GOautodial Open Source Omni-channel Contact Center Suite

Can you post the output of the following:

asterisk -rx "sip show peers" 

If it gets stuck to the Logging to your phone screen, this means that your webRTC phone is unable to register to Kamailio.

hi Demian
my case i'am able to login as agent001 and choose campaign but after phone webrtc is not connected
this my sip show peers

ivrstats*CLI> sip show peers
Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description
8001/8001                 (Unspecified)                            D  Yes        Yes            0        UNKNOWN
8002/8002                 (Unspecified)                            D  Yes        Yes            0        UNKNOWN
8003/8003                 (Unspecified)                            D  Yes        Yes            0        UNKNOWN
8004/8004                 (Unspecified)                            D  Yes        Yes            0        UNKNOWN
8005/8005                 (Unspecified)                            D  Yes        Yes            0        UNKNOWN
8006/8006                 (Unspecified)                            D  Yes        Yes            0        UNKNOWN
8007/8007                 (Unspecified)                            D  Yes        Yes            0        UNKNOWN
8008/8008                 (Unspecified)                            D  Yes        Yes            0        UNKNOWN
8009/8009                 (Unspecified)                            D  Yes        Yes            0        UNKNOWN
8010/8010                 (Unspecified)                            D  Yes        Yes            0        UNKNOWN
8011/8011                 (Unspecified)                            D  Yes        Yes            0        UNKNOWN
8012/8012                 (Unspecified)                            D  Yes        Yes            0        UNKNOWN
8013/8013                 (Unspecified)                            D  Yes        Yes            0        UNKNOWN
8014/8014                 (Unspecified)                            D  Yes        Yes            0        UNKNOWN
8015/8015                 (Unspecified)                            D  Yes        Yes            0        UNKNOWN
8016/8016                 (Unspecified)                            D  Yes        Yes            0        UNKNOWN
8017/8017                 (Unspecified)                            D  Yes        Yes            0        UNKNOWN
8018/8018                 (Unspecified)                            D  Yes        Yes            0        UNKNOWN
8019/8019                 (Unspecified)                            D  Yes        Yes            0        UNKNOWN
8020/8020                 (Unspecified)                            D  Yes        Yes            0        UNKNOWN
kamailio                  80.211.157.116                              Yes        Yes            5060     OK (1 ms)

In my case kamailio is showing UNREACHABLE but kamailio and asterisk both are running fine.
Mehdi doudou : can you please send you kamailio sip configuration, I am not sure what I am missing that it's not able to registered on Asterisk.

#!KAMAILIO
#
#!define WITH_MYSQL
#!define WITH_AUTH
#!define WITH_USRLOCDB
#!define WITH_NAT
#!define WITH_ANTIFLOOD
#
#
# Kamailio (OpenSER) SIP Server v5.0 - default configuration script
#     - web: http://www.kamailio.org
#     - git: http://sip-router.org
#
# Direct your questions about this file to: <[email protected]>
#
# Refer to the Core CookBook at http://www.kamailio.org/wiki/
# for an explanation of possible statements, functions and parameters.
#
# Several features can be enabled using '#!define WITH_FEATURE' directives:
#
# *** To run in debug mode: 
#     - define WITH_DEBUG
#
# *** To enable mysql: 
#     - define WITH_MYSQL
#
# *** To enable authentication execute:
#     - enable mysql
#     - define WITH_AUTH
#     - add users using 'kamctl'
#
# *** To enable IP authentication execute:
#     - enable mysql
#     - enable authentication
#     - define WITH_IPAUTH
#     - add IP addresses with group id '1' to 'address' table
#
# *** To enable persistent user location execute:
#     - enable mysql
#     - define WITH_USRLOCDB
#
# *** To enable presence server execute:
#     - enable mysql
#     - define WITH_PRESENCE
#
# *** To enable nat traversal execute:
#     - define WITH_NAT
#     - install RTPProxy: http://www.rtpproxy.org
#     - start RTPProxy:
#        rtpproxy -l _your_public_ip_ -s udp:localhost:7722
#     - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING
#
# *** To enable PSTN gateway routing execute:
#     - define WITH_PSTN
#     - set the value of pstn.gw_ip
#     - check route[PSTN] for regexp routing condition
#
# *** To enable database aliases lookup execute:
#     - enable mysql
#     - define WITH_ALIASDB
#
# *** To enable speed dial lookup execute:
#     - enable mysql
#     - define WITH_SPEEDDIAL
#
# *** To enable multi-domain support execute:
#     - enable mysql
#     - define WITH_MULTIDOMAIN
#
# *** To enable TLS support execute:
#     - adjust CFGDIR/tls.cfg as needed
#     - define WITH_TLS
#
# *** To enable XMLRPC support execute:
#     - define WITH_XMLRPC
#     - adjust route[XMLRPC] for access policy
#
# *** To enable anti-flood detection execute:
#     - adjust pike and htable=>ipban settings as needed (default is
#       block if more than 16 requests in 2 seconds and ban for 300 seconds)
#     - define WITH_ANTIFLOOD
#
# *** To block 3XX redirect replies execute:
#     - define WITH_BLOCK3XX
#
# *** To enable VoiceMail routing execute:
#     - define WITH_VOICEMAIL
#     - set the value of voicemail.srv_ip
#     - adjust the value of voicemail.srv_port
#
# *** To enhance accounting execute:
#     - enable mysql
#     - define WITH_ACCDB
#     - add following columns to database
#!ifdef ACCDB_COMMENT
  ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT '';
  ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT '';
  ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default '';
  ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT '';
  ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT '';
  ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT '';
  ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT '';
  ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT '';
  ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default '';
  ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT '';
  ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT '';
  ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT '';
#!endif

