[SR-Users] One way Audio with kamailio -Rtpproxy
master voip
voip.batsman at gmail.com
Wed Jun 18 16:42:03 CEST 2014
Hi can some one help me . I am using kamailio 4.1.3 and having one way
audio issue.
Scenario is
Pjsip 2.0 based UA (private IP)------->Router(publicIP)--------> Kamailio
(with RTP proxy)-----------> Third Party_Sip Server ------>PSTN
UA is registering fine with sip server but when i make a call they can
hear but i can not hear the other party .
What i found is in SDP first c=IN is private ip of UA and second c=IN is
public ip of UA i dont know if this is the cause
When I trace the call on sip server , sip server is sending RTP to UA
private ip so
it is not reaching kamailio back to forward to UA .
I am attaching the kamailio configuration file as well as SIP/SDP trace
user agent private ip : 192.168.1.4
User agent public ip : 61.61.61.61
Kamailio IP : 81.81.81.81
Sip Server IP: 71.71.71.71
(* ip addresses are not actual ips)
Please reply with detailed instructions to fix this isue
config file:
#!KAMAILIO
#
# Kamailio (OpenSER) SIP Server v4.1 - default configuration script
# - web: http://www.kamailio.org
# - git: http://sip-router.org
#
# Direct your questions about this file to: <sr-users at lists.sip-router.org>
#
# Refer to the Core CookBook at http://www.kamailio.org/wiki/
# for an explanation of possible statements, functions and parameters.
#
# Several features can be enabled using '#!define WITH_FEATURE' directives:
#
# *** To run in debug mode:
# - define WITH_DEBUG
#
# *** To enable mysql:
# - define WITH_MYSQL
#
# *** To enable authentication execute:
# - enable mysql
# - define WITH_AUTH
# - add users using 'kamctl'
#
# *** To enable IP authentication execute:
# - enable mysql
# - enable authentication
# - define WITH_IPAUTH
# - add IP addresses with group id '1' to 'address' table
#
# *** To enable persistent user location execute:
# - enable mysql
# - define WITH_USRLOCDB
#
# *** To enable presence server execute:
# - enable mysql
# - define WITH_PRESENCE
#
#!define WITH_NAT
# *** 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
#
# *** To enable PSTN gateway routing execute:
# - define WITH_PSTN
# - set the value of pstn.gw_ip
# - check route[PSTN] for regexp routing condition
#
# *** To enable database aliases lookup execute:
# - enable mysql
# - define WITH_ALIASDB
#
# *** To enable speed dial lookup execute:
# - enable mysql
# - define WITH_SPEEDDIAL
#
# *** To enable multi-domain support execute:
# - enable mysql
# - define WITH_MULTIDOMAIN
#
# *** To enable TLS support execute:
# - adjust CFGDIR/tls.cfg as needed
# - define WITH_TLS
#
# *** To enable XMLRPC support execute:
# - define WITH_XMLRPC
# - adjust route[XMLRPC] for access policy
#
# *** To enable anti-flood detection execute:
# - adjust pike and htable=>ipban settings as needed (default is
# block if more than 16 requests in 2 seconds and ban for 300 seconds)
# - define WITH_ANTIFLOOD
#
# *** To block 3XX redirect replies execute:
# - define WITH_BLOCK3XX
#
# *** To enable VoiceMail routing execute:
# - define WITH_VOICEMAIL
# - set the value of voicemail.srv_ip
# - adjust the value of voicemail.srv_port
#
# *** To enhance accounting execute:
# - enable mysql
# - define WITH_ACCDB
# - add following columns to database
#!ifdef ACCDB_COMMENT
ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT '';
ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT '';
ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default '';
ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT '';
ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT '';
ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT '';
ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT
'';
ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL
DEFAULT '
';
ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default
'';
ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL
DEFAULT '';
ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT
'';
ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL
DEFAULT '
';
#!endif
####### Include Local Config If Exists #########
import_file "kamailio-local.cfg"
