[SR-Users] Older Sip Phone ia hangup after 60sec with Realtime asterisk
Noriyuki Hayashi
nhayashi at wats.gr.jp
Mon Jan 6 08:50:48 CET 2014
Hello,
I am beginner using kamailio with much appreciated.
Only one sip-phone is hang up after 60 seconds problem.
This sip phone has no nat function at all.(SANYO SIP-2100)
Grand Stream is works fine with kamailio.
I would like give me your great advice with much appreciated.
Environment.
CentOS5.10, Asterisk-11.6.0 with PostgreSQL-9.2.5 as Realtime.
Kamailio-4.1.0
Only Asterisk and PostgreSQL with older sip phone works fine.
If Kamailio is running that registered is OK, But meetme(example) is hangup after 60 sec.
I do not know "reINVITE or RTP" problem.
Kamailio.cfg
#!KAMAILIO
#!enable postgresql
#!define WITH_PGSQL
#!define WITH_AUTH
#!define WITH_USRLOCDB
#!define WITH_ASTERISK
#!define WITH_NAT
#!define WITH_DISPATCHER
#!define WITH_ANTIFLOOD
#!define WITH_MULTIDOMAIN
#!define WITH_WITHINDLG
#!define WITH_DEBUG
#!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
####### Defined Values #########
# *** Value defines - IDs used later in config
#!ifdef WITH_PGSQL
# - database URL - used to connect to database server by modules such
# as: auth_db, acc, usrloc, a.s.o.
#!define DBURL "postgres://postgres:password@localhost/kamailio"
#!ifdef WITH_ASTERISK
#!define DBASTURL "postgres://asterisk:password@localhost/asterisk"
#!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 #########
#!ifdef WITH_DEBUG
debug=4
log_stderror=yes
#!else
debug=2
log_stderror=no
#!endif
memdbg=5
memlog=5
#log_facility=LOG_LOCAL0
log_facility=LOG_LOCAL7
fork=yes
children=4
check_via=no # (cmd. line: -v)
dns=off # (cmd. line: -r)
rev_dns=off # (cmd. line: -R)
/* uncomment the next line to disable TCP (default on) */
#disable_tcp=yes
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
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 */
#listen=udp:192.168.192.92
port=5060
mhomed=1
#!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 #########
#!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"
#!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
#!ifdef WITH_ASTERISK
asterisk1.bindip = "192.168.192.92" desc "Asterisk IP Address"
asterisk1.bindport = "5080" desc "Asterisk Port"
asterisk2.bindip = "192.168.192.93" desc "Asterisk IP Address"
asterisk2.bindport = "5080" desc "Asterisk Port"
kamailio.bindip = "192.168.192.92" desc "Kamailio IP Address"
kamailio.bindport = "5060" desc "Kamailio Port"
#!endif
####### Modules Section ########
# set paths to location of modules (to sources or installation folders)
#!ifdef WITH_SRCPATH
mpath="modules/"
#!else
mpath="/usr/lib64/kamailio/modules/"
#!endif
#!ifdef WITH_PGSQL
loadmodule "db_postgres.so"
#!endif
loadmodule "mi_fifo.so"
loadmodule "kex.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"
#!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
#!ifdef WITH_ASTERISK
loadmodule "uac.so"
#!endif
#!ifdef WITH_DISPATCHER
loadmodule "dispatcher.so"
loadmodule "sqlops.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)
#!ifdef WITH_ASTERISK
modparam("rr", "append_fromtag", 1)
#!else
modparam("rr", "append_fromtag", 0)
#!endif
# ----- registrar params -----
modparam("registrar", "method_filtering", 1)
/* uncomment the next line to disable parallel forking via location */
# modparam("registrar", "append_branches", 0)
modparam("registrar", "append_branches", 0)
/* uncomment the next line not to allow more than 10 contacts per AOR */
#modparam("registrar", "max_contacts", 10)
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", "calculate_ha1", yes)
modparam("auth_db", "load_credentials", "")
#!ifdef WITH_ASTERISK
modparam("auth_db", "user_column", "name")
