[OpenSIPS-Users] call_center module issues
Mikhail
forfx at yandex.ru
Sat Apr 20 14:09:13 EDT 2019
this is it:
#
# OpenSIPS residential configuration script
# by OpenSIPS Solutions <team at opensips-solutions.com>
#
# This script was generated via "make menuconfig", from
# the "Residential" scenario.
# You can enable / disable more features / functionalities by
# re-generating the scenario with different options.#
#
# Please refer to the Core CookBook at:
# http://www.opensips.org/Resources/DocsCookbooks
# for a explanation of possible statements, functions and parameters.
#
####### Global Parameters #########
log_level=3
log_stderror=no
log_facility=LOG_LOCAL0
children=4
/* uncomment the following lines to enable debugging */
#debug_mode=yes
/* uncomment the next line to enable the auto temporary blacklisting of
not available destinations (default disabled) */
#disable_dns_blacklist=no
/* uncomment the next line to enable IPv6 lookup after IPv4 dns
lookup failures (default disabled) */
#dns_try_ipv6=yes
/* comment the next line to enable the auto discovery of local aliases
based on reverse DNS on IPs */
auto_aliases=no
listen=udp:192.168.16.6 # CUSTOMIZE ME
listen=udp:111.111.111.111 # CUSTOMIZE ME
####### Modules Section ########
#set module path
mpath="/usr/lib64/opensips/modules"
#### SIGNALING module
loadmodule "signaling.so"
#### StateLess module
loadmodule "sl.so"
#### Transaction Module
loadmodule "tm.so"
modparam("tm", "fr_timeout", 5)
modparam("tm", "fr_inv_timeout", 30)
modparam("tm", "restart_fr_on_each_reply", 0)
modparam("tm", "onreply_avp_mode", 1)
#### Record Route Module
loadmodule "rr.so"
/* do not append from tag to the RR (no need for this script) */
modparam("rr", "append_fromtag", 0)
#### MAX ForWarD module
loadmodule "maxfwd.so"
#### SIP MSG OPerationS module
loadmodule "sipmsgops.so"
#### FIFO Management Interface
loadmodule "mi_fifo.so"
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
modparam("mi_fifo", "fifo_mode", 0666)
#### URI module
loadmodule "uri.so"
modparam("uri", "use_uri_table", 0)
#### MYSQL module
loadmodule "db_mysql.so"
#### HTTPD module
loadmodule "httpd.so"
modparam("httpd", "port", 8888)
#### USeR LOCation module
loadmodule "usrloc.so"
modparam("usrloc", "nat_bflag", "NAT")
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
#### REGISTRAR module
loadmodule "registrar.so"
modparam("registrar", "tcp_persistent_flag", "TCP_PERSISTENT")
/* uncomment the next line not to allow more than 10 contacts per AOR */
#modparam("registrar", "max_contacts", 10)
#### ACCounting module
loadmodule "acc.so"
/* what special events should be accounted ? */
modparam("acc", "early_media", 0)
modparam("acc", "report_cancels", 0)
/* by default we 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)
modparam("acc", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
modparam("acc", "extra_fields", "db: src_ip; dst_ip")
modparam("acc", "leg_fields", "db: caller; callee")
#### AUTHentication modules
loadmodule "auth.so"
loadmodule "auth_db.so"
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("auth_db|uri", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
modparam("auth_db", "load_credentials", "")
#### ALIAS module
loadmodule "alias_db.so"
modparam("alias_db", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
#### DIALOG module
loadmodule "dialog.so"
modparam("dialog", "dlg_match_mode", 1)
modparam("dialog", "default_timeout", 21600) # 6 hours timeout
modparam("dialog", "db_mode", 2)
modparam("dialog", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
#### DIALPLAN module
loadmodule "dialplan.so"
modparam("dialplan", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
#### DYNAMIC ROUTING module
loadmodule "drouting.so"
modparam("drouting", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
modparam("drouting", "use_domain", 0)
#### MI_HTTP module
loadmodule "mi_http.so"
loadmodule "mi_json.so"
loadmodule "proto_udp.so"
#### permissions
loadmodule "permissions.so"
modparam("permissions", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips")
#### call_center
loadmodule "b2b_entities.so"
loadmodule "b2b_logic.so"
modparam("b2b_logic", "script_scenario",
"/etc/opensips/scenario_callcenter.xml")
loadmodule "call_center.so"
modparam("call_center", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips")
modparam("call_center", "acc_db_url",
"mysql://opensips:opensipsrw@localhost/opensips")
####### Routing Logic ########
# main request routing logic
route{
if (!mf_process_maxfwd_header("10")) {
send_reply("483","Too Many Hops");
exit;
}
if (has_totag()) {
# handle hop-by-hop ACK (no routing required)
if ( is_method("ACK") && t_check_trans() ) {
t_relay();
exit;