####### Defined Values #########
# *** Value defines - IDs used later in config
#!ifdef WITH_MYSQL
# - database URL - used to connect to database server by modules such
# as: auth_db, acc, usrloc, a.s.o.
#!ifndef DBURL
#!define DBURL "mysql://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=2
log_stderror=no
#!else
debug=3
log_stderror=no
#!endif
memdbg=1
memlog=1
log_facility=LOG_LOCAL0
fork=yes
children=4
/* uncomment the next line to disable TCP (default on) */
disable_tcp=yes
/* uncomment the next line to disable the auto discovery of local aliases
based on reverse DNS on IPs (default on) */
#auto_aliases=no
/* add local domain aliases */
#alias="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:10.0.0.10:5060
/* port to listen to
* - can be specified more than once if needed to listen on many ports */
port=7878
#!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 (to sources or installation folders)
#!ifdef WITH_SRCPATH
mpath="modules/"
#!else
mpath="/usr/local/lib/kamailio/modules/"
#!endif
#!ifdef WITH_MYSQL
loadmodule "db_mysql.so"
#!endif
loadmodule "mi_fifo.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 "mi_rpc.so"
loadmodule "acc.so"
#!ifdef WITH_AUTH
loadmodule "auth.so"
loadmodule "auth_db.so"
#!ifdef WITH_IPAUTH
loadmodule "permissions.so"
#!endif
#!endif
#!ifdef WITH_ALIASDB
loadmodule "alias_db.so"
#!endif
#!ifdef WITH_SPEEDDIAL
loadmodule "speeddial.so"
#!endif
#!ifdef WITH_MULTIDOMAIN
loadmodule "domain.so"
#!endif
#!ifdef WITH_PRESENCE
loadmodule "presence.so"
loadmodule "presence_xml.so"
#!endif
#!ifdef WITH_NAT
loadmodule "nathelper.so"
loadmodule "rtpproxy.so"
loadmodule "nat_traversal.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 ---------------
# ----- mi_fifo params -----
modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")
# ----- 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 -----
# add value to ;lr param to cope with most of the UAs
modparam("rr", "enable_full_lr", 1)
# 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=$rd")
#!endif
# ----- usrloc params -----
/* enable DB persistency for location entries */
#!ifdef WITH_USRLOCDB
modparam("usrloc", "db_url", DBURL)
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "use_domain", MULTIDOMAIN)
#!endif
# ----- auth_db params -----
#!ifdef WITH_AUTH
modparam("auth_db", "db_url", DBURL)
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("auth_db", "load_credentials", "")
modparam("auth_db", "use_domain", MULTIDOMAIN)
# ----- permissions params -----
#!ifdef WITH_IPAUTH
modparam("permissions", "db_url", DBURL)
modparam("permissions", "db_mode", 1)
#!endif
#!endif
# ----- alias_db params -----
#!ifdef WITH_ALIASDB
modparam("alias_db", "db_url", DBURL)
modparam("alias_db", "use_domain", MULTIDOMAIN)
#!endif
# ----- speeddial params -----
#!ifdef WITH_SPEEDDIAL
modparam("speeddial", "db_url", DBURL)
modparam("speeddial", "use_domain", MULTIDOMAIN)
#!endif
# ----- domain params -----
#!ifdef WITH_MULTIDOMAIN
modparam("domain", "db_url", DBURL)
# register callback to match myself condition with domains list
modparam("domain", "register_myself", 1)
#!endif
#!ifdef WITH_PRESENCE
# ----- presence params -----
modparam("presence", "db_url", DBURL)
# ----- presence_xml params -----
modparam("presence_xml", "db_url", DBURL)
modparam("presence_xml", "force_active", 1)
#!endif
#!ifdef WITH_NAT
# ----- rtpproxy params -----
modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722")
# ----- nathelper params -----
modparam("nathelper", "natping_interval", 30)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_bflag", FLB_NATSIPPING)
modparam("nathelper", "sipping_from", "sip:pinger at kamailio.org")
# params needed for NAT traversal in other modules
modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)")
modparam("usrloc", "nat_bflag", FLB_NATB)
#!endif
#!ifdef WITH_TLS
# ----- tls params -----
modparam("tls", "config", "/usr/local/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", 7)
#!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 requests within SIP dialogs
route(WITHINDLG);
### only initial requests (no To tag)
t_check_trans();
