Having trouble Setup Kamailio as a load balancer for Asterisk servers using module dispatcher
9 Nov
2020
9 Nov
'20
6:16 p.m.
Hello Everyone.
I am having trouble to setting up kamailio for load balancing for asterisk
servers.
I am using module dispatcher for load balancing AND two different servers
for kamailio and asterisk.
I am doing it first time and have not exact idea about its all configs… but
i have researched and done so far, now need help from you guys. Currently,
trying to setup with the one kamailio and one asterisk then will go ahead
if I get any success…
using MariaDB MySQL for the dispatcher data.
Simply I want to do is :
1. Register my asterisk devices with kamailioIP (and I am able to do this
using below configs)
Devices are getting register with the asterisk from with kamailioIP
2. I want to receive calls on the registered devices with KamailioIP (and i
am not able to do this)
WORK FLOW:
KAMAILIO IP = XX.XX.XX.95 on port 5060 and Asterisk IP = XX.XX.XX.164 on
port 5060
SIP Device(ZOIPER) -------->>>>KAMAILIO SERVER(device register/load
balancer)--------->>> ASTERISK SERVER
Please have a look into my configuration files and let me know if anything
is missing or wrong in these. Or anything to add or remove from the confs…
my kamailio.cfg
#!KAMAILIO
#!define WITH_MYSQL
#!define WITH_AUTH
#!define WITH_ASTERISK
#define WITH_NAT
#!define WITH_DEBUG
#!define WITH_USRLOCDB
#
# Kamailio (OpenSER) SIP Server v3.3 - 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
#
# *** 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
####### 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://kamailio:kamailiorw@localhost/kamailio"
#!ifdef WITH_ASTERISK
#!define DBASTURL "mysql://kamailio:kamailiorw@localhost/kamailio"
#!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=yes
#!else
debug=2
log_stderror=no
#!endif
memdbg=5
memlog=5
log_facility=LOG_LOCAL7
log_name="kamailio"
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="sip.mydomain.com"
/* 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:0.0.0.0:5060
#listen=udp: KamailioIP: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 = "" desc "VoiceMail IP Address"
voicemail.srv_port = "5060" desc "VoiceMail Port"
#!endif
#!ifdef WITH_ASTERISK
asterisk.bindip = " ASTERISKIP.164" desc "Asterisk IP Address"
asterisk.bindport = "5060" desc "Asterisk Port"
kamailio.bindip = " KamailioIP .95" 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="/usr/local/kamailio-devel/lib64/kamailio/modules/"
#!else
mpath="/usr/local/kamailio-devel/lib64/kamailio/modules/"
#!endif
#!ifdef WITH_MYSQL
loadmodule "db_mysql.so"
#!endif
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 "acc.so"
loadmodule "sctp.so"
loadmodule "dialog.so"
loadmodule "siptrace.so"
#!ifdef WITH_AUTH
loadmodule "auth.so"
loadmodule "auth_db.so"
#!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"
loadmodule "dispatcher.so"
loadmodule "jsonrpcs.so"
loadmodule "htable.so"
# ---------------------- mod dispatcher params
------------------------------
modparam("dispatcher", "db_url", DBURL)
modparam("dispatcher", "table_name", "dispatcher")
modparam("dispatcher", "ds_ping_interval", 15)
modparam("dispatcher", "ds_ping_from", "sip:dispathcer@
KamailioIP ")
modparam("dispatcher", "force_dst", 1)
modparam("dispatcher", "ds_ping_method", "INFO")
#modparam("dispatcher", "ds_probing_mode", 1)
modparam("dispatcher", "ds_probing_threshold", 1)
#modparam("dispatcher", "ds_ping_reply_codes",
"class=2;code=403;code=404;code=484;code=488;code=481;class=3")
modparam("dispatcher", "ds_ping_reply_codes",
"class=2;class=3;class=4")
#modparam("dispatcher", "priority_col", "dstpriority")
#modparam("dispatcher", "dstid_avp", "$avp(dsdstid)")
# do failover
modparam("dispatcher", "flags", 2)
#modparam("dispatcher", "dst_avp", "$avp(AVP_DST)")
#modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)")
#modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)")
#!endif
# ----------------- setting module-specific parameters ---------------
modparam("siptrace", "db_url", DBURL)
modparam("siptrace", "trace_flag", 22)
modparam("siptrace", "trace_on", 1)
# ----- jsonrpcs params -----
modparam("jsonrpcs", "pretty_format", 1)
/* set the path to RPC fifo control file */
# modparam("jsonrpcs", "fifo_name",
"/run/kamailio/kamailio_rpc.fifo")
/* set the path to RPC unix socket control file */
# modparam("jsonrpcs", "dgram_socket",
"/run/kamailio/kamailio_rpc.sock")
#!ifdef WITH_JSONRPC
modparam("jsonrpcs", "transport", 7)
#!endif
# ---------------------- mod uac params ------------------------------
modparam("uac", "reg_contact_addr", "ASTERISKIP:5060")
modparam("uac", "reg_db_url",
"mysql://kamailio:kamailiorw@localhost
/kamailio")
modparam("uac", "reg_db_table", "uacreg")
modparam("uac", "reg_timer_interval", 60)
modparam("uac","restore_mode","auto")
modparam("uac", "restore_dlg", 1)
modparam("uac","auth_realm_avp","$avp(s:auth_realm_avp)")
modparam("uac","auth_username_avp","$avp(s:auth_username_avp)")
modparam("uac","auth_password_avp","$avp(s:auth_password_avp)")
