1.10.2021

RTPENGINE DTMF transcoding

Kamailio:

route[rtpengine_invite] {
        if (has_body("application/sdp"))
                rtpengine_manage("codec-mask=telephone-event transcode=PCMA always-transcode");
}

route[rtpengine_answer] {
         if (has_body("application/sdp"))
            rtpengine_manage("always-transcode");
}

Demo for transcoding from telephone-event – to in-band and vice versa

Full kamailio.cfg

#!KAMAILIO
#
# Kamailio (OpenSER) SIP Server v5.2 - default configuration script
#     - web: https://www.kamailio.org
#     - git: https://github.com/kamailio/kamailio
#
# Direct your questions about this file to: <sr-users@lists.kamailio.org>
#
# Refer to the Core CookBook at https://www.kamailio.org/wiki/
# for an explanation of possible statements, functions and parameters.
#
# Note: the comments can be:
#     - lines starting with #, but not the pre-processor directives,
#       which start with #!, like #!define, #!ifdef, #!endif, #!else, #!trydef,
#       #!subst, #!substdef, ...
#     - lines starting with //
#     - blocks enclosed in between /* */
#
# 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 block 401 and 407 authentication replies execute:
#     - define WITH_BLOCK401407
#
# *** 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://kamailio:kamailiorw@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

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

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

memdbg=5
memlog=5

log_facility=LOG_LOCAL0
log_prefix="{$mt $hdr(CSeq) $ci} "

/* number of SIP routing processes */
children=8

/* 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=yes

/* add local domain aliases */
# alias="sip.mydomain.com"

/* uncomment and configure the following line if you want Kamailio to
 * bind on a specific interface/port/proto (default bind on all available) */
 listen=udp:192.168.1.199:5090

#!ifdef WITH_TLS
enable_tls=yes
#!endif

/* 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 "PSTN 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

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

/* set paths to location of modules */
# mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/"

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

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"
loadmodule "counters.so"
loadmodule "rtpengine.so"
modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2223")
loadmodule "dialog.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 "rtpproxy.so"
#!endif

#!ifdef WITH_TLS
loadmodule "tls.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 ---------------


# ----- 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")

# ----- ctl params -----
/* set the path to RPC unix socket control file */
# modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl")

# ----- 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 10 contacts per AOR */
# modparam("registrar", "max_contacts", 10)
/* 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", 2)
modparam("usrloc", "use_domain", MULTIDOMAIN)
#!endif

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

# ----- 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
# ----- rtpproxy params -----
modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722")

# ----- nathelper params -----
modparam("nathelper", "natping_interval", 30)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_bflag", FLB_NATSIPPING)
modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org")

# 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")
#!endif

#!ifdef WITH_ANTIFLOOD
# ----- pike params -----
modparam("pike", "sampling_time_unit", 2)
modparam("pike", "reqs_density_per_unit", 16)
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);

        # NAT detection
        route(NATDETECT);

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

        # handle retransmissions
        if (!is_method("ACK")) {
                if(t_precheck_trans()) {
                        t_check_trans();
                        exit;
                }
                t_check_trans();
        }

        # handle requests within SIP dialogs
        route(WITHINDLG);

        ### only initial requests (no To tag)

        # 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")) {
                dlg_manage();
                route(rtpengine_invite);

                setflag(FLT_ACC); # do accounting
        }

        # 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);
}

# 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")) {
                if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE");
        }

        if (!t_relay()) {
                sl_reply_error();
        }
        exit;
}

# Per SIP request initial checks
route[REQINIT] {
#!ifdef WITH_ANTIFLOOD
        # flood detection 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;
                }
        }
#!endif
        if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") {
                # silent drop for scanners - uncomment next line if want to reply
                # sl_send_reply("200", "OK");
                exit;
        }

        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("17895", "7")) {
                xlog("Malformed SIP message from $si:$sp\n");
                exit;
        }
}

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

         if (has_body("application/sdp"))
                route(rtpengine_invite);

        # sequential request withing a dialog should
        # take the path determined by record-routing
        if (loose_route()) {
                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 statelessly
                        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")) {
                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);
        }

        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 authentication
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;
        }

#!else

        # authentication not enabled - do not relay at all to foreign networks
#       if(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;
}

# RTPProxy control and signaling 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(nat_uac_test("8")) {
                rtpproxy_manage("co");
        } else {
                rtpproxy_manage("cor");
        }