# authentication
# Below line commented by test-MATHUR as PC-2-PC calls are not
getting
through
#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"))
{
sl_send_reply("100", "Trying");
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);
}
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()) {
xlog("testM : Unable to relay !!!! :{ \n");
sl_reply_error();
}
exit;
}
# Per SIP request initial checks
route[REQINIT] {
#!ifdef WITH_ANTIFLOOD
# flood dection from same IP and traffic ban for a while
# be sure you exclude checking trusted peers, such as pstn gateways
# - local host excluded (e.g., loop to self)
if(src_ip!=myself)
{
if($sht(ipban=>$si)!=$null)
{
# ip is already blocked
xdbg("request from blocked IP - $rm from $fu
(IP:$si:$sp
)\n");
exit;
}
if (!pike_check_req())
{
xlog("L_ALERT","ALERT: pike blocking $rm from $fu
(IP:$s
i:$sp)\n");
$sht(ipban=>$si) = 1;
exit;
}
}
#!endif
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;
}
}
# 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()) {
route(DLGURI);
if (is_method("BYE")) {
#xlog("testM : Got BYE from some
PSTN/UAC.\n");
setflag(FLT_ACC); # do accounting ...
#xlog("testM : Did accounting.\n");
setflag(FLT_ACCFAILED); # ... even if the
transa
ction fails
#xlog("testM : Calling route{NATMANAGE}
from WI
THINDLG.\n");
#t_newtran();
#t_reply("200", "OK");
#xlog("testM : Called route{NATMANAGE} for
BYE
froom WITHINDLG.\n");
#exit;
}
else if ( is_method("ACK") ) {
# ACK is forwarded statelessy
route(NATMANAGE);
}
else if ( is_method("NOTIFY") ) {
# Add Record-Route for in-dialog NOTIFY as
per R
FC 6665.
record_route();
}
route(RELAY);
} else {
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
... i
gnore and discard
exit;
}
}
sl_send_reply("404","Not here");
}
exit;
}
}
# Handle SIP registrations
route[REGISTRAR] {
if (is_method("REGISTER"))
{
if(isflagset(FLT_NATS))
{
setbflag(FLB_NATB);
# uncomment next line to do SIP NAT pinging
## setbflag(FLB_NATSIPPING);
}
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 route
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;
}
# Authentication route
route[AUTH] {
#!ifdef WITH_AUTH
#!ifdef WITH_IPAUTH
if((!is_method("REGISTER")) && allow_source_address())
{
# source IP allowed
return;
}
#!endif
if (is_method("REGISTER") || from_uri==myself)
{
# authenticate requests
if (!auth_check("$fd", "subscriber", "1")) {
auth_challenge("$fd", "0");
exit;
}
# user authenticated - remove auth header
if(!is_method("REGISTER|PUBLISH"))
consume_credentials();
}
# if caller is not local subscriber, then check if it calls
# a local destination, otherwise deny, not an open relay here
if (from_uri!=myself && uri!=myself)
{
sl_send_reply("403","Not relaying");
exit;
}
#!endif
return;
}
# Caller NAT detection route
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
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)))
#{
# rtpproxy_manage();
# return;
#}
rtpproxy_manage("z90");
if (is_request()) {
if (!has_totag()) {
if(t_is_branch_route()) {
if ((isflagset(FLT_NATS) ||
isbflagset(FLB_NATB)
)) {
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)
{
append_hf("P-hint: outbound\r\n");
route(RELAY);
}
}
# PSTN GW routing
route[PSTN] {
#!ifdef WITH_PSTN
# check if PSTN GW IP is defined
if (strempty($sel(cfg_get.pstn.gw_ip))) {
xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not
defined\n"
);
return;
}
# route to PSTN dialed numbers starting with '+' or '00'
# (international format)
# - update the condition to match your dialing rules for PSTN
routing
if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$"))
return;
# only local users allowed to call
if(from_uri!=myself) {
sl_send_reply("403", "Not Allowed");
exit;
}
if (strempty($sel(cfg_get.pstn.gw_port))) {
$ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip);
} else {
$ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":"
+ $sel(cfg_get.pstn.gw_port);
}
route(RELAY);
exit;
#!endif
return;
}
# XMLRPC routing
#!ifdef WITH_XMLRPC
route[XMLRPC] {
# allow XMLRPC from localhost
if ((method=="POST" || method=="GET")
&& (src_ip==127.0.0.1)) {
# close connection only for xmlrpclib user agents (there is
a bu
g in
# xmlrpclib: it waits for EOF before interpreting the
response).
