Marc,
your configuration looks good except calling two times radius
accounting for BYEs (once in loose_routing and the second one in
route(3) right after proxying the request out (I assume that your
request will be detected as being generated from behind NAT due to
test flag 2 used in nat_uac_test(). I don't know whether this can
influence message processing, but in the case of failure I am not sure
that BYE will be sent out.
Can u check that once? Your radius server should tell u more about
that. Simplest way should be disabling accounting in both cases just
for tests.
Also, could be good if u would try to see whether the BYE follows
loose_route or goes out in if (!uri==myself) block. (xlog should help
u finding that out).
Cheers,
DanB
On 8/2/07, Marc LEURENT <lftsy(a)free.fr> wrote:
According to what I said, it is not a problem that the
phones answered with a SIP/2.0 481 CallLeg/Transaction Does Not Exist.
to SIP pings. What is important is that is will maintain the path through NAT
My only problem is with BYE requests that are not forwarded by OpenSER to the second
user
Is there something wrong in my openser.cfg???
#
# $Id: openser.cfg 1827 2007-03-12 15:22:53Z bogdan_iancu $
#
# simple quick-start config script
# Please refer to the Core CookBook at
http://www.openser.org/dokuwiki/doku.php
# for a explanation of possible statements, functions and parameters.
#
# ----------- global configuration parameters ------------------------
debug=7 # debug level (cmd line: -dddddddddd)
fork=no
log_stderror=yes # (cmd line: -E)
children=4
#alias=sd-7501.dedibox.fr
port=5060
#server_signature=yes
#tos=IPTOS_LOWDELAY
avp_aliases="day=i:101;time=i:102;can_uri=i:800;s_ip=i:801;billing_party=i:802;from_header=i:803;sip_proxy_ip=i:804"
#;pstnuser=i:805;pstnpassword=i:806:pstnrealm=i:807"
# ------------------ module loading ----------------------------------
#set module path
mpath="/usr/lib/openser/modules/"
# Uncomment this if you want to use SQL database
loadmodule "mysql.so"
loadmodule "sl.so" # Stateless Module
loadmodule "tm.so" # Transaction Module
loadmodule "rr.so" # Record-Route and Route Module
loadmodule "maxfwd.so" # Max-Forward processor Module
loadmodule "usrloc.so" # User Location Implementation Module
loadmodule "registrar.so" # SIP Registrat Implementation Module
(need usrloc)
loadmodule "textops.so" # Text Operation Module
loadmodule "mi_fifo.so" # FIFO transport layer implementation
for Management Interface
loadmodule "acc.so" # Accounting Module
loadmodule "avpops.so" # AVP Operation Module (user
preference)
loadmodule "uri.so" # Generic URI operation Module
loadmodule "auth.so" # Authentification Module
#loadmodule "auth_db.so" # Database-backend
Authentication mMdule
loadmodule "auth_radius.so" # RADIUS-backend Authentication Module
loadmodule "group_radius.so" # User-groups Module with
RADIUS-backend
#loadmodule "avp_radius.so" # RADIUS-backend for AVP loading
Module
#loadmodule "presence.so" # Presence server Module
#loadmodule "pua.so" # Common API for presence user agent
client
loadmodule "options.so" # OPTIONS server replier Module
loadmodule "xlog.so" # Advanced Logger Module
loadmodule "nathelper.so" # NAT Traversal Helper Module
#loadmodule "dispatcher.so" # Dispatcher (load-balancer) Module
loadmodule "uac.so" # User Agent Client
loadmodule "siptrace.so" # SipTrace module (storage of SIP
requests)
#loadmodule "exec.so" # Allows to start an external command
from a OpenSER script
