Kamailio as load balancer for multiple asterisk servers - linux

Question

I have three virtual servers running Ubuntu 14.04. On one server I installed Kamailio and on the others Asterisk. I want that the Kamailio server works as a load balancer and forwards the incoming calls to the asterisk servers (round robin).
I want to test it first with one asterisk server and if that works, I can add more for more performance.
I added my SIP provider credentials like this:
kamctl add test testpasswd
Then I added the asterisk server to the dispatcher table like this:
INSERT INTO dispatcher (setid,destination,flags,priority,attrs,description) VALUES (1,"sip:10.1.1.3:5060",0,0,"","Asteriskl-I");
I changed the sip.conf file on my asterisk server that it connects to my kamailio server and that seems to work.
My kamailio.cfg file looks like this:
#!KAMAILIO
#
# sample config file for dispatcher module
# - load balancing of VoIP calls with round robin
# - no TPC listening
# - don't dispatch REGISTER and presence requests
#
# Kamailio (OpenSER) SIP Server v3.2
# - web: http://www.kamailio.org
# - git: http://sip-router.org
#
# Direct your questions about this file to: sr-users#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
#
####### Global Parameters #########
#!define WITH_DEBUG
#!ifdef WITH_DEBUG
debug=4
log_stderror=yes
#!else
debug=2
log_stderror=no
#!endif
memdbg=5
memlog=5
log_facility=LOG_LOCAL0
fork=yes
children=4
/* comment the next line to enable TCP */
disable_tcp=yes
/* uncomment the next line to disable the auto discovery of local aliases
based on revers DNS on IPs (default on) */
auto_aliases=no
/* add local domain aliases */
# alias="mysipserver.com"
port=5060
/* uncomment and configure the following line if you want Kamailio to
bind on a specific interface/port/proto (default bind on all available) */
# listen=udp:127.0.0.1:5060
sip_warning=no
####### Modules Section ########
#set module path
mpath="/usr/local/lib64/kamailio/modules/"
# loadmodule "db_mysql.so"
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 "textops.so"
loadmodule "siputils.so"
loadmodule "xlog.so"
loadmodule "sanity.so"
# loadmodule "ctl.so"
loadmodule "mi_rpc.so"
loadmodule "acc.so"
loadmodule "dispatcher.so"
# ----------------- setting module-specific parameters ---------------
# ----- mi_fifo params -----
modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")
# ----- 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)
modparam("rr", "append_fromtag", 0)
# ----- acc params -----
modparam("acc", "log_flag", 1)
modparam("acc", "failed_transaction_flag", 3)
modparam("acc", "log_extra",
"src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si")
# ----- tm params -----
modparam("tm", "fr_timer", 2000)
modparam("tm", "fr_inv_timer", 40000)
# ----- dispatcher params -----
# modparam("dispatcher", "db_url",
# "mysql://kamailio:123456#localhost/kamailio")
modparam("dispatcher", "table_name", "dispatcher")
modparam("dispatcher", "flags", 2)
modparam("dispatcher", "dst_avp", "$avp(AVP_DST)")
modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)")
modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)")
####### Routing Logic ########
# main request routing logic
route {
# per request initial checks
route(REQINIT);
# handle requests within SIP dialogs
route(WITHINDLG);
### only initial requests (no To tag)
# CANCEL processing
if (is_method("CANCEL"))
{
if (t_check_trans())
t_relay();
exit;
}
t_check_trans();
# record routing for dialog forming requests (in case they are routed)
# - remove preloaded route headers
remove_hf("Route");
if (is_method("INVITE|SUBSCRIBE"))
record_route();
#if (is_method("INVITE"))
#{
# ds_select_domain("1","4");
# #sl_send_reply("300","Redirect");
# #t_relay();
# exit;
#}
# account only INVITEs
if (is_method("INVITE"))
{
setflag(1); # do accounting
}
# handle presence related requests
route(PRESENCE);
# handle registrations
route(REGISTRAR);
if ($rU==$null)
{
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}
# dispatch destinations
route(DISPATCH);
route(RELAY);
}
route[RELAY] {
if (!t_relay()) {
sl_reply_error();
}
exit;
}
# Per SIP request initial checks
route[REQINIT] {
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] {
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
if (is_method("BYE")) {
setflag(1); # do accounting ...
setflag(3); # ... even if the transaction fails
}
route(RELAY);
} else {
if (is_method("SUBSCRIBE") && uri == myself) {
# in-dialog subscribe requests
route(PRESENCE);
exit;
}
if ( is_method("ACK") ) {
if ( t_check_trans() ) {
# non loose-route, but stateful ACK;
# must be ACK after a 487 or e.g. 404 from upstream server
t_relay();
exit;
} else {
# ACK without matching transaction ... ignore and discard.
exit;
}
}
sl_send_reply("404","Not here");
}
exit;
}
}
# Handle SIP registrations
route[REGISTRAR] {
if(!is_method("REGISTER"))
return;
#sl_send_reply("404", "No registrar");
#t_relay();
if(!ds_select_dst("1", "4"))
{
sl_send_reply("404", "No registrar");
exit;
}
forward();
exit;
}
# Presence server route
route[PRESENCE] {
if(!is_method("PUBLISH|SUBSCRIBE"))
return;
sl_send_reply("404", "Not here");
exit;
}
# Dispatch requests
route[DISPATCH] {
# round robin dispatching on gateways group '1'
if(!ds_select_dst("1", "4"))
{
send_reply("404", "No destination");
exit;
}
xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n");
t_on_failure("RTF_DISPATCH");
return;
}
# Sample failure route
failure_route[RTF_DISPATCH] {
if (t_is_canceled()) {
exit;
}
# next DST - only for 500 or local timeout
if (t_check_status("500")
or (t_branch_timeout() and !t_branch_replied()))
{
if(ds_next_dst())
{
t_on_failure("RTF_DISPATCH");
route(RELAY);
exit;
}
}
}
If I connect my asterisk box directly to my SIP provider it works perfectly. But if I connect it to the kamailio server and the kamailio server to the SIP provider it doesn't.
I googled for hours and tried a lot of things and I really have no idea what I could try next... If anyone could help me I would be very happy!
Thank you very much and best regards

Answer 1

I added my SIP provider credentials like this:
kamctl add test testpasswd - This is wrong.
Check following link for detailed info how you should set-up SIP trunk on Kamailio, which are using username/password authentication:
http://lists.sip-router.org/pipermail/sr-users/2015-September/090001.html