if ($hdr(User-Agent) =~ "xmlrpclib")
set_reply_close();
set_reply_no_connect();
dispatch_rpc();
exit;
}
send_reply("403", "Forbidden");
exit;
}
#!endif
# route to voicemail server
route[TOVOICEMAIL] {
#!ifdef WITH_VOICEMAIL
if(!is_method("INVITE|SUBSCRIBE"))
return;
# check if VoiceMail server IP is defined
if (strempty($sel(cfg_get.voicemail.srv_ip))) {
xlog("SCRIPT: VoiceMail rotuing enabled but IP not
defined\n");
return;
}
if(is_method("INVITE")) {
if($avp(oexten)==$null)
return;
$ru = "sip:" + $avp(oexten) + "@" +
$sel(cfg_get.voicemail.srv_i
p)
+ ":" + $sel(cfg_get.voicemail.srv_port);
} else {
if($rU==$null)
return;
$ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip)
+ ":" + $sel(cfg_get.voicemail.srv_port);
}
route(RELAY);
exit;
#!endif
return;
}
# manage outgoing branches
branch_route[MANAGE_BRANCH] {
xdbg("new branch [$T_branch_idx] to $ru\n");
route(NATMANAGE);
}
# manage incoming replies
onreply_route[MANAGE_REPLY] {
xdbg("incoming reply\n");
if(status=~"[12][0-9][0-9]")
route(NATMANAGE);
}
# manage failure routing cases
failure_route[MANAGE_FAILURE] {
route(NATMANAGE);
if (t_is_canceled()) {
exit;
}
#!ifdef WITH_BLOCK3XX
# block call redirect based on 3xx replies.
if (t_check_status("3[0-9][0-9]")) {
t_reply("404","Not found");
exit;
}
#!endif
#!ifdef WITH_VOICEMAIL
# serial forking
# - route to voicemail on busy or no answer (timeout)
if (t_check_status("486|408")) {
$du = $null;
route(TOVOICEMAIL);
exit;
}
#!endif
}
------------------------------------------------------------------------------------------------------------------
SIP TRACE
REGISTER sip:71.71.71.71 SIP/2.0
Via: SIP/2.0/UDP 81.81.81.81:7878
;branch=z9hG4bK1b54.98ebaeef387ced8eb54624e6c7a90504.0
Via: SIP/2.0/UDP 192.168.1.4:58931
;received=61.61.61.61;rport=58931;branch=z9hG4bKPjI5YxCTUEIkkWEg0VswESOCq7lpvwYUkE
Max-Forwards: 69
From: "57778" <sip:test at 71.71.71.71>;tag=ODxcd4MPCfZc.o5M6gAZObocSE9ZpbyW
To: "57778" <sip:test at 71.71.71.71>
Call-ID: XsaPsPHKODzjjDr9TgogZFZs61hetHv7
CSeq: 62147 REGISTER
User-Agent: pj_arubaslim-16/r2
Contact: "57778" <sip:test at 192.168.1.4:58931;ob>
Expires: 900
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Content-Length: 0
P-hint: outbound
SIP/2.0 407 Proxy Authentication Required
CSeq: 62147 REGISTER
Via: SIP/2.0/UDP 81.81.81.81:7878
;branch=z9hG4bK1b54.98ebaeef387ced8eb54624e6c7a90504.0
Via: SIP/2.0/UDP 192.168.1.4:58931
;branch=z9hG4bKPjI5YxCTUEIkkWEg0VswESOCq7lpvwYUkE
From: "57778" <sip:test at 71.71.71.71>;tag=ODxcd4MPCfZc.o5M6gAZObocSE9ZpbyW
Call-ID: XsaPsPHKODzjjDr9TgogZFZs61hetHv7
To: "57778" <sip:test at 71.71.71.71>;tag=170604141058
Proxy-Authenticate: DIGEST realm="sip.mydomain.co",
nonce="140301389817101016806045824224"
Content-Length: 0
REGISTER sip:71.71.71.71 SIP/2.0
Via: SIP/2.0/UDP 81.81.81.81:7878
;branch=z9hG4bKfb54.1eaeef967bf46fb647eed0bb87a9406e.0
Via: SIP/2.0/UDP 192.168.1.4:58931
;received=61.61.61.61;rport=58931;branch=z9hG4bKPjRPPULjuny1lG1x2Qn0UumpeHPLGxqQHI
