Hi,
There are lots of parameters controlling the creation of nonce values on
a server, and I'm curious if there is a way to kind of "sync" them
between servers.
The use case would be to have a UA send for example its registration to
Proxy1. Proxy1 would challenge it, UA will send the registration again,
this time with credentials. Proxy1 would look up the user based on
$au/$ar in the subscriber table, and if it's not found, will look up the
responsible proxy from another table (with key being $au@$ar), forward
it to Proxy2, which then would be able authenticate the user.
The reason for this is that the auth credentials are unique across all
servers and reliably identify a user, whereas for example From could be
something else (e.g. in case of an IP-PBX sending a CLI in the
From-userpart).
Challenging the user on the second proxy again would theoretically be
possible, but if the UA gets a 401 twice (once from Proxy1, once from
Proxy2), it'll most likely pop up a password form for soft-clients, so I
want to avoid that.
Any ideas how to accomplish that?
Andreas
Hi everyone
I'm trying to integrate Asterisk with Kamailio for voicemail.
I tried to follow this tutorial:
http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb
BUT:
- I had to adapt it because I use LDAP authentication with Kamailio
- I had problems with Asterisk 10.7 (problems with chan_sip module
crashing) so I've installed Asterisk 11 on another VM
- we have high-availability with 2 Kamailio servers, with Kamailio
listening on TCP (constraint from our SSL gateway in front of Kamailio) on
a "virtual IP" (created by keep-alive): this VIP is not visible with
ifconfig, but you can see it with the command "ip addr sh eth0"
For now, we use Linphone on Windows as SIP clients to test.
If I don't define WITH_ASTERISK, calls work, I can call someone(a)domain.tld
However, if I define WITH_ASTERISK, calls fail (even with destination
registered and available) and I have these errors in the logfile:
Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [ut.h:333]:
no corresponding socket for af 2
Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm
[t_fwd.c:424]: ERROR: can't fwd to af 2, proto 1 (no corresponding
listening socket)
Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm
[t_fwd.c:1530]: ERROR: t_forward_nonack: failure to add branches
Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: INFO: <script>: reply
error
Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: sl
[sl_funcs.c:371]: ERROR: sl_reply_error used: I'm terribly sorry, server
error occurred (1/SL)
It seems to happen on the if (!t_relay()) line in ROUTE[RELAY]
192.168.14.25 is the real IP of the Kamailio server,
192.168.14.24 is the VIP of the Kamailio "cluster"
192.168.14.28 is the IP of the Mysql server
192.168.14.32 is the IP of the Asterisk server
I can't find why the relay doesn't work. I've tried to bypass the VIP and
have Kamailio listen on the real IP, but it still doesn't work: I don't
seem to have the same errors as above, but I don't see any traffic between
Kamailio and Asterisk.
What could be the problem? Thanks for your help
Christophe
Below is my kamailio.cfg:
#!WITH_DEBUG
#!KAMAILIO
#
# Kamailio (OpenSER) SIP Server v3.2 - default configuration script
# - web: http://www.kamailio.org
# - git: http://sip-router.org
#
# Direct your questions about this file to: <sr-users(a)lists.sip-router.org>
#
# Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php
# for an explanation of possible statements, functions and parameters.
#
# Several features can be enabled using '#!define WITH_FEATURE' directives:
#
# *** To run in debug mode:
# - define WITH_DEBUG
#
# *** To enable mysql:
# - define WITH_MYSQL
#
# *** To enable authentication execute:
# - enable mysql
# - define WITH_AUTH
# - add users using 'kamctl'
#
# *** To enable IP authentication execute:
# - enable mysql
# - enable authentication
# - define WITH_IPAUTH
# - add IP addresses with group id '1' to 'address' table
#
# *** To enable persistent user location execute:
# - enable mysql
# - define WITH_USRLOCDB
#
# *** To enable presence server execute:
# - enable mysql
# - define WITH_PRESENCE
#
# *** To enable nat traversal execute:
# - define WITH_NAT
# - install RTPProxy: http://www.rtpproxy.org
# - start RTPProxy:
# rtpproxy -l _your_public_ip_ -s udp:localhost:7722
#
###!define WITH_NAT
# *** To enable PSTN gateway routing execute:
# - define WITH_PSTN
# - set the value of pstn.gw_ip
# - check route[PSTN] for regexp routing condition
#
# *** To enable database aliases lookup execute:
# - enable mysql
# - define WITH_ALIASDB
#
# *** To enable speed dial lookup execute:
# - enable mysql
# - define WITH_SPEEDDIAL
#
# *** To enable multi-domain support execute:
# - enable mysql
# - define WITH_MULTIDOMAIN
#
# *** To enable TLS support execute:
# - adjust CFGDIR/tls.cfg as needed
# - define WITH_TLS
#
# *** To enable XMLRPC support execute:
# - define WITH_XMLRPC
# - adjust route[XMLRPC] for access policy
#
# *** To enable anti-flood detection execute:
# - adjust pike and htable=>ipban settings as needed (default is
# block if more than 16 requests in 2 seconds and ban for 300 seconds)
# - define WITH_ANTIFLOOD
#
# *** To block 3XX redirect replies execute:
# - define WITH_BLOCK3XX
#
# *** To enable VoiceMail routing execute:
# - define WITH_VOICEMAIL
# - set the value of voicemail.srv_ip
# - adjust the value of voicemail.srv_port
#
# *** To enhance accounting execute:
# - enable mysql
# - define WITH_ACCDB
# - add following columns to database
#!ifdef ACCDB_COMMENT
ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT '';
ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT '';
ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default '';
ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT '';
ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT '';
ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT '';
ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT
'';
ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL
DEFAULT '';
ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default
'';
ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL
DEFAULT '';
ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT
'';
ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL
DEFAULT '';
