Talant Blogs about VOIP
6.10.2021
opensips 3.1 TLS
Чтобы поднять рабочий сервер TLS-SIP На базе opensips 3.1 нужно учесть несколько моментов:
- Установить certbot (https://www.digitalocean.com/community/tutorials/how-to-secure-apache-with-let-s-encrypt-on-debian-10)
- Установить базу данных: apt install mariadb-server apache
- Установить opensips И opensips control panel
- https://apt.opensips.org/packages.php?v=3.1
- apt install opensips opensips-cli
- apt install opensips* (для ленивых конечно)
- установить сертификаты для своего домена
- Установить opensips control panel
- поправить файлик /var/www/html/opensips-cp/config/tools/system/tls_mgm/local.inc.php закомментировав validation для sip_domain И network_address
socket=udp:x.x.x.x:5060
socket=tcp:x.x.x.x:5060
socket=tls:x.x.x.x:5061
loadmodule "db_mysql.so"
loadmodule "proto_udp.so"
loadmodule "proto_tcp.so"
loadmodule "proto_tls.so"
## TLS specific settings
loadmodule "tls_mgm.so"
loadmodule "tls_openssl.so"
modparam("tls_mgm", "db_url", "mysql://opensips:opensipsrw@localhost/opensips")
6. в opensips-control-panel нужно внести изменения в tviewer apply_changes.php вместо require(“init.php”)
require("../../../../web/tools/".$_SESSION['branch']."/".$_SESSION['module_id']."/init.php");
FAQ:
ERROR:proto_tls:proto_tls_conn_init: no TLS client domain found
opensips не может найти через какой сокет установить соединение т.к. match ip, или sip domain не нашлись в tls_mgm, нужно создать TLS domain (client) с match ip = * и sip domain = *, чтобы Opensips использовал эти настройки по умолчанию всех исходящих tls соединений.
error:1417C086:SSL routines:tls_process_client_certificate:certificate verify failed
значит что выставлена проверка сертификатов, ее либо нужно отключить и перезагрузить Opensips либо загрузить на клиента сертификат для которого нужно загрузить сертификат CA на opensips.
INFO:tls_mgm:ssl_servername_cb: No domain found matching host: in servername extension
ERROR:proto_tls:tls_print_errstack: TLS errstack: error:1422E0EA:SSL routines:final_server_name:callback failed
sip_domain в параметрах указан конкретный, который не передается с сертификатом клиента
решением может быть – поставить * в sip_domain
ERROR:tls_mgm:load_tls_library: No TLS library module loaded
loadmodule “tls_openssl.so” – возможно не установлен этот модуль.
ERROR:tls_openssl:openssl_tls_conn_init: failed to create SSL structure (0:Success)
ERROR:tls_openssl:tls_print_errstack: TLS errstack: error:140BA0C3:SSL routines:SSL_new:null ssl ctx
ERROR:proto_tls:proto_tls_conn_clean: Failed to retrieve the tls_domain pointer in the SSL struct
TIPS: how to see all TLS messages (как посмотреть зашифрованный sip трафик)
opensips.cfg:
socket=hep_udp:127.0.0.1:5656
loadmodule "tracer.so"
# -- tracert --
modparam("tracer", "trace_on", 1)
modparam("tracer", "trace_id", "[tid]uri=hep:hep_dst")
loadmodule "proto_hep.so"
modparam("proto_hep", "hep_id", "[hep_dst] 127.0.0.1:5757;transport=udp;")
sngrep:
sngrep port 5757 -L udp:127.0.0.1:5757
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 = $null;
route(TOVOICEMAIL);
exit;
}
#!endif
}
route[rtpengine_invite] {
if (has_body("application/sdp"))
rtpengine_manage("codec-mask=telephone-event transcode=PCMA always-transcode");
}
route[rtpengine_answer] {
if (has_body("application/sdp"))
rtpengine_manage("always-transcode");
}
28.09.2021 kamailio rtpengine media timeout TIPs
- timeout will be raised only if both sides of rtp is silent
- you have to enable tcp at kamailio config (disable_tcp=no)
- you must have in kamailio cfg: listen=tcp:127.0.0.1:8090, loadmodule “xmlrpc.so”, and additional params:
loadmodule "xmlrpc.so"
modparam("xmlrpc", "route", "XMLRPCS")
modparam("xmlrpc", "url_skip", "^/sip")
modparam("xmlrpc", "url_match", "^/RPC2")
4. you have to add route XMLRPC:
route[XMLRPCS] {
xlog("L_ALERT","RPC recieved");
dispatch_rpc();
}
5. config rtpengine should have:
b2b-url = http://127.0.0.1:8090/RPC2 xmlrpc-format = 2
6. after restart kamailio and rtpengine you may to check you may do command from command line:
curl http://127.0.0.1:8090/RPC2
output should be like that:
<?xml version="1.0"?> <methodResponse> <fault> <value> <struct> <member> <name>faultCode</name> <value><int>400</int></value> </member> <member> <name>faultString</name> <value><string>Invalid XML-RPC Document</string></value> </member> </struct> </value> </fault> </methodResponse>20.09.2021
RTPENGINE + G729 DEBIAN 10.10 and Debian 11
apt update apt upgrade -y apt install -y linux-headers-$(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 ../*.deb8.08.2021
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");
}
}
Tags: dispatcher, Kamailio, openssl, sslv2, sslv3, tls
| Posted in kamailio, ssl\tls, Безопасность, Готовые решения | No Comments »
21.04.2021 | Posted in kamailio, ssl\tls, Безопасность, Готовые решения | No Comments »
Kamailio and Opensips
This is the list of completed solutions based on Kamailio\opensips.