Max-Forwards: 69
From: "57778" <sip:test at 71.71.71.71>;tag=ODxcd4MPCfZc.o5M6gAZObocSE9ZpbyW
To: "57778" <sip:test at 71.71.71.71>
Call-ID: XsaPsPHKODzjjDr9TgogZFZs61hetHv7
CSeq: 62148 REGISTER
User-Agent: pj_arubaslim-16/r2
Contact: "57778" <sip:test at 192.168.1.4:58931;ob>
Expires: 900
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Proxy-Authorization: Digest username="test", realm="sip.mydomain.co",
nonce="140301389817101016806045824224", uri="sip:71.71.71.71",
response="0ca765f266b6bfc8ea0ef00227272393"
Content-Length: 0
P-hint: outbound
SIP/2.0 200 OK
CSeq: 62148 REGISTER
Via: SIP/2.0/UDP 81.81.81.81:7878
;branch=z9hG4bKfb54.1eaeef967bf46fb647eed0bb87a9406e.0
Via: SIP/2.0/UDP 192.168.1.4:58931
;branch=z9hG4bKPjRPPULjuny1lG1x2Qn0UumpeHPLGxqQHI
From: "57778" <sip:test at 71.71.71.71>;tag=ODxcd4MPCfZc.o5M6gAZObocSE9ZpbyW
Call-ID: XsaPsPHKODzjjDr9TgogZFZs61hetHv7
To: "57778" <sip:test at 71.71.71.71>;tag=170604141059
Contact: "57778" <sip:test at 192.168.1.4:58931;ob>;expires=600
Expires: 600
Content-Length: 0
SUBSCRIBE sip:test at 71.71.71.71 SIP/2.0
Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes>
Via: SIP/2.0/UDP 81.81.81.81:7878
;branch=z9hG4bK0f2c.8c652a23fe5f790c4bb9216d48b80199.0
Via: SIP/2.0/UDP 192.168.1.4:58931
;received=61.61.61.61;rport=58931;branch=z9hG4bKPj.syjKlsSGhtf420D3Gs4.gYpLomA2oQq
Max-Forwards: 69
From: "57778" <sip:test at 71.71.71.71>;tag=hkjR9-fCrHBhiGYK1xpZGvZmEqjx6bIC
To: "57778" <sip:test at 71.71.71.71>
Contact: "57778" <sip:test at 192.168.1.4:58931;ob;alias=61.61.61.61~58931~1>
Call-ID: RvFl7GSaJwV6YVHkewSAJ4NjqH239DqB
CSeq: 10987 SUBSCRIBE
Event: message-summary
Expires: 3600
Supported: replaces, 100rel, timer, norefersub
Accept: application/simple-message-summary
Allow-Events: presence, message-summary, refer
User-Agent: pj_arubaslim-16/r2
Content-Length: 0
P-hint: outbound
SIP/2.0 401 Unauthorised
CSeq: 10987 SUBSCRIBE
Via: SIP/2.0/UDP 81.81.81.81:7878
;branch=z9hG4bK0f2c.8c652a23fe5f790c4bb9216d48b80199.0
Via: SIP/2.0/UDP 192.168.1.4:58931
;branch=z9hG4bKPj.syjKlsSGhtf420D3Gs4.gYpLomA2oQq
From: "57778" <sip:test at 71.71.71.71>;tag=hkjR9-fCrHBhiGYK1xpZGvZmEqjx6bIC
Call-ID: RvFl7GSaJwV6YVHkewSAJ4NjqH239DqB
To: "57778" <sip:test at 71.71.71.71>;tag=170605141000
Content-Length: 0
WWW-Authenticate: DIGEST realm="sip.mydomain.co",
nonce="140301390017101016806050024224"
Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes>
SUBSCRIBE sip:test at 71.71.71.71 SIP/2.0
Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes>
Via: SIP/2.0/UDP 81.81.81.81:7878
;branch=z9hG4bKef2c.315393ac3a4f54a9fd7fe71cb140b41b.0
Via: SIP/2.0/UDP 192.168.1.4:58931
;received=61.61.61.61;rport=58931;branch=z9hG4bKPjIFtOYolYGyRGCBo77jg6j38Co3oSuB6l
Max-Forwards: 69
From: "57778" <sip:test at 71.71.71.71>;tag=hkjR9-fCrHBhiGYK1xpZGvZmEqjx6bIC
To: "57778" <sip:test at 71.71.71.71>
Contact: "57778" <sip:test at 192.168.1.4:58931;ob;alias=61.61.61.61~58931~1>
