GOautodial Open Source Omni-channel Contact Center Suite

Hey everyone , i've read almost all the posts here , tried everything but this kamailio just won't start , i managed to start at first but with a self signed certificate now i'm using Let's encrypt with an ip&domain .Maybe i misconfigured something here is my codes .

This is a Centos7 Server From Digital Ocean , I followed the From Scratch installation & fixed a lot of ip's import databases manually and all that , all the posts have been really helpful

<kamailio.cfg>

<pre><code class="yaml">

#!KAMAILIO

#

#!define WITH_MYSQL

#!define WITH_AUTH

#!define WITH_USRLOCDB

#!define WITH_NAT

#!define WITH_ANTIFLOOD

#

#

# Kamailio (OpenSER) SIP Server v5.0 - default configuration script

#     - 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/wiki/

# 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

#     - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING

#

# *** 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

####### Include Local Config If Exists #########

import_file "kamailio-local.cfg" 

####### 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.

#!ifndef DBURL

#!define DBURL "mysql://kamailiou:kamailiou1234@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

#!substdef "!MY_IP_ADDR!165.22.15.111!g" 

#!substdef "!MY_DOMAIN!callcenter.siarum.com!g" 

#!substdef "!MY_WS_PORT!8081!g" 

#!substdef "!MY_WSS_PORT!4443!g" 

#!substdef "!MY_MSRP_PORT!9080!g" 

#!substdef "!MY_WS_ADDR!tcp:MY_IP_ADDR:MY_WS_PORT!g" 

#!substdef "!MY_WSS_ADDR!tls:MY_IP_ADDR:MY_WSS_PORT!g" 

#!substdef "!MY_MSRP_ADDR!tls:MY_IP_ADDR:MY_MSRP_PORT!g" 

#!substdef "!MSRP_MIN_EXPIRES!1800!g" 

#!substdef "!MSRP_MAX_EXPIRES!3600!g" 

#!define WITH_TLS

#!define WITH_WEBSOCKETS

#!define WITH_MSRP

####### Global Parameters #########

### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR

#!ifdef WITH_DEBUG

debug=4

log_stderror=no

#!else

debug=2

log_stderror=no

#!endif

memdbg=5

memlog=5

log_facility=LOG_LOCAL0

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="callcenter.siarum.com" 

alias="165.22.15.111"

/* 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:165.22.15.111:5060

listen=udp:callcenter.siarum.com: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

listen=MY_IP_ADDR

#!ifdef WITH_WEBSOCKETS

listen=MY_WS_ADDR

#!ifdef WITH_TLS

listen=MY_WSS_ADDR

#!endif

#!endif

#!ifdef WITH_MSRP

listen=MY_MSRP_ADDR

#!endif

listen=tls:MY_IP_ADDR:5061

tcp_connection_lifetime=3604

tcp_accept_no_cl=yes

tcp_rd_buf_size=16384

# 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 "tos.cloud.goautodial.com GW Address" 

pstn.gw_port = "" desc "PSTN GW Port" 

#!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

# don't advertise server headers

server_signature=no

sip_warning=0

####### Modules Section ########

# set paths to location of modules (to sources or installation folders)

#!ifdef WITH_SRCPATH

mpath="modules/" 

#!else

mpath="/usr/lib64/kamailio/modules/" 

#mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" 

#!endif

#!ifdef WITH_MYSQL

loadmodule "db_mysql.so" 

#!endif

#loadmodule "topoh.so" 

#loadmodule "mi_fifo.so" 

loadmodule "jsonrpcs.so" 

loadmodule "kex.so" 

loadmodule "corex.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" 

#!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 "rtpengine.so" 

#loadmodule "rtpproxy.so" 

#!endif

#!ifdef WITH_TLS

loadmodule "tls.so" 

#!endif

#!ifdef WITH_MSRP

loadmodule "msrp.so" 