#!endif
###!define WITH_ASTERISK
###!define WITH_VOICEMAIL
#!define WITH_LDAP
#!define WITH_AUTH
#!define WITH_MYSQL
####### Defined Values #########
# *** Value defines - IDs used later in config
#!ifdef WITH_MYSQL
# - database URL - used to connect to database server by modules such
# as: auth_db, acc, usrloc, a.s.o.
#!define DBURL "mysql://openserrw:openserrw@192.168.14.28/openser"
#!ifdef WITH_ASTERISK
#!define DBASTURL "mysql://asterisk:asteriskpwd@192.168.14.28/asterisk"
#!endif
#!endif
#!ifdef WITH_MULTIDOMAIN
# - the value for 'use_domain' parameters
#!define MULTIDOMAIN 1
#!else
#!define MULTIDOMAIN 0
#!endif
# - flags
# FLT_ - per transaction (message) flags
# FLB_ - per branch flags
#!define FLT_ACC 1
#!define FLT_ACCMISSED 2
#!define FLT_ACCFAILED 3
#!define FLT_NATS 5
#!define FLB_NATB 6
#!define FLB_NATSIPPING 7
####### Global Parameters #########
#!ifdef WITH_DEBUG
debug=4
log_stderror=no
#!else
debug=2
log_stderror=no
#!endif
memdbg=5
memlog=5
log_facility=LOG_LOCAL6
fork=yes
children=4
/* uncomment the next line to disable TCP (default on) */
#disable_tcp=yes
/* uncomment the next line to disable the auto discovery of local aliases
based on reverse DNS on IPs (default on) */
auto_aliases=no
/* add local domain aliases */
alias="mydomain.corp"
/* uncomment and configure the following line if you want Kamailio to
bind on a specific interface/port/proto (default bind on all available)
*/
listen=tcp:192.168.14.24:5060
#listen=tcp:192.168.14.25:5060
/* port to listen to
* - can be specified more than once if needed to listen on many ports */
#port=5060
#!ifdef WITH_TLS
enable_tls=yes
#!endif
# life time of TCP connection when there is no traffic
# - a bit higher than registration expires to cope with UA behind NAT
tcp_connection_lifetime=3605
####### Custom Parameters #########
# These parameters can be modified runtime via RPC interface
# - see the documentation of 'cfg_rpc' module.
#
# Format: group.id = value 'desc' description
# Access: $sel(cfg_get.group.id) or @cfg_get.group.id
#
#!ifdef WITH_PSTN
# PSTN GW Routing
#
# - pstn.gw_ip: valid IP or hostname as string value, example:
# pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address"
#
# - by default is empty to avoid misrouting
pstn.gw_ip = "" desc "PSTN GW Address"
#!endif
#!ifdef WITH_VOICEMAIL
# VoiceMail Routing on offline, busy or no answer
#
# - by default Voicemail server IP is empty to avoid misrouting
voicemail.srv_ip = "192.168.14.32" desc "VoiceMail IP Address"
voicemail.srv_port = "5060" desc "VoiceMail Port"
#!endif
#!ifdef WITH_ASTERISK
asterisk.bindip = "192.168.14.32" desc "Asterisk IP Address"
asterisk.bindport = "5060" desc "Asterisk Port"
kamailio.bindip = "192.168.14.24" desc "Kamailio IP Address"
kamailio.bindport = "5060" desc "Kamailio Port"
#!endif
####### Modules Section ########
# set paths to location of modules (to sources or installation folders)
#!ifdef WITH_SRCPATH
mpath="modules_k:modules"
#!else
mpath="/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/"
#!endif
#!ifdef WITH_MYSQL
loadmodule "db_mysql.so"
#!endif
loadmodule "mi_fifo.so"
loadmodule "kex.so"
loadmodule "tm.so"
loadmodule "tmx.so"
loadmodule "sl.so"
loadmodule "rr.so"
loadmodule "pv.so"
loadmodule "maxfwd.so"
loadmodule "usrloc.so"
loadmodule "registrar.so"
loadmodule "textops.so"
loadmodule "siputils.so"
loadmodule "xlog.so"
# loadmodule "sanity.so"
loadmodule "ctl.so"
loadmodule "cfg_rpc.so"
loadmodule "mi_rpc.so"
loadmodule "acc.so"
#!ifdef WITH_AUTH
loadmodule "auth.so"
loadmodule "auth_db.so" #chris
loadmodule "ldap.so"
modparam ("ldap", "config_file", "/etc/kamailio/ldap.cfg")
#!ifdef WITH_IPAUTH
loadmodule "permissions.so"
#!endif
#!endif
#!ifdef WITH_ALIASDB
loadmodule "alias_db.so"
#!endif
#!ifdef WITH_SPEEDDIAL
loadmodule "speeddial.so"
#!endif
#!ifdef WITH_MULTIDOMAIN
loadmodule "domain.so"
#!endif
#!ifdef WITH_PRESENCE
loadmodule "presence.so"
loadmodule "presence_xml.so"
#!endif
#!ifdef WITH_NAT
loadmodule "nathelper.so"
loadmodule "rtpproxy.so"
#!endif
#!ifdef WITH_TLS
loadmodule "tls.so"
#!endif
#!ifdef WITH_ANTIFLOOD
loadmodule "htable.so"