Call-ID: RvFl7GSaJwV6YVHkewSAJ4NjqH239DqB
CSeq: 10988 SUBSCRIBE
Event: message-summary
Expires: 3600
Supported: replaces, 100rel, timer, norefersub
Accept: application/simple-message-summary
Allow-Events: presence, message-summary, refer
User-Agent: pj_arubaslim-16/r2
Authorization: Digest username="test", realm="sip.mydomain.co",
nonce="140301390017101016806050024224", uri="sip:test at 71.71.71.71",
response="15ee3de757c18e084fca607438017093"
Content-Length: 0
P-hint: outbound
SIP/2.0 200 OK
CSeq: 10988 SUBSCRIBE
Via: SIP/2.0/UDP 81.81.81.81:7878
;branch=z9hG4bKef2c.315393ac3a4f54a9fd7fe71cb140b41b.0
Via: SIP/2.0/UDP 192.168.1.4:58931
;branch=z9hG4bKPjIFtOYolYGyRGCBo77jg6j38Co3oSuB6l
From: "57778" <sip:test at 71.71.71.71>;tag=hkjR9-fCrHBhiGYK1xpZGvZmEqjx6bIC
Call-ID: RvFl7GSaJwV6YVHkewSAJ4NjqH239DqB
To: "57778" <sip:test at 71.71.71.71>;tag=170605141001
Expires: 3600
Content-Length: 0
Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes>
NOTIFY sip:test at 71.71.71.71 SIP/2.0
CSeq: 100 NOTIFY
Via: SIP/2.0/UDP 71.71.71.71:5060
From: "57778" <sip:test at 71.71.71.71>;tag=170605141001
Call-ID: RvFl7GSaJwV6YVHkewSAJ4NjqH239DqB
To: "57778" <sip:test at 71.71.71.71>;tag=hkjR9-fCrHBhiGYK1xpZGvZmEqjx6bIC
Contact: <sip:71.71.71.71:5060;transport=udp>
Subscription-State: active
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 65
Messages-Waiting: no
Message-Account: sip:test at 71.71.71.71
SIP/2.0 404 Not here
CSeq: 100 NOTIFY
Via: SIP/2.0/UDP 71.71.71.71:5060;rport=5060
From: "57778" <sip:test at 71.71.71.71>;tag=170605141001
Call-ID: RvFl7GSaJwV6YVHkewSAJ4NjqH239DqB
To: "57778" <sip:test at 71.71.71.71>;tag=hkjR9-fCrHBhiGYK1xpZGvZmEqjx6bIC
Server: kamailio (4.1.3 (i386/linux))
Content-Length: 0
INVITE sip:919000000002 at 71.71.71.71 SIP/2.0
Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes>
Via: SIP/2.0/UDP 81.81.81.81:7878
;branch=z9hG4bKb76c.aa93d0b16ced02bdc00227054365c825.0
Via: SIP/2.0/UDP 192.168.1.4:58931
;received=61.61.61.61;rport=58931;branch=z9hG4bKPjcJJWEomCJmbTkvY8iXMdrmQ935aS1h1g
Max-Forwards: 69
From: "57778" <sip:test at 71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ
To: <sip:919000000002 at 71.71.71.71>
Contact: "57778" <sip:test at 192.168.1.4:58931;ob;alias=61.61.61.61~58931~1>
Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81
CSeq: 23382 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: pj_arubaslim-16/r2
Content-Type: application/sdp
Content-Length: 356
P-hint: outbound
v=0
o=- 3612003234 3612003234 IN IP4 81.81.81.81
s=pjmedia
c=IN IP4 192.168.1.4
t=0 0
m=audio 43314 RTP/AVP 3 18 0 8 101
c=IN IP4 81.81.81.81
a=rtcp:43315
a=sendrecv
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=nortpproxy:yes
SIP/2.0 407 Proxy Authentication Required
CSeq: 23382 INVITE
Via: SIP/2.0/UDP 81.81.81.81:7878
;branch=z9hG4bKb76c.aa93d0b16ced02bdc00227054365c825.0
Via: SIP/2.0/UDP 192.168.1.4:58931
;branch=z9hG4bKPjcJJWEomCJmbTkvY8iXMdrmQ935aS1h1g