        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 routing 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;
        }

        # normalize target number for pstn gateway
        # - convert leading 00 to +
        if (starts_with("$rU", "00")) {
                strip(2);
                prefix("+");
        }

        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($sel(cfg_get.voicemail.srv_ip))) {
                xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n");
                return;
        }
        if(is_method("INVITE")) {
                if($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(rtpengine_answer);
                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_BLOCK401407
        # block call redirect based on 401, 407 replies.
        if (t_check_status("401|407")) {
                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
}
route[rtpengine_invite] {
        if (has_body("application/sdp"))
                rtpengine_manage("codec-mask=telephone-event transcode=PCMA always-transcode");
}

route[rtpengine_answer] {
         if (has_body("application/sdp"))
            rtpengine_manage("always-transcode");
}

28.09.2021

kamailio rtpengine media timeout TIPs

  1. timeout will be raised only if both sides of rtp is silent
  2. you have to enable tcp at kamailio config (disable_tcp=no)
  3. you must have in kamailio cfg: listen=tcp:127.0.0.1:8090, loadmodule “xmlrpc.so”, and additional params:
loadmodule "xmlrpc.so"
modparam("xmlrpc", "route", "XMLRPCS")
modparam("xmlrpc", "url_skip", "^/sip")
modparam("xmlrpc", "url_match", "^/RPC2")

4. you have to add route XMLRPC:

route[XMLRPCS] {
  xlog("L_ALERT","RPC recieved");
  dispatch_rpc();
}

5. config rtpengine should have:

 b2b-url = http://127.0.0.1:8090/RPC2
 xmlrpc-format = 2

6. after restart kamailio and rtpengine you may to check you may do command from command line:

curl http://127.0.0.1:8090/RPC2

output should be like that:

<?xml version="1.0"?>
<methodResponse>
<fault>
<value>
<struct>
<member>
<name>faultCode</name>
<value><int>400</int></value>
</member>
<member>
<name>faultString</name>
<value><string>Invalid XML-RPC Document</string></value>
</member>
</struct>
</value>
</fault>
</methodResponse>

20.09.2021

RTPENGINE + G729 DEBIAN 10.10 and Debian 11

apt update
apt upgrade -y
apt install -y linux-headers-$(uname -r) linux-image-$(uname -r)

##reboot

mkdir /opt/rtpengine

cd /opt/rtpengine

apt install -qqy git curl mariadb-server libmosquitto-dev libwebsockets-dev python3-websockets apt-utils devscripts dpkg-dev debhelper iptables iptables-dev libcurl4-openssl-dev libglib2.0-dev libavcodec-extra libhiredis-dev libpcre3-dev libssl-dev libxmlrpc-core-c3-dev markdown zlib1g-dev module-assistant dkms gettext default-libmysqlclient-dev gperf libavcodec-dev libavfilter-dev libavformat-dev libavutil-dev libbencode-perl libcrypt-openssl-rsa-perl libcrypt-rijndael-perl libdigest-crc-perl libdigest-hmac-perl libevent-dev libio-multiplex-perl libio-socket-inet6-perl libjson-glib-dev libnet-interface-perl libpcap0.8-dev libsocket6-perl libswresample-dev libsystemd-dev nfs-common netcat-openbsd netcat unzip libconfig-tiny-perl libspandsp-dev**

git clone https://github.com/sipwise/rtpengine.git .
cp etc/rtpengine.sample.conf /etc/rtpengine/rtpengine.conf

VER=1.0.4

curl https://codeload.github.com/BelledonneCommunications/bcg729/tar.gz/$VER >bcg729_$VER.orig.tar.gz

tar zxf bcg729_$VER.orig.tar.gz

cd bcg729-$VER

git clone https://github.com/ossobv/bcg729-deb.git debian

dpkg-buildpackage -us -uc -sa

cd ../

dpkg -i libbcg729-*.deb


export DEBIAN_FRONTEND=noninteractive

apt-get update -qqy

mkdir -p ./debian/flavors

touch ./debian/flavors/no_ngcp


dpkg-checkbuilddeps

dpkg-buildpackage -b -us -uc

dpkg -i ../*.deb

**manual edit: /etc/rtpengine/rtpengine.conf: set your IPs.

If you want to add webrtc2sip feature use next lines

certbot certonly -d webrtc.domain.com

**manual edit: kamailiorc: uncomment db engine string

kamdbctl create kamailio 


download template kamailio.cfg, tls.cfg and kamailio_local.cfg 

(git clone https://bitbucket.org/erewin/webrtc2sip-template.git)

