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Docker and Goautodial , Registration Failed - Rejected

Added by beshoo beshoo 3 months ago

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', '');
?>

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

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: <> #
  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:")
  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 = :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:
;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::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.


Replies (34)

RE: Docker and Goautodial , Registration Failed - Rejected - Added by beshoo beshoo 3 months ago

I forgot to mention that when the agent tries to load the campaign and then logs into his dialer, he receives an error.

RE: Docker and Goautodial , Registration Failed - Rejected - Added by beshoo beshoo 3 months ago

Nothing happened when am running asterisk -r , no loges printed !

RE: Docker and Goautodial , Registration Failed - Rejected - Added by Demian Biscocho 3 months ago

Can you post your /etc/asterisk/sip-goautodial.conf?

RE: Docker and Goautodial , Registration Failed - Rejected - Added by beshoo beshoo 3 months ago

Demian Biscocho wrote in RE: Docker and Goautodial , Registration Failed - Rejected:

Can you post your /etc/asterisk/sip-goautodial.conf?

it seems kernel problem

RE: Docker and Goautodial , Registration Failed - Rejected - Added by beshoo beshoo 3 months ago

Demian Biscocho wrote in RE: Docker and Goautodial , Registration Failed - Rejected:

Can you post your /etc/asterisk/sip-goautodial.conf?

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

[root@v29878 conf.d]# cat /etc/asterisk/sip-goautodial.conf
[kamailio]
;encryption=yes ;uncomment for TLS encryption
disallow=all
allow=opus
allow=ulaw
type=friend
dtmfmode=rfc2833
context=default
qualify=yes
nat=force_rport,comedia
host=v29878.1blu.de. ;change me to my FQDN
insecure=port,invite

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

status -l asterisk.service
asterisk.service - Asterisk PBX and telephony daemon.
Loaded: loaded (/usr/lib/systemd/system/asterisk.service; disabled; vendor preset: disabled)
Active: active (running) since Thu 2023-12-07 09:09:10 PST; 55s ago
Process: 21410 ExecStop=/usr/sbin/asterisk -rx core stop now (code=exited, status=0/SUCCESS)
Main PID: 21415 (asterisk)
CGroup: /system.slice/asterisk.service
__21415 /usr/sbin/asterisk -f -C /etc/asterisk/asterisk.conf

Dec 07 09:09:11 v29878.1blu.de asterisk21415: [Dec 7 09:09:11] NOTICE21499: chan_sip.c:24586 handle_response_peerpoke: Peer 'kamailio' is now Reachable. (1ms / 2000ms)
Dec 07 09:09:11 v29878.1blu.de asterisk21415: [Dec 7 09:09:11] NOTICE21415: confbridge/conf_config_parser.c:2095 verify_default_profiles: Adding default_menu menu to app_confbridge
Dec 07 09:09:11 v29878.1blu.de asterisk21415: [Dec 7 09:09:11] NOTICE21415: cel_custom.c:97 load_config: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs.
Dec 07 09:09:11 v29878.1blu.de asterisk21415: [Dec 7 09:09:11] ERROR21415: pbx_dundi.c:5035 load_module: Unable to bind to 0.0.0.0 port 4520: Address already in use
Dec 07 09:09:11 v29878.1blu.de asterisk21415: [Dec 7 09:09:11] NOTICE21499: chan_sip.c:24586 handle_response_peerpoke: Peer 'ukrcall' is now Reachable. (75ms / 2000ms)
Dec 07 09:09:11 v29878.1blu.de asterisk21415: [Dec 7 09:09:11] ERROR21415: codec_dahdi.c:820 find_transcoders: Failed to open /dev/dahdi/transcode: No such file or directory
Dec 07 09:09:11 v29878.1blu.de asterisk21415: [Dec 7 09:09:11] NOTICE21415: app_conference.c:90 load_module: Loading app_konference module release=2.7
Dec 07 09:09:11 v29878.1blu.de asterisk21415: [Dec 7 09:09:11] ERROR21415: logger.c:1654 logger_queue_init: Unable to create queue log: Permission denied
Dec 07 09:09:13 v29878.1blu.de asterisk21415: [Dec 7 09:09:13] WARNING21476: db.c:332 ast_db_put: Couldn't execute statment: SQL logic error or missing database
Dec 07 09:09:23 v29878.1blu.de asterisk21415: [Dec 7 09:09:23] WARNING21458: db.c:332 ast_db_put: Couldn't execute statment: SQL logic error or missing database

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

RE: Docker and Goautodial , Registration Failed - Rejected - Added by beshoo beshoo 3 months ago

Demian Biscocho wrote in RE: Docker and Goautodial , Registration Failed - Rejected:

Can you post your /etc/asterisk/sip-goautodial.conf?

