GOautodial Open Source Omni-channel Contact Center Suite

Dear All,
First of all, thank you for this great software (I hope it will works with me)

After overcoming numerous challenges, I have successfully installed your program in a Docker container and linked my domain to the container.

However, I am having difficulty understanding how the program operates within a Docker container. For your reference, here is the command I used to run the container:

```
docker run -d -it --privileged -p 8090:443 -p 5060:5060/udp -p 5060:5060 -p 5070:5070/udp -p 5070:5070 -p 4443:4443 -v /etc/ssl/certs_backup:/etc/ssl/certs/ --name=callcenter 63aafe564c6b /usr/sbin/init
```

As you can see, I have shared ports 5060, 5070, and 4443. Therefore, any connection going to reply.viralcaption.com:[PORT] will be forwarded to the Docker container.

please note that the docker public Ip is " 172.17.0.2"

and all Ports are routed to 172.17.0.2 container

So the WebRTC has to work correctly.

Now get back to the configuration file, I search the forum and here is the settings you usually asked for:

asterisk -rx "sip show peers"

Name/username Host Dyn Forcerport Comedia ACL Port Status Description
kamailio 104.254.128.211 Yes Yes 5060 OK (155 ms)

Am not sure what is this 104.254.128.211 Ip address, it is not my server IP address at all! nor doker IP address

1. GoAdmin Server Settings
2. Asterisk CLI during agent logging in
3. nano /var/www/html/php/Config.php
4. nano /var/www/html/php/goCRMAPISettings.php
5. nano /etc/kamailio/kamailio.cfg
6. nano /etc/rtpengine/rtpengine.conf
7. nano /etc/asterisk/sip.conf
8. GoAdmin Administration Settings & GoWebRTC Settings
9. ifconfig output

1- I don't understand how to Asterisk CLI during agent logging in, since when i

asterisk -r

Nothing happened...

3. nano /var/www/html/php/Config.php
// database configuration
define('DB_USERNAME', 'goautodialu');
define('DB_PASSWORD', 'goautodialu1234');
define('DB_HOST', 'localhost');
define('DB_NAME', 'goautodial');
define('DB_PORT', '3306');
define('DB_NAME_ASTERISK', 'asterisk');
define('DB_USERNAME_KAMAILIO', 'kamailiou');
define('DB_PASSWORD_KAMAILIO', 'kamailiou1234');
define('DB_HOST_KAMAILIO', 'localhost');
define('DB_NAME_KAMAILIO', 'kamailio');
define('DB_PORT_KAMAILIO', '3306');

// other configuration parameters
define('CRM_ADMIN_EMAIL', 'admin@localhost.com');
?>

========================================

nano /var/www/html/php/goCRMAPISettings.php

define ('gourl', 'https://reply.viralcaption.com/goAPIv2');
define ('goUser', 'goAPI');
define ('responsetype', 'json');
define ('goPass', 'pWUc1vR0z/coUmzI.oxkLANmGD19c3m');

?>

========================================

cat /etc/kamailio/kamailio.cfg
#!KAMAILIO #
#!define WITH_MYSQL
#!define WITH_AUTH
#!define WITH_USRLOCDB
#!define WITH_NAT
#!define WITH_ANTIFLOOD # #

