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Need help urgently with goautodial V4 no audio issue

Added by White Star 25 days ago

Hi, We have installed the Goautodial V4 on our server, according to the CentOS7 guide. and have done all the changes mentioned there, But when we dial the number call gets connected but there is no sound on both sides, we have tried alot of methods but unable to solve it if we press hold button the person on phone end hears the music but he cannot hear the agent and agent cannot hear the person, we have tried this in manual dialing mode.

Any help in this regards will be helpfull, we have sorted the tls issue and we have installed our domain SSL certificates properly.

We are using our Freepbx sip trunk on goautodial.

both servers are inhouse.

I have attached my Kamailio.cfg , rtpengine.cfg

My go autodial server ip is : 192.168.10.25
My external domain is: mydomain1.dyndns.org

Here below is my Kamalio.cfg:
#!KAMAILIO #
#!define WITH_MYSQL
#!define WITH_AUTH
#!define WITH_USRLOCDB
#!define WITH_NAT
#!define WITH_ANTIFLOOD # #
Kamailio (OpenSER) SIP Server v5.0 - default configuration script
- web: http://www.kamailio.org
- git: http://sip-router.org #
Direct your questions about this file to: <sr-users@lists.sip-router.org> #
Refer to the Core CookBook at http://www.kamailio.org/wiki/
for an explanation of possible statements, functions and parameters. #
Several features can be enabled using '#!define WITH_FEATURE' directives: #
  • To run in debug mode:
    - define WITH_DEBUG #
  • To enable mysql:
    - define WITH_MYSQL #
  • To enable authentication execute:
    - enable mysql
    - define WITH_AUTH
    - add users using 'kamctl' #
  • To enable IP authentication execute:
    - enable mysql
    - enable authentication
    - define WITH_IPAUTH
    - add IP addresses with group id '1' to 'address' table #
  • To enable persistent user location execute:
    - enable mysql
    - define WITH_USRLOCDB #
  • To enable presence server execute:
    - enable mysql
    - define WITH_PRESENCE #
  • To enable nat traversal execute:
    - define WITH_NAT
    - install RTPProxy: http://www.rtpproxy.org
    - start RTPProxy:
    rtpproxy l your_public_ip -s udp:localhost:7722
    option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING #
  • To enable PSTN gateway routing execute:
    - define WITH_PSTN
    - set the value of pstn.gw_ip
    - check route[PSTN] for regexp routing condition #
  • To enable database aliases lookup execute:
    - enable mysql
    - define WITH_ALIASDB #
  • To enable speed dial lookup execute:
    - enable mysql
    - define WITH_SPEEDDIAL #
  • To enable multi-domain support execute:
    - enable mysql
    - define WITH_MULTIDOMAIN #
  • To enable TLS support execute:
    - adjust CFGDIR/tls.cfg as needed
    - define WITH_TLS #
  • To enable XMLRPC support execute:
    - define WITH_XMLRPC
    - adjust route[XMLRPC] for access policy #
  • To enable anti-flood detection execute:
    - adjust pike and htable=>ipban settings as needed (default is
    block if more than 16 requests in 2 seconds and ban for 300 seconds)
    - define WITH_ANTIFLOOD #
  • To block 3XX redirect replies execute:
    - define WITH_BLOCK3XX #
  • To enable VoiceMail routing execute:
    - define WITH_VOICEMAIL
    - set the value of voicemail.srv_ip
    - adjust the value of voicemail.srv_port #
  • To enhance accounting execute:
    - enable mysql
    - define WITH_ACCDB
    - add following columns to database
    #!ifdef ACCDB_COMMENT
    ALTER TABLE acc ADD COLUMN src_user VARCHAR 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
    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!192.168.10.25!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

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="192.168.10.25"
alias="mydomain1.dyndns.org"