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

####### Defined Values #########

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

# - flags
#   FLT_ - per transaction (message) flags
#    FLB_ - per branch flags
#!define FLT_ACC 1
#!define FLT_ACCMISSED 2
#!define FLT_ACCFAILED 3
#!define FLT_NATS 5

#!define FLB_NATB 6
#!define FLB_NATSIPPING 7

#!substdef "!MY_IP_ADDR!80.211.157.116!g" 
#!substdef "!MY_DOMAIN!crm.ivrstats.cloud!g" 
#!substdef "!MY_WS_PORT!8081!g" 
#!substdef "!MY_WSS_PORT!4443!g" 
#!substdef "!MY_MSRP_PORT!9080!g" 
#!substdef "!MY_WS_ADDR!tcp:MY_IP_ADDR:MY_WS_PORT!g" 
#!substdef "!MY_WSS_ADDR!tls:MY_IP_ADDR:MY_WSS_PORT!g" 
#!substdef "!MY_MSRP_ADDR!tls:MY_IP_ADDR:MY_MSRP_PORT!g" 
#!substdef "!MSRP_MIN_EXPIRES!1800!g" 
#!substdef "!MSRP_MAX_EXPIRES!3600!g" 

#!define WITH_TLS
#!define WITH_WEBSOCKETS
#!define WITH_MSRP

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

### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR
#!ifdef WITH_DEBUG
debug=4
log_stderror=no
#!else
debug=2
log_stderror=no
#!endif

memdbg=5
memlog=5

log_facility=LOG_LOCAL0

fork=yes
children=4

/* uncomment the next line to disable TCP (default on) */
#disable_tcp=yes

/* uncomment the next line to disable the auto discovery of local aliases
   based on reverse DNS on IPs (default on) */
#auto_aliases=no

/* add local domain aliases */
alias="crm.ivrstats.cloud" 

/* 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:80.211.157.116:5060

/* 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

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

####### Custom Parameters #########

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

#!ifdef WITH_PSTN
# PSTN GW Routing
#
# - pstn.gw_ip: valid IP or hostname as string value, example:
# pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" 
#
# - by default is empty to avoid misrouting
pstn.gw_ip = "" desc "tos.cloud.goautodial.com GW Address" 
pstn.gw_port = "" desc "PSTN GW Port" 
#!endif

#!ifdef WITH_VOICEMAIL
# VoiceMail Routing on offline, busy or no answer
#
# - by default Voicemail server IP is empty to avoid misrouting
voicemail.srv_ip = "" desc "VoiceMail IP Address" 
voicemail.srv_port = "5060" desc "VoiceMail Port" 
#!endif

# don't advertise server headers
server_signature=no
sip_warning=0

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

# set paths to location of modules (to sources or installation folders)
#!ifdef WITH_SRCPATH
mpath="modules/" 
#!else
mpath="/usr/lib64/kamailio/modules/" 
#mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" 
#!endif

#!ifdef WITH_MYSQL
loadmodule "db_mysql.so" 
#!endif

#loadmodule "topoh.so" 
#loadmodule "mi_fifo.so" 
loadmodule "jsonrpcs.so" 
loadmodule "kex.so" 
loadmodule "corex.so" 
loadmodule "tm.so" 
loadmodule "tmx.so" 
loadmodule "sl.so" 
loadmodule "rr.so" 
loadmodule "pv.so" 
loadmodule "maxfwd.so" 
loadmodule "usrloc.so" 
loadmodule "registrar.so" 
loadmodule "textops.so" 
loadmodule "siputils.so" 
loadmodule "xlog.so" 
loadmodule "sanity.so" 
loadmodule "ctl.so" 
loadmodule "cfg_rpc.so" 
loadmodule "acc.so" 

#!ifdef WITH_AUTH
loadmodule "auth.so" 
loadmodule "auth_db.so" 
#!ifdef WITH_IPAUTH
loadmodule "permissions.so" 
#!endif
#!endif

#!ifdef WITH_ALIASDB
loadmodule "alias_db.so" 
#!endif

#!ifdef WITH_SPEEDDIAL
loadmodule "speeddial.so" 
#!endif

#!ifdef WITH_MULTIDOMAIN
loadmodule "domain.so" 
#!endif

#!ifdef WITH_PRESENCE
loadmodule "presence.so" 
loadmodule "presence_xml.so" 
#!endif

#!ifdef WITH_NAT
loadmodule "nathelper.so" 
loadmodule "rtpengine.so" 
#loadmodule "rtpproxy.so" 
#!endif

#!ifdef WITH_TLS
loadmodule "tls.so" 
#!endif

#!ifdef WITH_MSRP
loadmodule "msrp.so" 
#loadmodule "htable.so" 
loadmodule "cfgutils.so" 
#!endif

#!ifdef WITH_WEBSOCKETS
loadmodule "xhttp.so" 
loadmodule "websocket.so" 
loadmodule "sdpops.so" 
loadmodule "textopsx.so" 
loadmodule "dialog.so" 
loadmodule "sst.so" 
#!endif

#!ifdef WITH_ANTIFLOOD
loadmodule "htable.so" 
loadmodule "pike.so" 
#!endif

#!ifdef WITH_XMLRPC
loadmodule "xmlrpc.so" 
#!endif

#!ifdef WITH_DEBUG
loadmodule "debugger.so" 
#!endif

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

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

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

# ----- jsonrpcs params -----
modparam("jsonrpcs", "pretty_format", 1)
/* set the path to RPC fifo control file */
modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo")
/* set the path to RPC unix socket control file */
modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock")