1. Kamailio (OpenSER) SIP Server v5.0 - default configuration script
2. - web: http://www.kamailio.org
3. - git: http://sip-router.org #
4. Direct your questions about this file to: <sr-users@lists.sip-router.org> #
5. Refer to the Core CookBook at http://www.kamailio.org/wiki/
6. for an explanation of possible statements, functions and parameters. #
7. Several features can be enabled using '#!define WITH_FEATURE' directives: #
8. *** To run in debug mode:
9. - define WITH_DEBUG #
10. *** To enable mysql:
11. - define WITH_MYSQL #
12. *** To enable authentication execute:
13. - enable mysql
14. - define WITH_AUTH
15. - add users using 'kamctl' #
16. *** To enable IP authentication execute:
17. - enable mysql
18. - enable authentication
19. - define WITH_IPAUTH
20. - add IP addresses with group id '1' to 'address' table #
21. *** To enable persistent user location execute:
22. - enable mysql
23. - define WITH_USRLOCDB #
24. *** To enable presence server execute:
25. - enable mysql
26. - define WITH_PRESENCE #
27. *** To enable nat traversal execute:
28. - define WITH_NAT
29. - install RTPProxy: http://www.rtpproxy.org
30. - start RTPProxy:
31. rtpproxy -l your_public_ip -s udp:localhost:7722
32. - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING #
33. *** To enable PSTN gateway routing execute:
34. - define WITH_PSTN
35. - set the value of pstn.gw_ip
36. - check route[PSTN] for regexp routing condition #
37. *** To enable database aliases lookup execute:
38. - enable mysql
39. - define WITH_ALIASDB #
40. *** To enable speed dial lookup execute:
41. - enable mysql
42. - define WITH_SPEEDDIAL #
43. *** To enable multi-domain support execute:
44. - enable mysql
45. - define WITH_MULTIDOMAIN #
46. *** To enable TLS support execute:
47. - adjust CFGDIR/tls.cfg as needed
48. - define WITH_TLS #
49. *** To enable XMLRPC support execute:
50. - define WITH_XMLRPC
51. - adjust route[XMLRPC] for access policy #
52. *** To enable anti-flood detection execute:
53. - adjust pike and htable=>ipban settings as needed (default is
54. block if more than 16 requests in 2 seconds and ban for 300 seconds)
55. - define WITH_ANTIFLOOD #
56. *** To block 3XX redirect replies execute:
57. - define WITH_BLOCK3XX #
58. *** To enable VoiceMail routing execute:
59. - define WITH_VOICEMAIL
60. - set the value of voicemail.srv_ip
61. - adjust the value of voicemail.srv_port #
62. *** To enhance accounting execute:
63. - enable mysql
64. - define WITH_ACCDB
65. - add following columns to database
    #!ifdef ACCDB_COMMENT
    ALTER TABLE acc ADD COLUMN src_user VARCHAR NOT NULL DEFAULT '';
    ALTER TABLE acc ADD COLUMN src_domain VARCHAR NOT NULL DEFAULT '';
    ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default '';
    ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR NOT NULL DEFAULT '';
    ALTER TABLE acc ADD COLUMN dst_user VARCHAR NOT NULL DEFAULT '';
    ALTER TABLE acc ADD COLUMN dst_domain VARCHAR NOT NULL DEFAULT '';
    ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR NOT NULL DEFAULT '';
    ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR 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 NOT NULL DEFAULT '';
    ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR NOT NULL DEFAULT '';
    ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR NOT NULL DEFAULT '';
    #!endif

1. Include Local Config If Exists #########
   import_file "kamailio-local.cfg"

1. Defined Values #########

1. *** Value defines - IDs used later in config
   #!ifdef WITH_MYSQL
2. - database URL - used to connect to database server by modules such
3. as: auth_db, acc, usrloc, a.s.o.
   #!ifndef DBURL
   #!define DBURL "mysql://kamailiou:kamailiou1234@localhost/kamailio"
   #!endif
   #!endif
   #!ifdef WITH_MULTIDOMAIN
4. - the value for 'use_domain' parameters
   #!define MULTIDOMAIN 1
   #!else
   #!define MULTIDOMAIN 0
   #!endif

1. - flags
2. FLT_ - per transaction (message) flags
3. 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!172.17.0.2!g"
#!substdef "!MY_DOMAIN!vaglxc01.goautodial.com!g"
#!substdef "!MY_WS_PORT!8080!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

1. Global Parameters #########

1. 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="172.17.0.2"
alias="vaglxc01.goautodial.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:127.0.0.1:5060
listen=udp:172.17.0.2:5060

/* 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

tcp_connection_lifetime=3604
tcp_accept_no_cl=yes
tcp_rd_buf_size=16384

1. life time of TCP connection when there is no traffic
2. - a bit higher than registration expires to cope with UA behind NAT
   #tcp_connection_lifetime=3605

1. Custom Parameters #########

1. These parameters can be modified runtime via RPC interface
2. - see the documentation of 'cfg_rpc' module. #
3. Format: group.id = value 'desc' description
4. Access: $sel(cfg_get.group.id) or @cfg_get.group.id #

#!ifdef WITH_PSTN

1. PSTN GW Routing #
2. - pstn.gw_ip: valid IP or hostname as string value, example:
3. pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" #
4. - 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

