i am trying to route all calls to twilio through kamailio proxy. with my config file, call gets connected and automatically drops after about 30 seconds. This is because ACK sent to twilio for 200 OK was not correct. Twilio expects ACK with ruri same as contact in 200 OK response, but kamailio sent was different. How to fix this error?
200 OK response
kvE@.H|k
SIP/2.0 200 OK
To: <sip:4803513158@54.69.159.69>;tag=52642973_6772d868_f5ff7ec8-2860-4963-894d-4686de291299
Via: SIP/2.0/UDP 54.69.159.69:5060;branch=z9hG4bKcdde.3a10daa4c4dfb45218165e7003bc1925.0
Via: SIP/2.0/UDP 64.2.142.90;branch=z9hG4bKcdde.0da1e6a5.0
Via: SIP/2.0/UDP 64.2.142.153:5060;received=64.2.142.153;branch=z9hG4bK0c18e9b5;rport=5060
Record-Route: <sip:107.21.211.20:5060;lr;ftag=as0b111d7b>
Record-Route: <sip:54.69.159.69:5060;lr=on>
Record-Route: <sip:64.2.142.90;lr=on>
CSeq: 102 INVITE
Call-ID: 3e944c2d4543707439a08bd63612841f@64.2.142.153
From: "+14088271419" <sip:4088271419@64.2.142.153>;tag=as0b111d7b
Contact: <sip:10.108.170.11:5060>
Content-Type: application/sdp
X-Twilio-CallSid: CA2d519006d553f3acf89f1dd2fef77cc3
Content-Length: 247
v=0
o=- 1132985615 1132985615 IN IP4 54.82.64.195
s=session
c=IN IP4 54.82.64.195
t=0 0
m=audio 13800 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
ACK forwarded to twilio
E@d
kCACK sip:107.21.211.20:5060;lr;ftag=as0b111d7b SIP/2.0
Via: SIP/2.0/UDP 54.69.159.69:5060;branch=z9hG4bKcdde.95210bf832aa38aeeccc727ba583b66f.0
Via: SIP/2.0/UDP 54.69.159.69:5060;branch=z9hG4bKcdde.0a606545da03734a8b332edc9b21079f.0
Via: SIP/2.0/UDP 64.2.142.90;branch=z9hG4bKcdde.0da1e6a5.2
Via: SIP/2.0/UDP 64.2.142.153:5060;received=64.2.142.153;branch=z9hG4bK32a73a53;rport=5060
From: "+14088271419" <sip:4088271419@64.2.142.153>;tag=as0b111d7b
To: <sip:4803513158@64.2.142.90:5060>;tag=52642973_6772d868_f5ff7ec8-2860-4963-894d-4686de291299
Contact: <sip:4088271419@64.2.142.153>
Call-ID: 3e944c2d4543707439a08bd63612841f@64.2.142.153
CSeq: 102 ACK
User-Agent: packetrino
Max-Forwards: 67
Content-Length: 0
this is my config file.(ip and account names are changed)
#!KAMAILIO
#
# sample config file for dispatcher module
# - load balancing of VoIP calls with round robin
# - no TPC listening
# - don't dispatch REGISTER and presence requests
# Direct your questions about this file to: sr-users@lists.sip-router.org
#
#
# *** To run in debug mode:
# - define WITH_DEBUG
#
#!ifndef DBURL
#!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio"
#!endif
####### 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
fork=yes
children=4
/* comment the next line to enable TCP */
disable_tcp=yes
/* uncomment the next line to disable the auto discovery of local aliases
based on revers DNS on IPs (default on) */
auto_aliases=no
/* add local domain aliases */
alias="54.69.159.69"
port=5060
/* 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:172.31.22.10:5060 advertise 54.69.159.69:5060
#advertised_address="54.69.159.69"
#advertised_port=5060
sip_warning=no
####### Modules Section ########
#set module path
mpath="/usr/local/lib64/kamailio/modules_k/:/usr/local/lib64/kamailio/modules/"
loadmodule "db_mysql.so"
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 "textops.so"
loadmodule "siputils.so"
loadmodule "xlog.so"
loadmodule "sanity.so"
loadmodule "ctl.so"
loadmodule "mi_rpc.so"
loadmodule "acc.so"
loadmodule "dispatcher.so"
loadmodule "pdt.so"
loadmodule "nathelper.so"
