Reverse Proxy Asterisk - SIP Provider - sr-users

24 Jul 2019

Hi all,

I've tried to create a reverse proxy to forward incoming request that came
from SIP provider to Asterisk PBX and forward the requests from asterisk to
kamailio then sip provider.
What i get is that I see the invite, but is like no ACK.
Thanks in advance.
M

kamailio.cfg:

#!KAMAILIO
#

####### Defined Values #########
# - 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
debug=3
log_stderror=yes
memdbg=5
memlog=5

log_facility=LOG_LOCAL0
log_prefix="{$mt $hdr(CSeq) $ci} "
children=1

server_id = 10
xavp_via_params = "via"
disable_tcp=yes
auto_aliases=no
listen=udp:0.0.0.0:5060

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

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 "textops.so"
loadmodule "siputils.so"
loadmodule "xlog.so"
loadmodule "sanity.so"
loadmodule "ctl.so"
loadmodule "cfg_rpc.so"
loadmodule "acc.so"
loadmodule "counters.so"

# ----------------- setting module-specific parameters ---------------

# ----- jsonrpcs params -----
modparam("jsonrpcs", "pretty_format", 1)
modparam("jsonrpcs", "fifo_name",
"/var/run/kamailio/kamailio_rpc.fifo")
modparam("jsonrpcs", "dgram_socket",
"/var/run/kamailio/kamailio_rpc.sock")
modparam("ctl", "binrpc",
"unix:/var/run/kamailio/kamailio_ctl")

# ----- tm params -----
modparam("tm", "failure_reply_mode", 3)
modparam("tm", "fr_timer", 30000)
modparam("tm", "fr_inv_timer", 120000)
modparam("rr", "enable_full_lr", 0)
modparam("rr", "append_fromtag", 0)
modparam("acc", "early_media", 0)
modparam("acc", "report_ack", 0)
modparam("acc", "report_cancels", 0)
modparam("acc", "detect_direction", 0)
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)

####### 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();
}

# handle requests within SIP dialogs
route(WITHINDLG);

# record routing for dialog forming requests (in case they are routed)
remove_hf("Route");
if (is_method("INVITE|SUBSCRIBE|REFER")) {
record_route();
}

# account only INVITEs
if (is_method("INVITE")) {
setflag(FLT_ACC);
sl_send_reply("100","Trying");

if ($si == "172.16.16.1") {
sl_send_reply("183","Incoming session from Avoxi");
rewritehost("10.1.1.10");
#exit;
}
else if ($si == "10.1.1.10"){
# receiving response from client
sl_send_reply("183","Outgoing session to Avoxi");
#rewritehost("172.16.16.1");
drop;
exit;
}
else {
sl_send_reply("500","No configured IP!");
drop;
exit;
}
}

if ($rU==$null) {
sl_send_reply("484","Address Incomplete");
exit;
}

# received from main server - send to client and add via tokens for anycast
handling
via_add_srvid("1");
$xavp(via=>node) = "10.1.1.4";
via_add_xavp_params("1");
route(RELAY);
exit;
}

# 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] {
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("1511", "7")) {
xlog("Malformed SIP message from $si:$sp\n");
exit;
}

if ($si == "10.1.1.4") {
                xlog("L_WARN", "$ci|end|dropping message");
                exit;
    }

}

# Handle requests within SIP dialogs
route[WITHINDLG] {
if (!has_totag()) return;
if (loose_route()) {
if (is_method("BYE")) {
setflag(FLT_ACC);
setflag(FLT_ACCFAILED);
} else if ( is_method("NOTIFY") ) {
record_route();
}
route(RELAY);
exit;
}

if ( is_method("ACK") ) {
if ( t_check_trans() ) {
route(RELAY);
exit;
} else {
exit;
}
}
sl_send_reply("400","Loop detected");
exit;
}

# TM manage for outgoing branches
branch_route[MANAGE_BRANCH] {
xdbg("new branch [$T_branch_idx] to $ru\n");
}

# TM manage for incoming replies
onreply_route[MANAGE_REPLY] {
xdbg("incoming reply\n");
}

# TM manage for failure routing cases
failure_route[MANAGE_FAILURE] {
if (t_is_canceled()) exit;
}

asterisk - sip.conf

[opensips-trunk](sip-provider)
fromdomain=10.1.1.10
host=10.1.1.4
context=from-trunk
type=friend
insecure=invite,port
trunk=yes