I note this in kamailio

_____________________
it enabled
Dec 07 09:26:51 v29878.1blu.de /usr/sbin/kamailio23234: INFO: rtpengine [rtpengine.c:2209]: rtpp_test(): rtp proxy <udp:127.0.0.1:5066> found, support for it enabled
Dec 07 09:26:51 v29878.1blu.de /usr/sbin/kamailio23239: INFO: rtpengine [rtpengine.c:2209]: rtpp_test(): rtp proxy <udp:127.0.0.1:5066> found, support for it enabled
Dec 07 09:26:51 v29878.1blu.de /usr/sbin/kamailio23238: INFO: rtpengine [rtpengine.c:2209]: rtpp_test(): rtp proxy <udp:127.0.0.1:5066> found, support for it enabled
Dec 07 09:26:51 v29878.1blu.de /usr/sbin/kamailio23240: INFO: rtpengine [rtpengine.c:2209]: rtpp_test(): rtp proxy <udp:127.0.0.1:5066> found, support for it enabled
Dec 07 09:26:51 v29878.1blu.de /usr/sbin/kamailio23241: INFO: rtpengine [rtpengine.c:2209]: rtpp_test(): rtp proxy <udp:127.0.0.1:5066> found, support for it enabled
Dec 07 09:27:28 v29878.1blu.de /usr/sbin/kamailio23237: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14094416:SSL routines:ssl3_read_bytes:sslv3 alert certificate unknown
Dec 07 09:27:28 v29878.1blu.de /usr/sbin/kamailio23237: ERROR: <core> [core/tcp_read.c:1352]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f213a0e0218 r: 0x7f213a0e0298
Dec 07 09:27:32 v29878.1blu.de /usr/sbin/kamailio23238: INFO: <script>: WebSocket connection from 5.0.9.229:6646 has closed

++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++

RE: Docker and Goautodial , Registration Failed - Rejected - Added by beshoo beshoo 3 months ago

Wittie Manansala wrote in RE: Docker and Goautodial , Registration Failed - Rejected:

Demian already replied to you on https://goautodial.org/boards/3/topics/21973.

Yes , Thank you so much for your time , I moved to a KVM VPS and installed it from scratch

All works as a system, but when itride to call my number. it just keep saying

And time out eventually

Unforchnitly the

Show nothing when i try to login to agent or do the call!

so am am not sure what I am missing here!

RE: Docker and Goautodial , Registration Failed - Rejected - Added by Wittie Manansala 3 months ago

From what I see, the IP for Kamailio in your 'sip show peers' has changed. You should have followed the guide at https://goautodial.org/projects/goautodialce/wiki/Goautodial_Getting_Started_Guidev4 so we could assist you.

Please also provide the changes you've made so we can follow along. Also, avoid posting details such as your server's public IP or VOIP configurations.

RE: Docker and Goautodial , Registration Failed - Rejected - Added by beshoo beshoo 3 months ago

Wittie Manansala wrote in RE: Docker and Goautodial , Registration Failed - Rejected:

From what I see, the IP for Kamailio in your 'sip show peers' has changed. You should have followed the guide at https://goautodial.org/projects/goautodialce/wiki/Goautodial_Getting_Started_Guidev4 so we could assist you.

Please also provide the changes you've made so we can follow along. Also, avoid posting details such as your server's public IP or VOIP configurations.

Apologies for any inconvenience. I’ve initiated a new thread as I’ve transitioned to a new KVM VPS. I want to assure you that I’ve adhered to the instructions.

However, I’m puzzled by the appearance of an unfamiliar IP in Kamailio’s “sip show peers”. I didn’t configure that IP; the Kamailio settings contain my personal IP address, which I’ve just verified.

It’s perplexing how an IP, seemingly associated with the VPS provider "hivelocity.net", is displayed in the “sip show peers”.

I attached my Kamailio conf

RE: Docker and Goautodial , Registration Failed - Rejected - Added by beshoo beshoo 2 months ago

Wittie Manansala wrote in RE: Docker and Goautodial , Registration Failed - Rejected:

From what I see, the IP for Kamailio in your 'sip show peers' has changed. You should have followed the guide at https://goautodial.org/projects/goautodialce/wiki/Goautodial_Getting_Started_Guidev4 so we could assist you.

Please also provide the changes you've made so we can follow along. Also, avoid posting details such as your server's public IP or VOIP configurations.

PING GOautodial.org (104.254.128.211)!!!

this IP belong to GOautodial.org!