systemctl start rtpengine
systemctl start kamailio

To make the same on Debian 11

#!/usr/bin/sh
apt update
apt upgrade

#here you have reboot

#!/usr/bin/sh
mkdir /opt/rtpengine

cd /opt/rtpengine

apt-get install git curl -y

apt-get install libmosquitto-dev libwebsockets-dev python3-websockets apt-utils dpkg-dev debhelper iptables libxtables-dev libip6tc-dev libip4tc-dev libcurl4-openssl-dev libglib2.0-dev libavcodec-extra libhiredis-dev libpcre3-dev libssl-dev libxmlrpc-core-c3-dev markdown zlib1g-dev module-assistant dkms gettext default-libmysqlclient-dev gperf libavcodec-dev libavfilter-dev libavformat-dev libavutil-dev libbencode-perl libcrypt-openssl-rsa-perl libcrypt-rijndael-perl libdigest-crc-perl libdigest-hmac-perl libevent-dev libio-multiplex-perl libio-socket-inet6-perl libjson-glib-dev libnet-interface-perl libpcap0.8-dev libsocket6-perl libswresample-dev libsystemd-dev nfs-common netcat-openbsd netcat unzip libconfig-tiny-perl libspandsp-dev
#debian 11
apt install libxtables-dev libip6tc-dev libip4tc-dev libiptc-dev

apt install -y linux-headers-$(uname -r) linux-image-$(uname -r)

git clone https://github.com/sipwise/rtpengine.git .


VER=1.0.4

curl https://codeload.github.com/BelledonneCommunications/bcg729/tar.gz/$VER >bcg729_$VER.orig.tar.gz

tar zxf bcg729_$VER.orig.tar.gz

cd bcg729-$VER

git clone https://github.com/ossobv/bcg729-deb.git debian

dpkg-buildpackage -us -uc -sa

cd ../

dpkg -i libbcg729-*.deb


export DEBIAN_FRONTEND=noninteractive

apt-get update -qqy

mkdir -p ./debian/flavors

touch ./debian/flavors/no_ngcp


dpkg-checkbuilddeps

dpkg-buildpackage -b -us -uc

dpkg -i ../*.deb

8.08.2021

Kamailio TLS-UDP with dispatcher and used old openssl 1.0.2 (sslv2\3 support)

This project is how to convert TLS-UDP with kamailio. Problem is that modern Unix (Ubuntu and debian have only modern openssl library so it’s not support ssv2\3 protocol). To make it works you have to

  • do this steps at vanilla system.
  • get source of kamailio
  • for compiling using /make include_modules=”tls”/
  • rtpengine should be installed as usual
  • then everything should go as usual

Benefits from this configs is that working for inbound\outbound calls and use Dispatcher.
this is example of dispatcher file:

#
# dispatcher destination sets (groups)
#

# line format
# setid(int) destination(sip uri) flags(int,opt) priority(int,opt) attributes(str,opt)


1 sip:PEER1:5070;transport=udp 0 0 socket=udp:KAM_SOCKET_UDP:5070
1 sip:PERR2;transport=udp 0 0 socket=udp:KAM_SOcKET_UDP:5070

2 sip:TLS_CARRIER:5061;transport=tls 0 0 socket=tls:TLS_SOCKET_KAM:5061

Example of kamailio config you may get here.

#!KAMAILIO
#
############################################################
# *** Value defines - IDs used later in config
#!ifdef WITH_DEBUG
#!define DBGLEVEL 3
#!else
#!define DBGLEVEL 2
#!endif
#!define DS_LIST "/usr/local/etc/kamailio/dispatcher.list"
#!define LISTEN_UDP_PRIVATE udp:LOCAL_INTERFACE_IP:5070
#!define FLAG_FROM_ASTERISK 10
#!define FLAG_FROM_PEER 11


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

#	FLB_ - per branch flags
#!define FLB_NATB 6
#!define FLB_NATSIPPING 7

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

/* LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR, ... */
debug=DBGLEVEL