modparam("auth_db", "password_column", "sippasswd")
modparam("auth_db", "db_url", DBASTURL)
modparam("auth_db", "version_table", 0)
#!else
modparam("auth_db", "db_url", DBURL)
modparam("auth_db", "password_column", "password")
modparam("auth_db", "use_domain", MULTIDOMAIN)
#!endif
# ----- 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
# ----- speedial 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", "$avp(RECEIVED)")
modparam("usrloc", "nat_bflag", FLB_NATB)
#!endif
#!ifdef WITH_TLS
# ----- tls params -----
modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg")
#!endif
#!ifdef WITH_ANTIFLOOD
# ----- pike params -----
modparam("pike", "sampling_time_unit", 2)
modparam("pike", "reqs_density_per_unit", 16)
modparam("pike", "remove_latency", 4)
# ----- htable params -----
# ip ban htable with autoexpire after 5 minutes
modparam("htable", "htable", "ipban=>size=8;autoexpire=300;")
#!endif
#!ifdef WITH_XMLRPC
# ----- xmlrpc params -----
modparam("xmlrpc", "route", "XMLRPC");
modparam("xmlrpc", "url_match", "^/RPC")
#!endif
#!ifdef WITH_DEBUG
# ----- debugger params -----
modparam("debugger", "cfgtrace", 1)
#!endif
#!ifdef WITH_DISPATCHER
# ----- dispatcher params -----
modparam("dispatcher", "db_url", DBURL)
modparam("dispatcher", "table_name", "dispatcher")
modparam("dispatcher", "flags", 2)
modparam("dispatcher", "dst_avp", "$avp(AVP_DST)")
modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)")
modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)")
modparam("sqlops","sqlcon", "ca=>postgres://asterisk:password@localhost/kamailio")
#!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);
# handle requests within SIP dialogs
route(WITHINDLG);
### only initial requests (no To tag)
# CANCEL processing
if (is_method("CANCEL"))
{
if (t_check_trans())
t_relay();
exit;
}
t_check_trans();
# authentication
route(AUTH);
# record routing for dialog forming requests (in case they are routed)
# - remove preloaded route headers
remove_hf("Route");
if (is_method("INVITE|SUBSCRIBE"))
record_route();
# account only INVITEs
if (is_method("INVITE"))
{
setflag(FLT_ACC); # do accounting
}
# 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);
}
route[RELAY] {
# enable additional event routes for forwarded requests
# - serial forking, RTP relaying handling, a.s.o.
if (is_method("INVITE|SUBSCRIBE")) {
t_on_branch("MANAGE_BRANCH");
t_on_reply("MANAGE_REPLY");
}
if (is_method("INVITE")) {
t_on_failure("MANAGE_FAILURE");
}
if (!t_relay()) {
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 $fu (IP:$si:$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()) {
if (is_method("BYE")) {
setflag(FLT_ACC); # do accounting ...
setflag(FLT_ACCFAILED); # ... even if the transaction fails
}
if ( is_method("ACK") ) {
# ACK is forwarded statelessy
route(NATMANAGE);
}
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
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"))
{
if(isflagset(FLT_NATS))
{
setbflag(FLB_NATB);
# uncomment next line to do SIP NAT pinging
## setbflag(FLB_NATSIPPING);
}
if (!save("location"))
sl_reply_error();
#!ifdef WITH_ASTERISK
route(REGFWD);
#!endif
exit;
}
}
# USER location service
route[LOCATION] {
#!ifdef WITH_SPEEDIAL
# 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
#!ifdef WITH_ASTERISK
if(is_method("INVITE") && (!route(FROMASTERISK))) {
# if new call from out there - send to Asterisk
# - non-INVITE request are routed directly by Kamailio
# - traffic from Asterisk is routed also directy by Kamailio
route(TOASTERISK);
exit;
}
#!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);
}
}
# Presence server route
route[PRESENCE] {
if(!is_method("PUBLISH|SUBSCRIBE"))
return;