#loadmodule "htable.so" 

loadmodule "cfgutils.so" 

#!endif

#!ifdef WITH_WEBSOCKETS

loadmodule "xhttp.so" 

loadmodule "websocket.so" 

loadmodule "sdpops.so" 

loadmodule "textopsx.so" 

loadmodule "dialog.so" 

loadmodule "sst.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

# ----------------- setting module-specific parameters ---------------

# ---- topoh params -----

#modparam("topoh", "mask_key", "Gu3ssWh@T1tS2016")

#modparam("topoh", "mask_ip", "10.0.0.1")

#modparam("topoh", "mask_callid", 1)

# ----- mi_fifo params -----

#modparam("mi_fifo", "fifo_name", "/var/run/kamailio/kamailio_fifo")

# ----- jsonrpcs params -----

modparam("jsonrpcs", "pretty_format", 1)

/* set the path to RPC fifo control file */

modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo")

/* set the path to RPC unix socket control file */

modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock")

# ----- 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)

# ----- rr params -----

# set next param to 1 to add value to ;lr param (helps with some UAs)

modparam("rr", "enable_full_lr", 0)

# do not append from tag to the RR (no need for this script)

modparam("rr", "append_fromtag", 0)

# ----- 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 100 contacts per AOR */

modparam("registrar", "max_contacts", 100)

# 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", 1)

modparam("usrloc", "use_domain", MULTIDOMAIN)

modparam("usrloc", "timer_interval", 60)

modparam("usrloc", "timer_procs", 4)

#!endif

# ----- auth_db params -----

#!ifdef WITH_AUTH

modparam("auth_db", "db_url", DBURL)

modparam("auth_db", "calculate_ha1", 0)

modparam("auth_db", "password_column", "ha1")

modparam("auth_db", "load_credentials", "")

modparam("auth_db", "use_domain", MULTIDOMAIN)

modparam("auth", "nonce_count", 1) # enable nonce_count support

modparam("auth", "qop", "auth")    # enable qop=auth

modparam("auth", "nonce_expire", 60)

modparam("auth", "nonce_auth_max_drift", 2)

# For REGISTER requests we hash the Request-URI, Call-ID, and source IP of the

# request into the nonce string. This ensures that the generated credentials

# cannot be used with another registrar, user agent with another source IP

# address or Call-ID. Note that user agents that change Call-ID with every

# REGISTER message will not be able to register if you enable this.

modparam("auth", "auth_checks_register", 11)

# For dialog-establishing requests (such as the original INVITE, OPTIONS, etc)

# we hash the Request-URI and source IP. Hashing Call-ID and From tags takes

# some extra precaution, because these checks could render some UA unusable.

modparam("auth", "auth_checks_no_dlg", 9)

# For mid-dialog requests, such as re-INVITE, we can hash source IP and

# Request-URI just like in the previous case. In addition to that we can hash

# Call-ID and From tag because these are fixed within a dialog and are

# guaranteed not to change. This settings effectively restrict the usage of

# generated credentials to a single user agent within a single dialog.

modparam("auth", "auth_checks_in_dlg", 15)

# ----- 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

# ----- speeddial 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

# ----- rtpengine params -----

modparam("rtpengine", "rtpengine_sock", "udp:165.22.15.111:5066")

modparam("rtpengine", "rtpengine_disable_tout", 20)

#modparam("rtpengine", "db_url", DBURL)

# ----- 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", "/etc/kamailio/tls.cfg")

modparam("tls", "private_key", "/etc/letsencrypt/live/callcenter.siarum.com/privkey.pem")

modparam("tls", "certificate", "/etc/letsencrypt/live/callcenter.siarum.com/fullchain.pem")

#modparam("tls", "ca_list", "/etc/httpd/certs/essentialSSL/wildcard.goautodial.com.ca-bundle")

#!endif

#!ifdef WITH_WEBSOCKETS

# ----- nathelper params -----

modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)")