loadmodule "pike.so"
#!endif
#!ifdef WITH_XMLRPC
loadmodule "xmlrpc.so"
#!endif
#!ifdef WITH_DEBUG
loadmodule "debugger.so"
#!endif
#!ifdef WITH_ASTERISK
loadmodule "uac.so"
#!endif
# ----------------- setting module-specific parameters ---------------
# ----- mi_fifo params -----
#modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")
modparam("mi_fifo", "fifo_name", "/tmp/kamailio_tmp")
# ----- tm params -----
# auto-discard branches from previous serial forking leg
modparam("tm", "failure_reply_mode", 3)
# default retransmission timeout: 30sec
modparam("tm", "fr_timer", 30000)
# default invite retransmission timeout after 1xx: 120sec
modparam("tm", "fr_inv_timer", 120000)
# suppress the check for the CSEQ method
# modparam("sanity", "default_checks", 967)
# ----- rr params -----
# add value to ;lr param to cope with most of the UAs
modparam("rr", "enable_full_lr", 1)
# do not append from tag to the RR (no need for this script)
#edit asterisk
#!ifdef WITH_ASTERISK
modparam("rr", "append_fromtag", 1)
#!else
modparam("rr", "append_fromtag", 0)
#!endif
# ----- registrar params -----
modparam("registrar", "method_filtering", 1)
/* uncomment the next line to disable parallel forking via location */
# modparam("registrar", "append_branches", 0)
/* uncomment the next line not to allow more than 10 contacts per AOR */
modparam("registrar", "max_contacts", 256)
# max value for expires of registrations
modparam("registrar", "max_expires", 3600)
# ----- acc params -----
/* what special events should be accounted ? */
modparam("acc", "early_media", 0)
modparam("acc", "report_ack", 0)
modparam("acc", "report_cancels", 0)
/* by default ww do not adjust the direct of the sequential requests.
if you enable this parameter, be sure the enable "append_fromtag"
in "rr" module */
modparam("acc", "detect_direction", 0)
/* account triggers (flags) */
modparam("acc", "log_flag", FLT_ACC)
modparam("acc", "log_missed_flag", FLT_ACCMISSED)
modparam("acc", "log_extra",
"src_user=$fU;src_domain=$fd;src_ip=$si;"
"dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
modparam("acc", "failed_transaction_flag", FLT_ACCFAILED)
/* enhanced DB accounting */
#!ifdef WITH_ACCDB
modparam("acc", "db_flag", FLT_ACC)
modparam("acc", "db_missed_flag", FLT_ACCMISSED)
modparam("acc", "db_url", DBURL)
modparam("acc", "db_extra",
"src_user=$fU;src_domain=$fd;src_ip=$si;"
"dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
#!endif
# ----- usrloc params -----
/* enable DB persistency for location entries */
#!ifdef WITH_USRLOCDB
modparam("usrloc", "db_url", DBURL)
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "use_domain", MULTIDOMAIN)
#!endif
#chris commented out this part
# ----- auth_db params -----
#!ifdef WITH_AUTH
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "load_credentials", "")
#!ifdef WITH_ASTERISK
modparam("auth_db", "user_column", "username")
modparam("auth_db", "password_column", "sippasswd")
modparam("auth_db", "db_url", DBASTURL)
modparam("auth_db", "version_table", 0)
#!else
modparam("auth_db", "db_url", DBURL)
modparam("auth_db", "password_column", "password")
modparam("auth_db", "use_domain", MULTIDOMAIN)
#!endif
# ----- permissions params -----
#!ifdef WITH_IPAUTH
modparam("permissions", "db_url", DBURL)
modparam("permissions", "db_mode", 1)
#!endif
#!endif
# ----- alias_db params -----
#!ifdef WITH_ALIASDB
modparam("alias_db", "db_url", DBURL)
modparam("alias_db", "use_domain", MULTIDOMAIN)
#!endif
# ----- speedial params -----
#!ifdef WITH_SPEEDDIAL
modparam("speeddial", "db_url", DBURL)
modparam("speeddial", "use_domain", MULTIDOMAIN)
#!endif
# ----- domain params -----
#!ifdef WITH_MULTIDOMAIN
modparam("domain", "db_url", DBURL)
# use caching
modparam("domain", "db_mode", 1)
# register callback to match myself condition with domains list
modparam("domain", "register_myself", 1)
#!endif
#!ifdef WITH_PRESENCE
# ----- presence params -----
modparam("presence", "db_url", DBURL)
# ----- presence_xml params -----
modparam("presence_xml", "db_url", DBURL)
modparam("presence_xml", "force_active", 1)
#!endif
#!ifdef WITH_NAT