/* set to 'yes' to print log messages to terminal or use '-E' cli option */
log_stderror=no

memdbg=5
memlog=5

log_facility=LOG_LOCAL0
log_prefix="{$mt $hdr(CSeq) $ci} "

children=8

auto_aliases=no

server_signature=no

alias="DOMAIN_NAME"

listen=udp:LOCAL_INTERFACE_IP:5070
listen=tls:LOCAL_INTERFACE_IP:5061 advertise EXTERNAL_IP:5061

tcp_connection_lifetime=3605

tcp_max_connections=20000

tcp_accept_no_cl=yes

enable_tls=yes
tls_max_connections=20000

enable_sctp=no

####### Modules Section ########
mpath="/usr/local/lib64/kamailio/modules/"


#loadmodule "db_mysql.so"
loadmodule "kex.so"
loadmodule "corex.so"
loadmodule "tm.so"
loadmodule "pike.so"
loadmodule "htable.so"
loadmodule "nathelper.so"
loadmodule "tmx.so"
loadmodule "sl.so"
loadmodule "rr.so"
loadmodule "pv.so"
loadmodule "maxfwd.so"
#loadmodule "registrar"
#loadmodule "usrloc.so"
loadmodule "textops.so"
loadmodule "textopsx.so"
loadmodule "dialog.so"
loadmodule "tls.so"
loadmodule "siputils.so"
loadmodule "xlog.so"
loadmodule "sanity.so"
loadmodule "ctl.so"
loadmodule "cfg_rpc.so"
loadmodule "acc.so"
loadmodule "counters.so"
loadmodule "dispatcher.so"
loadmodule "outbound.so"
loadmodule "rtpengine.so"
loadmodule "path.so"


####Module Specific Parameters####
modparam("rr", "enable_double_rr", 1)

modparam("tls", "config", "/etc/kamailio/tls.cfg")
modparam("path", "use_received", 1)


modparam("acc", "log_flag", FLT_ACC)
modparam("acc", "failed_transaction_flag", FLT_ACCFAILED)
modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si")

modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2223")
modparam("pike", "sampling_time_unit", 2)
modparam("pike", "reqs_density_per_unit", 16)
modparam("pike", "remove_latency", 4)
modparam("htable", "htable", "ipban=>size=8;autoexpire=300;")


modparam("dispatcher", "list_file", DS_LIST)
#modparam("dispatcher", "db_url", DBURL)
modparam("dispatcher", "table_name", "dispatcher")
modparam("dispatcher", "flags", 2)
modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=480;code=404")
modparam("dispatcher", "ds_probing_mode", 1)
modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=480;code=404")
modparam("dispatcher", "force_dst", 1)
modparam("dispatcher", "ds_ping_interval", 20)
modparam("dispatcher", "ds_ping_from", "sip:keepalive@smsglobal.com")
modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=480;code=404")
modparam("nathelper", "received_avp", "$avp(s:rcv)")


###Routing Logic

request_route {

	# per request initial checks
	route(REQINIT);
	
	# CANCEL processing
	if (is_method("CANCEL")) {
		if (t_check_trans()) {
			route(RELAY);
		}
		exit;
	}

	# handle retransmissions
	if (!is_method("ACK")) {
		if(t_precheck_trans()) {
			t_check_trans();
			exit;
		}
		t_check_trans();
	}

        route(CHECK_SOURCE_IP);

	# handle requests within SIP dialogs
	route(WITHINDLG);

	### only initial requests (no To tag)

	# 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
	}

	# 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
	route(DISPATCH);
}


route[RELAY] {
    if (is_method("INVITE")) {
        if(!t_is_set("failure_route")) {
            t_on_failure("MANAGE_FAILURE");
        }
    }

    if (isflagset(FLAG_FROM_PEER)) {
	    xlog("L_INFO","seems call from $si goig from PEER");
	    
    } else {
            
            xlog("L_INFO","Relaying TO TLS\n ");
    }


    if (!t_relay()) {
            sl_reply_error();
    }
    #exit;

}

# Per SIP request initial checks
route[REQINIT] {
	if (!mf_process_maxfwd_header("10")) {
		sl_send_reply("483","Too Many Hops");
		exit;
	}

	if(!sanity_check("1511", "7")) {
		xlog("Malformed SIP message from $si:$sp\n");
		exit;
	}
	force_rport();
	