/* 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:192.168.10.25:5060
listen=udp:mydomain1.dyndns.org: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=1
#!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:192.168.10.25:5066")
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:")
params needed for NAT traversal in other modules
modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)")
modparam("usrloc", "nat_bflag", FLB_NATB)
#!endif
#!ifdef WITH_TLS
----- tls params -----
modparam("tls", "tls_method", "TLSv1")
modparam("tls", "certificate", "/etc/kamailio/mydomain1.crt")
modparam("tls", "private_key", "/etc/kamailio/mydomain1.key")
modparam("tls", "verify_certificate",0 )
modparam("tls", "require_certificate",0 )
#modparam("tls", "config", "/etc/kamailio/tls.cfg")
#modparam("tls", "private_key", "/etc/httpd/certs/essentialSSL/wildcard.goautodial.com.key")
#modparam("tls", "certificate", "/etc/httpd/certs/essentialSSL/wildcard.goautodial.com.crt")
#modparam("tls", "ca_list", "/etc/httpd/certs/essentialSSL/wildcard.goautodial.com.ca-bundle")
#!endif
#!ifdef WITH_WEBSOCKETS
----- nathelper params -----
modparam("nathelper|registrar", "received_avp", "$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==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 ($hdr(User-Agent) =~ "xmlrpclib")
set_reply_close();
set_reply_no_connect();
dispatch_rpc();
exit;
}
send_reply("403", "Forbidden");
exit;
}
#!endif
Routing to voicemail server
route[TOVOICEMAIL] {
#!ifdef WITH_VOICEMAIL
if(!is_method("INVITE|SUBSCRIBE"))
return;
check if VoiceMail server IP is defined
if (strempty($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 =~ "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;
- 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
Here below is my rtpengine.cfg

[rtpengine]

#table = 0
no-fallback = false
for userspace forwarding only:
table = -1
a single interface:
interface = 192.168.10.25
separate multiple interfaces with semicolons:
interface = internal/12.23.34.45;external/23.34.45.54
for different advertised address:
interface = 12.23.34.45!23.34.45.56
#listen-ng = 127.0.0.1:5066
listen-ng = 192.168.10.25:5066
#listen-ng = mydomain1.dyndns.org:5066
listen-tcp = 25060
listen-udp = 12222
timeout = 60
silent-timeout = 3600
tos = 184
#control-tos = 184
delete-delay = 30
final-timeout = 10800
foreground = false
pidfile = /var/run/ngcp-rtpengine-daemon.pid
num-threads = 16
port-min = 30000
port-max = 50000
max-sessions = 5000
recording-dir = /var/spool/rtpengine
recording-method = proc
recording-format = raw
redis = 127.0.0.1:6379/5
redis-write = :6379/42
redis-num-threads = 8
no-redis-required = false
redis-expires = 86400
redis-allowed-errors = -1
redis-disable-time = 10
redis-cmd-timeout = 0
redis-connect-timeout = 1000
b2b-url = http://127.0.0.1:8090/
xmlrpc-format = 0
log-level = 6
log-stderr = false
log-facility = daemon
log-facility-cdr = local0
log-facility-rtcp = local1
graphite = 127.0.0.1:9006
graphite-interval = 60
graphite-prefix = foobar.
homer = 123.234.345.456:65432
homer-protocol = udp
homer-id = 2001
sip-source = false
dtls-passive = false
[rtpengine-testing]
table = -1
interface = 10.15.20.121
listen-ng = 2223
foreground = true
log-stderr = true
log-level = 7


Replies (11)

RE: Need help urgently with goautodial V4 no audio issue - Added by Leopoldo Martinez 25 days ago

Hi, your error in kamailio.cfg is in this line

#!substdef "!MY_DOMAIN!vaglxc01.goautodial.com!g"

you should put this

#!substdef "!MY_DOMAIN!mydomain1.dyndns.org!g"

and here you can left the original aliasses and add your new aliasses below

/* add local domain aliases */
alias="192.168.10.25"
alias="vaglxc01.goautodial.com"
alias="mydomain1.dyndns.org"
alias="dynds.org"

I think, this should solve your problem. tell me if it works. Good Luck

PD: Be sure that your server's name is the same than your domain, it make the things easier

RE: Need help urgently with goautodial V4 no audio issue - Added by White Star 24 days ago

Hi tried the above changes but still no solution

RE: Need help urgently with goautodial V4 no audio issue - Added by White Star 24 days ago

Thsi is the core issue im facing when trying to dial call from webrtc

file.c:774 ast_openstream_full: File confbridge-join does not exist in any format

RE: Need help urgently with goautodial V4 no audio issue - Added by White Star 24 days ago

Have solved the file issue also, but still no audio on webrtc, softphone is having audio. below is my astrisk log for call mader from webrtc