# Note: leaving NAT pings turned off here as nathelper is _only_ being used for

#       WebSocket connections.  NAT pings are not needed as WebSockets have

#       their own keep-alives.

modparam("dialog", "dlg_flag", 10)

modparam("dialog", "track_cseq_updates", 0)

modparam("dialog", "dlg_match_mode", 2)

modparam("dialog", "timeout_avp", "$avp(i:10)")

# Set the sst modules timeout_avp to be the same value

modparam("sst", "timeout_avp", "$avp(i:10)")

modparam("sst", "sst_flag", 11)

#!endif

#!ifdef WITH_MSRP

# ----- htable params -----

modparam("htable", "htable", "msrp=>size=8;autoexpire=MSRP_MAX_EXPIRES;")

#!endif

#!ifdef WITH_ANTIFLOOD

# ----- pike params -----

modparam("pike", "sampling_time_unit", 2)

modparam("pike", "reqs_density_per_unit", 32)

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)

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 {

    # per request initial checks

    route(REQINIT);

#!ifdef WITH_WEBSOCKETS

    if (nat_uac_test(64)) {

        # Do NAT traversal stuff for requests from a WebSocket

        # connection - even if it is not behind a NAT!

        # This won't be needed in the future if Kamailio and the

        # WebSocket client support Outbound and Path.

        force_rport();

        if (is_method("REGISTER")) {

            fix_nated_register();

        } else {

            if (!add_contact_alias()) {

                xlog("L_ERR", "Error aliasing contact <$ct>\n");

                sl_send_reply("400", "Bad Request");

                exit;

            }

        }

    }

#!endif

    # NAT detection

    route(NATDETECT);

    # CANCEL processing

    if (is_method("CANCEL")) {

        if (t_check_trans()) {

            route(RELAY);

        }

        exit;

    }

    # handle requests within SIP dialogs

    route(WITHINDLG);

    ### only initial requests (no To tag)

    # handle retransmissions

    if(t_precheck_trans()) {

        t_check_trans();

        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

        setflag(10); # set the dialog flag

        setflag(11); # Set the sst flag

    }

    if (is_method("UPDATE")) {

        setflag(FLT_ACC); # do accounting

        setflag(10); # set the dialog flag

        setflag(11); # Set the sst flag

    }

    # 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);

}

# Wrapper for relaying requests

route[RELAY] {

    # enable additional event routes for forwarded requests

    # - serial forking, RTP relaying handling, a.s.o.

    if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) {

        if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH");

    }

    if (is_method("INVITE|SUBSCRIBE|UPDATE")) {

        if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY");

    }

    if (is_method("INVITE")) {

        dlg_manage();

        route(SETUP_BY_TRANSPORT);

        if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE");

    }

    if (!t_relay()) {

        sl_reply_error();

    }

    exit;

}

route[SETUP_BY_TRANSPORT] {

    if ($ru =~ "transport=ws") {

        xlog("L_INFO", "Request going to WS");

        if(sdp_with_transport("RTP/SAVPF")) {

            xlog("L_INFO", "RTP/SAVPF detected");

            rtpengine_manage("force trust-address replace-origin replace-session-connection ICE=force");

            t_on_reply("REPLY_WS_TO_WS");

            return;

        }

        rtpengine_manage("trust-address replace-origin replace-session-connection ICE=force RTP/SAVPF rtcp-mux-offer rtcp-mux-accept SDES-off");

        t_on_reply("REPLY_FROM_WS");

    }

    else if ($proto =~ "ws") {

        xlog("L_INFO", "Request coming from WS");

        rtpengine_manage("RTP/AVP");

        t_on_reply("REPLY_TO_WS");

    }

    else {

        xlog("L_INFO", "This is a classic phone call");

        rtpengine_manage("trust-address replace-origin replace-session-connection RTP/AVP");

        t_on_reply("MANAGE_CLASSIC_REPLY");

