Hello List,
I've been using SER with RADIUS successfully now for a few months and am
very pleased with the result. It's used for authenticating users accessing
our gateways.
I know have a new requirement to extend this to provide authentication for
remote domains.
The setup being as follows.
We've got SER running with FreeRADIUS, then at the remote sites we will have
the same plus Asterisk that is to act as a local gateway.
I've configured the local FreeRADIUS instance to proxy the requests for the
remote SIP domains to the remote RADIUS server. Unfortunately this doesn't
work and I'm not sure why.
The SUA gets asked by the remote SIP proxy to authenticate, it then forwards
the INVITE to the local SER instance which then gets the LOCAL RADIUS to do
another auth. This doesn't work. However if I disable the local auth and
leave the remote auth enabled it works fine.
Has anyone successfully managed to get proxied radius auth to work?
My other question is to do with getting SER to send the INVITE to a
different gateway if the primary one is at capacity/out of action? Is there
an example of this sort of config?
Kind Regards,
Alan
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Telco Electronics Limited
6-8 Oxford Court
Brackley
Northants
NN13 7XY
Tel 07000 701999
Fax 07000 701777
It is a ZyXEL Phone connected to a ZyXEL Wlan Router.
Sending INVITE
INVITE => SER => ASTERISK
ASTERISK send "Session Progress" everything is fine:
ASTERISK sends "OK"
Session starts. I can hear my Voice. After 6 Seconds
Session breaks up.
Something seems to be strange with the ACK the ZyXEL
Phones sends:
PHONE => PROXY
ACK sip:0xxxxx@21.1.7.9 SIP/2.0
Via:SIP/2.0/UDP 21.28.17.18:5060;branch=z9hGddbkb8a7ca191bf4fd
From:Zyxel<sip:1111111@domain.de;user=phone>;tag=C2E2D6CF845EA11AEBC2
To:<sip:0xxxxxxxxx@domain.de;user=phone>;tag=as4f89971f
Call-ID:15482-E655-54C3-B488-1D32D8DF7653@21.28.17.18
CSeq:1 ACK
User-Agent: ZyXEL P2000W VoIP Wi-Fi Phone
Route:<sip:49xxxxx@21.1.7.8;ftag=C2E2D6CF845EA11AEBC2;lr=on>,<sip:492405690929@21.28.17.18:5060>
When loggin with debug = 10 , i do not see some errors in syslog.
The Gateway then sends 6 more "OK" and the terminate by sending "BYE".
The Phone answers with "200 OK" .. ?
Regards,
Markus
Hi all,
I solved my problem SEND IM, missed calls, Voicemail.
I do hope that could help somebody !!!
However when I call an offline user1 he received a missed calls from
user2 but user1 don't receive email notification in his mailbox although
voicemail system answer the user2's call???
Harry
---------------------------------------------------------
# native SIP destinations are handled using our USRLOC DB
lookup("aliases");
if (!lookup("location")) {
if (! t_newtran()) {
sl_reply_error();
break;
};
t_reply("100","Trying -- just wait a minute !");
if(method=="INVITE"){
acc_log_request("404 missed call\n");
acc_db_request("404 missed call",
"missed_calls");
setflag(3);
if(!vm("/tmp/am_fifo","voicemail")){
log("could not contact the answer machine\n");
t_reply("500","could not contact the answer machine");
};
if(method=="BYE" || method=="CANCEL"){
if(!vm("/tmp/am_fifo","bye")){
log("could not contact the answer machine\n");
t_reply("500","could not contact the answer machine");
};
};
break;
};
# we do not care about anything else but MESSAGEs
if (!method=="MESSAGE") {
if (!t_reply("404", "Not found")) {
sl_reply_error();
};
break;
};
log("MESSAGE received -> storing using MSILO\n");
# MSILO - storing as offline message
if (m_store("0")) {
log("MSILO: offline message stored\n");
if (!t_reply("202", "Accepted")) {
sl_reply_error();
};
}else{
log("MSILO: offline message NOT stored\n");
if (!t_reply("503", "Service Unavailable")) {
sl_reply_error();
};
};
break;
};
# if the downstream UA does not support MESSAGE requests
# go to failure_route[1]
t_on_failure("1");
t_relay();
break;
};
# forward to current uri now; use stateful forwarding; that
# works reliably even if we forward from TCP to UDP
if (!t_relay()) {
sl_reply_error();
};
}
failure_route[1] {
# forwarding failed -- check if the request was a MESSAGE
if (!method=="MESSAGE")
{
break;
};
log(1,"MSILO: the downstream UA does not support MESSAGE requests
...\n");
# we have changed the R-URI with the contact address -- ignore it now
if (m_store("1"))
{
log("MSILO: offline message stored\n");
t_reply("202", "Accepted for delivery");
}else{
log("MSILO: offline message NOT stored\n");
t_reply("503", "Service Unavailable");
};
}
I have added a user with some domain as ip address and second with domain as
domain name. They are not able to contact each other.....
and how can i enable multiple domain.....
regards
_________________________________________________________________
Need quick cash? http://go.msnserver.com/IN/46923.asp Click here !
I have version mysql-3.23.58-1.9 installed but seem to remember a thread
about a later version being more compatible with some of the functions in
SER.
Thanks in advance..
Regards,
Steve
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Hi,
I have a few questions about multiple domains and call
from/to pstn. I don't want to change the ser.cfg
script everytime a new domain is added. How can I
support the following features?
1. Every domain has their own 3 or 4 digit dial plan.
When they call PSTN, their caller id should be the
either a prefix plus their internal number or a
generic receptionist number, but not the 3/4 digit
internal number. Different domains have different
prefix. Is there a way to do this without changing the
script for every domain?
2. Should interdomain calls be prohibited? Is it
preferred to send the call to PSTN and hairpin back? I
know it consumes more resources but would it provide
better privacy? If that's the case, how to prevent
interdomain calls, i.e. how to verify from and to URI
have the same host parts?
3. When PSTN calls come in, should I use prefix2domain
in pdt module to convert the PSTN number to a local
domain number? Any pitfalls?
This list has been great! Thanks for all your help.
Best Regards,
Richard
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