}
# sequential request within a dialog should
# take the path determined by record-routing
if ( !loose_route() ) {
# we do record-routing for all our traffic, so we should not
# receive any sequential requests without Route hdr.
send_reply("404","Not here");
exit;
}
# validate the sequential request against dialog
if ( $DLG_status!=NULL && !validate_dialog() ) {
xlog("In-Dialog $rm from $si (callid=$ci) is not valid
according to dialog\n");
## exit;
}
if (is_method("BYE")) {
# do accounting even if the transaction fails
##do_accounting("db","failed"); #no need for cdr
}
# route it out to whatever destination was set by loose_route()
# in $du (destination URI).
route(relay);
exit;
}
# CANCEL processing
if (is_method("CANCEL")) {
if (t_check_trans())
t_relay();
exit;
}
# absorb retransmissions, but do not create transaction
t_check_trans();
if ( !(is_method("REGISTER") ) ) {
if (is_myself("$si") && is_myself("$rd")) {
xlog("-- it looks like call from call_center module --");
setflag(call_center_internal);
} else {
if (is_myself("$fd")) {
# authenticate if from local subscriber
# authenticate all initial non-REGISTER request that
pretend to be
# generated by local subscriber (domain from FROM URI
is local)
if (!proxy_authorize("", "subscriber")) {
proxy_challenge("", "0");
exit;
}
if (!db_check_from()) {
send_reply("403","Forbidden auth ID");
exit;
}
consume_credentials();
# caller authenticated
} else {
# if caller is not local, then called number must be local
if (!is_myself("$rd")) {
xlog("-- F");
send_reply("403","Relay Forbidden");
exit;
}
# if caller is not local, then check ip - address
if(!check_source_address("0")){
send_reply("403","Forbidden for You");
exit;
}
}
}
}
xlog("-- V1 --[$rm / $fu / $tu / $ru / $ci]");
# preloaded route checking
if (loose_route()) {
xlog("L_ERR",
"Attempt to route with preloaded Route's [$fu/$tu/$ru/$ci]");
if (!is_method("ACK"))
send_reply("403","Preload Route denied");
exit;
}
# record routing
if (!is_method("REGISTER|MESSAGE"))
record_route();
# account only INVITEs
if (is_method("INVITE")) {
# create dialog with timeout
if ( !create_dialog("B") ) {
send_reply("500","Internal Server Error");
exit;
}
do_accounting("db", "cdr");
}
if (!is_myself("$rd")) {
append_hf("P-hint: outbound\r\n");
route(relay);
}
# requests for my domain
if (is_method("PUBLISH|SUBSCRIBE")) {
send_reply("503", "Service Unavailable");
exit;
}
if (is_method("REGISTER")) {
# authenticate the REGISTER requests
if (!www_authorize("", "subscriber")) {
www_challenge("", "0");
exit;
}
if (!db_check_to()) {
send_reply("403","Forbidden auth ID");
exit;
}
if (!save("location"))
sl_reply_error();
exit;
}
if ($rU==NULL) {
# request with no Username in RURI
send_reply("484","Address Incomplete");
exit;
}
$acc_extra(src_ip) = $si; # source IP of the request
$acc_leg(caller) = $fu;
$acc_leg(callee) = $ru;
# apply DB based aliases
alias_db_lookup("dbaliases");
#Dial plan processing
dp_translate("0","$rU/$rU", "$avp(attrs)");
xlog("rU: $rU , avp(attrs): $avp(attrs)");
if ($avp(attrs)=="pstn") {
#route to pstn
route(pstn);
}
if ($avp(attrs)=="media") {
#route to media server
route(media);
}
if ($avp(attrs)=="ivr") {
#route to ivr
route(ivr);
}
#all other calls are local ($avp(attrs)=="usrloc")
#Route to usrloc
route(lookup);
send_reply("420", "Invalid Extension V2");
exit;
#######
}
route[relay] {
# for INVITEs enable some additional helper routes
if (is_method("INVITE")) {
t_on_branch("per_branch_ops");
t_on_reply("handle_nat");
t_on_failure("missed_call");
}
if (!t_relay()) {
send_reply("500","Internal Error");
}
exit;
}
route[lookup] {
# do lookup with method filtering
if (!lookup("location", "m")) {
switch ($retcode) {
case -1:
case -3:
t_newtran();
t_reply("404", "Not Found");
exit;
case -2:
sl_send_reply("405", "Method Not Allowed");
exit;
}
}
do_accounting("db","missed");
serialize_branches(1);
next_branches();
route(relay);
exit;
}
route[pstn] {
#---- PSTN route ----#
if(!do_routing("0","1")){
send_reply("503", "No rules found matching the URI prefix");
exit;
}
# flag 10 - route to pstn
setflag(10);
route(relay);
}
route[media] {