	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|sipcli|sipvicious|VaxSIPUserAgent") {
		# silent drop for scanners - uncomment next line if want to reply
		# sl_send_reply("200", "OK");
		exit;
	}
	if(is_method("OPTIONS") && uri==myself && $rU==$null) {
		sl_send_reply("200","Keepalive");
		exit;
	}



}

route[CHECK_SOURCE_IP] {
    if(ds_is_from_list("1")) {
            setflag(FLAG_FROM_ASTERISK);
    } else {
            setflag(FLAG_FROM_PEER);
    }
}

# Handle requests within SIP dialogs
route[WITHINDLG] {
	if (has_totag()) {
		# sequential request withing a dialog should
		# take the path determined by record-routing
		if (loose_route()) {
			if (is_method("BYE")) {
				setflag(FLT_ACC); # do accounting ...
				setflag(FLT_ACCFAILED); # ... even if the transaction fails
			}
			route(RELAY);
		} else {
			if (is_method("SUBSCRIBE") && uri == myself) {
				# in-dialog subscribe requests
				exit;
			}
			if ( is_method("ACK") ) {
				if ( t_check_trans() ) {
					# non loose-route, but stateful ACK;
					# must be ACK after a 487 or e.g. 404 from upstream server
					t_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;

	sl_send_reply("404", "Not Acceptable");
	exit;
}

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

	sl_send_reply("404", "Not Acceptable");
	exit;
}

# Dispatch requests
route[DISPATCH] {
        # round robin dispatching on gateways group '1'
        # record routing for dialog forming requests (in case they are routed)
        # - remove preloaded route headers
        remove_hf("Route");
        if (is_method("INVITE|REFER")) {
                record_route();
                if (has_body("application/sdp")) {
                    if (rtpengine_offer()) {
                            t_on_reply("1");
                    }
                } else {
                    t_on_reply("2");
                }

                if (isflagset(FLAG_FROM_PEER)) {
		    xlog("L_INFO","Call from $si seems from PEER");
                    if(!ds_select_domain("1", "4")) {
                            send_reply("404", "No destination");
                            exit;
                    }

                }

                if (isflagset(FLAG_FROM_ASTERISK)) {
		    xlog("L_INFO","Call from $si seems from ASTERISK");
                    if(!ds_select_domain("2", "4")) {

                            send_reply("404", "No destination");
                            exit;
                    }
		    xlog("L_INFO","Call from $si seems from ASTERISK [$du] [$ru]");
		    
                }
		    

		    xlog("L_INFO","DESTINATION is $du");
        }

        if (is_method("ACK") && has_body("application/sdp")) {
                rtpengine_answer("force");
        }

        route(RELAY);
}


failure_route[RTF_DISPATCH] {
	if (t_is_canceled()) {
		exit;
	}
	# next DST - only for 500 or local timeout
	if (t_check_status("500")
			or (t_branch_timeout() and !t_branch_replied())) {
		if(ds_next_dst()) {
			xdbg("--- SCRIPT: retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n");
			t_on_failure("RTF_DISPATCH");
			route(RELAY);
			exit;
		}
	}
}

onreply_route[1] {
        if (has_body("application/sdp")) {
                rtpengine_answer("force");
        }
}

onreply_route[2] {
        if (has_body("application/sdp")) {
                rtpengine_offer("force");
        }
}

31.07.2021

install opensips 3.1 Debian 10 + RTPPROXY 2.2

apt update
apt upgrade -y
apt-key adv --keyserver keyserver.ubuntu.com --recv-keys 049AD65B
echo "deb https://apt.opensips.org buster 3.1-releases" >/etc/apt/sources.list.d/opensips.list
echo "deb https://apt.opensips.org buster cli-nightly" >/etc/apt/sources.list.d/opensips-cli.list
apt update
apt install opensips*
apt install mariadb-server
opensips-cli -> database create opensips
apt install build-essential letsencrypt -y
(rtpproxy manual: https://www.rtpproxy.org/doc/master/user_manual.html#idm650)
useradd rtpproxy
cd /usr/src
git clone -b master https://github.com/sippy/rtpproxy.git
git -C rtpproxy submodule update --init --recursive
cd rtpproxy
./configure
make clean all
make install
put this content to /lib/systemd/system/rtpproxy.service 
-----
[Unit]
Description=RTPProxy media server
After=network.target
Requires=network.target

[Service]
Type=simple
PIDFile=/var/run/rtpproxy/rtpproxy.pid
Environment='OPTIONS= -f -L 4096 -l 0.0.0.0 -m 10000 -M 20000 -d INFO:LOG_LOCAL5'