# ----------------- setting module-specific parameters ---------------
# -- maxfwd params --
modparam("maxfwd", "max_limit", 10) # Default is 256 | 10 in the
functions
# -- sl params --
#modparam("sl", "enable_stats", 1)
# -- mi_fifo params --
modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo")
# -- usrloc params --
# Uncomment this if you want to use SQL database
modparam("usrloc", "db_mode", 1) # Write
instantaneously in the DB
modparam("usrloc", "db_url",
"mysql://openser:test@127.0.0.1/openser")
modparam("usrloc", "timer_interval", 10)
modparam("usrloc", "nat_bflag" , 3)
# -- rr params --
modparam("rr", "enable_full_lr", 1) # add value to ;lr
param to make some broken UAs happy
# -- siptrace params --
modparam("siptrace", "db_url",
"mysql://openser:test@127.0.0.1/openser")
modparam("siptrace", "table", "sip_trace") #
Default value "sip_trace"
modparam("siptrace", "trace_on", 1)
# -- registrar params --
modparam("registrar", "default_expires", 1800)
modparam("registrar", "received_avp", "$avp(i:42)")
# -- nathelper params --
modparam("nathelper", "rtpproxy_disable", 1)
modparam("nathelper", "sipping_bflag", 5)
modparam("nathelper", "natping_interval", 10)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_method", "OPTIONS")
modparam("nathelper", "received_avp", "$avp(i:42)")
# Same Value as the registrar module
modparam("nathelper", "sipping_from",
"sip:pinger@sd-7501.dedibox.fr")
# -- auth params --
#modparam("auth", "secret", "johndoessecretphrase")
# Default is random => don't set it
#modparam("auth", "nonce_expire", 300)
# Time before nounce expiration
modparam("auth_radius", "radius_config",
"/etc/radiusclient-ng/radiusclient.conf")
# -- group_radius params --
modparam("group_radius", "radius_config",
"/etc/radiusclient-ng/radiusclient.conf")
modparam("group_radius", "use_domain", 1) # username@domain
will be used for lookup
# -- avp_radius parameter --
#modparam("avp_radius", "radius_config",
"/etc/radiusclient-ng/radiusclient.conf")
# -- acc params (with radius )--
modparam("acc", "radius_config",
"/etc/radiusclient-ng/radiusclient.conf")
modparam("acc", "radius_flag", 1)
modparam("acc", "radius_missed_flag", 2)
modparam("acc", "early_media", 1)
modparam("acc", "report_cancels", 1)
#modparam("acc", "report_ack", 0)
modparam("acc", "detect_direction", 1)
#modparam("acc", "log_flag", 1) # number of the flag which
will be used to mark messages for accounting
#modparam("acc", "log_level", 1) # Set the reporting log
level
#modparam("acc", "log_missed_flag", 2) #
#modparam("acc", "failed_transaction_flag", 2)
modparam("acc", "service_type", 15) # Radius service type used
for accounting : 15 = (SIP)
#modparam("acc", "radius_extra",
"Sip-Src-IP=$si;Sip-Src-Port=$sp")
# ATTENTION: DO NOT PUT ; at the end of the radius_extra attribute
modparam("acc", "radius_extra", "Sip-Src-IP=$si;
Sip-Src-Port=$sp;
Canonical-URI=$avp(can_uri);
Billing-Party=$avp(billing_party);
SIP-Proxy-IP=$avp(sip_proxy_ip);
User-Agent=$ua
")
#Billing-Party=$avp(billing_party)
#From-Header=$hdr(from);
#User-Name=$fU;
#From-Header=$avp(from_header);
#Digest-Realm=$fd
#Sip-From-Tag=$avp(from_header);
#SIP-Method=$rm;
# ------------------------- request routing logic -------------------
# main routing logic
route{
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
};
if (msg:len >= 2048 ) {
sl_send_reply("513", "Message too big");
exit;
};
# NAT detection
route(2);
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol
if (!method=="REGISTER") {
record_route();
};