# ----- rtpproxy params -----
modparam("rtpproxy", "rtpproxy_sock", "udp:192.168.14.25:22222")
# ----- nathelper params -----
modparam("nathelper", "natping_interval", 30)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_bflag", FLB_NATSIPPING)
modparam("nathelper", "sipping_from", "sip:pinger@teopad-toip.corp")
# params needed for NAT traversal in other modules
modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)")
modparam("usrloc", "nat_bflag", FLB_NATB)
#!endif
#!ifdef WITH_TLS
# ----- tls params -----
modparam("tls", "config", "/etc/kamailio/tls.cfg")
#!endif
#!ifdef WITH_ANTIFLOOD
# ----- pike params -----
modparam("pike", "sampling_time_unit", 2)
modparam("pike", "reqs_density_per_unit", 16)
modparam("pike", "remove_latency", 4)
# ----- htable params -----
# ip ban htable with autoexpire after 5 minutes
modparam("htable", "htable", "ipban=>size=8;autoexpire=300;")
#!endif
#!ifdef WITH_XMLRPC
# ----- xmlrpc params -----
modparam("xmlrpc", "route", "XMLRPC");
modparam("xmlrpc", "url_match", "^/RPC")
#!endif
#!ifdef WITH_DEBUG
# ----- debugger params -----
modparam("debugger", "cfgtrace", 1)
#!endif
####### Routing Logic ########
# Main SIP request routing logic
# - processing of any incoming SIP request starts with this route
# - note: this is the same as route { ... }
request_route {
# per request initial checks
route(REQINIT);
xlog("L_INFO","apres REQINIT");
# NAT detection
route(NATDETECT);
xlog("L_INFO","apres NATDETECT");
# handle requests within SIP dialogs
route(WITHINDLG);
xlog("L_INFO","apres WITHINDLG");
### only initial requests (no To tag)
# CANCEL processing
if (is_method("CANCEL"))
{
if (t_check_trans())
t_relay();
exit;
}
t_check_trans();
xlog("L_INFO","apres t_check_trans");
# authentication
route(AUTH);
xlog("L_INFO","apres AUTH");
# record routing for dialog forming requests (in case they are routed)
# - remove preloaded route headers
remove_hf("Route");
if (is_method("INVITE|SUBSCRIBE"))
record_route();
xlog("L_INFO","apres RECORD ROUTE");
# account only INVITEs
if (is_method("INVITE"))
{
setflag(FLT_ACC); # do accounting
}
xlog("L_INFO","apres INVITE");
# dispatch requests to foreign domains
route(SIPOUT);
xlog("L_INFO","apres SIPOUT");
### requests for my local domains
# handle presence related requests
route(PRESENCE);
xlog("L_INFO","apres PRESENCE");
# handle registrations
route(REGISTRAR);
xlog("L_INFO","apres REGISTRAR");
if ($rU==$null)
{
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}
# dispatch destinations to PSTN
route(PSTN);
xlog("L_INFO","apres PSTN");
# user location service
route(LOCATION);
xlog("L_INFO","apres LOCATION");
route(RELAY);
xlog("L_INFO","apres RELAY");
}
route[RELAY] {
# enable additional event routes for forwarded requests
# - serial forking, RTP relaying handling, a.s.o.
xlog("L_INFO","Dans route relay");
if (is_method("INVITE|SUBSCRIBE")) {
xlog("L_INFO","avant manage branch");
t_on_branch("MANAGE_BRANCH");
xlog("L_INFO","avant manage reply");
t_on_reply("MANAGE_REPLY");
}
if (is_method("INVITE")) {
xlog("L_INFO","avant manage failure");
t_on_failure("MANAGE_FAILURE");
}
if (!t_relay()) {
xlog("L_INFO","reply error");
sl_reply_error();
}
exit;
}
# Per SIP request initial checks
route[REQINIT] {
#!ifdef WITH_ANTIFLOOD
# flood dection from same IP and traffic ban for a while
# be sure you exclude checking trusted peers, such as pstn gateways
# - local host excluded (e.g., loop to self)
if(src_ip!=myself)
{
if($sht(ipban=>$si)!=$null)
{
# ip is already blocked
xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n");
exit;
}
if (!pike_check_req())
{
xlog("L_ALERT","ALERT: pike blocking $rm from $fu
(IP:$si:$sp)\n");
$sht(ipban=>$si) = 1;
exit;
}
}
#!endif
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
}
# if(!sanity_check("1511", "7"))
# {
# xlog("Malformed SIP message from $si:$sp\n");
# exit;
# }
}
# Handle requests within SIP dialogs
route[WITHINDLG] {
xlog("L_INFO","Dans WITHINDLG");
if (has_totag()) {
xlog("L_INFO","dans has totag");
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
xlog("L_INFO","looseroute");
if (is_method("BYE")) {
setflag(FLT_ACC); # do accounting ...