# ----- mi_fifo params -----
#modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")
# ----- 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)
modparam("tm", "ruri_matching", 0)
# ----- 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)
#!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", 10)
# max value for expires of registrations
modparam("registrar", "max_expires", 3600)
# set it to 1 to enable GRUU
modparam("registrar", "gruu_enabled", 0)
# ----- 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)
modparam("usrloc", "matching_mode", 1)
modparam("usrloc", "db_insert_update", 1)
#!endif
# ----- auth_db params -----
#!ifdef WITH_AUTH
modparam("auth_db", "db_url", DBURL)
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", "secret")
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)
modparam("permissions", "trusted_table", "trusted")
#!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)
# 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:127.0.0.1:7722")
# ----- 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@kamailio.org")
# params needed for NAT traversal in other modules
modparam("nathelper|registrar", "received_avp",
"$avp(RECEIVED)")
modparam("usrloc", "nat_bflag", FLB_NATB)
#!endif
#!ifdef WITH_TLS
# ----- tls params -----
modparam("tls", "config",
"/usr/local/etc/kamailio/tls.cfg")
#!endif
#!ifdef WITH_ANTIFLOOD
# ----- pike params -----
modparam("pike", "sampling_time_unit", 2)
modparam("pike", "reqs_density_per_unit", 16)
modparam("pike", "remove_latency", 4)
# ----- htable params -----
# ip ban htable with autoexpire after 5 minutes
#modparam("htable", "htable",
"ipban=>size=8;autoexpire=300;")
modparam("htable", "htable", "stats=>size=6;")
#!endif
#!ifdef WITH_XMLRPC
# ----- xmlrpc params -----
modparam("xmlrpc", "route", "XMLRPC");
modparam("xmlrpc", "url_match", "^/RPC")
#!endif
#!ifdef WITH_DEBUG
# ----- debugger params -----
#modparam("debugger", "cfgtrace", 1)
#modparam("debugger", "breakpoint", 1)
modparam("debugger", "log_level_name", "exec")
#!endif
####### Routing Logic ########
# Main SIP request routing logic
# - processing of any incoming SIP request starts with this route
# - note: this is the same as route { ... }
request_route {
route(DISPATCH);
# per request initial checks
route(REQINIT);
# NAT detection
route(NATDETECT);
# handle requests within SIP dialogs
route(WITHINDLG);
### only initial requests (no To tag)
if(is_method("OPTIONS"))
{
sl_send_reply("200","Keepalive");
exit;
}
if(is_method("NOTIFY"))
{
sl_send_reply("200","OK");
exit;
}
# CANCEL processing
if (is_method("CANCEL"))
{
if (t_check_trans())
t_relay();
exit;
}
t_check_trans();
# authentication
route(AUTH);
# record routing for dialog forming requests (in case they are routed)
# - remove preloaded route headers
remove_hf("Route");
if (is_method("INVITE|SUBSCRIBE"))
record_route();
# account only INVITEs
if (is_method("INVITE"))
{
setflag(FLT_ACC); # do accounting
}
# dispatch requests to foreign domains
route(SIPOUT);
### requests for my local domains
# handle presence related requests
route(PRESENCE);
# handle registrations
route(REGISTRAR);
if ($rU==$null)
{
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}
# dispatch destinations to PSTN
route(PSTN);
# user location service
route(LOCATION);
route(RELAY);
}
route[RELAY] {
# enable additional event routes for forwarded requests
# - serial forking, RTP relaying handling, a.s.o.
if (is_method("INVITE|SUBSCRIBE")) {
t_on_branch("MANAGE_BRANCH");
t_on_reply("MANAGE_REPLY");
}
if (is_method("INVITE")) {
t_on_failure("MANAGE_FAILURE");
}
if (!t_relay()) {
sl_reply_error();
}
exit;
}
# Per SIP request initial checks
route[REQINIT] {
#!ifdef WITH_ANTIFLOOD
# flood dection from same IP and traffic ban for a while
# be sure you exclude checking trusted peers, such as pstn gateways
# - local host excluded (e.g., loop to self)
if(src_ip!=myself)
{
if($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] {
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
if (is_method("BYE")) {
setflag(FLT_ACC); # do accounting ...
setflag(FLT_ACCFAILED); # ... even if the transaction fails
}
if ( is_method("ACK") ) {
# ACK is forwarded statelessy
route(NATMANAGE);
}
route(RELAY);
} else {
if (is_method("SUBSCRIBE") && uri == myself) {
# in-dialog subscribe requests
route(PRESENCE);
exit;
}
if ( is_method("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
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"))
{
if(isflagset(FLT_NATS))
{
setbflag(FLB_NATB);
# uncomment next line to do SIP NAT pinging
## setbflag(FLB_NATSIPPING);
}
if (!save("location"))
sl_reply_error();
#!ifdef WITH_ASTERISK
route(REGFWD);
#!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
#!ifdef WITH_ASTERISK
if(is_method("INVITE") && (!route(FROMASTERISK))) {
# if new call from out there - send to Asterisk
# - non-INVITE request are routed directly by Kamailio
# - traffic from Asterisk is routed also directy by Kamailio
append_hf("FromIp: $src_ip\r\n", "Call-ID");
route(TOASTERISK);