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

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

# ----- registrar params -----
modparam("registrar", "method_filtering", 1)
/* uncomment the next line to disable parallel forking via location */
modparam("registrar", "append_branches", 0)
/* uncomment the next line not to allow more than 100 contacts per AOR */
modparam("registrar", "max_contacts", 100)
# max value for expires of registrations
modparam("registrar", "max_expires", 3600)
# set it to 1 to enable GRUU
modparam("registrar", "gruu_enabled", 0)

# ----- acc params -----
/* what special events should be accounted ? */
modparam("acc", "early_media", 0)
modparam("acc", "report_ack", 0)
modparam("acc", "report_cancels", 0)
/* by default ww do not adjust the direct of the sequential requests.
   if you enable this parameter, be sure the enable "append_fromtag" 
   in "rr" module */
modparam("acc", "detect_direction", 0)
/* account triggers (flags) */
modparam("acc", "log_flag", FLT_ACC)
modparam("acc", "log_missed_flag", FLT_ACCMISSED)
modparam("acc", "log_extra", 
    "src_user=$fU;src_domain=$fd;src_ip=$si;" 
    "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
modparam("acc", "failed_transaction_flag", FLT_ACCFAILED)
/* enhanced DB accounting */
#!ifdef WITH_ACCDB
modparam("acc", "db_flag", FLT_ACC)
modparam("acc", "db_missed_flag", FLT_ACCMISSED)
modparam("acc", "db_url", DBURL)
modparam("acc", "db_extra",
    "src_user=$fU;src_domain=$fd;src_ip=$si;" 
    "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
#!endif

# ----- usrloc params -----
/* enable DB persistency for location entries */
#!ifdef WITH_USRLOCDB
modparam("usrloc", "db_url", DBURL)
modparam("usrloc", "db_mode", 1)
modparam("usrloc", "use_domain", MULTIDOMAIN)
modparam("usrloc", "timer_interval", 60)
modparam("usrloc", "timer_procs", 4)
#!endif

# ----- auth_db params -----
#!ifdef WITH_AUTH
modparam("auth_db", "db_url", DBURL)
modparam("auth_db", "calculate_ha1", 0)
modparam("auth_db", "password_column", "ha1")
modparam("auth_db", "load_credentials", "")
modparam("auth_db", "use_domain", MULTIDOMAIN)

modparam("auth", "nonce_count", 1) # enable nonce_count support
modparam("auth", "qop", "auth")    # enable qop=auth
modparam("auth", "nonce_expire", 60)
modparam("auth", "nonce_auth_max_drift", 2)

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

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

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

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

#!endif

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

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

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

#!ifdef WITH_PRESENCE
# ----- presence params -----
modparam("presence", "db_url", DBURL)

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

#!ifdef WITH_NAT
# ----- rtpengine params -----
modparam("rtpengine", "rtpengine_sock", "udp:80.211.157.116:5066")
modparam("rtpengine", "rtpengine_disable_tout", 20)
#modparam("rtpengine", "db_url", DBURL)

# ----- nathelper params -----
modparam("nathelper", "natping_interval", 30)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_bflag", FLB_NATSIPPING)
modparam("nathelper", "sipping_from", "sip:[email protected]")

# params needed for NAT traversal in other modules
modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)")
modparam("usrloc", "nat_bflag", FLB_NATB)
#!endif

#!ifdef WITH_TLS
# ----- tls params -----
modparam("tls", "config", "/etc/kamailio/tls.cfg")
#modparam("tls", "private_key", "/etc/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
# ----- nathelper params -----
modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)")
# Note: leaving NAT pings turned off here as nathelper is _only_ being used for
#       WebSocket connections.  NAT pings are not needed as WebSockets have
#       their own keep-alives.
modparam("dialog", "dlg_flag", 10)
modparam("dialog", "track_cseq_updates", 0)
modparam("dialog", "dlg_match_mode", 2)

modparam("dialog", "timeout_avp", "$avp(i:10)")
# Set the sst modules timeout_avp to be the same value
modparam("sst", "timeout_avp", "$avp(i:10)")
modparam("sst", "sst_flag", 11)
#!endif

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

#!ifdef WITH_ANTIFLOOD
# ----- pike params -----
modparam("pike", "sampling_time_unit", 2)
modparam("pike", "reqs_density_per_unit", 32)
modparam("pike", "remove_latency", 4)

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

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

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

####### Routing Logic ########

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

    # per request initial checks
    route(REQINIT);

#!ifdef WITH_WEBSOCKETS
    if (nat_uac_test(64)) {
        # Do NAT traversal stuff for requests from a WebSocket
        # connection - even if it is not behind a NAT!
        # This won't be needed in the future if Kamailio and the
        # WebSocket client support Outbound and Path.
        force_rport();
        if (is_method("REGISTER")) {
            fix_nated_register();
        } else {
            if (!add_contact_alias()) {
                xlog("L_ERR", "Error aliasing contact <$ct>\n");
                sl_send_reply("400", "Bad Request");
                exit;
            }
        }
    }
#!endif

    # NAT detection
    route(NATDETECT);

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

    # handle requests within SIP dialogs
    route(WITHINDLG);

    ### only initial requests (no To tag)

    # handle retransmissions
    if(t_precheck_trans()) {
        t_check_trans();
        exit;
    }
    t_check_trans();

    # authentication
    route(AUTH);

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

    # account only INVITEs
    if (is_method("INVITE")) {
        setflag(FLT_ACC); # do accounting
        setflag(10); # set the dialog flag
        setflag(11); # Set the sst flag
    }

    if (is_method("UPDATE")) {
        setflag(FLT_ACC); # do accounting
        setflag(10); # set the dialog flag
        setflag(11); # Set the sst flag
    }