#---- Route to media servers ----#
xlog("route to media servers");
}
route[ivr] {
#---- IVR route ----#
xlog("route to IVR");
if (!cc_handle_call("prq_test1")) {
xlog("route to IVR failed, retcode: $retcode");
send_reply("403","Cannot handle call");
exit;
}
exit;
}
branch_route[per_branch_ops] {
xlog("new branch at $ru\n");
}
onreply_route[handle_nat] {
xlog("incoming reply\n");
if ( $rs >= 200 )
$acc_extra(dst_ip) = $si;
}
failure_route[missed_call] {
do_accounting("db","missed");
if(isflagset(10)){
if (use_next_gw()) {
xlog ("next gateway $ru \n");
t_relay();
exit;
} else {
t_reply("503", "Service not available, no more gateways");
exit;
}
}
if (t_was_cancelled()) {
exit;
}
next_branches();
# if we've got any more branches arm the failure route
if ($rc != 2) {
t_on_failure("missed_call");
}
if (!t_relay()) {
send_reply("500","Internal Error");
};
if ( $avp(redirect_uri)!=NULL ) {
# set the new destination
$ru = $avp(redirect_uri);
# create a new call leg
acc_new_leg();
# new caller is the callee of the previous leg
$acc_leg(caller) = $(acc_leg(callee)[-2]);
# new callee is the new destination
$acc_leg(callee) = $ru;
t_on_failure("missed_call");
route(lookup);
exit;
}
# uncomment the following lines if you want to block client
# redirect based on 3xx replies.
##if (t_check_status("3[0-9][0-9]")) {
##t_reply("404","Not found");
## exit;
##}
}
local_route {
if (is_method("BYE") && $DLG_dir=="UPSTREAM") {
acc_db_request("200 Dialog Timeout", "acc");
}
}
Mikhail Laba
20.04.2019 19:47, David Villasmil пишет:
> You’re not pasting the config scenario
>
> On Sat, 20 Apr 2019 at 16:30, Mikhail <forfx at yandex.ru
> <mailto:forfx at yandex.ru>> wrote:
>
> Hi Alex,
>
> my config file is here: http://video2dv.com/download/opensips.cfg
>
> Mikhail Laba
>
>
>
> 20.04.2019 15:30, Alexander Jankowsky пишет:
> > Hello Mikhail,
> >
> > Can you copy and show us your opensips config file.
> > Maybe it is something simple and very basic that someone will
> quickly recognise.
> >
> > Alex
> >
> > -----Original Message-----
> > From: Users [mailto:users-bounces at lists.opensips.org
> <mailto:users-bounces at lists.opensips.org>] On Behalf Of Mikhail
> > Sent: Saturday, 20 April 2019 8:17 PM
> > To: users at lists.opensips.org <mailto:users at lists.opensips.org>
> > Subject: [OpenSIPS-Users] call_center module issues
> >
> > Hello!
> >
> > I'm trying to setup call_center module for the first time and
> found the following problems:
> >
> >
> > #1. When a call is in the queue and waiting for the free agent,
> opensips
> > makes a call to a media server and caller hear music - that's is ok,
> >
> > but when a free agent found and opensips is calling to him, the
> call to
> > mediaserver is ended and caller hear nothing.
> >
> > If agent did not answer, opensips sets up a new call to mediaserver.
> >
> > This is very strange behavior, because normally the caller,
> while he is
> > in queue, should hear music without interrupt until one of the
> agents
> > answer the call. Exactly answer, but not ringing.
> >
> >
> > #2. After the agent answer the call it's not possible to put call on
> > hold, both agent and caller can't do it. In sip dialog there is
> a "400
> > Not Acceptable"
> >
> >
> > Of course it's possible that I made a wrong logic in opensips
> config,
> > anyway Is this behavior was hardcoded in call_center module? How
> to fix it?
> >
> >
> > Mikhail Laba
> >
> >
> >
> > _______________________________________________
> > Users mailing list
> > Users at lists.opensips.org <mailto:Users at lists.opensips.org>
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> >
> > _______________________________________________
> > Users mailing list
> > Users at lists.opensips.org <mailto:Users at lists.opensips.org>
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org <mailto:Users at lists.opensips.org>
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> --
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> <mailto:david.villasmil.work at gmail.com>
> phone: +34669448337
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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