KAMAILIO:
WEBRTC2SIP server – using kamailio + rtpengine, makes possible to use webphone based on webrtc to using common SIP servers
- transfer between SIP and WEBRTC protocols and vice versa.
- transconding g729 to g711
LoadBalancer – using only kamailio without DBs, makes loadbalancing between asterisks servers.
SBC – using kamailio and sql DBs for manage incoming traffic to asterisk, placed before asterisk on same PC.
- insert custom headers
- block hackers traffic
- manage CPS and monitoring tools
- control PDD of calls
OPENSIPS:
Redirect server – working with billing software JERA. Makes simultaneously outbound calls based on redirect messages from billing.
- control CPS for customers
- block unwanted traffic
- monitoring online calls with custom dashboard
- make CDRs for insert into billing
Failover server – makes possible to switch between 2 opensips servers without loosing online calls.
9.12.2020sipdump per day. compressed. heplify.
Ниже представлен скрипт для установки сервиса systemd сбора sip пакетов в папку /var/log/sipdump по дням. в дальнейшем можно распаковать файлы и и пробежаться по ним sngrep.
!/usr/bin/sh echo "Instaiiling sipdump have started: \n" yum install wget git -y apt install wget git -y cd /usr/src/ mkdir sipdump cd sipdump wget https://github.com/sipcapture/heplify/releases/download/1.62/heplify chmod 760 heplify cp heplify /usr/bin rm heplify mkdir /var/log/sipdump echo " [Unit] Description=Yooxy sipdump After=network.target ConditionPathExists=/var/log/sipdump [Service] WorkingDirectory=/var/log/sipdump ExecStart=/usr/bin/heplify -dim OPTIONS,NOTIFY -wf /var/log/sipdump -rt 1440 -zf -sl ExecReload=/bin/kill -HUP $MAINPID KillMode=process Restart=on-failure [Install] WantedBy=multi-user.target Alias=sipdump.service" > sipdump.service chmod 664 sipdump.service cp sipdump.service /etc/systemd/system/sipdump.service rm sipdump.service systemctl daemon-reload systemctl start sipdump echo "Script ending \n"
Tags: heplify, sipdump
| Posted in Asterisk, freeswitch, kamailio, opensips, Готовые решения | No Comments »
8.12.2020 | Posted in Asterisk, freeswitch, kamailio, opensips, Готовые решения | No Comments »
kamailio. siremis. xmlrpc. jsonrpc.
xmlrpc работает через порты, которые используются и для SIP. Файлы настройки протоколов для siremis
siremis/modules/sipadmin/service/
jsonrpc может работать через разные транспорты, по умолчанию работается через Unixsock.нужные параметры в конфиге kamailio:
<UnixSockLocal name="unixsocklocal" address="/var/run/siremis/siremis_rpc.sock" timeout="3.0"/><!-- kamailio.cfg: modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock") --><!-- kamailio.cfg: modparam("jsonrpcs", "dgram_mode", 0666) --> <UnixSockRemote name="unixsockremote" address="/var/run/kamailio/kamailio_rpc.sock" timeout="3.0"/>
TIPs: Возможны проблемы с разрешениями поправляется выставлением разрешения на каталог /var/run/kamailio 701 (добавить поиск для остальных пользователей) ну и сам файл sock должен быть доступен для чтения\записи
30.11.2020kamailio. Rtpproxy not apply on re-invite.
При реинвайте не применяется rtpproxy, использовал rtpproxy_manage. Проблема была в том, что при реинвайте провайдер отправлял ответ с уже включенным a=nortpproxy в sdp. соответственно kamilio просто игнорил этот ответ. полечилось добавлением в конфиг такой строчки:
modparam("rtpproxy", "nortpproxy_str", "")
| Posted in Asterisk, kamailio, rtpengine, Эксперт | No Comments »