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
#!ifdef WITH_ASTERISK
# do not auth traffic from Asterisk - trusted!
if(route(FROMASTERISK))
return;
#!endif
#!ifdef WITH_IPAUTH
if((!is_method("REGISTER")) && allow_source_address())
{
# source IP allowed
return;
}
if (is_method("REGISTER") || from_uri==myself){
if (!allow_trusted()) {
xlog('wokred here for ipauth check');
sl_send_reply("403", "Forbidden");
}
}
#!endif
if (is_method("REGISTER") || from_uri==myself)
{
# authenticate requests
#!ifdef WITH_ASTERISK
if (!auth_check("$fd", "sip_users", "1")) {
#!else
if (!auth_check("$fd", "subscriber", "1")) {
#!endif
auth_challenge("$fd", "0");
exit;
}
# user authenticated - remove auth header
if(!is_method("REGISTER|PUBLISH"))
consume_credentials();
}
# if caller is not local subscriber, then check if it calls
# a local destination, otherwise deny, not an open relay here
if (from_uri!=myself && uri!=myself)
{
sl_send_reply("403","Not relaying");
exit;
}
#!endif
return;
}
# Caller NAT detection route
route[NATDETECT] {
#!ifdef WITH_NAT
force_rport();
if (nat_uac_test("19")) {
if (is_method("REGISTER")) {
fix_nated_register();
} else {
fix_nated_contact();
}
setflag(FLT_NATS);
}
#!endif
return;
}
# RTPProxy control
route[NATMANAGE] {
#!ifdef WITH_NAT
if (is_request()) {
if(has_totag()) {
if(check_route_param("nat=yes")) {
setbflag(FLB_NATB);
}
}
}
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
return;
rtpproxy_manage();
if (is_request()) {
if (!has_totag()) {
add_rr_param(";nat=yes");
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
fix_nated_contact();
}
}
#!endif
return;
}
# Routing to foreign domains
route[SIPOUT] {
if (!uri==myself)
{
append_hf("P-hint: outbound\r\n");
route(RELAY);
}
}
# PSTN GW routing
route[PSTN] {
#!ifdef WITH_PSTN
# check if PSTN GW IP is defined
if (strempty($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 = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip)
+ ":" + $sel(cfg_get.voicemail.srv_port);
route(RELAY);
exit;
#!endif
return;
}
# manage outgoing branches
branch_route[MANAGE_BRANCH] {
xdbg("new branch [$T_branch_idx] to $ru\n");
route(NATMANAGE);
}
# manage incoming replies
onreply_route[MANAGE_REPLY] {
xdbg("incoming reply\n");
if(status=~"[12][0-9][0-9]")
route(NATMANAGE);
}
# manage failure routing cases
failure_route[MANAGE_FAILURE] {
route(NATMANAGE);
if (t_is_canceled()) {
exit;
}
if(t_check_status("401")) {
uac_auth();
}
#!ifdef WITH_BLOCK3XX
# block call redirect based on 3xx replies.
if (t_check_status("3[0-9][0-9]")) {
t_reply("404","Not found");
exit;
}
#!endif
#!ifdef WITH_VOICEMAIL
# serial forking
# - route to voicemail on busy or no answer (timeout)
if (t_check_status("486|408")) {
route(TOVOICEMAIL);
exit;
}
t_relay();
#!endif
}
# Test if coming from Asterisk
route[FROMASTERISK] {
# if($si==$sel(cfg_get.asterisk.bindip)
# && $sp==$sel(cfg_get.asterisk.bindport))
# return 1;
# return -1;
# dispatch: is this asterisk ip ?
if (ds_is_from_list("1"))
return 1;
return -1;
}
# Send to Asterisk
route[TOASTERISK] {
# $du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":"
# + $sel(cfg_get.asterisk.bindport);
# append_hf("X-Asterisk-Context: dexter-phones\r\n");
# route(RELAY);
# exit;
# select asterisk port
append_hf("X-Asterisk-Context: dexter-phones\r\n");
route(DISPATCH);
}
route[DISPATCH] {
# round robin dispatching on gateways group '1'
#xlog("<<<<<<<<<<<<<<<<trying for
dispatch>>>>>>>>>>>>>>>>>>>");
#if (is_method("INVITE")) {
# dst_select( "GROUP", "HASH METHOD")
#ds_select_dst("1","4");
#sl_send_reply("100","Trying");
# forward();
# exit();
#}
if (!ds_select_domain("1", "4")) {
send_reply("404", "No destination");
exit;
}
t_on_failure("RTF_DISPATCH");
route(RELAY);
exit;
}
# Forward REGISTER to Asterisk
route[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)
+ ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n";
if($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();
}
“kamctl dispatcher dump” looks like this:
{
"jsonrpc": "2.0",
"result": {
"NRSETS": 1,
"RECORDS": [{
"SET": {
"ID": 1,
"TARGETS": [{
"DEST": {
"URI": "sip:ASTERISKIP:5060",
"FLAGS": "AP",
"PRIORITY": 1,
"ATTRS": {
"BODY": "rweight=50;weight=50;cc=1",
"DUID": "",
"MAXLOAD": 0,
"WEIGHT": 50,
"RWEIGHT": 50,
"SOCKET": "",
"SOCKNAME": "",
"OBPROXY": ""
}
}
}]
}
}]
},
"id": 13403
}
And also have set trunk in ASTERISK sip.conf
register => kamailioSIP:XXXX@KamailioIP:5060
[kamailio]
type=friend
host= KamailioIP
port=5060
transport=udp
allow=all
allow=gsm
allow=alaw,ulaw
insecure=invite,port
Permit = 0.0.0.0/0.0.0.0
sipdebug=yes
username=kamailioSIP
canreinvite=no
secret=XXXX
qualify=yes
dtmfmode=auto
when i Register my asterisk devices with kamailioIP (and I am able to do
this)
Devices are getting register with the asterisk from with kamailioIP
asterisk -r : (here 8101 and 8100 is registered with Kamailio IP )
[Nov 9 10:36:02] -- Registered SIP '8101' at XX.XX.XX.95:5060
[Nov 9 10:36:02] -- Unregistered SIP '8101'
[Nov 9 10:36:03] -- Registered SIP '8101' at XX.XX.XX..95:5060
[Nov 9 10:36:03] == Manager 'listencron' logged on from 127.0.0.1
[Nov 9 10:36:04] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 9 10:36:07] -- Unregistered SIP '8100'
[Nov 9 10:36:09] -- Registered SIP '8100' at XX.XX.XX..95:5060
[Nov 9 10:36:09] > Saved useragent "Z 3.15.40006 rv2.8.20" for peer 8100
[Nov 9 10:36:09] NOTICE[8453]: chan_sip.c:23869 handle_response_peerpoke:
Peer '8100' is now
Reachable. (51ms / 2000ms)
when i dial 8100 to 8101 i get in my asteris -r
NOTICE[8453][C-0000000b]: chan_sip.c:26002 handle_request_invite: Call from
'8101' (95.217.223.95:5060) to extension '8100' rejected because
extension
not found in context 'none-dial'.