1. VoiceMail Routing on offline, busy or no answer #
2. - 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

1. don't advertise server headers
   server_signature=no
   sip_warning=0

1. Modules Section ########

1. 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

1. ----------------- setting module-specific parameters ---------------

1. ---- topoh params -----
   #modparam("topoh", "mask_key", "Gu3ssWh@T1tS2016")
   #modparam("topoh", "mask_ip", "10.0.0.1")
   #modparam("topoh", "mask_callid", 1)

1. ----- mi_fifo params -----
   #modparam("mi_fifo", "fifo_name", "/var/run/kamailio/kamailio_fifo")

1. ----- 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")

1. ----- tm params -----
2. auto-discard branches from previous serial forking leg
   modparam("tm", "failure_reply_mode", 3)
3. default retransmission timeout: 30sec
   modparam("tm", "fr_timer", 30000)
4. default invite retransmission timeout after 1xx: 120sec
   modparam("tm", "fr_inv_timer", 120000)

1. ----- rr params -----
2. set next param to 1 to add value to ;lr param (helps with some UAs)
   modparam("rr", "enable_full_lr", 0)
3. do not append from tag to the RR (no need for this script)
   modparam("rr", "append_fromtag", 0)

1. ----- 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)
2. max value for expires of registrations
   modparam("registrar", "max_expires", 3600)
3. set it to 1 to enable GRUU
   modparam("registrar", "gruu_enabled", 0)

1. ----- 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

1. ----- 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

1. ----- 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)

1. For REGISTER requests we hash the Request-URI, Call-ID, and source IP of the
2. request into the nonce string. This ensures that the generated credentials
3. cannot be used with another registrar, user agent with another source IP
4. address or Call-ID. Note that user agents that change Call-ID with every
5. REGISTER message will not be able to register if you enable this.
   modparam("auth", "auth_checks_register", 11)

1. For dialog-establishing requests (such as the original INVITE, OPTIONS, etc)
2. we hash the Request-URI and source IP. Hashing Call-ID and From tags takes
3. some extra precaution, because these checks could render some UA unusable.
   modparam("auth", "auth_checks_no_dlg", 9)

1. For mid-dialog requests, such as re-INVITE, we can hash source IP and
2. Request-URI just like in the previous case. In addition to that we can hash
3. Call-ID and From tag because these are fixed within a dialog and are
4. guaranteed not to change. This settings effectively restrict the usage of
5. generated credentials to a single user agent within a single dialog.
   modparam("auth", "auth_checks_in_dlg", 15)

1. ----- permissions params -----
   #!ifdef WITH_IPAUTH
   modparam("permissions", "db_url", DBURL)
   modparam("permissions", "db_mode", 1)
   #!endif

#!endif

1. ----- alias_db params -----
   #!ifdef WITH_ALIASDB
   modparam("alias_db", "db_url", DBURL)
   modparam("alias_db", "use_domain", MULTIDOMAIN)
   #!endif

1. ----- speeddial params -----
   #!ifdef WITH_SPEEDDIAL
   modparam("speeddial", "db_url", DBURL)
   modparam("speeddial", "use_domain", MULTIDOMAIN)
   #!endif

1. ----- domain params -----
   #!ifdef WITH_MULTIDOMAIN
   modparam("domain", "db_url", DBURL)
2. register callback to match myself condition with domains list
   modparam("domain", "register_myself", 1)
   #!endif

#!ifdef WITH_PRESENCE

1. ----- presence params -----
   modparam("presence", "db_url", DBURL)

1. ----- presence_xml params -----
   modparam("presence_xml", "db_url", DBURL)
   modparam("presence_xml", "force_active", 1)
   #!endif

#!ifdef WITH_NAT

1. ----- rtpengine params -----
   modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:5066")
   modparam("rtpengine", "rtpengine_disable_tout", 20)
   #modparam("rtpengine", "db_url", DBURL)

1. ----- 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")

1. 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

1. ----- tls params -----
   modparam("tls", "config", "/etc/kamailio/tls.cfg")
   #modparam("tls", "private_key", "/etc/httpd/certs/essentialSSL/wildcard.goautodial.com.key")
   #modparam("tls", "certificate", "/etc/httpd/certs/essentialSSL/wildcard.goautodial.com.crt")
   #modparam("tls", "ca_list", "/etc/httpd/certs/essentialSSL/wildcard.goautodial.com.ca-bundle")
   #!endif