loadmodule "avpops.so"
loadmodule "htable.so"
# ----------------- setting module-specific parameters ---------------
# ----- mi_fifo params -----
modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")
# ----- 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)
# ----- acc params -----
modparam("acc", "log_flag", 1)
modparam("acc", "failed_transaction_flag", 3)
modparam("acc", "log_extra",
"src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si")
# ----- tm params -----
modparam("tm", "fr_timer", 2000)
modparam("tm", "fr_inv_timer", 40000)
# ----- dispatcher params -----
modparam("dispatcher", "db_url", DBURL)
modparam("dispatcher", "table_name", "dispatcher")
modparam("dispatcher", "flags_col", "flags")
modparam("dispatcher", "priority_col", "priority")
modparam("dispatcher", "dst_avp", "$avp(AVP_DST)")
modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)")
modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)")
modparam("dispatcher", "force_dst", 1)
#-------pdt--------------
modparam("pdt", "db_table", "pdt")
modparam("pdt", "domain_column", "sdomain")
modparam("pdt", "prefix_column", "prefix")
modparam("pdt", "domain_column", "domain")
#-----------------nat-----------
modparam("nathelper", "natping_interval", 0)
modparam("nathelper", "ping_nated_only", 0)
#modparam("nathelper", "sipping_bflag", FLB_NATSIPPING)
modparam("nathelper", "sipping_from", "sip:172.31.22.10:5060")
#-----------acc------------
modparam("acc", "report_ack", 1)
####### Routing Logic ########
# main request routing logic
route {
# per request initial checks
route(REQINIT);
# handle requests within SIP dialogs
route(WITHINDLG);
### only initial requests (no To tag)
# CANCEL processing
if (is_method("CANCEL"))
{
if (t_check_trans())
t_relay();
exit;
}
t_check_trans();
# record routing for dialog forming requests (in case they are routed)
# - remove preloaded route headers
remove_hf("Route");
if (is_method("INVITE|SUBSCRIBE"))
#add_rr_param(";nat=yes");
record_route_advertised_address("54.69.159.69:5060");
#record_route();
# account only INVITEs
if (is_method("INVITE"))
{
setflag(1); # do accounting
}
# handle presence related requests
route(PRESENCE);
# handle registrations
route(REGISTRAR);
if ($rU==$null)
{
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}
# dispatch destinations
route(DISPATCH);
}
route[RELAY] {
if (!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;
}
}
# Handle requests within SIP dialogs
route[WITHINDLG] {
t_check_trans();
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
if (is_method("BYE")) {
setflag(1); # do accounting ...
setflag(3); # ... even if the transaction fails
}
route(RELAY);
} else {
if (is_method("SUBSCRIBE") && uri == myself) {
# in-dialog subscribe requests
route(PRESENCE);
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", "No registrar");
exit;
}
# Presence server route
route[PRESENCE] {
if(!is_method("PUBLISH|SUBSCRIBE"))
return;
sl_send_reply("404", "Not here");
exit;
}
# Dispatch requests
route[DISPATCH] {
if(prefix2domain("2", "0")) {
$ru = "sip:" + $rU + "@" + $rd;
t_relay();
exit;
}
# round robin dispatching on gateways group '1'
if(!ds_select_dst("1", "4"))
{
send_reply("404", "No destination");
exit;
}
xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n");
t_on_failure("RTF_DISPATCH");
route(RELAY);
exit;
}
# 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;
}
}
}