and not able to call extension to extension,
PLEASE HELP to do this, I have search a lot and tried so much of stuff in
configs but no success still kindly guide me if anything I am missing to do
in the process or anything I am doing wrong.
I will be eagerly waiting for any help or pointer…
Thanks in advance
10 Nov
10 Nov
9:39 a.m.
Hello,
this looks somehow like a dialplan issue from Asterisk side:
when i dial 8100 to 8101 i get in my asteris -r
NOTICE[8453][C-0000000b]: chan_sip.c:26002 handle_request_invite: Call from '8101'
(95.217.223.95:5060<http://95.217.223.95:5060>) to extension '8100' rejected
because extension not found in context 'none-dial'.
and not able to call extension to extension,
See if you find something in the net with the quoted error message. Maybe if the issue is
more from Asterisk side, try with a simple example without the Kamailio and then adding
the Kamailio again when its works from Asterisk side.
Cheers,
Henning
--
Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com<https://gilawa.com/>
From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Usama Shaikh
Sent: Monday, November 9, 2020 6:16 PM
To: sr-users(a)lists.kamailio.org
Subject: [SR-Users] Having trouble Setup Kamailio as a load balancer for Asterisk servers
using module dispatcher
Hello Everyone.
I am having trouble to setting up kamailio for load balancing for asterisk servers.
I am using module dispatcher for load balancing AND two different servers for kamailio and
asterisk.
I am doing it first time and have not exact idea about its all configs… but i have
researched and done so far, now need help from you guys. Currently, trying to setup with
the one kamailio and one asterisk then will go ahead if I get any success…
using MariaDB MySQL for the dispatcher data.
Simply I want to do is :
1. Register my asterisk devices with kamailioIP (and I am able to do this using below
configs)
Devices are getting register with the asterisk from with kamailioIP
2. I want to receive calls on the registered devices with KamailioIP (and i am not able to
do this)
WORK FLOW:
KAMAILIO IP = XX.XX.XX.95 on port 5060 and Asterisk IP = XX.XX.XX.164 on port 5060
SIP Device(ZOIPER) -------->>>>KAMAILIO SERVER(device register/load
balancer)--------->>> ASTERISK SERVER
Please have a look into my configuration files and let me know if anything is missing or
wrong in these. Or anything to add or remove from the confs…
my kamailio.cfg
#!KAMAILIO
#!define WITH_MYSQL
#!define WITH_AUTH
#!define WITH_ASTERISK
#define WITH_NAT
#!define WITH_DEBUG
#!define WITH_USRLOCDB
#
# Kamailio (OpenSER) SIP Server v3.3 - default configuration script
# - web: http://www.kamailio.org<http://www.kamailio.org/>
# - git: http://sip-router.org<http://sip-router.org/>
#
# Direct your questions about this file to:
<sr-users@lists.sip-router.org<mailto: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
#
# *** 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<http://www.rtpproxy.org/>
# - start RTPProxy:
# rtpproxy -l _your_public_ip_ -s udp:localhost:7722
#
# *** 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
####### 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://kamailio:kamailiorw@localhost/kamailio"
#!ifdef WITH_ASTERISK
#!define DBASTURL "mysql://kamailio:kamailiorw@localhost/kamailio"
#!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=yes
#!else
debug=2
log_stderror=no
#!endif
memdbg=5
memlog=5
log_facility=LOG_LOCAL7
log_name="kamailio"
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="sip.mydomain.com<http://sip.mydomain.com>"
/* 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:0.0.0.0:5060<http://0.0.0.0:5060>
#listen=udp: KamailioIP: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<http://group.id> = value 'desc' description
# Access: $sel(cfg_get.group.id<http://cfg_get.group.id>) or
@cfg_get.group.id<http://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 = "" desc "VoiceMail IP Address"
voicemail.srv_port = "5060" desc "VoiceMail Port"
#!endif
#!ifdef WITH_ASTERISK
asterisk.bindip = " ASTERISKIP.164" desc "Asterisk IP Address"
asterisk.bindport = "5060" desc "Asterisk Port"
kamailio.bindip = " KamailioIP .95" 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="/usr/local/kamailio-devel/lib64/kamailio/modules/"
#!else
mpath="/usr/local/kamailio-devel/lib64/kamailio/modules/"
#!endif
#!ifdef WITH_MYSQL
loadmodule "db_mysql.so"
#!endif
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 "acc.so"
loadmodule "sctp.so"
loadmodule "dialog.so"
loadmodule "siptrace.so"
#!ifdef WITH_AUTH
loadmodule "auth.so"
loadmodule "auth_db.so"
#!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"
loadmodule "dispatcher.so"
loadmodule "jsonrpcs.so"
loadmodule "htable.so"
# ---------------------- mod dispatcher params ------------------------------
modparam("dispatcher", "db_url", DBURL)
modparam("dispatcher", "table_name", "dispatcher")
modparam("dispatcher", "ds_ping_interval", 15)
modparam("dispatcher", "ds_ping_from", "sip:dispathcer@
KamailioIP <sip:dispathcer@%20KamailioIP%20> ")
modparam("dispatcher", "force_dst", 1)