#!ifdef WITH_WEBSOCKETS

1. ----- nathelper params -----
   modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)")
2. Note: leaving NAT pings turned off here as nathelper is only being used for
3. WebSocket connections. NAT pings are not needed as WebSockets have
4. 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)")

1. 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

1. ----- htable params -----
   modparam("htable", "htable", "msrp=>size=8;autoexpire=MSRP_MAX_EXPIRES;")
   #!endif

#!ifdef WITH_ANTIFLOOD

1. ----- pike params -----
   modparam("pike", "sampling_time_unit", 2)
   modparam("pike", "reqs_density_per_unit", 32)
   modparam("pike", "remove_latency", 4)

1. ----- htable params -----
2. ip ban htable with autoexpire after 5 minutes
3. modparam("htable", "htable", "ipban=>size=8;autoexpire=300;")
   #!endif

#!ifdef WITH_XMLRPC

1. ----- xmlrpc params -----
   modparam("xmlrpc", "route", "XMLRPC");
   modparam("xmlrpc", "url_match", "^/RPC")
   #!endif

#!ifdef WITH_DEBUG

1. ----- debugger params -----
   modparam("debugger", "cfgtrace", 1)
   modparam("debugger", "log_level_name", "exec")
   #!endif

1. Routing Logic ########

1. Main SIP request routing logic
2. - processing of any incoming SIP request starts with this route
3. - note: this is the same as route { ... }
   request_route {
   1. 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

1. NAT detection
   route(NATDETECT);

1. CANCEL processing
   if (is_method("CANCEL")) {
   if (t_check_trans()) {
   route(RELAY);
   }
   exit;
   }

1. handle requests within SIP dialogs
   route(WITHINDLG);

1. only initial requests (no To tag)

1. handle retransmissions
   if(t_precheck_trans()) {
   t_check_trans();
   exit;
   }
   t_check_trans();

1. authentication
   route(AUTH);

1. record routing for dialog forming requests (in case they are routed)
2. - remove preloaded route headers
   remove_hf("Route");
   if (is_method("INVITE|SUBSCRIBE"))
   record_route();

1. 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
}

1. dispatch requests to foreign domains
   route(SIPOUT);

1. requests for my local domains

1. handle presence related requests
   route(PRESENCE);

1. handle registrations
   route(REGISTRAR);

if ($rU==$null) { # request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}

1. dispatch destinations to PSTN
   route(PSTN);

1. user location service
   route(LOCATION);
   route(RELAY);
   }

1. 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);
        }
}

1. 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;
   }
   }

1. Handle requests within SIP dialogs
   route[WITHINDLG] {
   if (!has_totag()) return;
   1. sequential request withing a dialog should
   2. 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;
   }

1. 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;
   }

1. 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;
}
}

1. when routing via usrloc, log the missed calls also
   if (is_method("INVITE")) {
   setflag(FLT_ACCMISSED);
   }

1. t_on_failure("UA_FAILURE");
   route(RELAY);
   exit;
   }

1. 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

1. if presence enabled, this part will not be executed
   if (is_method("PUBLISH") || $rU==$null) {
   sl_send_reply("404", "Not here");
   exit;
   }
   return;
   }

1. 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();
}

1. if caller is not local subscriber, then check if it calls
2. 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;
}

1. 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;
   }

1. 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;
   }

1. URI update for dialog requests
   route[DLGURI] {
   #!ifdef WITH_NAT
   if(!isdsturiset()) {
   handle_ruri_alias();
   }
   #!endif
   return;
   }

1. Routing to foreign domains
   route[SIPOUT] {
   if (uri==myself) return;

   append_hf("P-hint: outbound\r\n");
   route(RELAY);
   exit;
   }

1. PSTN GW routing
   route[PSTN] {
   #!ifdef WITH_PSTN # check if PSTN GW IP is defined
   if (strempty($sel(cfg_get.pstn.gw_ip))) {
   xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n");
   return;
   }
   1. route to PSTN dialed numbers starting with '+' or '00'
   2. (international format)
   3. - update the condition to match your dialing rules for PSTN routing
      if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$"))
      return;
   1. 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;
   }