setflag(FLT_ACCFAILED); # ... even if the transaction fails
}
if ( is_method("ACK") ) {
xlog("L_INFO","ack");
# ACK is forwarded statelessy
route(NATMANAGE);
}
xlog("L_INFO","relay");
route(RELAY);
} else {
xlog("L_INFO","else");
if (is_method("SUBSCRIBE") && uri == myself) {
# in-dialog subscribe requests
xlog("L_INFO","subscribe avant presence");
route(PRESENCE);
xlog ("L_INFO","apres presence");
exit;
}
if ( is_method("ACK") ) {
xlog("L_INFO","else ack");
if ( t_check_trans() ) {
# no loose-route, but stateful ACK;
# must be an ACK after a 487
# or e.g. 404 from upstream server
xlog("L_INFO","else ack avant relay");
t_relay();
xlog("L_INFO","else ack apres relay");
exit;
} else {
# ACK without matching transaction ... ignore and
discard
xlog("L_INFO","else final");
exit;
}
}
sl_send_reply("404","Not here");
}
exit;
}
}
# Handle SIP registrations
route[REGISTRAR] {
if (is_method("REGISTER"))
{
if(isflagset(FLT_NATS))
{
setbflag(FLB_NATB);
# uncomment next line to do SIP NAT pinging
setbflag(FLB_NATSIPPING);
}
if (!save("location"))
sl_reply_error();
#edit asterisk
#!ifdef WITH_ASTERISK
xlog ("L_INFO","avant regfwd dans registrar");
route(REGFWD);
xlog ("L_INFO","apres regfwd dans registrar");
#!endif
exit;
}
}
# USER location service
route[LOCATION] {
#!ifdef WITH_SPEEDIAL
# search for short dialing - 2-digit extension
if($rU=~"^[0-9][0-9]$")
if(sd_lookup("speed_dial"))
route(SIPOUT);
#!endif
#!ifdef WITH_ALIASDB
# search in DB-based aliases
if(alias_db_lookup("dbaliases"))
route(SIPOUT);
#!endif
#edit asterisk
#!ifdef WITH_ASTERISK
if(is_method("INVITE") && (!route(FROMASTERISK))) {
#if new call from out there - send to Asterisk
# - non-INVITE requests are routed directly by Kamailio
# - traffic from Asterisk is router also directly by Kamailio
xlog ("L_INFO", "avant toasterisk dans location");
route(TOASTERISK);
xlog ("L_INFO", "apres toasterisk dans location");
exit;
}
#!endif
$avp(oexten) = $rU;
if (!lookup("location")) {
$var(rc) = $rc;
route(TOVOICEMAIL);
t_newtran();
switch ($var(rc)) {
case -1:
case -3:
send_reply("404", "Not Found");
exit;
case -2:
send_reply("405", "Method Not Allowed");
exit;
}
}
# when routing via usrloc, log the missed calls also
if (is_method("INVITE"))
{
setflag(FLT_ACCMISSED);
}
}
# Presence server route
route[PRESENCE] {
if(!is_method("PUBLISH|SUBSCRIBE"))
return;
#!ifdef WITH_PRESENCE
if (!t_newtran())
{
sl_reply_error();
exit;
};
if(is_method("PUBLISH"))
{
handle_publish();
t_release();
}
else
if( is_method("SUBSCRIBE"))
{
handle_subscribe();
t_release();
}
exit;
#!endif
# if presence enabled, this part will not be executed
if (is_method("PUBLISH") || $rU==$null)
{
sl_send_reply("404", "Not here");
exit;
}
return;
}
# Authentication route
route[AUTH] {
#!ifdef WITH_AUTH
#edit asterisk
#!ifdef WITH_ASTERISK
#do not auth traffic from Asterisk: trusted!