# subsequent messages withing a dialog should take the
# path determined by record-routing
if (loose_route()) { # mark routing logic in request
append_hf("P-hint: rr-enforced\r\n");
if(is_method("BYE")) { # log it all the time
acc_rad_request("200 ok");
acc_log_request("200 ok");
}
route(1);
};
# Set the acc flags
if(is_method("INVITE") && !has_totag()) {
xlog("L_INFO", "I AM SETTING THE FLAGS FOR RADIUS
\r\n");
$avp(can_uri) = $ru; # SIP Request's URI
$avp(billing_party) = $fu; # From URI
$avp(from_header) = $fU; # From URI username
$avp(sip_proxy_ip) = $Ri; # Received IP address
setflag(1); # radius_flag
setflag(2); # radius_missed_flag
};
# Functions when calling other domains
if (!uri==myself) {
# check if user is allowed to do voip calls to other domains
# if(is_method("INVITE|MESSAGE")) {
# if (radius_is_user_in("From", "voip")) {
# sl_send_reply("403", "Forbidden
VoIP");
# exit;
# };
# };
# mark routing logic in request
append_hf("P-hint: outbound\r\n");
route(1);
};
# if the request is for other domain use UsrLoc
# (in case, it does not work, use the following command
# with proper names and addresses in it)
if (uri==myself) {
if (method=="REGISTER") {
sip_trace();
xlog("L_INFO", "$fU IS TRYING TO REGISTER
\r\n");
if (!radius_www_authorize("sd-7501.dedibox.fr")) {
www_challenge("sd-7501.dedibox.fr",
"0"); # qop set to 1
xlog("L_INFO", "WWW_CHALLENGE of $si
FAILED \r\n");
exit;
};
#if (isflagset(5)) {
if (isbflagset(3)) {
#setflag(6);
# if you want OPTIONS natpings uncomment next
# setflag(7); # Deprecated
setbflag(5); # Set Flag for SIP PINGING
};
save("location");
xlog("L_INFO", "SAVE LOCATION OF $si \r\n");
exit;
};
if (!lookup("location")) {
# log to acc as missed call
acc_rad_request("404 Not Found");
acc_log_request("404 Not Found");
xlog("L_DBG", "ACC RADIUS: 404 NOT FOUND FOR $si
\r\n");
sl_send_reply("404", "Not Found");
exit;
};
append_hf("P-hint: usrloc applied\r\n");
};
route(1);
}
## Generic Forward
route[1] {
if (subst_uri('/(sip:.*);nat=yes/\1/')){
#setflag(6); # Deprecated, for version 1.1
setbflag(3);
};
#if (isflagset(5)||isflagset(6)) {
if (isbflagset(3)) {
route(3);
}
if (!t_relay()) {
sl_reply_error();
};
exit;
}
# NAT Detection
route[2]{
force_rport();
if (nat_uac_test("19")) {
xlog("!!!!!!!!! NAT UAC TEST 19 SUCEDEED \r\n");
if (method=="REGISTER") {
fix_nated_register();
} else {
fix_nated_contact();
};
#setflag(5); Deprecated
setbflag(3);
};
}
## Route for natted contact
route[3] {
if (is_method("BYE|CANCEL")) {
# Ajout Maison
acc_rad_request("200 ok");
acc_log_request("200 ok");
#unforce_rtp_proxy();
} else if (is_method("INVITE")){
#force_rtp_proxy();
t_on_failure("1");
};
#if (isflagset(5))
if (isbflagset(3)){
search_append('Contact:.*sip:[^>[:cntrl:]]*',
';nat=yes');
}
t_on_reply("1");
}
## Failure Route 1
failure_route[1] {
xlog("!!!!!!!!! ON FAILURE ROUTE \r\n");
#if (isflagset(6) || isflagset(5)) {
if (isbflagset(3)) {
#unforce_rtp_proxy();
}
}
## Reply route
onreply_route[1] {
xlog("!!!!!!!!! ON REPLY ROUTE \r\n");
#if ((isflagset(5) || isflagset(6)) &&
status=~"(183)|(2[0-9][0-9])") {
if (isbflagset(3) && status=~"(183)|(2[0-9][0-9])") {
#force_rtp_proxy();
}
search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');
#if (isflagset(6)) {
if (isbflagset(3)) {
xlog("!!!!!!!!! ON REPLY ROUTE / FIX NATED CONTACT \r\n");
fix_nated_contact();
}
exit;
}
Dan-Cristian Bogos a écrit :
Hi Marc,
it will help more if u will post full dialog from INVITE to BYE coming
in and out from the server. It is important to check whether openser
forwards the BYEs and if the end party listens on that port.