modparam("dispatcher", "ds_ping_method", "INFO")
#modparam("dispatcher", "ds_probing_mode", 1)
modparam("dispatcher", "ds_probing_threshold", 1)
#modparam("dispatcher", "ds_ping_reply_codes",
"class=2;code=403;code=404;code=484;code=488;code=481;class=3")
modparam("dispatcher", "ds_ping_reply_codes",
"class=2;class=3;class=4")
#modparam("dispatcher", "priority_col", "dstpriority")
#modparam("dispatcher", "dstid_avp", "$avp(dsdstid)")
# do failover
modparam("dispatcher", "flags", 2)
#modparam("dispatcher", "dst_avp", "$avp(AVP_DST)")
#modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)")
#modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)")
#!endif
# ----------------- setting module-specific parameters ---------------
modparam("siptrace", "db_url", DBURL)
modparam("siptrace", "trace_flag", 22)
modparam("siptrace", "trace_on", 1)
# ----- jsonrpcs params -----
modparam("jsonrpcs", "pretty_format", 1)
/* set the path to RPC fifo control file */
# modparam("jsonrpcs", "fifo_name",
"/run/kamailio/kamailio_rpc.fifo")
/* set the path to RPC unix socket control file */
# modparam("jsonrpcs", "dgram_socket",
"/run/kamailio/kamailio_rpc.sock")
#!ifdef WITH_JSONRPC
modparam("jsonrpcs", "transport", 7)
#!endif
# ---------------------- mod uac params ------------------------------
modparam("uac", "reg_contact_addr", "ASTERISKIP:5060")
modparam("uac", "reg_db_url",
"mysql://kamailio:kamailiorw@localhost/kamailio")
modparam("uac", "reg_db_table", "uacreg")
modparam("uac", "reg_timer_interval", 60)
modparam("uac","restore_mode","auto")
modparam("uac", "restore_dlg", 1)
modparam("uac","auth_realm_avp","$avp(s:auth_realm_avp)")
modparam("uac","auth_username_avp","$avp(s:auth_username_avp)")
modparam("uac","auth_password_avp","$avp(s:auth_password_avp)")
# ----- mi_fifo params -----
#modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")
# ----- 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)
modparam("tm", "ruri_matching", 0)
# ----- 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)
#!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", 10)
# max value for expires of registrations
modparam("registrar", "max_expires", 3600)
# set it to 1 to enable GRUU
modparam("registrar", "gruu_enabled", 0)
# ----- 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)
modparam("usrloc", "matching_mode", 1)
modparam("usrloc", "db_insert_update", 1)
#!endif
# ----- auth_db params -----
#!ifdef WITH_AUTH
modparam("auth_db", "db_url", DBURL)
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", "secret")
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)
modparam("permissions", "trusted_table", "trusted")
#!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)
# 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:127.0.0.1:7722<http://127.0.0.1:7722>")
# ----- 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@kamailio.org<mailto:sip%3Apinger@kamailio.org>")
# params needed for NAT traversal in other modules
modparam("nathelper|registrar", "received_avp",
"$avp(RECEIVED)")
modparam("usrloc", "nat_bflag", FLB_NATB)
#!endif
#!ifdef WITH_TLS
# ----- tls params -----
modparam("tls", "config",
"/usr/local/etc/kamailio/tls.cfg")
#!endif
#!ifdef WITH_ANTIFLOOD
# ----- pike params -----
modparam("pike", "sampling_time_unit", 2)
modparam("pike", "reqs_density_per_unit", 16)
modparam("pike", "remove_latency", 4)
# ----- htable params -----
# ip ban htable with autoexpire after 5 minutes
#modparam("htable", "htable",
"ipban=>size=8;autoexpire=300;")
modparam("htable", "htable", "stats=>size=6;")
#!endif
#!ifdef WITH_XMLRPC
# ----- xmlrpc params -----
modparam("xmlrpc", "route", "XMLRPC");
modparam("xmlrpc", "url_match", "^/RPC")
#!endif
#!ifdef WITH_DEBUG
# ----- debugger params -----
#modparam("debugger", "cfgtrace", 1)
#modparam("debugger", "breakpoint", 1)
modparam("debugger", "log_level_name", "exec")
#!endif
####### Routing Logic ########
# Main SIP request routing logic
# - processing of any incoming SIP request starts with this route
# - note: this is the same as route { ... }
request_route {
route(DISPATCH);
# per request initial checks
route(REQINIT);
# NAT detection
route(NATDETECT);
# handle requests within SIP dialogs
route(WITHINDLG);
### only initial requests (no To tag)
if(is_method("OPTIONS"))
{
sl_send_reply("200","Keepalive");
exit;
}
if(is_method("NOTIFY"))
{
sl_send_reply("200","OK");
exit;
}
# CANCEL processing
if (is_method("CANCEL"))
{
if (t_check_trans())
t_relay();
exit;
}
t_check_trans();
# authentication
route(AUTH);
# record routing for dialog forming requests (in case they are routed)
# - remove preloaded route headers
remove_hf("Route");
if (is_method("INVITE|SUBSCRIBE"))
record_route();
# account only INVITEs
if (is_method("INVITE"))
{
setflag(FLT_ACC); # do accounting
}
# dispatch requests to foreign domains
route(SIPOUT);
### requests for my local domains
# handle presence related requests
route(PRESENCE);
# handle registrations
route(REGISTRAR);
if ($rU==$null)
{
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}
# dispatch destinations to PSTN
route(PSTN);
# user location service
route(LOCATION);
route(RELAY);