    # dispatch requests to foreign domains
    route(SIPOUT);

    ### requests for my local domains

    # handle presence related requests
    route(PRESENCE);

    # handle registrations
    route(REGISTRAR);

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

    # dispatch destinations to PSTN
    route(PSTN);

    # user location service
    route(LOCATION);
    route(RELAY);
}

# Wrapper for relaying requests
route[RELAY] {
    # enable additional event routes for forwarded requests
    # - serial forking, RTP relaying handling, a.s.o.
    if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) {
        if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH");
    }
    if (is_method("INVITE|SUBSCRIBE|UPDATE")) {
        if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY");
    }
    if (is_method("INVITE")) {
        dlg_manage();
        route(SETUP_BY_TRANSPORT);
        if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE");
    }
    if (!t_relay()) {
        sl_reply_error();
    }
    exit;
}

route[SETUP_BY_TRANSPORT] {
    if ($ru =~ "transport=ws") {
        xlog("L_INFO", "Request going to WS");
        if(sdp_with_transport("RTP/SAVPF")) {
            xlog("L_INFO", "RTP/SAVPF detected");
            rtpengine_manage("force trust-address replace-origin replace-session-connection ICE=force");
            t_on_reply("REPLY_WS_TO_WS");
            return;
        }
        rtpengine_manage("trust-address replace-origin replace-session-connection ICE=force RTP/SAVPF rtcp-mux-offer rtcp-mux-accept SDES-off");
        t_on_reply("REPLY_FROM_WS");
    }
    else if ($proto =~ "ws") {
        xlog("L_INFO", "Request coming from WS");
        rtpengine_manage("RTP/AVP");
        t_on_reply("REPLY_TO_WS");
    }
    else {
        xlog("L_INFO", "This is a classic phone call");
        rtpengine_manage("trust-address replace-origin replace-session-connection RTP/AVP");
        t_on_reply("MANAGE_CLASSIC_REPLY");
    }
}

onreply_route[REPLY_WS_TO_WS] {
    xlog("L_INFO", "WS to WS");
    if(status=~"[12][0-9][0-9]") {
        rtpengine_manage("force trust-address replace-origin replace-session-connection ICE=force");
        route(NATMANAGE);
    }
}

onreply_route[REPLY_FROM_WS] {
    xlog("L_INFO", "Reply from webrtc client: $rs");
    if(status=~"[12][0-9][0-9]") {
        rtpengine_manage("trust-address replace-origin replace-session-connection ICE=remove RTP/AVP rtcp-mux-offer rtcp-mux-accept SDES-off");
        route(NATMANAGE);
    }
}

onreply_route[REPLY_TO_WS] {
    xlog("L_INFO", "Reply from softphone: $rs");

    if (t_check_status("183")) {
        change_reply_status("180", "Ringing");
        remove_body();
        exit;
    }

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

    rtpengine_manage("froc+SP");
    route(NATMANAGE);
}

onreply_route[MANAGE_CLASSIC_REPLY] {
    xlog("L_INFO", "Boring reply from softphone: $rs");

    if(status=~"[12][0-9][0-9]") {
        xlog("L_INFO", "rtpengine_manage - trust-address replace-origin replace-session-connection RTP/AVP");
        rtpengine_manage("trust-address replace-origin replace-session-connection RTP/AVP");    
                route(NATMANAGE);
    }
}

# Per SIP request initial checks
route[REQINIT] {
#!ifdef WITH_ANTIFLOOD
    # flood dection from same IP and traffic ban for a while
    # be sure you exclude checking trusted peers, such as pstn gateways
    # - local host excluded (e.g., loop to self)
    if(src_ip!=myself) {
        if($sht(ipban=>$si)!=$null) {
            # ip is already blocked
            xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n");
            exit;
        }
        if (!pike_check_req()) {
            xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n");
            $sht(ipban=>$si) = 1;
            exit;
        }
    }
    if($ua =~ "friendly-scanner") {
        sl_send_reply("200", "OK");
        exit;
    }
#!endif

    if (!mf_process_maxfwd_header("10")) {
        sl_send_reply("483","Too Many Hops");
        exit;
    }

    if(is_method("OPTIONS") && uri==myself && $rU==$null) {
        sl_send_reply("200","Keepalive");
        exit;
    }

    if(!sanity_check("1511", "7")) {
        xlog("Malformed SIP message from $si:$sp\n");
        exit;
    }
}

# Handle requests within SIP dialogs
route[WITHINDLG] {
    if (!has_totag()) return;

    # sequential request withing a dialog should
    # take the path determined by record-routing
    if (loose_route()) {
#!ifdef WITH_WEBSOCKETS
        if ($du == "") {
            if (!handle_ruri_alias()) {
                xlog("L_ERR", "Bad alias <$ru>\n");
                sl_send_reply("400", "Bad Request");
                exit;
            }
        }
#!endif
        route(DLGURI);
        if (is_method("BYE")) {
            setflag(FLT_ACC); # do accounting ...
            setflag(FLT_ACCFAILED); # ... even if the transaction fails
        }
        else if ( is_method("ACK") ) {
            # ACK is forwarded statelessy
            route(NATMANAGE);
        }
        else if ( is_method("NOTIFY") ) {
            # Add Record-Route for in-dialog NOTIFY as per RFC 6665.
            record_route();
        }
        route(RELAY);
        exit;
    }

    if (is_method("SUBSCRIBE") && uri == myself) {
        # in-dialog subscribe requests
        route(PRESENCE);
        exit;
    }
    if ( is_method("ACK") ) {
        if ( t_check_trans() ) {
            # no loose-route, but stateful ACK;
            # must be an ACK after a 487
            # or e.g. 404 from upstream server
            route(RELAY);
            exit;
        } else {
            # ACK without matching transaction ... ignore and discard
            exit;
        }
    }
    sl_send_reply("404","Not here");
    exit;
}