#!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_ASTERISK
# do not auth traffic from Asterisk - trusted!
if(route(FROMASTERISK))
return;
#!endif
#!ifdef WITH_IPAUTH
if((!is_method("REGISTER")) && allow_source_address())
{
# source IP allowed
return;
}
#!endif
if (is_method("REGISTER") || from_uri==myself)
{
# authenticate requests
#!ifdef WITH_ASTERISK
if (!auth_check("$fd", "sip_devices", "1")) {
#!else
if (!auth_check("$fd", "subscriber", "1")) {
#!endif
auth_challenge("$fd", "0");
exit;
}
# user authenticated - remove auth header
if(!is_method("REGISTER|PUBLISH"))
consume_credentials();
}
# if caller is not local subscriber, then check if it calls
# a local destination, otherwise deny, not an open relay here
if (from_uri!=myself && uri!=myself)
{
sl_send_reply("403","Not relaying");
exit;
}
#!endif
return;
}
# Caller NAT detection route
route[NATDETECT] {
#!ifdef WITH_NAT
force_rport();
if (nat_uac_test("19")) {
if (is_method("REGISTER")) {
fix_nated_register();
} else {
fix_nated_contact();
}
setflag(FLT_NATS);
}
#!endif
return;
}
# RTPProxy control
route[NATMANAGE] {
#!ifdef WITH_NAT
if (is_request()) {
if(has_totag()) {
if(check_route_param("nat=yes")) {
setbflag(FLB_NATB);
}
}
}
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
return;
rtpproxy_manage();
if (is_request()) {
if (!has_totag()) {
add_rr_param(";nat=yes");
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
fix_nated_contact();
}
}
#!endif
return;
}
# Routing to foreign domains
route[SIPOUT] {
if (!uri==myself)
{
append_hf("P-hint: outbound\r\n");
route(RELAY);
}
}
# PSTN GW routing
route[PSTN] {
#!ifdef WITH_PSTN
# check if PSTN GW IP is defined
if (strempty($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;
}
$ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip);
route(RELAY);
exit;
#!endif
return;
}
# XMLRPC routing
#!ifdef WITH_XMLRPC
route[XMLRPC] {
# allow XMLRPC from localhost
if ((method=="POST" || method=="GET")
&& (src_ip==127.0.0.1)) {
# close connection only for xmlrpclib user agents (there is a bug in
# xmlrpclib: it waits for EOF before interpreting the response).
if ($hdr(User-Agent) =~ "xmlrpclib")
set_reply_close();
set_reply_no_connect();
dispatch_rpc();
exit;
}
send_reply("403", "Forbidden");
exit;
}
#!endif
# route to voicemail server
route[TOVOICEMAIL] {
#!ifdef WITH_VOICEMAIL
if(!is_method("INVITE"))
return;
# check if VoiceMail server IP is defined
if (strempty($sel(cfg_get.voicemail.srv_ip))) {
xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n");
return;
}
if($avp(oexten)==$null)
return;
$ru = "sip:" + $avp(oexten) + "@" + $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 $ru\n");
route(NATMANAGE);
}
# manage incoming replies
onreply_route[MANAGE_REPLY] {
xdbg("incoming reply\n");
if(status=~"[12][0-9][0-9]")
route(NATMANAGE);
}
# manage failure routing cases
failure_route[MANAGE_FAILURE] {
route(NATMANAGE);
if (t_is_canceled()) {
exit;
}
#!ifdef WITH_BLOCK3XX
# block call redirect based on 3xx replies.
if (t_check_status("3[0-9][0-9]")) {
t_reply("404","Not found");
exit;
}
#!endif
#!ifdef WITH_VOICEMAIL
# serial forking
# - route to voicemail on busy or no answer (timeout)
if (t_check_status("486|408")) {
route(TOVOICEMAIL);
exit;
}
#!endif
}
#!ifdef WITH_ASTERISK
# Test if coming from Asterisk
route[FROMASTERISK] {
if(ds_is_from_list())
{
return 1;
} else {
return -1;
}
}
# Send to Asterisk
route[TOASTERISK] {
if($(au{s.len})<=5)
{
$var(setid) = 0;
xlog("SCRIPT: Connected Asterisk #0 - using set $var(setid) \n");
} else {
$var(setid) = 9;
xlog("SCRIPT: Connected Asterisk #9 - using set $var(setid) \n");
}
# failover dispatching on set determined above
if(!ds_select_dst($var(setid), "8"))
{
send_reply("404", "No destination");
exit;
}
t_on_failure("RTF_DISPATCH");
route(RELAY);
exit;
}
# Forward REGISTER to Asterisk
route[REGFWD] {
if(!is_method("REGISTER"))
{
return;
}
if($(au{s.len})<=5)
{
$var(rip) = $sel(cfg_get.asterisk1.bindip);