From: "57778" <sip:test at 71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ
Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81
To: <sip:919000000002 at 71.71.71.71>;tag=1706051410107236868006249
Contact: <sip:71.71.71.71:5060;transport=udp>
Proxy-Authenticate: DIGEST realm="sip.mydomain.co",
nonce="140301391017101016806051024224"
Content-Length: 0
Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes>
ACK sip:919000000002 at 71.71.71.71 SIP/2.0
Via: SIP/2.0/UDP 81.81.81.81:7878
;branch=z9hG4bKb76c.aa93d0b16ced02bdc00227054365c825.0
Max-Forwards: 69
From: "57778" <sip:test at 71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ
To: <sip:919000000002 at 71.71.71.71>;tag=1706051410107236868006249
Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81
CSeq: 23382 ACK
Content-Length: 0
INVITE sip:919000000002 at 71.71.71.71 SIP/2.0
Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes>
Via: SIP/2.0/UDP 81.81.81.81:7878
;branch=z9hG4bKc76c.caf535d51d5f83fc6b4139b411ebc727.0
Via: SIP/2.0/UDP 192.168.1.4:58931
;received=61.61.61.61;rport=58931;branch=z9hG4bKPjmFPFtuAR124gIuWOHue6pEiRcHdc5wkC
Max-Forwards: 69
From: "57778" <sip:test at 71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ
To: <sip:919000000002 at 71.71.71.71>
Contact: "57778" <sip:test at 192.168.1.4:58931;ob;alias=61.61.61.61~58931~1>
Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81
CSeq: 23383 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: pj_arubaslim-16/r2
Proxy-Authorization: Digest username="test", realm="sip.mydomain.co",
nonce="140301391017101016806051024224", uri="sip:919000000002 at 71.71.71.71",
response="4edf5e2efd1e8e03bdbe4a693fdfee4c"
Content-Type: application/sdp
Content-Length: 356
P-hint: outbound
v=0
o=- 3612003234 3612003234 IN IP4 81.81.81.81
s=pjmedia
c=IN IP4 192.168.1.4
t=0 0
m=audio 43314 RTP/AVP 3 18 0 8 101
c=IN IP4 81.81.81.81
a=rtcp:43315
a=sendrecv
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=nortpproxy:yes
SIP/2.0 183 Session Progress
CSeq: 23383 INVITE
Via: SIP/2.0/UDP 81.81.81.81:7878
;branch=z9hG4bKc76c.caf535d51d5f83fc6b4139b411ebc727.0
Via: SIP/2.0/UDP 192.168.1.4:58931
;branch=z9hG4bKPjmFPFtuAR124gIuWOHue6pEiRcHdc5wkC
From: "57778" <sip:test at 71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ
Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81
To: <sip:919000000002 at 71.71.71.71>;tag=1706051410107236868006249
Contact: <sip:71.71.71.71:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 252
Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes>
v=0
o=SipSwitch 6248 7248 IN IP4 71.71.71.71
s=VoipSIP
i=Audio Session
c=IN IP4 71.71.71.71
t=0 0
m=audio 6248 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
SIP/2.0 180 Ringing
CSeq: 23383 INVITE
Via: SIP/2.0/UDP 81.81.81.81:7878
;branch=z9hG4bKc76c.caf535d51d5f83fc6b4139b411ebc727.0
Via: SIP/2.0/UDP 192.168.1.4:58931
;branch=z9hG4bKPjmFPFtuAR124gIuWOHue6pEiRcHdc5wkC