# Handle SIP registrations
route[REGISTRAR] {
    if (!is_method("REGISTER")) return;

    if(isflagset(FLT_NATS)) {
        setbflag(FLB_NATB);
#!ifdef WITH_NATSIPPING
        # do SIP NAT pinging
        setbflag(FLB_NATSIPPING);
#!endif
    }
    if (!save("location", "0x04"))
        sl_reply_error();
    exit;
}

# User location service
route[LOCATION] {

#!ifdef WITH_SPEEDDIAL
    # search for short dialing - 2-digit extension
    if($rU=~"^[0-9][0-9]$")
        if(sd_lookup("speed_dial"))
            route(SIPOUT);
#!endif

#!ifdef WITH_ALIASDB
    # search in DB-based aliases
    if(alias_db_lookup("dbaliases"))
        route(SIPOUT);
#!endif

    $avp(oexten) = $rU;
    if (!lookup("location")) {
        $var(rc) = $rc;
        route(TOVOICEMAIL);
        t_newtran();
        switch ($var(rc)) {
            case -1:
            case -3:
                send_reply("404", "Not Found");
                exit;
            case -2:
                send_reply("405", "Method Not Allowed");
                exit;
        }
    }

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

    # t_on_failure("UA_FAILURE");
    route(RELAY);
    exit;
}

# Presence server processing
route[PRESENCE] {
    if(!is_method("PUBLISH|SUBSCRIBE"))
        return;

    if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") {
        route(TOVOICEMAIL);
        # returns here if no voicemail server is configured
        sl_send_reply("404", "No voicemail service");
        exit;
    }

#!ifdef WITH_PRESENCE
    if (!t_newtran()) {
        sl_reply_error();
        exit;
    }

    if(is_method("PUBLISH")) {
        handle_publish();
        t_release();
    } else if(is_method("SUBSCRIBE")) {
        handle_subscribe();
        t_release();
    }
    exit;
#!endif

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

# IP authorization and user uthentication
route[AUTH] {
#!ifdef WITH_AUTH

#!ifdef WITH_IPAUTH
    if((!is_method("REGISTER")) && allow_source_address()) {
        # source IP allowed
        return;
    }
#!endif

    if (is_method("REGISTER") || from_uri==myself)
    {
        # authenticate requests
        if (!auth_check("$fd", "subscriber", "1")) {
            auth_challenge("$fd", "0");
            exit;
        }        
        # user authenticated - remove auth header
        if(!is_method("REGISTER|PUBLISH"))
            consume_credentials();
    }
    # if caller is not local subscriber, then check if it calls
    # a local destination, otherwise deny, not an open relay here
    if (from_uri!=myself && uri!=myself) {
        sl_send_reply("403","Not relaying");
        exit;
    }

#!endif
    return;
}

# Caller NAT detection
route[NATDETECT] {
#!ifdef WITH_NAT
    force_rport();
    if (nat_uac_test("19")) {
        if (is_method("REGISTER")) {
            fix_nated_register();
        } else {
            if(is_first_hop())
                set_contact_alias();
        }
        setflag(FLT_NATS);
    }
#!endif
    return;
}

# RTPengine control and singaling updates for NAT traversal
route[NATMANAGE] {
#!ifdef WITH_NAT
    if (is_request()) {
        if(has_totag()) {
            if(check_route_param("nat=yes")) {
                setbflag(FLB_NATB);
            }
        }
    }
    if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
        return;

    if (is_request()) {
        if (!has_totag()) {
            if(t_is_branch_route()) {
                add_rr_param(";nat=yes");
            }
        }
    }
    if (is_reply()) {
        if(isbflagset(FLB_NATB)) {
            if(is_first_hop())
                set_contact_alias();
        }
    }
#!endif
    return;
}

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

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

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

# PSTN GW routing
route[PSTN] {
#!ifdef WITH_PSTN
    # check if PSTN GW IP is defined
    if (strempty($sel(cfg_get.pstn.gw_ip))) {
        xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n");
        return;
    }

    # route to PSTN dialed numbers starting with '+' or '00'
    #     (international format)
    # - update the condition to match your dialing rules for PSTN routing
    if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$"))
        return;

    # only local users allowed to call
    if(from_uri!=myself) {
        sl_send_reply("403", "Not Allowed");
        exit;
    }

    if (strempty($sel(cfg_get.pstn.gw_port))) {
        $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip);
    } else {
        $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" 
                    + $sel(cfg_get.pstn.gw_port);
    }

    route(RELAY);
    exit;
#!endif

    return;
}

# XMLRPC routing
#!ifdef WITH_XMLRPC
route[XMLRPC] {
    # allow XMLRPC from localhost
    if ((method=="POST" || method=="GET")
            && (src_ip==80.211.157.116)) {
        # close connection only for xmlrpclib user agents (there is a bug in
        # xmlrpclib: it waits for EOF before interpreting the response).
        if ($hdr(User-Agent) =~ "xmlrpclib")
            set_reply_close();
        set_reply_no_connect();
        dispatch_rpc();
        exit;
    }
    send_reply("403", "Forbidden");
    exit;
}
#!endif