1. 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

1. Routing to voicemail server
   route[TOVOICEMAIL] {
   #!ifdef WITH_VOICEMAIL
   if(!is_method("INVITE|SUBSCRIBE"))
   return;
   1. 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;
   }

1. Manage outgoing branches
   branch_route[MANAGE_BRANCH] {
   xdbg("new branch [$T_branch_idx] to $ru\n");
   route(NATMANAGE);
   }

1. Manage incoming replies
   onreply_route[MANAGE_REPLY] {
   xdbg("incoming reply\n");
   if(status=~"[12][0-9][0-9]")
   route(NATMANAGE);
   }

1. 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") {

1. Validate Host - make sure the client is using the correct
2. alias for WebSockets
3. 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))) {
4. xlog("L_WARN", "Bad host $hdr(Host)\n");
5. xhttp_reply("403", "Forbidden", "", "");
6. exit;
   #}

1. Optional... validate Origin - make sure the client is from an
2. authorised website. For example, #
3. if ($hdr(Origin) != "http://communicator.MY_DOMAIN"
4. && $hdr(Origin) != "https://communicator.MY_DOMAIN") {
5. xlog("L_WARN", "Unauthorised client $hdr(Origin)\n");
6. xhttp_reply("403", "Forbidden", "", "");
7. exit;
8. }

1. Optional... perform HTTP authentication

1. ws_handle_handshake() exits (no further configuration file
2. 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 =~ "18⁰³") {
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;

1. - 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

================================================

cat /etc/rtpengine/rtpengine.conf
[rtpengine]

table = 0

1. no-fallback = false
   1. for userspace forwarding only:
2. table = -1

1. a single interface:
   interface = 172.17.0.2
2. separate multiple interfaces with semicolons:

1. interface = internal/12.23.34.45;external/23.34.45.54
   1. for different advertised address:
2. interface = 12.23.34.45!23.34.45.56

listen-ng = 127.0.0.1:5066

1. listen-tcp = 25060
2. listen-udp = 12222

timeout = 60
silent-timeout = 3600
tos = 184
#control-tos = 184

1. delete-delay = 30
2. final-timeout = 10800

1. foreground = false
2. pidfile = /var/run/ngcp-rtpengine-daemon.pid
3. num-threads = 16

port-min = 30000
port-max = 50000

1. max-sessions = 5000

1. recording-dir = /var/spool/rtpengine
2. recording-method = proc
3. recording-format = raw

1. redis = 127.0.0.1:6379/5
2. redis-write = password@12.23.34.45:6379/42
3. redis-num-threads = 8
4. no-redis-required = false
5. redis-expires = 86400
6. redis-allowed-errors = -1
7. redis-disable-time = 10
8. redis-cmd-timeout = 0
9. redis-connect-timeout = 1000

1. b2b-url = http://127.0.0.1:8090/
2. xmlrpc-format = 0

1. log-level = 6
2. log-stderr = false
3. log-facility = daemon
4. log-facility-cdr = local0
5. log-facility-rtcp = local1

1. graphite = 127.0.0.1:9006
2. graphite-interval = 60
3. graphite-prefix = foobar.

1. homer = 172.17.0.2:65432
2. homer-protocol = udp
3. homer-id = 2001

1. sip-source = false
2. dtls-passive = false

#[rtpengine-testing]
#table = -1
#interface = 10.15.20.121
#listen-ng = 2223
#foreground = true
#log-stderr = true
#log-level = 7