}
route[RELAY] {
# enable additional event routes for forwarded requests
# - serial forking, RTP relaying handling, a.s.o.
if (is_method("INVITE|SUBSCRIBE")) {
t_on_branch("MANAGE_BRANCH");
t_on_reply("MANAGE_REPLY");
}
if (is_method("INVITE")) {
t_on_failure("MANAGE_FAILURE");
}
if (!t_relay()) {
sl_reply_error();
}
exit;
}
# Per SIP request initial checks
route[REQINIT] {
#!ifdef WITH_ANTIFLOOD
# flood dection from same IP and traffic ban for a while
# be sure you exclude checking trusted peers, such as pstn gateways
# - local host excluded (e.g., loop to self)
if(src_ip!=myself)
{
if($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] {
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
if (is_method("BYE")) {
setflag(FLT_ACC); # do accounting ...
setflag(FLT_ACCFAILED); # ... even if the transaction fails
}
if ( is_method("ACK") ) {
# ACK is forwarded statelessy
route(NATMANAGE);
}
route(RELAY);
} else {
if (is_method("SUBSCRIBE") && uri == myself) {
# in-dialog subscribe requests
route(PRESENCE);
exit;
}
if ( is_method("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
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"))
{
if(isflagset(FLT_NATS))
{
setbflag(FLB_NATB);
# uncomment next line to do SIP NAT pinging
## setbflag(FLB_NATSIPPING);
}
if (!save("location"))
sl_reply_error();
#!ifdef WITH_ASTERISK
route(REGFWD);
#!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
#!ifdef WITH_ASTERISK
if(is_method("INVITE") && (!route(FROMASTERISK))) {
# if new call from out there - send to Asterisk
# - non-INVITE request are routed directly by Kamailio
# - traffic from Asterisk is routed also directy by Kamailio
append_hf("FromIp: $src_ip\r\n", "Call-ID");
route(TOASTERISK);
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
#!ifdef WITH_ASTERISK
# do not auth traffic from Asterisk - trusted!
if(route(FROMASTERISK))
return;
#!endif
#!ifdef WITH_IPAUTH
if((!is_method("REGISTER")) && allow_source_address())
{
# source IP allowed
return;
}
if (is_method("REGISTER") || from_uri==myself){
if (!allow_trusted()) {
xlog('wokred here for ipauth check');
sl_send_reply("403", "Forbidden");
}
}
#!endif
if (is_method("REGISTER") || from_uri==myself)
{
# authenticate requests
#!ifdef WITH_ASTERISK
if (!auth_check("$fd", "sip_users", "1")) {
#!else
if (!auth_check("$fd", "subscriber", "1")) {
#!endif
auth_challenge("$fd", "0");
exit;
}
# user authenticated - remove auth header
if(!is_method("REGISTER|PUBLISH"))
consume_credentials();
}
# if caller is not local subscriber, then check if it calls
# a local destination, otherwise deny, not an open relay here
if (from_uri!=myself && uri!=myself)
{
sl_send_reply("403","Not relaying");
exit;
}
#!endif
return;
}
# Caller NAT detection route
route[NATDETECT] {
#!ifdef WITH_NAT
force_rport();
if (nat_uac_test("19")) {
if (is_method("REGISTER")) {
fix_nated_register();
} else {
fix_nated_contact();
}
setflag(FLT_NATS);
}
#!endif
return;
}
# RTPProxy control
route[NATMANAGE] {
#!ifdef WITH_NAT
if (is_request()) {
if(has_totag()) {
if(check_route_param("nat=yes")) {
setbflag(FLB_NATB);
}
}
}
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
return;
rtpproxy_manage();
if (is_request()) {
if (!has_totag()) {
add_rr_param(";nat=yes");
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
fix_nated_contact();
}
}
#!endif
return;
}
# Routing to foreign domains
route[SIPOUT] {
if (!uri==myself)
{
append_hf("P-hint: outbound\r\n");
route(RELAY);
}
}
# PSTN GW routing
route[PSTN] {
#!ifdef WITH_PSTN
# check if PSTN GW IP is defined
if (strempty($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 = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip)
+ ":" + $sel(cfg_get.voicemail.srv_port);
route(RELAY);
exit;
#!endif
return;
}
# manage outgoing branches
branch_route[MANAGE_BRANCH] {
xdbg("new branch [$T_branch_idx] to $ru\n");
route(NATMANAGE);
}
# manage incoming replies
onreply_route[MANAGE_REPLY] {
xdbg("incoming reply\n");
if(status=~"[12][0-9][0-9]")
route(NATMANAGE);
}
# manage failure routing cases
failure_route[MANAGE_FAILURE] {
route(NATMANAGE);
if (t_is_canceled()) {
exit;
}
if(t_check_status("401")) {
uac_auth();