-- Called 8600051@default
-- Executing [8600051@default:1] Konference("Local/8600051@default-0000001e;2", "8600051,R") in new stack
-- Local/8600051@default-0000001e;1 answered
-- Executing [9990521234567@default:1] AGI("Local/8600051@default-0000001e;1", "agi://127.0.0.1:4577/call_log") in new stack
Manager 'sendcron' logged off from 127.0.0.1
-- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=26475602))
-- &lt;Local/8600051@default-0000001e;1&gt;AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing [9990521234567@default:2] Dial("Local/8600051@default-0000001e;1", "SIP/0521234567@FreePBX,,tTo") in new stack
Using SIP RTP CoS mark 5
-- Called SIP/0521234567@FreePBX
-- SIP/FreePBX-00000013 is making progress passing it to Local/8600051@default-0000001e;1
-- SIP/FreePBX-00000013 answered Local/8600051@default-0000001e;1
-- Channel SIP/FreePBX-00000013 joined 'simple_bridge' basic-bridge &lt;fe1fb947-c494-4d1d-a1bd-a819eda8b2a5&gt;
-- Channel Local/8600051@default-0000001e;1 joined 'simple_bridge' basic-bridge &lt;fe1fb947-c494-4d1d-a1bd-a819eda8b2a5&gt;
-- Channel SIP/FreePBX-00000013 left 'simple_bridge' basic-bridge &lt;fe1fb947-c494-4d1d-a1bd-a819eda8b2a5&gt;
-- Channel Local/8600051@default-0000001e;1 left 'simple_bridge' basic-bridge &lt;fe1fb947-c494-4d1d-a1bd-a819eda8b2a5&gt;
Spawn extension (default, 9990521234567, 2) exited non-zero on 'Local/8600051@default-0000001e;1'
-- Executing [h@default:1] AGI("Local/8600051@default-0000001e;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----25-----25-----SIP 200 OK)") in new stack
-- &lt;Local/8600051@default-0000001e;1&gt;AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----25-----25-----SIP 200 OK) completed, returning 0
[Jun 20 00:52:39] WARNING[18883][C-00000031]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
-- Executing [h@default:1] AGI("Local/8600051@default-0000001e;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
-- &lt;Local/8600051@default-0000001e;2&gt;AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0
Manager 'sendcron' logged on from 127.0.0.1
[Jun 20 00:52:43] NOTICE[18973]: manager.c:4309 action_hangup: Request to hangup non-existent channel: SIP/FreePBX-00000013

RE: Need help urgently with goautodial V4 no audio issue - Added by White Star 24 days ago

My rtp is giving this error:

ngcp-rtpengine.service - LSB: NGCP rtpengine
Loaded: loaded (/etc/rc.d/init.d/ngcp-rtpengine; bad; vendor preset: disabled)
Active: failed (Result: exit-code) since Sat 2020-06-20 01:56:50 +04; 20s ago
Docs: man:systemd-sysv-generator(8)
Process: 1998 ExecStop=/etc/rc.d/init.d/ngcp-rtpengine stop (code=exited, status=0/SUCCESS)
Process: 2861 ExecStart=/etc/rc.d/init.d/ngcp-rtpengine start (code=exited, status=1/FAILURE)
Main PID: 25734 (code=exited, status=0/SUCCESS)

Jun 20 01:56:50 globalarc.dyndns.org ngcp-rtpengine2861: ip6tables: No chain/target/match by that name.
Jun 20 01:56:50 globalarc.dyndns.org rtpengine2889: INFO: Generating new DTLS certificate
Jun 20 01:56:50 globalarc.dyndns.org rtpengine2889: ERR: FAILED TO CREATE KERNEL TABLE 0 (No such file or directory), KERNEL FORWARDING DISABLED
Jun 20 01:56:50 globalarc.dyndns.org ngcp-rtpengine2861: Starting rtpengine: [1592603810.978941] ERR: FAILED TO CREATE KERNEL TABLE 0 (No such file or d...DISABLED
Jun 20 01:56:50 globalarc.dyndns.org ngcp-rtpengine2861: [1592603810.978968] CRIT: Userspace fallback disallowed - exiting
Jun 20 01:56:50 globalarc.dyndns.org ngcp-rtpengine2861: [FAILED]
Jun 20 01:56:50 globalarc.dyndns.org systemd1: ngcp-rtpengine.service: control process exited, code=exited status=1
Jun 20 01:56:50 globalarc.dyndns.org systemd1: Failed to start LSB: NGCP rtpengine.
Jun 20 01:56:50 globalarc.dyndns.org systemd1: Unit ngcp-rtpengine.service entered failed state.
Jun 20 01:56:50 globalarc.dyndns.org systemd1: ngcp-rtpengine.service failed.
Hint: Some lines were ellipsized, use -l to show in full.