$uac_req(method)="REGISTER";
$uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk1.bindport);
} else {
$var(rip) = $sel(cfg_get.asterisk2.bindip);
$uac_req(method)="REGISTER";
$uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk2.bindport);
}
$uac_req(furi)="sip:" + $au + "@" + $var(rip);
$uac_req(turi)="sip:" + $au + "@" + $var(rip);
$uac_req(hdrs)="Contact: <sip:" + $au + "@"
+ $sel(cfg_get.kamailio.bindip)
+ ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n";
if($sel(contact.expires) != $null)
$uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n";
else
$uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n";
uac_req_send();
}
# Sample failure route
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())
{
t_on_failure("RTF_DISPATCH");
route(RELAY);
exit;
}
}
}
#!endif
*** Test call to meetme Logs. ****
sip1*CLI> sip set debug on
sip1*CLI> SIP Debugging re-enabled
sip1*CLI> sip set debug on
sip1*CLI>
Name/username Host Dyn Forcerport ACL Port Status Description Realtime
99206/99206 192.168.192.92 D N 5060 OK (515 ms) Cached RT
1 sip peers [Monitored: 1 online, 0 offline Unmonitored: 0 online, 0 offline]
sip1*CLI> -- Executing [901 at 99:1] Answer("SIP/99206-00000000", "")
Audio is at 15506
sip1*CLI> Adding codec 100003 (ulaw) to SDP
sip1*CLI> Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
sip1*CLI>
<--- Reliably Transmitting (NAT) to 192.168.192.92:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.192.92;branch=z9hG4bKac02.538bb4a63719b3291fac2770ec3b5b31.0
Via: SIP/2.0/UDP 192.168.192.190:5060;rport=5060;branch=z9hG4bK-52cad077-29932-3be9
Record-Route: <sip:192.168.192.92;lr=on;ftag=bec0a8c0-13c4-52cad076-29850-7422;nat=yes>
From: Richard Nough<sip:99206 at 192.168.192.92>;tag=bec0a8c0-13c4-52cad076-29850-7422
To: <sip:901 at 192.168.192.92>;tag=as7cd1f3fc
Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404 at 192.168.192.92
CSeq: 2 INVITE
Session-Expires: 120;refresher=uas
Contact: <sip:901 at 192.168.192.92:5080>
Content-Type: application/sdp
Require: timer
Content-Length: 284
v=0
o=root 729993436 729993436 IN IP4 192.168.192.92
s=Asterisk PBX 11.6.0
c=IN IP4 192.168.192.92
t=0 0
m=audio 15506 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
sip1*CLI> Retransmitting #1 (NAT) to 192.168.192.92:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.192.92;branch=z9hG4bKac02.538bb4a63719b3291fac2770ec3b5b31.0
Via: SIP/2.0/UDP 192.168.192.190:5060;rport=5060;branch=z9hG4bK-52cad077-29932-3be9
Record-Route: <sip:192.168.192.92;lr=on;ftag=bec0a8c0-13c4-52cad076-29850-7422;nat=yes>
From: Richard Nough<sip:99206 at 192.168.192.92>;tag=bec0a8c0-13c4-52cad076-29850-7422
To: <sip:901 at 192.168.192.92>;tag=as7cd1f3fc
Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404 at 192.168.192.92
CSeq: 2 INVITE
Session-Expires: 120;refresher=uas
Contact: <sip:901 at 192.168.192.92:5080>
Content-Type: application/sdp
Require: timer
Content-Length: 284
v=0
o=root 729993436 729993436 IN IP4 192.168.192.92
s=Asterisk PBX 11.6.0
c=IN IP4 192.168.192.92
t=0 0
m=audio 15506 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
sip1*CLI>
<--- SIP read from UDP:192.168.192.92:5060 --->
ACK sip:901 at 192.168.192.92:5080 SIP/2.0
From: Richard Nough<sip:99206 at 192.168.192.92>;tag=bec0a8c0-13c4-52cad076-29850-7422
To: <sip:901 at 192.168.192.92>;tag=as7cd1f3fc
Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404 at 192.168.192.92
CSeq: 2 ACK
Via: SIP/2.0/UDP 192.168.192.92;branch=z9hG4bKac02.9f229e8f27e7aa8e3fad6cc83135f434.0
Via: SIP/2.0/UDP 192.168.192.190:5060;rport=5060;branch=z9hG4bK-52cad077-29a6e-306e
Max-Forwards: 16
Contact: <sip:99206 at 192.168.192.190:5060>
Proxy-Authorization: Digest username="99206", realm="192.168.192.92", nonce="UspTFFLKUehXxuLrJ9AbLwT69Jtg5MFQ", uri="sip:901 at 192.168.192.92", response="8d49c10f184381801aac43fc45b15117", algorithm=MD5
Content-Length:0
<------------->
--- (11 headers 0 lines) ---