}
#!ifdef WITH_BLOCK3XX
# block call redirect based on 3xx replies.
if (t_check_status("3[0-9][0-9]")) {
t_reply("404","Not found");
exit;
}
#!endif
#!ifdef WITH_VOICEMAIL
# serial forking
# - route to voicemail on busy or no answer (timeout)
if (t_check_status("486|408")) {
route(TOVOICEMAIL);
exit;
}
t_relay();
#!endif
}
# Test if coming from Asterisk
route[FROMASTERISK] {
# if($si==$sel(cfg_get.asterisk.bindip)
# && $sp==$sel(cfg_get.asterisk.bindport))
# return 1;
# return -1;
# dispatch: is this asterisk ip ?
if (ds_is_from_list("1"))
return 1;
return -1;
}
# Send to Asterisk
route[TOASTERISK] {
# $du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":"
# + $sel(cfg_get.asterisk.bindport);
# append_hf("X-Asterisk-Context: dexter-phones\r\n");
# route(RELAY);
# exit;
# select asterisk port
append_hf("X-Asterisk-Context: dexter-phones\r\n");
route(DISPATCH);
}
route[DISPATCH] {
# round robin dispatching on gateways group '1'
#xlog("<<<<<<<<<<<<<<<<trying for
dispatch>>>>>>>>>>>>>>>>>>>");
#if (is_method("INVITE")) {
# dst_select( "GROUP", "HASH METHOD")
#ds_select_dst("1","4");
#sl_send_reply("100","Trying");
# forward();
# exit();
#}
if (!ds_select_domain("1", "4")) {
send_reply("404", "No destination");
exit;
}
t_on_failure("RTF_DISPATCH");
route(RELAY);
exit;
}
# Forward REGISTER to Asterisk
route[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) + ":" + $sel(cfg_get.kamailio.bindport) +
"<sip:%22%20+%20$au%20+%20%22@%22%20+%20$sel(cfg_get.kamailio.bindip)%20+%20%22:%22%20+%20$sel(cfg_get.kamailio.bindport)%20+%20%22>>\r\n";
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();
}
“kamctl dispatcher dump” looks like this:
{
"jsonrpc": "2.0",
"result": {
"NRSETS": 1,
"RECORDS": [{
"SET": {
"ID": 1,
"TARGETS": [{
"DEST": {
"URI": "sip:ASTERISKIP:5060",
"FLAGS": "AP",
"PRIORITY": 1,
"ATTRS": {
"BODY": "rweight=50;weight=50;cc=1",
"DUID": "",
"MAXLOAD": 0,
"WEIGHT": 50,
"RWEIGHT": 50,
"SOCKET": "",
"SOCKNAME": "",
"OBPROXY": ""
}
}
}]
}
}]
},
"id": 13403
}
And also have set trunk in ASTERISK sip.conf
register => kamailioSIP:XXXX@KamailioIP:5060
[kamailio]
type=friend
host= KamailioIP
port=5060
transport=udp
allow=all
allow=gsm
allow=alaw,ulaw
insecure=invite,port
Permit = 0.0.0.0/0.0.0.0<http://0.0.0.0/0.0.0.0>
sipdebug=yes
username=kamailioSIP
canreinvite=no
secret=XXXX
qualify=yes
dtmfmode=auto
when i Register my asterisk devices with kamailioIP (and I am able to do this)
Devices are getting register with the asterisk from with kamailioIP asterisk -r : (here
8101 and 8100 is registered with Kamailio IP )
[Nov 9 10:36:02] -- Registered SIP '8101' at XX.XX.XX.95:5060
[Nov 9 10:36:02] -- Unregistered SIP '8101'
[Nov 9 10:36:03] -- Registered SIP '8101' at XX.XX.XX..95:5060
[Nov 9 10:36:03] == Manager 'listencron' logged on from 127.0.0.1
[Nov 9 10:36:04] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 9 10:36:07] -- Unregistered SIP '8100'
[Nov 9 10:36:09] -- Registered SIP '8100' at XX.XX.XX..95:5060
[Nov 9 10:36:09] > Saved useragent "Z 3.15.40006 rv2.8.20" for peer 8100
[Nov 9 10:36:09] NOTICE[8453]: chan_sip.c:23869 handle_response_peerpoke: Peer
'8100' is now
Reachable. (51ms / 2000ms)
when i dial 8100 to 8101 i get in my asteris -r
NOTICE[8453][C-0000000b]: chan_sip.c:26002 handle_request_invite: Call from '8101'
(95.217.223.95:5060<http://95.217.223.95:5060>) to extension '8100' rejected
because extension not found in context 'none-dial'.
and not able to call extension to extension,
PLEASE HELP to do this, I have search a lot and tried so much of stuff in configs but no
success still kindly guide me if anything I am missing to do in the process or anything I
am doing wrong.
I will be eagerly waiting for any help or pointer…
Thanks in advance
9:41 a.m.