RE: Docker and Goautodial , Registration Failed - Rejected - Added by beshoo beshoo 2 months ago

Wittie Manansala wrote in RE: Docker and Goautodial , Registration Failed - Rejected:

From what I see, the IP for Kamailio in your 'sip show peers' has changed. You should have followed the guide at https://goautodial.org/projects/goautodialce/wiki/Goautodial_Getting_Started_Guidev4 so we could assist you.

Please also provide the changes you've made so we can follow along. Also, avoid posting details such as your server's public IP or VOIP configurations.

I replace vaglxc01.goautodial.com with my domain...

But still I can not do any outpund call!

RE: Docker and Goautodial , Registration Failed - Rejected - Added by Wittie Manansala 2 months ago

Aside from following the steps outlined here: https://goautodial.org/projects/goautodialce/wiki/Goautodial_Getting_Started_Guidev4, could you also provide us with the adjustments you made? What specific settings or configurations did you change so we can follow along?

RE: Docker and Goautodial , Registration Failed - Rejected - Added by beshoo beshoo 2 months ago

Wittie Manansala wrote in RE: Docker and Goautodial , Registration Failed - Rejected:

Aside from following the steps outlined here: https://goautodial.org/projects/goautodialce/wiki/Goautodial_Getting_Started_Guidev4, could you also provide us with the adjustments you made? What specific settings or configurations did you change so we can follow along?

I did a fresh ISO installation and followed the documentation precisely

Then I replaced all vaglxc01.goautodial.com with my server domian from the following files

/etc/asterisk/sip-goautodial.conf
/etc/hostname
/etc/kamailio/kamailio.cfg
/etc/kamailio/kamctlrc

and removed vaglxc01.goautodial.com from
/etc/hosts

Reboot the server, now I can see

kamailio attached to my IP Address.

Now , I notes a new strange problem, after importing the list ... when I do call, the phone number is empty


And the log of the call as following:

And I still can not do any outbound call!

RE: Docker and Goautodial , Registration Failed - Rejected - Added by beshoo beshoo 2 months ago

I rebooted the server , the calling problem (Empty Number) has been resolved by it self...

But I can not do any outbound call till now ...here is the log

RE: Docker and Goautodial , Registration Failed - Rejected - Added by beshoo beshoo 2 months ago

I am not sure if this is related to the self-signed SSL certificate

RE: Docker and Goautodial , Registration Failed - Rejected - Added by beshoo beshoo 2 months ago

/etc/asterisk/sip-goautodial.conf

[kamailio]
;encryption=yes ;uncomment for TLS encryption
disallow=all
allow=opus
allow=ulaw
type=friend
dtmfmode=rfc2833
context=default
qualify=yes
nat=force_rport,comedia
host=v29878.1blu.de ;change me to my FQDN
insecure=port,invite

[root@v29878 download]# 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

RE: Docker and Goautodial , Registration Failed - Rejected - Added by beshoo beshoo 2 months ago

UPDATE.

I add my subdomain

to the /etc/hosts

now I tried to make a call, I heard a message from ayto dile saying that am the only one at this conference
when I call the number:

it is a great improvement but still can not call LOL

RE: Docker and Goautodial , Registration Failed - Rejected - Added by beshoo beshoo 2 months ago

UPDATE 2, sorry for the updates but am trying to put you on the situation!

RE: Docker and Goautodial , Registration Failed - Rejected - Added by beshoo beshoo 2 months ago

chan_sip.c:23990 handle_response_invite: Failed to authenticate on INVITE to '"S2312120727058600051" 

RE: Docker and Goautodial , Registration Failed - Rejected - Added by Wittie Manansala 2 months ago

When you log in, do you hear the voice prompt? You are currently the only person in this conference.

If you heard the voice prompt, it means there's no issue with your server setup. The problem might lie in your VOIP settings if you can't make calls. You can coordinate with your VOIP Provider to check if they see any call attempts coming from your server

RE: Docker and Goautodial , Registration Failed - Rejected - Added by beshoo beshoo 2 months ago

Yes i heard it one time only
You are currently the only person in this conference.

But is this sound has to be played each time?

Btw, i tested the voip via zoiper and it works just fine...

W

RE: Docker and Goautodial , Registration Failed - Rejected - Added by Wittie Manansala 2 months ago

The voice prompt (You are currently the only person in this conference) should always play when you log in as an agent. It's an indication or signal that the dialer is properly synced with the extensions.

RE: Docker and Goautodial , Registration Failed - Rejected - Added by beshoo beshoo 2 months ago

Yes, i thought on each page reload.
Well, yes it played each time.

But outbound call never get success.
And my voip is working over zoiper...

Am not sure what am missing here.

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