    }

}

onreply_route[REPLY_WS_TO_WS] {

    xlog("L_INFO", "WS to WS");

    if(status=~"[12][0-9][0-9]") {

        rtpengine_manage("force trust-address replace-origin replace-session-connection ICE=force");

        route(NATMANAGE);

    }

}

onreply_route[REPLY_FROM_WS] {

    xlog("L_INFO", "Reply from webrtc client: $rs");

    if(status=~"[12][0-9][0-9]") {

        rtpengine_manage("trust-address replace-origin replace-session-connection ICE=remove RTP/AVP rtcp-mux-offer rtcp-mux-accept SDES-off");

        route(NATMANAGE);

    }

}

onreply_route[REPLY_TO_WS] {

    xlog("L_INFO", "Reply from softphone: $rs");

    if (t_check_status("183")) {

        change_reply_status("180", "Ringing");

        remove_body();

        exit;

    }

    if(!(status=~"[12][0-9][0-9]"))

        return;

    rtpengine_manage("froc+SP");

    route(NATMANAGE);

}

onreply_route[MANAGE_CLASSIC_REPLY] {

    xlog("L_INFO", "Boring reply from softphone: $rs");

    if(status=~"[12][0-9][0-9]") {

        xlog("L_INFO", "rtpengine_manage - trust-address replace-origin replace-session-connection RTP/AVP");

        rtpengine_manage("trust-address replace-origin replace-session-connection RTP/AVP");    

                route(NATMANAGE);

    }

}

# 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;

        }

    }

    if($ua =~ "friendly-scanner") {

        sl_send_reply("200", "OK");

        exit;

    }

#!endif

    if (!mf_process_maxfwd_header("10")) {

        sl_send_reply("483","Too Many Hops");

        exit;

    }

    if(is_method("OPTIONS") && uri==myself && $rU==$null) {

        sl_send_reply("200","Keepalive");

        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()) return;

    # sequential request withing a dialog should

    # take the path determined by record-routing

    if (loose_route()) {

#!ifdef WITH_WEBSOCKETS

        if ($du == "") {

            if (!handle_ruri_alias()) {

                xlog("L_ERR", "Bad alias <$ru>\n");

                sl_send_reply("400", "Bad Request");

                exit;

            }

        }

#!endif

        route(DLGURI);

        if (is_method("BYE")) {

            setflag(FLT_ACC); # do accounting ...

            setflag(FLT_ACCFAILED); # ... even if the transaction fails

        }

        else if ( is_method("ACK") ) {

            # ACK is forwarded statelessy

            route(NATMANAGE);

        }

        else if ( is_method("NOTIFY") ) {

            # Add Record-Route for in-dialog NOTIFY as per RFC 6665.

            record_route();

        }

        route(RELAY);

        exit;

    }

    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

            route(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;

    if(isflagset(FLT_NATS)) {

        setbflag(FLB_NATB);

#!ifdef WITH_NATSIPPING

        # do SIP NAT pinging

        setbflag(FLB_NATSIPPING);

#!endif

    }

    if (!save("location", "0x04"))

        sl_reply_error();

    exit;

}

# User location service

route[LOCATION] {

#!ifdef WITH_SPEEDDIAL

    # 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

    $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);

    }

    # t_on_failure("UA_FAILURE");

    route(RELAY);

    exit;

}

# Presence server processing

route[PRESENCE] {

    if(!is_method("PUBLISH|SUBSCRIBE"))

        return;

    if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") {

        route(TOVOICEMAIL);

        # returns here if no voicemail server is configured

        sl_send_reply("404", "No voicemail service");

        exit;

    }

#!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;

}

# IP authorization and user uthentication

route[AUTH] {

#!ifdef WITH_AUTH

#!ifdef WITH_IPAUTH

    if((!is_method("REGISTER")) && allow_source_address()) {

        # source IP allowed

        return;