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

    # check if VoiceMail server IP is defined
    if (strempty($sel(cfg_get.voicemail.srv_ip))) {
        xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n");
        return;
    }
    if(is_method("INVITE")) {
        if($avp(oexten)==$null)
            return;
        $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip)
                + ":" + $sel(cfg_get.voicemail.srv_port);
    } else {
        if($rU==$null)
            return;
        $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip)
                + ":" + $sel(cfg_get.voicemail.srv_port);
    }
    route(RELAY);
    exit;
#!endif

    return;
}

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

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

# Manage failure routing cases
failure_route[MANAGE_FAILURE] {
    route(NATMANAGE);

    if (t_is_canceled()) {
        exit;
    }

#!ifdef WITH_BLOCK3XX
    # block call redirect based on 3xx replies.
    if (t_check_status("3[0-9][0-9]")) {
        t_reply("404","Not found");
        exit;
    }
#!endif

#!ifdef WITH_VOICEMAIL
    # serial forking
    # - route to voicemail on busy or no answer (timeout)
    if (t_check_status("486|408")) {
        $du = $null;
        route(TOVOICEMAIL);
        exit;
    }
#!endif
}

#!ifdef WITH_WEBSOCKETS
onreply_route {
    if ((($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT)
        && !(proto == WS || proto == WSS)) || $Rp == MY_MSRP_PORT) {
        xlog("L_WARN", "SIP response received on $Rp\n");
        drop;
        exit;
    }

    if (nat_uac_test(64)) {
        # Do NAT traversal stuff for replies to a WebSocket connection
        # - even if it is not behind a NAT!
        # This won't be needed in the future if Kamailio and the
        # WebSocket client support Outbound and Path.
        add_contact_alias();
    }
}

event_route[xhttp:request] {
    set_reply_close();
    set_reply_no_connect();

    if ($Rp != MY_WS_PORT
#!ifdef WITH_TLS
        && $Rp != MY_WSS_PORT
#!endif
    ) {
        xlog("L_WARN", "HTTP request received on $Rp\n");
        xhttp_reply("403", "Forbidden", "", "");
        exit;
    }

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

    if ($hdr(Upgrade)=~"websocket" 
            && $hdr(Connection)=~"Upgrade" 
            && $rm=~"GET") {

        # Validate Host - make sure the client is using the correct
        # alias for WebSockets
        # Sasa: commented out, see http://sip-router.1086192.n5.nabble.com/Testing-the-Websocket-module-with-sipml5-org-td65069.html
        #if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) {
        #    xlog("L_WARN", "Bad host $hdr(Host)\n");
        #    xhttp_reply("403", "Forbidden", "", "");
        #    exit;
        #}

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

        # Optional... perform HTTP authentication

        # ws_handle_handshake() exits (no further configuration file
        # processing of the request) when complete.
        if (ws_handle_handshake())
        {
            # Optional... cache some information about the
            # successful connection
            exit;
        }
    }

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

event_route[websocket:closed] {
    xlog("L_INFO", "WebSocket connection from $si:$sp has closed\n");
}

failure_route[UA_FAILURE] {
    xlog("L_INFO", "Triggered UA_FAILURE\n");
    if (t_check_status("488") && sdp_content()) {
        if (sdp_get_line_startswith("$avp(mline)", "m=")) {
            if ($avp(mline) =~ "SAVPF") {
                $avp(rtpengine_offer_flags) = "froc-sp";
                $avp(rtpengine_answer_flags) = "froc+SP";
            } else {
                $avp(rtpengine_offer_flags) = "froc+SP";
                $avp(rtpengine_answer_flags) = "froc-sp";
            }
        }
        append_branch();
        rtpengine_offer($avp(rtpengine_offer_flags));
        t_on_reply("RTPPROXY_REPLY");
        route(RELAY);
    }
}

onreply_route[RTPPROXY_REPLY] {
    xlog("L_INFO", "Triggered RTPPROXY_REPLY\n");
    if (status =~ "18[03]") {
        change_reply_status(180, "Ringing");
        remove_body();
    } else if (status =~ "2[0-9][0-9]" && sdp_content()) {
        rtpengine_answer($avp(rtpengine_answer_flags));
    }
}
#!endif

#!ifdef WITH_MSRP
event_route[msrp:frame-in] {
    msrp_reply_flags("1");

    if ((($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT)
        && !(proto == WS || proto == WSS)) && $Rp != MY_MSRP_PORT) {
        xlog("L_WARN", "MSRP request received on $Rp\n");
        msrp_reply("403", "Action-not-allowed");
        exit;
    }

    if (msrp_is_reply()) {
        msrp_relay();
    } else if($msrp(method)=="AUTH") {
        if($msrp(nexthops)>0) {
            msrp_relay();
            exit;
        }

        if (!www_authenticate("MY_DOMAIN", "subscriber",
                    "$msrp(method)")) {
            if (auth_get_www_authenticate("MY_DOMAIN", "1",
                            "$var(wauth)")) {
                msrp_reply("401", "Unauthorized",
                            "$var(wauth)");
            } else {
                msrp_reply("500", "Server Error");
            }
            exit;
        }

        if ($hdr(Expires) != $null) {
            $var(expires) = (int) $hdr(Expires);
            if ($var(expires) < MSRP_MIN_EXPIRES) {
                msrp_reply("423", "Interval Out-of-Bounds",
                    "Min-Expires: MSRP_MIN_EXPIRES\r\n");
                exit;
            } else if ($var(expires) > MSRP_MAX_EXPIRES) {
                msrp_reply("423", "Interval Out-of-Bounds",
                    "Max-Expires: MSRP_MAX_EXPIRES\r\n");
                exit;
            }
        } else {
            $var(expires) = MSRP_MAX_EXPIRES;
        }