Hello,
this looks somehow like a dialplan issue from Asterisk side:
when i dial 8100 to 8101 i get in my asteris -r
NOTICE[8453][C-0000000b]: chan_sip.c:26002 handle_request_invite: Call from '8101'
(95.217.223.95:5060<http://95.217.223.95:5060>) to extension '8100' rejected
because extension not found in context 'none-dial'.
and not able to call extension to extension,
See if you find something in the net with the quoted error message. Maybe if the issue is
more from Asterisk side, try with a simple example without the Kamailio and then adding
the Kamailio again when its works from Asterisk side.
Cheers,
Henning
--
Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com<https://gilawa.com/>
From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Usama Shaikh
Sent: Monday, November 9, 2020 6:16 PM
To: sr-users(a)lists.kamailio.org
Subject: [SR-Users] Having trouble Setup Kamailio as a load balancer for Asterisk servers
using module dispatcher
Hello Everyone.
I am having trouble to setting up kamailio for load balancing for asterisk servers.
I am using module dispatcher for load balancing AND two different servers for kamailio and
asterisk.
I am doing it first time and have not exact idea about its all configs… but i have
researched and done so far, now need help from you guys. Currently, trying to setup with
the one kamailio and one asterisk then will go ahead if I get any success…
using MariaDB MySQL for the dispatcher data.
Simply I want to do is :
1. Register my asterisk devices with kamailioIP (and I am able to do this using below
configs)
Devices are getting register with the asterisk from with kamailioIP
2. I want to receive calls on the registered devices with KamailioIP (and i am not able to
do this)
WORK FLOW:
KAMAILIO IP = XX.XX.XX.95 on port 5060 and Asterisk IP = XX.XX.XX.164 on port 5060
SIP Device(ZOIPER) -------->>>>KAMAILIO SERVER(device register/load
balancer)--------->>> ASTERISK SERVER
Please have a look into my configuration files and let me know if anything is missing or
wrong in these. Or anything to add or remove from the confs…
my kamailio.cfg
[….]
“kamctl dispatcher dump” looks like this:
{
"jsonrpc": "2.0",
"result": {
"NRSETS": 1,
"RECORDS": [{
"SET": {
"ID": 1,
"TARGETS": [{
"DEST": {
"URI": "sip:ASTERISKIP:5060",
"FLAGS": "AP",
"PRIORITY": 1,
"ATTRS": {
"BODY": "rweight=50;weight=50;cc=1",
"DUID": "",
"MAXLOAD": 0,
"WEIGHT": 50,
"RWEIGHT": 50,
"SOCKET": "",
"SOCKNAME": "",
"OBPROXY": ""
}
}
}]
}
}]
},
"id": 13403
}
And also have set trunk in ASTERISK sip.conf
register => kamailioSIP:XXXX@KamailioIP:5060
[kamailio]
type=friend
host= KamailioIP
port=5060
transport=udp
allow=all
allow=gsm
allow=alaw,ulaw
insecure=invite,port
Permit = 0.0.0.0/0.0.0.0<http://0.0.0.0/0.0.0.0>
sipdebug=yes
username=kamailioSIP
canreinvite=no
secret=XXXX
qualify=yes
dtmfmode=auto
when i Register my asterisk devices with kamailioIP (and I am able to do this)
Devices are getting register with the asterisk from with kamailioIP asterisk -r : (here
8101 and 8100 is registered with Kamailio IP )
[Nov 9 10:36:02] -- Registered SIP '8101' at XX.XX.XX.95:5060
[Nov 9 10:36:02] -- Unregistered SIP '8101'
[Nov 9 10:36:03] -- Registered SIP '8101' at XX.XX.XX..95:5060
[Nov 9 10:36:03] == Manager 'listencron' logged on from 127.0.0.1
[Nov 9 10:36:04] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 9 10:36:07] -- Unregistered SIP '8100'
[Nov 9 10:36:09] -- Registered SIP '8100' at XX.XX.XX..95:5060
[Nov 9 10:36:09] > Saved useragent "Z 3.15.40006 rv2.8.20" for peer 8100
[Nov 9 10:36:09] NOTICE[8453]: chan_sip.c:23869 handle_response_peerpoke: Peer
'8100' is now
Reachable. (51ms / 2000ms)
when i dial 8100 to 8101 i get in my asteris -r
NOTICE[8453][C-0000000b]: chan_sip.c:26002 handle_request_invite: Call from '8101'
(95.217.223.95:5060<http://95.217.223.95:5060>) to extension '8100' rejected
because extension not found in context 'none-dial'.
and not able to call extension to extension,
PLEASE HELP to do this, I have search a lot and tried so much of stuff in configs but no
success still kindly guide me if anything I am missing to do in the process or anything I
am doing wrong.
I will be eagerly waiting for any help or pointer…
Thanks in advance
11 Nov
11 Nov
7:48 p.m.
Hi Henning,
Thanks a lot for your response.
i searched on the internet and tried it, and got some success in this…
(and what i did was to put some of the conf dialplans from general context
to my none-dial context) and i successfully got call when i login agent and
able to listen the
“conf-onlyperson.gsm” the only person audio… and also i was able to call
8101 to 8102 (but i dont know why not able to call vice versa)
but the issue now is (i suppose) now only at asterisk side because now again
i am not able to call which was working before please help me in this and if
you have idea why the behavior is intermittent so please point out the
stuff…Preformatted text
now my extensions.conf look like this is :
and now my sip.conf is
these are the logs in my asterisk -r while i am dialing and calling not
working
--
Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html
1271
days inactive
1273
days old
3 comments
3 participants
participants (3)
-
Henning Westerholt -
Usama Shaikh -
Usama_Shaikh