From: "57778" <sip:test at 71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ
Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81
To: <sip:919000000002 at 71.71.71.71>;tag=1706051410107236868006249
Contact: <sip:71.71.71.71:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 252
Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes>
v=0
o=SipSwitch 6248 7248 IN IP4 71.71.71.71
s=VoipSIP
i=Audio Session
c=IN IP4 71.71.71.71
t=0 0
m=audio 6248 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
SIP/2.0 200 OK
CSeq: 23383 INVITE
Via: SIP/2.0/UDP 81.81.81.81:7878
;branch=z9hG4bKc76c.caf535d51d5f83fc6b4139b411ebc727.0
Via: SIP/2.0/UDP 192.168.1.4:58931
;branch=z9hG4bKPjmFPFtuAR124gIuWOHue6pEiRcHdc5wkC
From: "57778" <sip:test at 71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ
Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81
To: <sip:919000000002 at 71.71.71.71>;tag=1706051410107236868006249
Contact: <sip:71.71.71.71:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 252
Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes>
v=0
o=SipSwitch 6248 7248 IN IP4 71.71.71.71
s=VoipSIP
i=Audio Session
c=IN IP4 71.71.71.71
t=0 0
m=audio 6248 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
ACK sip:71.71.71.71:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 81.81.81.81:7878
;branch=z9hG4bKc76c.e624c28002d41f48ae76813b26e516f3.0
Via: SIP/2.0/UDP 192.168.1.4:58931
;received=61.61.61.61;rport=58931;branch=z9hG4bKPjJ8XsuToNj.92GHJLiuK-JZ5T5PpfTbhr
Max-Forwards: 69
From: "57778" <sip:test at 71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ
To: <sip:919000000002 at 71.71.71.71>;tag=1706051410107236868006249
Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81
CSeq: 23383 ACK
Content-Length: 0
BYE sip:71.71.71.71:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 81.81.81.81:7878
;branch=z9hG4bK976c.6ca2249268e6f9592a6948becb29c474.0
Via: SIP/2.0/UDP 192.168.1.4:58931
;received=61.61.61.61;rport=58931;branch=z9hG4bKPjJy-pnGAhtqwqV5uRixN-mvDWOTPLupLU
Max-Forwards: 69
From: "57778" <sip:test at 71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ
To: <sip:919000000002 at 71.71.71.71>;tag=1706051410107236868006249
Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81
CSeq: 23384 BYE
User-Agent: pj_arubaslim-16/r2
Content-Length: 0
SIP/2.0 200 OK
CSeq: 23384 BYE
Via: SIP/2.0/UDP 81.81.81.81:7878
;branch=z9hG4bK976c.6ca2249268e6f9592a6948becb29c474.0
Via: SIP/2.0/UDP 192.168.1.4:58931
;received=61.61.61.61;rport=58931;branch=z9hG4bKPjJy-pnGAhtqwqV5uRixN-mvDWOTPLupLU
From: "57778" <sip:test at 71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ
Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81
To: <sip:919000000002 at 71.71.71.71>;tag=1706051410107236868006249
Contact: <sip:71.71.71.71:5060;transport=udp>
Content-Length: 0
Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes>
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