ps
200 ok
0 22531 attendant
1 22532 fifo server
2 22533 receiver child=0 sock= 212.97.222.xxx:5060
3 22534 receiver child=1 sock= 212.97.222.xxx:5060
4 22535 receiver child=2 sock= 212.97.222.xxx:5060
5 22536 receiver child=3 sock= 212.97.222.xxx:5060
6 22537 timer
7 22538 tcp receiver
8 22539 tcp receiver
9 22540 tcp receiver
10 22541 tcp receiver
11 22542 tcp main process
And as far as I can see the same command on 0.8.14 yields almost the
same output, except that the sock has a number like '0' and a @ sign is
between scok and ip (eg: sock=0 @ ip-adress)! Can this be the cause of
my troubles?
I'm afraid that I'm not that much into Unix and Sockets!
Any and all help will be greatly appreciated!
Kind regards
Martin
Hi list !
I have a Problem with the appending of an rpid with the
functions avp_load_radius(), append_rpid_hf().
In my Radius DB i have an SIP-AVP field which is filled with the Value
rpid:[NUMBER]
When i run manuall the Radius Reply-Query i get the correct value for
SIP-AVP.
Now, i reconfigured my ser.cfg so that an avp_load_radius() is executed
after an incoming INVITE.
My Radius is answering on the command and sends the correct SIP-AVP
Value for the User.
Now my Problem:
I can't append the RPID or anything else to the SIP-Header after the
answer from the Radius.
So the call is going out without the rpid in the Header.
Can anyone help me ?
Here is the Configure-Snip:
if (method=="INVITE") { # here comes the invite
if (!radius_proxy_authorize("")) { # for the digest avp_load_radius it seems that i must use an proxy auth
proxy_challenge("","0"); # and an proxy challenge
if(!avp_load_radius("digest")) { # here i get the SIP-AVP from the radius
log(1,"Failure with the AVP-Value\n"); # if it fails i'll get a message
break; # go out
}
avp_print(); # print all avp's (this command is never executed :( )
append_hf ("Remote-Party-ID: 111222333\r\n"); # append this silly row - it is never executed
append_rpid_hf("", ";party=calling;id-type=subscriber;screen=yes"); # append the rpid from SIP-AVP - and this command is also never executed
break; # go out
};
route(3); # go to route block 3
break; # end
}
I tried also this snippet without the breaks and proxy-commands but it doesn't work.
I don't know what is wrong - please help me :s
Thanks !
Dirk
I am trying to route a call from a non-routable 10.x.x.x network to a
phone on the internet using ser 0.9.0 and mediaproxy 1.31. Aka: PSTN to
SIP from my asterisk to an x-lite
I found some references indicating that I should call:
if ( src_ip = IP_GATEWAY)
{
force_rport();
fix_contact();
use_media_proxy();
};
Supposedly, you need the force_rport and the fix_contact to put media
proxy in the middle of the two rtp sessions, but I have et to see this
actually work. Has anyone been able to pass a call through media proxy
from a GW on a non-routable to a routable network, or even between two
ip's on the same network as the mediaproxy?
If so, please share your configs.
-----Original Message-----
From: bogdan aux [mailto:aux1d@yahoo.com]
Sent: Friday, May 13, 2005 6:16 AM
To: serusers(a)lists.iptel.org
Subject: [Serusers] Mediaproxy in LAN (?)
I want to route RTP packets between 2 computers in the
same network through Mediaproxy. I know this may have
no sense, but it's just a test configuration.
The call use_media_proxy() seems to have no effect,
the RTP packets go directly from one computer to
another.
My guess is mediaproxy makes a test on the IPs (they
are 192.168.147.2 and 192.168.147.3) and refuses to
proxy the call.
Is mediaproxy working just between different networks
or it's an error in my configuration file?
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Hello,
is it possible to use avp_write on the responses in failure route?
If yes, Contact header field could be evaluated and pushed into RURI,
then routing via route(x) to a new destination and finally solve the 3xx
forwarding issue.
Regards,
Martin
Hi SER Users !
Is there a mean to check that the URI submitted in 'TO' header during
registration is allowed for a user in the uri table or SER(if URI and username
are not the same) ??
I tried with function check_to() but it doesn't work. REGISTERs are rejected
because username does not match with URI. However it's written in documentation
that this function should performs a check in the uri table.
Is there any other me to perform this verification ?
Thanks for your help !
Hi list !
I have a Problem with the appending of an rpid with the
functions avp_load_radius(), append_rpid_hf().
In my Radius DB i have an SIP-AVP field which is filled with the Value
rpid:[NUMBER]
When i run manuall the Radius Reply-Query i get the correct value for
SIP-AVP.
Now, i reconfigured my ser.cfg so that an avp_load_radius() is executed
after an incoming INVITE.
My Radius is answering on the command and sends the correct SIP-AVP
Value for the User.
Now my Problem:
I can't append the RPID or anything else to the SIP-Header after the
answer from the Radius.
So the call is going out without the rpid in the Header.
Can anyone help me ?
Here is the Configure-Snip:
if (method=="INVITE") { # here comes the invite
if (!radius_proxy_authorize("")) { # for the digest avp_load_radius it
seems that i must use an proxy auth
proxy_challenge("","0"); # and an proxy challenge
if(!avp_load_radius("digest")) { # here i get the SIP-AVP from the
radius
log(1,"Failure with the AVP-Value\n"); # if it fails i'll get a
message
break; # go out
}
avp_print(); # print all avp's (this command is never executed :( )
append_hf ("Remote-Party-ID: 111222333\r\n"); # append this silly row
- it is never executed
append_rpid_hf("", ";party=calling;id-type=subscriber;screen=yes"); #
append the rpid from SIP-AVP - and this command is also never executed
break; # go out
};
route(3); # go to route block 3
break; # end
}
I tried also this snippet without the breaks and proxy-commands but it
doesn't work.
I don't know what is wrong - please help me :s
Thanks !
Dirk
Hello everybody,
is it possible to check if a user is already registered or authenticated?
I want to allow only registered users call to PSTN destinations.
Regards
Bastian
Hello,
I am a student in DCU University in Dublin, Ireland, and i was down loading
the SIP Express Router onto a Sun Blade and i cant seem to permanently set
the environment variable "SIP_DOMAIN" to my localhost. I use the followng
commands in the terminal to set the variable SIP_DOMAIN to the localhost
and it works but when i reboot the Sun Blade and type in "#echo ${SIP_DOMAIN}"
i dont get back localhost. If you have any solutions to this problem, i would
be delighted to hear of them.
#SIP_DOMAIN=localhost
#export SIP_DOMAIN
#echo ${SIP_DOMAIN}
Regards,
Diarmuid.
hello.
I have a Problem by login in Servweb.
If i klick on the login button i get this errormessages:
error in SQL query, line: 30
What's wrong?
thanks