sip1*CLI>
<--- SIP read from UDP:192.168.192.92:5060 --->
ACK sip:901 at 192.168.192.92:5080 SIP/2.0
From: Richard Nough<sip:99206 at 192.168.192.92>;tag=bec0a8c0-13c4-52cad076-29850-7422
To: <sip:901 at 192.168.192.92>;tag=as7cd1f3fc
Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404 at 192.168.192.92
CSeq: 2 ACK
Via: SIP/2.0/UDP 192.168.192.92;branch=z9hG4bKac02.9f229e8f27e7aa8e3fad6cc83135f434.0
Via: SIP/2.0/UDP 192.168.192.190:5060;rport=5060;branch=z9hG4bK-52cad077-29a6e-306e
Max-Forwards: 16
Contact: <sip:99206 at 192.168.192.190:5060>
Proxy-Authorization: Digest username="99206", realm="192.168.192.92", nonce="UspTFFLKUehXxuLrJ9AbLwT69Jtg5MFQ", uri="sip:901 at 192.168.192.92", response="8d49c10f184381801aac43fc45b15117", algorithm=MD5
Content-Length:0
<------------->
--- (11 headers 0 lines) ---
sip1*CLI> -- Executing [901 at 99:2] Wait("SIP/99206-00000000", "1")
sip1*CLI> > 0x17aa0bd0 -- Probation passed - setting RTP source address to 192.168.192.190:17096
sip1*CLI> -- Executing [901 at 99:3] Authenticate("SIP/99206-00000000", "5963")
sip1*CLI> -- <SIP/99206-00000000> Playing 'agent-pass.gsm' (language 'ja')
sip1*CLI> -- <SIP/99206-00000000> Playing 'auth-thankyou.gsm' (language 'ja')
sip1*CLI> -- Executing [901 at 99:4] MeetMe("SIP/99206-00000000", "99901,pM")
== Parsing '/etc/asterisk/meetme.conf': Found
sip1*CLI> -- Created MeetMe conference 1023 for conference '99901'
sip1*CLI> -- <SIP/99206-00000000> Playing 'conf-onlyperson.gsm' (language 'ja')
sip1*CLI> -- Started music on hold, class 'default', on SIP/99206-00000000
sip1*CLI> -- Stopped music on hold on SIP/99206-00000000
sip1*CLI> -- Started music on hold, class 'default', on SIP/99206-00000000
sip1*CLI> Audio is at 15506
Adding codec 100003 (ulaw) to SDP
Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.192.92:5060:
INVITE sip:99206 at 192.168.192.190:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.192.92:5080;branch=z9hG4bK71165cd1;rport
Max-Forwards: 70
From: <sip:901 at 192.168.192.92>;tag=as7cd1f3fc
To: Richard Nough<sip:99206 at 192.168.192.92>;tag=bec0a8c0-13c4-52cad076-29850-7422
Contact: <sip:901 at 192.168.192.92:5080>
Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404 at 192.168.192.92
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.6.0
Session-Expires: 120;refresher=uac
Min-SE: 90
Allow: INVITE, ACK, CANCEL, BYE
X-asterisk-Info: SIP re-invite (Session-Timers)
Content-Type: application/sdp
Content-Length: 284
v=0
o=root 729993436 729993436 IN IP4 192.168.192.92
s=Asterisk PBX 11.6.0
c=IN IP4 192.168.192.92
t=0 0
m=audio 15506 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
sip1*CLI>
<--- SIP read from UDP:192.168.192.92:5060 --->
SIP/2.0 404 Not here
Via: SIP/2.0/UDP 192.168.192.92:5080;branch=z9hG4bK71165cd1;rport=5080
From: <sip:901 at 192.168.192.92>;tag=as7cd1f3fc
To: Richard Nough<sip:99206 at 192.168.192.92>;tag=bec0a8c0-13c4-52cad076-29850-7422
Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404 at 192.168.192.92
CSeq: 102 INVITE
Server: kamailio (4.1.0 (x86_64/linux))
Content-Length: 0
---
> [INSERT INTO cdr ("calldate","clid","src","dst","dcontext","channel","lastapp","lastdata","duration","billsec","disposition","amaflags","accountcode","uniqueid") VALUES ('2014-01-06 15:49:12','"Richard Nough" <99206>','99206','901','99','SIP/99206-00000000','MeetMe','99901,pM',60,60,'ANSWERED',3,'nori at wats','1388990952.0')]
<--- SIP read from UDP:192.168.192.92:5060 --->
SIP/2.0 404 Not here
Via: SIP/2.0/UDP 192.168.192.92:5080;branch=z9hG4bK339a9eb8;rport=5080
From: <sip:901 at 192.168.192.92>;tag=as7cd1f3fc
To: Richard Nough<sip:99206 at 192.168.192.92>;tag=bec0a8c0-13c4-52cad076-29850-7422
Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404 at 192.168.192.92
CSeq: 103 BYE
Server: kamailio (4.1.0 (x86_64/linux))
Content-Length: 0
I hope you have a great 2014.
Kind regards,
Nori
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