xlog ("L_INFO", "avant if route fromasterisk");
if(route(FROMASTERISK))
return;
#!endif
if (is_method("REGISTER"))
# {
# # authenticate the REGISTER requests (uncomment to enable auth)
# if (!www_authorize("$td", "subscriber"))
# {
# www_challenge("$td", "0");
# exit;
# }
#
# if ($au!=$tU)
# {
# sl_send_reply("403","Forbidden auth ID");
# exit;
# }
{
#edit asterisk
##!ifdef WITH_ASTERISK
# xlog ("L_INFO", "dans auth / authcheck sipusers");
# if (!auth_check("$fd","sipusers","1"))
##!else
if(is_present_hf("Authorization"))
##!endif
{
# ldap search
if
(!ldap_search("ldap://sipaccounts/OU=SIP,OU=Utilisateurs,DC=teopad-toip,DC=corp?teopad-Sip-Username,teopadSipPassword?one?(teopad-Sip-Username=$fU)"))
# if
(!ldap_search("ldap://sipaccounts/OU=SIP,OU=Utilisateurs,DC=teopad-toip,DC=corp?sAMAccountName,?one?(sAMAccountName=$fU)"))
{
switch ($retcode)
{
case -1:
# no LDAP entry found
sl_send_reply("404", "User Not Found");
xlog("L_INFO", "ldap_search: NO found [$retcode]
entries for (uid=$fU)");
exit;
case -2:
# internal error
sl_send_reply("500", "Internal server error");
exit;
default:
exit;
}
}
ldap_result("teopad-Sip-Username/$avp(s:username)");
ldap_result("teopadSipPassword/$avp(s:password)");
xlog("L_INFO", "ldap_search: found [$retcode] entries for
(uid=$fU)");
if(!pv_www_authenticate("$td", "$avp(s:password)", "0")) {
xlog ("L_INFO", "ldap pv_authenticate failed") ;
www_challenge("$td","1");
exit;
}
save("location");
sl_send_reply("200", "ok");
xlog ("L_INFO", "ldap pv_authenticate ok") ;
exit;
} else {
www_challenge("$td","1");
exit;
}
} else {
#!ifdef WITH_IPAUTH
if(allow_source_address())
{
# source IP allowed
return;
}
#!endif
# # authenticate if from local subscriber
if (from_uri==myself)
{
# if (!proxy_authorize("$fd", "subscriber")) {
# proxy_challenge("$fd", "0");
# exit;
# }
if (is_method("PUBLISH"))
{
xlog ("L_INFO", "au = $au") ;
xlog ("L_INFO", "fU = $fU") ;
xlog ("L_INFO", "tU = $tU") ;
xlog ("L_INFO", "fd = $fd") ;
xlog ("L_INFO", "rd = $rd") ;
if ($au!=$fU || $au!=$tU) {
sl_send_reply("403","Forbidden auth ID au!=fu ou
au!=tu");
exit;
}
if ($au!=$rU) {
sl_send_reply("403","Forbidden R-URI");
exit;
}
#!ifdef WITH_MULTIDOMAIN
if ($fd!=$rd) {
sl_send_reply("403","Forbidden R-URI domain");
exit;
}
#!endif
} else {
xlog ("L_INFO", "au = $au") ;
xlog ("L_INFO", "fU = $fU") ;
# if ($au!=$fU) {
# sl_send_reply("403","Forbidden auth ID au!=fu");
# exit;
# }
}
#consume_credentials();
# caller authenticated
} else {
# caller is not local subscriber, then check if it calls
# a local destination, otherwise deny, not an open relay here
if (!uri==myself)
{
sl_send_reply("403","Not relaying");
exit;
}
}
}
#!endif
return;
}
# Caller NAT detection route
route[NATDETECT] {
#!ifdef WITH_NAT
force_rport();
if (nat_uac_test("19")) {
if (is_method("REGISTER")) {
fix_nated_register();
} else {
fix_nated_contact();
}
setflag(FLT_NATS);
}
#!endif
return;
}
# RTPProxy control
route[NATMANAGE] {
#!ifdef WITH_NAT
if (is_request()) {
if(has_totag()) {
if(check_route_param("nat=yes")) {
setbflag(FLB_NATB);
}
}
}
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
return;
rtpproxy_manage();
#rtpproxy_manage("co","82.127.95.167");
if (is_request()) {
if (!has_totag()) {
add_rr_param(";nat=yes");
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
fix_nated_contact();
}
}
#!endif
return;
}
# Routing to foreign domains
route[SIPOUT] {
if (!uri==myself)
{
append_hf("P-hint: outbound\r\n");
route(RELAY);
}
}
# PSTN GW routing
route[PSTN] {
#!ifdef WITH_PSTN
# check if PSTN GW IP is defined
if (strempty($sel(cfg_get.pstn.gw_ip))) {
xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n");
return;
}
# route to PSTN dialed numbers starting with '+' or '00'
# (international format)
# - update the condition to match your dialing rules for PSTN routing
if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$"))
return;
# only local users allowed to call
if(from_uri!=myself) {
sl_send_reply("403", "Not Allowed");
exit;
}
$ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip);
route(RELAY);
exit;
#!endif
return;
}
# XMLRPC routing
#!ifdef WITH_XMLRPC
route[XMLRPC] {
# allow XMLRPC from localhost
if ((method=="POST" || method=="GET")