        $var(cnt) = $var(cnt) + 1;
        pv_printf("$var(sessid)", "s.$(pp).$(var(cnt)).$(RANDOM)");
        $sht(msrp=>$var(sessid)::srcaddr) = $msrp(srcaddr);
        $sht(msrp=>$var(sessid)::srcsock) = $msrp(srcsock);
        $shtex(msrp=>$var(sessid)) = $var(expires) + 5;
        # - Use-Path: the MSRP address for server + session id
        $var(hdrs) = "Use-Path: msrps://MY_IP_ADDR:MY_MSRP_PORT/" 
                    + $var(sessid) + ";tcp\r\n" 
                    + "Expires: " + $var(expires) + "\r\n";
        msrp_reply("200", "OK", "$var(hdrs)");
    } else if ($msrp(method)=="SEND" || $msrp(method)=="REPORT") {
        if ($msrp(nexthops)>1) {
            if ($msrp(method)!="REPORT") {
                msrp_reply("200", "OK");
            }
            msrp_relay();
            exit;
        }
        $var(sessid) = $msrp(sessid);
        if ($sht(msrp=>$var(sessid)::srcaddr) == $null) {
            # one more hop, but we don't have address in htable
            msrp_reply("481", "Session-does-not-exist");
            exit;
        } else if ($msrp(method)!="REPORT") {
            msrp_reply("200", "OK");
        }
        msrp_relay_flags("1");
        msrp_set_dst("$sht(msrp=>$var(sessid)::srcaddr)",
                "$sht(msrp=>$var(sessid)::srcsock)");
        msrp_relay();
    } else {
        msrp_reply("501", "Request-method-not-understood");
    }
}
#!endif

check if your KAMAILIO is running too

yes Kamailio is running and my configuration looks fine, but Kamailio is showing unreachable on asterisk CLI and not able to login the agent.
Any idea, why?

No sip are registering but problem is with web socket connection for webRTC. I am getting error on browser console as error in connection establishment websocket on any port.
I used 4443,10443 etc.

Please suggest.

Vijay Prakash wrote:

yes Kamailio is running and my configuration looks fine, but Kamailio is showing unreachable on asterisk CLI and not able to login the agent.
Any idea, why?

Make sure that Asterisk and Kamailio are running on different ports. UDP 5070 for Asterisk and UDP 5060 for Kamailio.

Demian Biscocho wrote:

Vijay Prakash wrote:

yes Kamailio is running and my configuration looks fine, but Kamailio is showing unreachable on asterisk CLI and not able to login the agent.
Any idea, why?

Make sure that Asterisk and Kamailio are running on different ports. UDP 5070 for Asterisk and UDP 5060 for Kamailio.

Now All services are running fine, but not sure why web socket connection is giving problem. Do you think that may be certificate issue.
I am using let's encrypt certificate and freenom for the domain name.

I think we are close now, please help.

Thank you

Unlikely to be a certificate issue since your SSL certificate is valid. Your domain name is also good. Test with a softphone (using the phone credentials of the agent created in v4) and see if you can register it to your v4 server.

I tried with that, soft phone registers successfully on asterisk but while login it says that phone is not connected.

Can anybody help with this?

Vijay Prakash wrote:

I tried with that, soft phone registers successfully on asterisk but while login it says that phone is not connected.

It should register to Kamailio not Asterisk. Your phone extension PROTOCOL should be set to EXTERNAL (default).

GOautodial 4 ISO installer is available now (though the "Getting Started Guide" to be posted soon). More details here: https://goautodial.org/boards/20/topics/14072.

Hi,
Autodial campaign does not dialing call auto, However whether i registered sip id on zoiper and unchecked Webrtpc then I am able to dial manual call and this step auto call does not working.

As in first case when i dial the number without soft phone then error is occurred on kamailio service, PFA screen shots,
Kindly assist on this how should I make the call the without soft phone (zoiper)

Demian Lizandro Biscocho wrote:

Vijay Prakash wrote:

yes Kamailio is running and my configuration looks fine, but Kamailio is showing unreachable on asterisk CLI and not able to login the agent.
Any idea, why?

Make sure that Asterisk and Kamailio are running on different ports. UDP 5070 for Asterisk and UDP 5060 for Kamailio.

How to change the port ?

I have change the port in /etc/asterisk/sip.config. but the still show the same 5060 in asterisk CLI

Try also to change it at /etc/kamailio/kamailio.cfg

Wittie Manansala wrote:

Try also to change it at /etc/kamailio/kamailio.cfg

Could you please share the line number where i need to change UDP port number in kamailio?

hi,
when I set my certificates in the tls.cfg file, I restart kamailio and give me certificates error (newgoautodial systemd1: Failed to start Kamailio (OpenSER) - the Open Source SIP Server.)
certificates also work correctly because they are generated with certbot and the page is secure in https: //.

Then when I set in the file kamailio.cfg listen = udp: 127.0.0.1: 5060 it listens on 127.0.0.1 but when I go to change this parameter and insert my server audience it always gives me 127.0.0.1 why does this happen?
Yours sincerely

Hi Fonzo,

May we know what installation guide you've? Please also provide us the error shows in your kamailio

Thanks

Hi Fonzo,
May we know what installation guide you've? Please also provide us the error shows in your kamailio
Thanks

I solved this point, I installed the certificates manually, while those of the certbot procedure .. gave errors.
I managed to register kamailio with host = ip of my server.
I registered my carrier on port 5070.
I followed this procedure https://goautodial.net/d/4-goautodial-version-4-scratch-install-on-linux.
now my mistake is I create the campaign I load a test list, when I enter the campaign with the agent it gives me the microphone, but I don't hear any voice and no ring.
returns the error in the attached image, honestly I don't understand the error you should log into my server, instead make a call from my supplier.