Restart=always
RestartSec=5

ExecStartPre=-/bin/mkdir /var/run/rtpproxy
ExecStartPre=-/bin/chown rtpproxy:rtpproxy /var/run/rtpproxy

ExecStart=/usr/local/bin/rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -s udp:127.0.0.1:22222 \
 -u rtpproxy:rtpproxy -n udp:127.0.0.1:22223 $OPTIONS

ExecStop=/usr/bin/pkill -F /var/run/rtpproxy/rtpproxy.pid

ExecStopPost=-/bin/rm -R /var/run/rtpproxy

StandardOutput=syslog
StandardError=syslog
SyslogIdentifier=rtpproxy
SyslogFacility=local5

TimeoutStartSec=10
TimeoutStopSec=10

[Install]
WantedBy=multi-user.target
---------------------------

30.07.2021

Front End.

Для создания телефонных проектов частно нужен веб интерфейс, скажем сейчас я даже не знаю ни одного проекта без предоставления веб-интерфейса. Все знают что есть фреймворки, и простые вещи такие как показать таблички из одной базы, что-то загрузить что-то отфильтровать – просто. Сложности начинаются когда надо действительно внедрить что-то что вы не знаете как делать силами фреймворка, а знать фреймворк хорошо это быть полноценным веб разработчиком. На этом этапе нам не нужно быть веб-разработчиками, но мы должны давать клиента хорошего качества вебстраницу на которой он сможет управлять своими телефонными сервисами.

Итак, я остановился, на том, что в проектах буду использовать:

Это простые системы, хорошо задокументированные. При помощи них легко создавать страницы для управления таблицами, загрузками или выгрузкой файлов и даже рисовать графики.

А шаблонизатор smarty позволяет разделить код и дизайн, чтобы фрон-энд разработчик в дальнейшем смог, делать любой дизайн.

Примеры:

26.07.2021

To deploy some Voip service

  • What server you want to use:
    • Standalone (physically server)
    • Dedicated Server
    • Virtual Machine (AWS,GCP,DO and so on)
  • Do you have requirements for OS and related software:
    • Debian\Centos\Ubuntu
    • Mysql/PostgreSQL
  • What Voip trunk you have or how you have connected to voice services:
    • I have two trunks with a-z routes and cheap prices.
  • What amount of calls you expect
    • normal: less 5cps, less: 100 concurrent channels,
    • high load: less 20 cps, less 1000 cc,
    • highest unlimited cps, unlimited cc,

29.05.2021

opensips realtime failover HA example with keepalive.

статья присутствует на испанском.

| Posted in opensips | No Comments »
5.05.2021

freepbx 15 automatic sync for high availability. part 1.

Part 1
main aim is using internal features of freepbx to make sync between two PBXs. Automatic sync will be made by back&restore feature in FREEPBX.

STEP: 1. add asterisk public key from master FREEPBX to backup server. master public key for asterisk user you can find in menu “Backup&Restore -> Global Setting”

Master asterisk public key

Copy this key to /home/asterisk/.ssh/authorized_keys at Backup machine.

STEP: 2. Add new filestore with SSH connect to backup server

Create filestore with ssh access to backup server

STEP: 3. example of ssh parameters for ssh connect

STEP 4: create new backup in menu “Backup&Restore”. use any modules you need and set crontime you wish. for example i will be use one time per day. Better you set “Delete After Runs” 1 so only 1 file will be in backup server.

Now run it and check if new file with backup created in /home/asterisk dir at backup server.

STEP5: add cronjob at backup: something like that: crontab -u asterisk -e
0 1 * * * fwconsole backup –restore /home/asterisk/*.tar.gz

22.04.2021

Аренда серверов со скидками

Мы предоставляем в аренду полноценные виртуальные АТС (freepbx\Issabel\FusionPBX) по запросу на электронный адрес.

  • для тестов 500 рублей\мес. (2 cpu\2 гб\20Гб HDD\external ip\domain 3 level),
  • для работы 1500 рублей\мес. (2 cpu\4 гб\60Гб HDD\external ip\domain 3 level),
  • для работы 3500 рублей\мес. (4 cpu\4 гб\120Гб HDD\external ip\domain 2 level),

Скидки у партнеров: для краткосрочной аренды и проведение тестов самое оно!