Hello again,
Still getting familiar with kamailio, and I'm wondering about the AUTH
route in the example configuration file. Here's a reducted-simplified
version of it for reference (from git master, without IP AUTH and comments):
______________________________________________
route[AUTH] {
if (is_method("REGISTER") || from_uri==myself) {
if (!auth_check("$fd", "subscriber", "1")) {
auth_challenge("$fd", "0");
exit;
}
if(!is_method("REGISTER|PUBLISH"))
consume_credentials();
}
if (from_uri!=myself && uri!=myself) {
sl_send_reply("403","Not relaying");
exit;
}
return;
}
______________________________________________
So the way I see it, what happens is the following:
* All REGISTERs will be challenged
* All SIP messages with kamailio's aliases in the "From" header URI will be
challenged
* All SIP messages with no reference to kamailio's aliases in both R-URI
and "From" header URI will be dropped
The question is, what about messages that do not enter either of the two
conditionals? For example, I expect the following to be very common:
* Method: INVITE
* R-URI: myself
* From: username@"UAC's local IP address" (not myself)
* To: myself
So in the example above, the auth route will return without either having
challenged or dropped the request, am I correct? This is because:
* For the challenge: Method is not REGISTER and "From URI" is not one of
kamailio host's aliases (cumulatively)
* For dropping after sending 403: "From URI" is not one of kamailio's
host's aliases (which calculates to true) but R-URI is "myself"
So I'm guessing we're expecting the challenge to come from elsewhere in
cases like the example above? Or is there something else I'm missing here?
Thanks!
BR,
George
Hi,
Can anyone comment on issue #1168?
https://github.com/kamailio/kamailio/issues/1168
Thanks,
Christoph
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Hello,
I tried to understand the concept of first hop in "replies"
In the documentation ;
For incoming SIP replies, it means that top Record-Route URI is 'myself' and source address is not matching it (to avoid detecting in case of local loops).
Note that it does not detect spirals, which can have the condition for replies true also in the case of additional SIP reply receival.
Here's an incoming reply received on 217.112.180.10.
Via: SIP/2.0/UDP 217.112.180.10;branch=z9hG4bK723f.db3510ce15ac9bb6467e115ecc2fe2a7.0.
Via: SIP/2.0/UDP 217.112.180.9;rport=5060;branch=z9hG4bK723f.741041093fec570744afcf44f2194581.0.
Via: SIP/2.0/UDP 217.110.38.229:5060;rport=5060;branch=z9hG4bK0eB7070489d338025ed.
Record-Route: <sip:217.112.180.9:5060;lr=on;ftag=7B84E4CBh3;did=4cf.c94;nat=yes>.
Record-Route: <sip:217.112.180.10:5060;lr=on;ftag=7B84E4CBh3;did=4cf.448>.
The Record-Route is 217.112.180.9 but the first proxy is 217.112.180.10.
Hi All,
Hoping for some help as a beginner. I need to configure a sip proxy server
so that it redirects all processes to a freeswitch server on another
network.
SIP UACs are on an isolated network and have only a connection to the sip
proxy. All I need the proxy to do is pass on all requests to the the
freeswitch UAS on a different network, as it will handle registering etc.
Is there a basic setup for this type of behaviour?
Many thanks,
Aaron
Hello i have problem with my kamailio sip server on rpi 3b i configured
tls with lets encrypt when i start kamailio with #!define WITH_AUTH its
always return 401 error but without tls working .
if i start kamailio without #!define WITH_AUTH its return
[500] reason [I'm terribly sorry, server error occurred (7/SL)] for
proxy [<sip:jurek.tk:5061;transport=tls>]
How to repair it ?
Hello. I found how to implement lua routimg logic with
cfgengine but didn't found how to include cimple route in lua at the cfg
file
Presume i have routing logic and want to include
my_super_custom_route written with Kemo on lua. How i need to set up my
config for it work with both engines?
Hi,
for a project I need the option to do a reverse lookup of an IP
address and use the value from it in a variable.
As far as I could see the ipops module only allows SRV and A record
lookups. Is there anything I missed?
And if there is nothing inside Kamailio, how could a good approach look like?
Thanks in advance,
Sebastian
Hello. I'm using almost default kamailio.cfg with nathelper, rtpproxy,
PSTN dialing and other default features. I have following scheme:
FXS gateway -> (NAT) -> Kamailio -> FreeSwitch
Voice calls work perfectly, no one-direction sound and so on.
But with fax machine there was a problem. In wireshark I saw UDPTL from
FS to Kamailio IP, and no UDPTL at all to FXS gw. I've guessed that
something's wrong with a re-INVITE request processing in Kamailio. I've
added this to WITHINDLG route:
else if ( is_method("INVITE") ) {
record_route();
}
And problem was magically solved, T.38 now works OK. Is this OK solution
or this may break something elsewhere?
Thanks.
How to make Kamailio write "normal" CDRs? acc cdr module writes start
time and end time in unixtime format, that is very uncomfortable.
Fnd stcond question - how to make Kamailio to write CDR in ordinary text
file?
Hello All,
I'm trying to code something on the on_reply part to do something if I receive a 183 ( Session in progress ) , followed immediately by a "180"
I tried to sleep 1 second the 183 and recheck if I received before a 180 but seems the "sleep" function sleeps everything ( also the received 180 )
Is a developper can confirm that ?
The goal is to detect if a 180 follow the 183.