Cheers,
DanB
On 8/2/07, Marc LEURENT <lftsy(a)free.fr> wrote:
> STUN seems to be working
>
> In openserctl ul show, I have:
> AOR:: 103 Contact:: sip:103@82.127.0.79:1028;user=phone Q=
> AOR:: 101 Contact:: sip:101@82.127.0.79:1313;user=phone Q=
>
>
> And tcpdump -i eth0 -n port 5060 returns:
> 13:47:44.375374 IP 88.191.45.91.5060 > 82.127.0.79.1027: SIP, length: 241
> 13:47:44.375396 IP 88.191.45.91.5060 > 82.127.0.79.1312: SIP, length: 241
> 13:47:44.422471 IP 82.127.0.79.1027 > 88.191.45.91.5060: SIP, length: 276
> 13:47:44.426415 IP 82.127.0.79.1312 > 88.191.45.91.5060: SIP, length: 275
>
>
> And my ngrep returns
>
>
> #
> U 88.191.45.91:5060 -> 82.127.0.79:1027
> OPTIONS sip:82.127.0.79:1027 SIP/2.0.
> Via: SIP/2.0/UDP 88.191.45.91:5060;branch=0.
> From: sip:pinger@sd-7501.dedibox.fr;tag=7ad21f92.
> To: sip:82.127.0.79:1027.
> Call-ID: 90f2eea1-5c41f342-a91(a)88.191.45.91.
> CSeq: 1 OPTIONS.
> Content-Length: 0.
> .
>
> #
> U 88.191.45.91:5060 -> 82.127.0.79:1312
> OPTIONS sip:82.127.0.79:1312 SIP/2.0.
> Via: SIP/2.0/UDP 88.191.45.91:5060;branch=0.
> From: sip:pinger@sd-7501.dedibox.fr;tag=8ad21f92.
> To: sip:82.127.0.79:1312.
> Call-ID: 90f2eea1-6c41f342-a91(a)88.191.45.91.
> CSeq: 1 OPTIONS.
> Content-Length: 0.
> .
>
> #
> U 82.127.0.79:1027 -> 88.191.45.91:5060
> SIP/2.0 481 CallLeg/Transaction Does Not Exist.
> Via: SIP/2.0/UDP 88.191.45.91:5060;branch=0.
> From: <sip:pinger@sd-7501.dedibox.fr>;tag=7ad21f92.
> To: <sip:82.127.0.79:1027>;tag=c0a80101-1db9be2.
> Call-ID: 90f2eea1-5c41f342-a91(a)88.191.45.91.
> CSeq: 1 OPTIONS.
> Content-Length: 0.
> .
>
> #
> U 82.127.0.79:1312 -> 88.191.45.91:5060
> SIP/2.0 481 CallLeg/Transaction Does Not Exist.
> Via: SIP/2.0/UDP 88.191.45.91:5060;branch=0.
> From: <sip:pinger@sd-7501.dedibox.fr>;tag=8ad21f92.
> To: <sip:82.127.0.79:1312>;tag=c0a80101-573ff0.
> Call-ID: 90f2eea1-6c41f342-a91(a)88.191.45.91.
> CSeq: 1 OPTIONS.
> Content-Length: 0.
>
>
>
>
>
>
> Iñaki Baz Castillo a écrit :
>> El Thursday 02 August 2007 12:25:07 Marc LEURENT escribió:
>>> #
>>> U 82.127.0.79:1312 -> 88.191.45.91:5060
>>> BYE sip:103@82.127.0.79:1027 SIP/2.0.
>>> Via: SIP/2.0/UDP 82.127.0.79:1313;branch=z9hG4bK8030359792092547043.
>>> From:
"101"<sip:101@sip.leurent.eu:5060;user=phone>;tag=c0a80101-4c5eed.
>>> To: <sip:103@sip.leurent.eu:5060;user=phone>;tag=c0a80101-1d0bb0d.
>>> Call-ID: 66464a0-c0a80101-0-1f(a)192.168.95.4.
>>> CSeq: 2 BYE.
>>> Max-Forwards: 70.
>>> Route: <sip:88.191.45.91:5060;lr=on;ftag=c0a80101-4c5eed>.
>>> User-Agent: THOMSON ST2030 hw0 fw1.50 00-0E-50-4E-AF-AE.
>>> Content-Length: 0.
>> What more is after this message?
>> Did you try tcpdump to monitorize to with IP:port are the messages sent?
>>
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