    }

#!endif

    if (is_method("REGISTER") || from_uri==myself)

    {

        # authenticate requests

        if (!auth_check("$fd", "subscriber", "1")) {

            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[NATDETECT] {

#!ifdef WITH_NAT

    force_rport();

    if (nat_uac_test("19")) {

        if (is_method("REGISTER")) {

            fix_nated_register();

        } else {

            if(is_first_hop())

                set_contact_alias();

        }

        setflag(FLT_NATS);

    }

#!endif

    return;

}

# RTPengine control and singaling updates for NAT traversal

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;

    if (is_request()) {

        if (!has_totag()) {

            if(t_is_branch_route()) {

                add_rr_param(";nat=yes");

            }

        }

    }

    if (is_reply()) {

        if(isbflagset(FLB_NATB)) {

            if(is_first_hop())

                set_contact_alias();

        }

    }

#!endif

    return;

}

# URI update for dialog requests

route[DLGURI] {

#!ifdef WITH_NAT

    if(!isdsturiset()) {

        handle_ruri_alias();

    }

#!endif

    return;

}

# Routing to foreign domains

route[SIPOUT] {

    if (uri==myself) return;

    append_hf("P-hint: outbound\r\n");

    route(RELAY);

    exit;

}

# 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;

    }

    if (strempty($sel(cfg_get.pstn.gw_port))) {

        $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip);

    } else {

        $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" 

                    + $sel(cfg_get.pstn.gw_port);

    }

    route(RELAY);

    exit;

#!endif

    return;

}

# XMLRPC routing

#!ifdef WITH_XMLRPC

route[XMLRPC] {

    # allow XMLRPC from localhost

    if ((method=="POST" || method=="GET")

            && (src_ip==165.22.15.111)) {

        # 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

# Routing to voicemail server

route[TOVOICEMAIL] {

#!ifdef WITH_VOICEMAIL

    if(!is_method("INVITE|SUBSCRIBE"))

        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(is_method("INVITE")) {

        if($avp(oexten)==$null)

            return;

        $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip)

                + ":" + $sel(cfg_get.voicemail.srv_port);

    } else {

        if($rU==$null)

            return;

        $ru = "sip:" + $rU + "@" + $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;

    }

#!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")) {

        $du = $null;

        route(TOVOICEMAIL);

        exit;

    }

#!endif

}

#!ifdef WITH_WEBSOCKETS

onreply_route {

    if ((($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT)

        && !(proto == WS || proto == WSS)) || $Rp == MY_MSRP_PORT) {

        xlog("L_WARN", "SIP response received on $Rp\n");

        drop;

        exit;

    }

    if (nat_uac_test(64)) {

        # Do NAT traversal stuff for replies to a WebSocket connection

        # - even if it is not behind a NAT!

        # This won't be needed in the future if Kamailio and the

        # WebSocket client support Outbound and Path.

        add_contact_alias();

    }

}

event_route[xhttp:request] {

    set_reply_close();

    set_reply_no_connect();

    if ($Rp != MY_WS_PORT

#!ifdef WITH_TLS

        && $Rp != MY_WSS_PORT

#!endif

    ) {

        xlog("L_WARN", "HTTP request received on $Rp\n");

        xhttp_reply("403", "Forbidden", "", "");

        exit;

    }

    xlog("L_DBG", "HTTP Request Received\n");

    if ($hdr(Upgrade)=~"websocket" 

            && $hdr(Connection)=~"Upgrade" 

            && $rm=~"GET") {

        # Validate Host - make sure the client is using the correct

        # alias for WebSockets

        # Sasa: commented out, see http://sip-router.1086192.n5.nabble.com/Testing-the-Websocket-module-with-sipml5-org-td65069.html