&& (src_ip==127.0.0.1)) {
# close connection only for xmlrpclib user agents (there is a bug in
# xmlrpclib: it waits for EOF before interpreting the response).
if ($hdr(User-Agent) =~ "xmlrpclib")
set_reply_close();
set_reply_no_connect();
dispatch_rpc();
exit;
}
send_reply("403", "Forbidden");
exit;
}
#!endif
# route to voicemail server
route[TOVOICEMAIL] {
#!ifdef WITH_VOICEMAIL
if(!is_method("INVITE"))
return;
# check if VoiceMail server IP is defined
if (strempty($sel(cfg_get.voicemail.srv_ip))) {
xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n");
return;
}
if($avp(oexten)==$null)
return;
$ru = "tcp:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip)
+ ":" + $sel(cfg_get.voicemail.srv_port);
xlog("L_INFO","tovoicemail ru: $ru");
route(RELAY);
exit;
#!endif
return;
}
# manage outgoing branches
branch_route[MANAGE_BRANCH] {
xdbg("new branch [$T_branch_idx] to $ru\n");
route(NATMANAGE);
}
# manage incoming replies
onreply_route[MANAGE_REPLY] {
xlog("L_INFO","dans managereply");
xdbg("incoming reply\n");
if(status=~"[12][0-9][0-9]"){
xlog("L_INFO","avant route natmanage");
route(NATMANAGE);
}
}
# manage failure routing cases
failure_route[MANAGE_FAILURE] {
route(NATMANAGE);
if (t_is_canceled()) {
exit;
}
#!ifdef WITH_BLOCK3XX
# block call redirect based on 3xx replies.
if (t_check_status("3[0-9][0-9]")) {
t_reply("404","Not found");
exit;
}
#!endif
#!ifdef WITH_VOICEMAIL
# serial forking
# - route to voicemail on busy or no answer (timeout)
if (t_check_status("486|408")) {
route(TOVOICEMAIL);
exit;
}
#!endif
}
#edit asterisk
#!ifdef WITH_ASTERISK
# Test if coming from Asterisk
route[FROMASTERISK] {
xlog ("L_INFO", "Dans FROMASTERISK? $si / $sp");
if($si==$sel(cfg_get.asterisk.bindip)
&& $sp==$sel(cfg_get.asterisk.bindport))
return 1;
return -1;
}
# Send to Asterisk
route[TOASTERISK] {
$du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":"
+ $sel(cfg_get.asterisk.bindport);
xlog ("L_INFO", "Dans TOASTERISK $du") ;
xlog ("L_INFO", "Juste avant route relay");
route(RELAY);
exit;
}
# Forward REGISTER to Asterisk
route[REGFWD] {
xlog("L_INFO", "Dans REGFWD");
if(!is_method("REGISTER"))
{
return;
}
$var(rip) = $sel(cfg_get.asterisk.bindip);
$uac_req(method)="REGISTER";
$uac_req(ruri)="sip:" + $var(rip) + ":" +
$sel(cfg_get.asterisk.bindport);
$uac_req(furi)="sip:" + $au + "@" + $var(rip);
$uac_req(turi)="sip:" + $au + "@" + $var(rip);
$uac_req(hdrs)="Contact: <sip:" + $au + "@"
+ $sel(cfg_get.kamailio.bindip)
+ ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n";
xlog("L_INFO","avant if dans regfwd");
if($sel(contact.expires) != $null)
$uac_req(hdrs)= $uac_req(hdrs) + "Expires: " +
$sel(contact.expires) + "\r\n";
else
$uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) +
"\r\n";
uac_req_send();
}
#!endif
Hi All,
I've been away from Kamailio for a few months, but recently I started
working with it again. I could use a hand in a small problem:
I'm using include_file statement to give some structure to my config
file, but when checking the config I keep getting error message:
0(4290) : <core> [cfg.lex:1554]: included file name must be between
quotes: "defined_modules.cfg"
I'm using Kamailio 3.3.2 (version: kamailio 3.3.2 (x86_64/linux)
78d4b1) on Ubuntu.
And here's the beginnign of the config file:
#!KAMAILIO
#
include_file "defined_modules.cfg"
include_file "defined_values.cfg"
include_file "global_parameters.cfg"
Those look like quotes to me, but could the problem be something like
encoding of the config file itself, something in the included files,
or is there something else I can check? I don't have extra whitespaces
etc, perhaps a typo I just don't see?
Cheers,
Pirjo
Hi,
I'm using dialog module and I set some dlg_var for every dialog. Is it
possible to get this variable value with MI command somehow ? I know to get
list of dialogs using dlg_list only. Thanks!
Mino
Hi, I am trying to figure out how much memory a Kamailio server should have to support an X number of users, considering that I am running Kamailio, MySQL and RTPproxy on the same machine. I am planning to use Kamailio at medium/large enterprises, not at service providers but I got the following questions:
- Will Kamailio dynamically allocate the system memory it needs or I have to tune up the memory it takes? I.e.: If I use 8G memory, will it take what it needs or I need to adjust it somewhere? - How much memory is reasonable for 10K, 50K and 100K users in such "enterprise" scenarios? - Can you reference any documentation from where I can get this information? Thanks, Moacir
Dear All
When I get INVITE (received in UDP), I send it to another proxy in TLS.
For this purpose I have modified kamailio.cfg route section , please refer
the lines in red.
Just before sending in TLS, I modify the IP address in SDP, so that media
packets will pass through RTP proxy running
When INVITE is reached the next proxy, I see the IP address is changed . So
far so good.
Next I am trying to modify the IP address in 200 OK to INVITE.