RE: Need help urgently with goautodial V4 no audio issue - Added by White Star 24 days ago

White Star wrote:

My rtp is giving this error:

ngcp-rtpengine.service - LSB: NGCP rtpengine
Loaded: loaded (/etc/rc.d/init.d/ngcp-rtpengine; bad; vendor preset: disabled)
Active: failed (Result: exit-code) since Sat 2020-06-20 01:56:50 +04; 20s ago
Docs: man:systemd-sysv-generator(8)
Process: 1998 ExecStop=/etc/rc.d/init.d/ngcp-rtpengine stop (code=exited, status=0/SUCCESS)
Process: 2861 ExecStart=/etc/rc.d/init.d/ngcp-rtpengine start (code=exited, status=1/FAILURE)
Main PID: 25734 (code=exited, status=0/SUCCESS)

Jun 20 01:56:50 mydomain1.dyndns.org ngcp-rtpengine2861: ip6tables: No chain/target/match by that name.
Jun 20 01:56:50 mydomain1.dyndns.org rtpengine2889: INFO: Generating new DTLS certificate
Jun 20 01:56:50 mydomain1.dyndns.org rtpengine2889: ERR: FAILED TO CREATE KERNEL TABLE 0 (No such file or directory), KERNEL FORWARDING DISABLED
Jun 20 01:56:50 mydomain1.dyndns.org ngcp-rtpengine2861: Starting rtpengine: [1592603810.978941] ERR: FAILED TO CREATE KERNEL TABLE 0 (No such file or d...DISABLED
Jun 20 01:56:50 mydomain1.dyndns.org ngcp-rtpengine2861: [1592603810.978968] CRIT: Userspace fallback disallowed - exiting
Jun 20 01:56:50 mydomain1.dyndns.org ngcp-rtpengine2861: [FAILED]
Jun 20 01:56:50 mydomain1.dyndns.org systemd1: ngcp-rtpengine.service: control process exited, code=exited status=1
Jun 20 01:56:50 mydomain1.dyndns.org systemd1: Failed to start LSB: NGCP rtpengine.
Jun 20 01:56:50 mydomain1.dyndns.org systemd1: Unit ngcp-rtpengine.service entered failed state.
Jun 20 01:56:50 mydomain1.dyndns.org systemd1: ngcp-rtpengine.service failed.
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RE: Need help urgently with goautodial V4 no audio issue - Added by Demian Lizandro Biscocho 22 days ago

Looks like your RTPengine is misconfigured. Copy the default configuration again from /usr/src/goautodial/etc/rtpengine/.

Replace with your public IP address

### a single interface:
interface = 123.234.345.456

It's recommended that your server has a public IP address whether via your router DMZ or an actual one.

RE: Need help urgently with goautodial V4 no audio issue - Added by White Star 10 days ago

Hi i sorted the issue, i reinstalled with the fresh iso and durring installation i configured my network card to turn on automatically, every thing was preconfigured after the install, all is working but im having issue now with call transfer, i dont see a dropdown to select agent it only shows ingroup name.

RE: Need help urgently with goautodial V4 no audio issue - Added by White Star 10 days ago


.
Here in teh attached image if you can see, i can select the closer group then next to closer group there is no drop down coming to select the agent name , only local closer button is showing and when i click on it it gives me error to select agent.

I have already selected agents in the Agent direct queue and the same group is selected in transfer and inbound in the campaign, but agents inside the group are not showing in the popup.

transfer error.png (24.8 KB) transfer error.png transfererror

RE: Need help urgently with goautodial V4 no audio issue - Added by White Star 10 days ago

Also in smtp setting when i fill the settings and click on submit after reload the field are empty no setting is saved. these two issues are only pain right now which is holding my complete live deployment.

RE: Need help urgently with goautodial V4 no audio issue - Added by Jackie Alfonso 5 days ago

White Star wrote:


.
Here in teh attached image if you can see, i can select the closer group then next to closer group there is no drop down coming to select the agent name , only local closer button is showing and when i click on it it gives me error to select agent.

I have already selected agents in the Agent direct queue and the same group is selected in transfer and inbound in the campaign, but agents inside the group are not showing in the popup.

Have you tried creating a new ingroup? please take note that the agent should select the ingroup so they received the local transfer call.

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