        #if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) {

        #    xlog("L_WARN", "Bad host $hdr(Host)\n");

        #    xhttp_reply("403", "Forbidden", "", "");

        #    exit;

        #}

        # Optional... validate Origin - make sure the client is from an

        # authorised website.  For example,

        #

        # if ($hdr(Origin) != "http://communicator.MY_DOMAIN" 

        #     && $hdr(Origin) != "https://communicator.MY_DOMAIN") {

        #    xlog("L_WARN", "Unauthorised client $hdr(Origin)\n");

        #    xhttp_reply("403", "Forbidden", "", "");

        #    exit;

        # }

        # Optional... perform HTTP authentication

        # ws_handle_handshake() exits (no further configuration file

        # processing of the request) when complete.

        if (ws_handle_handshake())

        {

            # Optional... cache some information about the

            # successful connection

            exit;

        }

    }

    xhttp_reply("404", "Not Found", "", "");

}

event_route[websocket:closed] {

    xlog("L_INFO", "WebSocket connection from $si:$sp has closed\n");

}

failure_route[UA_FAILURE] {

    xlog("L_INFO", "Triggered UA_FAILURE\n");

    if (t_check_status("488") && sdp_content()) {

        if (sdp_get_line_startswith("$avp(mline)", "m=")) {

            if ($avp(mline) =~ "SAVPF") {

                $avp(rtpengine_offer_flags) = "froc-sp";

                $avp(rtpengine_answer_flags) = "froc+SP";

            } else {

                $avp(rtpengine_offer_flags) = "froc+SP";

                $avp(rtpengine_answer_flags) = "froc-sp";

            }

        }

        append_branch();

        rtpengine_offer($avp(rtpengine_offer_flags));

        t_on_reply("RTPPROXY_REPLY");

        route(RELAY);

    }

}

onreply_route[RTPPROXY_REPLY] {

    xlog("L_INFO", "Triggered RTPPROXY_REPLY\n");

    if (status =~ "18[03]") {

        change_reply_status(180, "Ringing");

        remove_body();

    } else if (status =~ "2[0-9][0-9]" && sdp_content()) {

        rtpengine_answer($avp(rtpengine_answer_flags));

    }

}

#!endif

#!ifdef WITH_MSRP

event_route[msrp:frame-in] {

    msrp_reply_flags("1");

    if ((($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT)

        && !(proto == WS || proto == WSS)) && $Rp != MY_MSRP_PORT) {

        xlog("L_WARN", "MSRP request received on $Rp\n");

        msrp_reply("403", "Action-not-allowed");

        exit;

    }

    if (msrp_is_reply()) {

        msrp_relay();

    } else if($msrp(method)=="AUTH") {

        if($msrp(nexthops)>0) {

            msrp_relay();

            exit;

        }

        if (!www_authenticate("MY_DOMAIN", "subscriber",

                    "$msrp(method)")) {

            if (auth_get_www_authenticate("MY_DOMAIN", "1",

                            "$var(wauth)")) {

                msrp_reply("401", "Unauthorized",

                            "$var(wauth)");

            } else {

                msrp_reply("500", "Server Error");

            }

            exit;

        }

        if ($hdr(Expires) != $null) {

            $var(expires) = (int) $hdr(Expires);

            if ($var(expires) < MSRP_MIN_EXPIRES) {

                msrp_reply("423", "Interval Out-of-Bounds",

                    "Min-Expires: MSRP_MIN_EXPIRES\r\n");

                exit;

            } else if ($var(expires) > MSRP_MAX_EXPIRES) {

                msrp_reply("423", "Interval Out-of-Bounds",

                    "Max-Expires: MSRP_MAX_EXPIRES\r\n");

                exit;

            }

        } else {

            $var(expires) = MSRP_MAX_EXPIRES;

        }

        $var(cnt) = $var(cnt) + 1;

        pv_printf("$var(sessid)", "s.$(pp).$(var(cnt)).$(RANDOM)");

        $sht(msrp=>$var(sessid)::srcaddr) = $msrp(srcaddr);