However it is not taking effect.
I hope we need to modify in onreply_route function. Something is wrong
here. I am receiving 200 OK for INVITE in TLS. Next I am sending 200 OK for
INVITE to endpoints using UDP. Not sure if onreply_route is being executed
or not. Can somebody give me pointer whats wrong here.
*# Sample onreply route
onreply_route[REPLY_ONE] {
xdbg("incoming reply\n");
#!ifdef WITH_NAT
if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB))
&& status=~"(183)|(2[0-9][0-9])") {
force_rtp_proxy("r");
}
if (isbflagset(FLB_NATB)) {
fix_nated_contact();
}
#!endif
}
*
*
# Main SIP request routing logic
# - processing of any incoming SIP request starts with this route
route {
# per request initial checks
route(REQINIT);
setflag(FLT_NATS);
# NAT detection
route(NAT);
if(is_method("PUBLISH"))
{
loose_route();
t_relay();
exit;
}
if(is_method("REGISTER"))
{
t_relay_to("tls:115.114.48.75:443");
exit();
}
if(is_method("INVITE|BYE|CANCEL|SUBSCRIBE|REFER|NOTIFY"))
{
xdbg("incoming request\n");
route(RTPPROXY);
t_relay_to("tls:115.114.48.75:443");
exit();
}
*Thannks
Kamal
Hi,
I am using kamailio-3.3.2 on Ubuntu 11.04 x64 and when trying to connect
via TLS from my Yealink phone kamailio crashes with the following backtrace:
Loaded symbols for /lib/x86_64-linux-gnu/libnss_files.so.2
Core was generated by `/opt/kamailio/sbin/kamailio'.
Program terminated with signal 11, Segmentation fault.
#0 handle_ser_child (p=0x7fcbaea2eba0, fd_i=-1) at tcp_main.c:3575
3575 if (unlikely(p->unix_sock<=0)){
(gdb) bt
#0 handle_ser_child (p=0x7fcbaea2eba0, fd_i=-1) at tcp_main.c:3575
#1 0x000000000051e5bc in send2child (tcpconn=0x7f8c001597d8, ev=<value
optimized out>, fd_i=-1) at tcp_main.c:3975
#2 handle_tcpconn_ev (tcpconn=0x7f8c001597d8, ev=<value optimized out>,
fd_i=-1) at tcp_main.c:4310
#3 0x0000000000527dbc in io_wait_loop_epoll () at io_wait.h:1092
#4 tcp_main_loop () at tcp_main.c:4656
#5 0x00000000004726f5 in main_loop () at main.c:1727
#6 0x000000000047402e in main (argc=<value optimized out>, argv=<value
optimized out>) at main.c:2546
(gdb)
The manual I am following is
http://www.kamailio.org/dokuwiki/doku.php/tls:create-certificates
Thanks,
Den
Hi,
Can Kamailio act as SMTP server to receive emails and forward those to SIP
users as MESSAGEs?
It would be nice if postfix or some other rich SMTP server could function
as module of kamailio using all the services provided by kamailio scripting.
Krish Kura
Hi everyone
I'm trying to integrate Asterisk with Kamailio for voicemail.
I tried to follow this tutorial:
http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb
BUT:
- I had to adapt it because I use LDAP authentication with Kamailio
- I had problems with Asterisk 10.7 (problems with chan_sip module
crashing) so I've installed Asterisk 11 on another VM
- we have high-availability with 2 Kamailio servers, with Kamailio
listening on TCP (constraint from our SSL gateway in front of Kamailio) on
a "virtual IP" (created by keep-alive): this VIP is not visible with
ifconfig, but you can see it with the command "ip addr sh eth0"
For now, we use Linphone on Windows as SIP clients to test.
If I don't define WITH_ASTERISK, calls work, I can call someone(a)domain.tld
However, if I define WITH_ASTERISK, calls fail (even with destination
registered and available) and I have these errors in the logfile:
Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [ut.h:333]:
no corresponding socket for af 2
Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm
[t_fwd.c:424]: ERROR: can't fwd to af 2, proto 1 (no corresponding
listening socket)
Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm
[t_fwd.c:1530]: ERROR: t_forward_nonack: failure to add branches
Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: INFO: <script>: reply
error
Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: sl
[sl_funcs.c:371]: ERROR: sl_reply_error used: I'm terribly sorry, server
error occurred (1/SL)
It seems to happen on the if (!t_relay()) line in ROUTE[RELAY]
192.168.14.25 is the real IP of the Kamailio server,
192.168.14.24 is the VIP of the Kamailio "cluster"
192.168.14.28 is the IP of the Mysql server
192.168.14.32 is the IP of the Asterisk server
I can't find why the relay doesn't work. I've tried to bypass the VIP and
have Kamailio listen on the real IP, but it still doesn't work: I don't
seem to have the same errors as above, but I don't see any traffic between
Kamailio and Asterisk.
My kamailio.cfg is available here as it's too big for the list:
http://pastebin.com/zhncJ6j1
What could be the problem? Thank you very much for your help
Christophe