==============================================

cat /etc/asterisk/sip.conf
[general]
context=trunkinbound ; Default context for incoming calls
allowguest=no ; Allow or reject guest calls (default is yes)
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
;realm=mydomain.tld ; Realm for digest authentication
bindport=5070 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;domain=mydomain.tld ; Set default domain for this host
;pedantic=yes ; Enable checking of tags in headers,
;tos_sip=cs3 ; Sets TOS for SIP packets.
;tos_audio=ef ; Sets TOS for RTP audio packets.
;tos_video=af41 ; Sets TOS for RTP video packets.
;maxexpiry=3600 ; Maximum allowed time of incoming registrations
;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
;defaultexpiry=120 ; Default length of incoming/outgoing registration
;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
;checkmwi=10 ; Default time between mailbox checks for peers
;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
;vmexten=voicemail ; dialplan extension to reach mailbox sets the
disallow=all ; First disallow all codecs
allow=ulaw ; Allow codecs in order of preference
allow=gsm
mohinterpret=default
mohsuggest=default
language=en ; Default language setting for all users/peers
relaxdtmf=yes ; Relax dtmf handling
trustrpid = no ; If Remote-Party-ID should be trusted
sendrpid = yes ; If Remote-Party-ID should be sent
progressinband=no ; If we should generate in-band ringing always
;useragent=Asterisk PBX ; Allows you to change the user agent string
;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
;compactheaders = yes ; send compact sip headers.
videosupport=no ; Turn on support for SIP video. You need to turn this on
;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
callevents=yes ; generate manager events when sip ua
;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches
;regcontext=sipregistrations
rtptimeout=20 ; Terminate call if 60 seconds of no RTP or RTCP activity
rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
;sipdebug = yes ; Turn on SIP debugging by default, from
;recordhistory=yes ; Record SIP history by default
;dumphistory=yes ; Dump SIP history at end of SIP dialogue
;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
notifyringing = yes ; Notify subscriptions on RINGING state (default: no)
notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
limitonpeers = yes ; Apply call limits on peers only. This will improve
;t38pt_udptl = yes ; Default false
;register => 1234:password@mysipprovider.com
;registertimeout=20 ; retry registration calls every 20 seconds (default)
;registerattempts=10 ; Number of registration attempts before we give up
;externip = 192.168.1.1 ; Address that we're going to put in outbound SIP
;externhost=test.test.com ; Alternatively you can specify a domain
;externrefresh=10 ; How often to refresh externhost if
;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
;localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
;localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
nat=force_rport,comedia ; Global NAT settings (Affects all peers and users)
canreinvite=no ; Asterisk by default tries to redirect the
;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
;rtsavesysname=yes ; Save systemname in realtime database at registration
;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
;ignoreregexpire=yes ; Enabling this setting has two functions:
;domain=mydomain.tld,mydomain-incoming
;domain=1.2.3.4 ; Add IP address as local domain
;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains
;autodomain=yes ; Turn this on to have Asterisk add local host
;fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
jbmaxsize = 100 ; Max length of the jitterbuffer in milliseconds.
jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
qualify=yes ; By default, qualify all peers at 2000ms
limitonpeer = yes ; enable call limit on a per peer basis, different from limitonpeers
session-timers=refuse ; Refuse WebRTC session timers

#include sip-vicidial.conf
#include sip-goautodial.conf

; register SIP account on remote machine if using SIP trunks
; register => testSIPtrunk:test@10.10.10.16:5060
;
; setup account for SIP trunking:
; [SIPtrunk]
; disallow=all
; allow=ulaw
; allow=alaw
; type=friend
; username=testSIPtrunk
; secret=test
; host=10.10.10.16
; dtmfmode=inband
; qualify=1000

==================================================================
GoAdmin Administration Settings & GoWebRTC Settings

==========================================

ifconfig
eth0: flags=4163<UP,BROADCAST,RUNNING,MULTICAST> mtu 1500
inet 172.17.0.2 netmask 255.255.0.0 broadcast 0.0.0.0
ether 02:42:ac:11:00:02 txqueuelen 0 (Ethernet)
RX packets 24009 bytes 8315178 (7.9 MiB)
RX errors 0 dropped 0 overruns 0 frame 0
TX packets 25358 bytes 19021925 (18.1 MiB)
TX errors 0 dropped 0 overruns 0 carrier 0 collisions 0

lo: flags=73<UP,LOOPBACK,RUNNING> mtu 65536
inet 127.0.0.1 netmask 255.0.0.0
loop txqueuelen 1000 (Local Loopback)
RX packets 31591 bytes 14254207 (13.5 MiB)
RX errors 0 dropped 0 overruns 0 frame 0
TX packets 31591 bytes 14254207 (13.5 MiB)
TX errors 0 dropped 0 overruns 0 carrier 0 collisions 0

====================================================

I note this : am not sure if it is related!
rtpengine
[1700624745.233173] ERR: FAILED TO CREATE KERNEL TABLE 0 (No such file or directory), KERNEL FORWARDING DISABLED

Please help.