        $sht(msrp=>$var(sessid)::srcsock) = $msrp(srcsock);

        $shtex(msrp=>$var(sessid)) = $var(expires) + 5;

        # - Use-Path: the MSRP address for server + session id

        $var(hdrs) = "Use-Path: msrps://MY_IP_ADDR:MY_MSRP_PORT/" 

                    + $var(sessid) + ";tcp\r\n" 

                    + "Expires: " + $var(expires) + "\r\n";

        msrp_reply("200", "OK", "$var(hdrs)");

    } else if ($msrp(method)=="SEND" || $msrp(method)=="REPORT") {

        if ($msrp(nexthops)>1) {

            if ($msrp(method)!="REPORT") {

                msrp_reply("200", "OK");

            }

            msrp_relay();

            exit;

        }

        $var(sessid) = $msrp(sessid);

        if ($sht(msrp=>$var(sessid)::srcaddr) == $null) {

            # one more hop, but we don't have address in htable

            msrp_reply("481", "Session-does-not-exist");

            exit;

        } else if ($msrp(method)!="REPORT") {

            msrp_reply("200", "OK");

        }

        msrp_relay_flags("1");

        msrp_set_dst("$sht(msrp=>$var(sessid)::srcaddr)",

                "$sht(msrp=>$var(sessid)::srcsock)");

        msrp_relay();

    } else {

        msrp_reply("501", "Request-method-not-understood");

    }

}

#!endif

</code></pre>

<tls.cfg>

<pre><code class="yaml">

#

# $Id$

#

# Example Kamailio TLS Configuration File

#

# This is the default server domain, settings

# in this domain will be used for all incoming

# connections that do not match any other server

# domain in this configuration file.

#

# We do not enable anything else than TLSv1

# over the public internet. Clients do not have

# to present client certificates by default.

#

[server:default]

method = TLSv1 

verify_certificate = no  

require_certificate = no

private_key = /etc/letsencrypt/live/callcenter.siarum.com/privkey.pem

certificate = /etc/letsencrypt/live/callcenter.siarum.com/fullchain.pem

#ca_list = /etc/letsencrypt/live/callcenter.siarum.com/fullchain.pem

#ca_list = /etc/ssl/certs/ca-bundle.crt

#crl = ./modules/tls/crl.pem

# This is the default client domain, settings

# in this domain will be used for all outgoing

# TLS connections that do not match any other

# client domain in this configuration file.

# We require that servers present valid certificate.

#

[client:default]

verify_certificate = no 

require_certificate = no 

# This is an example server domain for TLS connections

# received from the loopback interface. We allow

# the use of SSLv2 and SSLv3 protocols here, we do

# not require that clients present client certificates

# but if they present it it must be valid. We also use

# a special certificate and CA list for loopback

# interface.

#

#[server:127.0.0.1:5061]

#method = SSLv23

#verify_certificate = yes

#require_certificate = no

#private_key = ./modules/tls/local_key.pem

#certificate = ./modules/tls/local_cert.pem

#verify_depth = 3

#ca_list = local_ca.pem

#crl = local_crl.pem

# Special settings for the iptel.org public SIP

# server. We do not verify the certificate of the

# server because it can be expired. The server

# implements authentication using SSL client

# certificates so configure the client certificate

# that was given to use by iptel.org staff here.

#

#[client:195.37.77.101:5061]