Yours sincerely

I don't understand this: the 9940 @ mycarrier
the agent's phone number is 4793019940, then to log in to the system why do you use my carrier? shouldn't he use kamailio?

Hi,

Our steps are posted here:

Scratch install:
https://goautodial.org/projects/goautodialce/wiki/Version_4_How_To_Install_Goautodial_From_Scratch_using_CentOS_7X

Update latest version:
https://goautodial.org/projects/goautodialce/wiki/HOWTO_Update_latest_version_via_Github

For checking visit below site specially on Administration Gui Settings and Configuration Files to check section:
https://goautodial.org/projects/goautodialce/wiki/Goautodial_Getting_Started_Guidev4

Thanks

hi
I have already followed the standard step-by-step guide, I find myself with the same problems see asterisk console log, when I log in with agent I choose the campaign (number assigned to agent 2567674969), then the item does not start 'you are the only one in this conference'. unlike the previous ones that worked very well at the first configuration. I would like to have some clarification as a result of solving the problem, also because unlike the others it is visually beautiful:
I might seem trivial in order to understand how it works I ask you trivial questions:
Administrator - setting - base URL: my IP, or my hostname?
 Administrator - webrtc setting:
WebRTC Websocket Host / IP: my IP, or my hostname?
WebRTC SIP Host / IP: my IP, or my hostname?
Kamailio Domain: my IP, or my hostname?
Setting -server - serverip: 127.0.0.1, my IP, or my hostname?
Setting - carrier - serverip: 127.0.0.1, my IP, or my hostname?
Telephony - Edit Campaign - Carrier to use for Campaign: mycarrier or custom dial prefix?
File - /etc/kamailio/tls.cfg: set my certificate (private_key, certificate, ca_list) or leave the default ones?
File - /etc/kamailio/tls.cfg SIP_DOMAIN: leave vaglxc01.goautodial.com or put my hostname in place of vaglxc01.goautodial.com?
File - /etc/kamailio/kamailio.cfg:
alias = "my_ip" alias = "vaglxc01.goautodial.com" or set my hostname?
listen = udp: 127.0.0.1: 5060 listen = udp: my_ip: 5060?
#! substdef "! MY_IP_ADDR! my_ip! g" #! substdef "! MY_DOMAIN! vaglxc01.goautodial.com! g" or put my hostname in place of vaglxc01.goautodial.com?
nano /var/www/html/php/goCRMAPISettings.php: https: // my_ip or my hostname / goAPIv2?

/ etc / hosts:
leave it like this or put my hostname in place of vaglxc01.goautodial.com?
127.0.0.1 localhost localhost.localdomain localhost4 localhost4.localdomain4 vaglxc01.goautodial.com
:: 1 localhost localhost.localdomain localhost6 localhost6.localdomain6

/etc/asterisk/sip-goautodial.conf: host = vaglxc01.goautodial.com; change me to my FQDN?

thanks in advance for your patience

fonzo de laia wrote:

hi
I have already followed the standard step-by-step guide, I find myself with the same problems see asterisk console log, when I log in with agent I choose the campaign (number assigned to agent 2567674969), then the item does not start 'you are the only one in this conference'. unlike the previous ones that worked very well at the first configuration. I would like to have some clarification as a result of solving the problem, also because unlike the others it is visually beautiful:
I might seem trivial in order to understand how it works I ask you trivial questions:
Administrator - setting - base URL: my IP, or my hostname?
 Administrator - webrtc setting:
WebRTC Websocket Host / IP: my IP, or my hostname?
WebRTC SIP Host / IP: my IP, or my hostname?
Kamailio Domain: my IP, or my hostname?
Setting -server - serverip: 127.0.0.1, my IP, or my hostname?
Setting - carrier - serverip: 127.0.0.1, my IP, or my hostname?
Telephony - Edit Campaign - Carrier to use for Campaign: mycarrier or custom dial prefix?
File - /etc/kamailio/tls.cfg: set my certificate (private_key, certificate, ca_list) or leave the default ones?
File - /etc/kamailio/tls.cfg SIP_DOMAIN: leave vaglxc01.goautodial.com or put my hostname in place of vaglxc01.goautodial.com?
File - /etc/kamailio/kamailio.cfg:
alias = "my_ip" alias = "vaglxc01.goautodial.com" or set my hostname?
listen = udp: 127.0.0.1: 5060 listen = udp: my_ip: 5060?
#! substdef "! MY_IP_ADDR! my_ip! g" #! substdef "! MY_DOMAIN! vaglxc01.goautodial.com! g" or put my hostname in place of vaglxc01.goautodial.com?
nano /var/www/html/php/goCRMAPISettings.php: https: // my_ip or my hostname / goAPIv2?

/ etc / hosts:
leave it like this or put my hostname in place of vaglxc01.goautodial.com?
127.0.0.1 localhost localhost.localdomain localhost4 localhost4.localdomain4 vaglxc01.goautodial.com
:: 1 localhost localhost.localdomain localhost6 localhost6.localdomain6

/etc/asterisk/sip-goautodial.conf: host = vaglxc01.goautodial.com; change me to my FQDN?

thanks in advance for your patience

Hi,

I would recommend to setup your server without SSL Cert first. Once everything works fine (meaning your able to login, create campaign and make calls) next step setup your SSL Cert.

Without SSL Cert your private IP should be configure on your server. Just follow the sample settings posted here https://goautodial.org/projects/goautodialce/wiki/Goautodial_Getting_Started_Guidev4 and read "Administration Gui Settings and Configuration Files to check" Section:

Thanks

hi,
ok, left everything as a guide and it works, the only things changed are my FQDN in the kamailio options, only in the file /etc/asterisk/sip-goautodial.conf I put host = localhost, and everything went well.
thank you