#verify_certificate = no

#certificate = ./modules/tls/iptel_client.pem

#private_key = ./modules/tls/iptel_key.pem

#ca_list = ./modules/tls/iptel_ca.pem

#crl = ./modules/tls/iptel_crl.pem

</code></pre>

sip-goautodial.conf

<pre><code class="yaml">

[kamailio]

encryption=yes 

disallow=all

allow=opus

allow=ulaw

type=friend

dtmfmode=rfc2833

context=default

qualify=yes

nat=force_rport,comedia

host=callcenter.siarum.com 

insecure=port,invite

</code></pre>

kamctlrc

<pre><code class="yaml">

# The Kamailio configuration file for the control tools.

#

# Here you can set variables used in the kamctl and kamdbctl setup

# scripts. Per default all variables here are commented out, the control tools

# will use their internal default values.

## your SIP domain

SIP_DOMAIN=callcenter.siarum.com

## chrooted directory

# $CHROOT_DIR="/path/to/chrooted/directory"

## database type: MYSQL, PGSQL, ORACLE, DB_BERKELEY, DBTEXT, or SQLITE

# by default none is loaded

#

# If you want to setup a database with kamdbctl, you must at least specify

# this parameter.

DBENGINE=MYSQL

## database host

DBHOST=localhost

## database host

DBPORT=3306

## database name (for ORACLE this is TNS name)

DBNAME=kamailio

# database path used by dbtext, db_berkeley or sqlite

# DB_PATH="/usr/local/etc/kamailio/dbtext"

## database read/write user

DBRWUSER="kamailio"

## password for database read/write user

DBRWPW="kamailiorw"

## database read only user

DBROUSER="kamailioro"

## password for database read only user

DBROPW="kamailioro"

## database access host (from where is kamctl used)

# DBACCESSHOST=192.168.0.1

## database super user (for ORACLE this is 'scheme-creator' user)

DBROOTUSER="root"

## password for database super user

## - important: this is insecure, targeting the use only for automatic testing

## - known to work for: mysql

# DBROOTPW="dbrootpw"

## database character set (used by MySQL when creating database)

#CHARSET="latin1"

## user name column

# USERCOL="username"

# SQL definitions

# If you change this definitions here, then you must change them

# in db/schema/entities.xml too.

# FIXME

# FOREVER="2030-05-28 21:32:15"

# DEFAULT_Q="1.0"

# Program to calculate a message-digest fingerprint

# MD5="md5sum"

# awk tool

# AWK="awk"

# gdb tool

# GDB="gdb"

# If you use a system with a grep and egrep that is not 100% gnu grep compatible,

# e.g. solaris, install the gnu grep (ggrep) and specify this below.

#

# grep tool

# GREP="grep"

# egrep tool

# EGREP="egrep"

# sed tool

# SED="sed"

# tail tool

# LAST_LINE="tail -n 1"

# expr tool

# EXPR="expr"

# Describe what additional tables to install. Valid values for the variables

# below are yes/no/ask. With ask (default) it will interactively ask the user

# for an answer, while yes/no allow for automated, unassisted installs.

#

# If to install tables for the modules in the EXTRA_MODULES variable.

# INSTALL_EXTRA_TABLES=ask

# If to install presence related tables.

# INSTALL_PRESENCE_TABLES=ask

# If to install uid modules related tables.

# INSTALL_DBUID_TABLES=ask

# Define what module tables should be installed.

# If you use the postgres database and want to change the installed tables, then you

# must also adjust the STANDARD_TABLES or EXTRA_TABLES variable accordingly in the

# kamdbctl.base script.

# Kamailio standard modules

# STANDARD_MODULES="standard acc lcr domain group permissions registrar usrloc msilo

#                   alias_db uri_db speeddial avpops auth_db pdt dialog dispatcher

#                   dialplan"

# Kamailio extra modules

# EXTRA_MODULES="imc cpl siptrace domainpolicy carrierroute userblacklist htable purple sca"

## type of aliases used: DB - database aliases; UL - usrloc aliases

## - default: none

# ALIASES_TYPE="DB"

## control engine: RPCFIFO

## - default RPCFIFO

CTLENGINE="RPCFIFO"

## path to FIFO file for engine RPCFIFO

RPCFIFOPATH="/var/run/kamailio/kamailio_rpc.fifo"

## check ACL names; default on (1); off (0)

# VERIFY_ACL=1